From thangappan143 at gmail.com  Sat Aug  1 00:18:12 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Sat, 1 Aug 2009 12:48:12 +0530
Subject: [Freeswitch-users] ODBC problem
Message-ID: <7aa29e790908010018k71502d49ob69665a2d48b2047@mail.gmail.com>

While installing mod_lcr I got the following problem.
freeswitch at debian> load mod_lcr
API CALL [load(mod_lcr)] output:
-ERR [module load file routine returned an error]

freeswitch at debian> 2009-08-01 18:12:15 [INFO] mod_lcr.c:522
lcr_load_config() odbc_dsn is 192.168.1.222:freeswitch:freeswitch
2009-08-01 18:12:15 [INFO] mod_lcr.c:536 lcr_load_config() dsn is
"192.168.1.222", user is "freeswitch", and password is "freeswitch"
2009-08-01 18:12:15 [ERR] switch_odbc.c:164 switch_odbc_handle_connect()
STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not
found, and no default driver specified

2009-08-01 18:12:15 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot Open ODBC
Database!
2009-08-01 18:12:15 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to load lcr
config file
2009-08-01 18:12:15 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_lcr.so
**Module load routine returned an error**

What could the error?
How can I resolve it?

-- 
Regards,
Thangappan.M
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From dome at tel.co.th  Sat Aug  1 01:17:42 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 1 Aug 2009 15:17:42 +0700
Subject: [Freeswitch-users] ODBC problem
In-Reply-To: <7aa29e790908010018k71502d49ob69665a2d48b2047@mail.gmail.com>
References: <7aa29e790908010018k71502d49ob69665a2d48b2047@mail.gmail.com>
Message-ID: <8ccbff060908010117j6e2fe01ale4b72a12e70203a8@mail.gmail.com>

please check
/etc/odbc.ini
/opdbinst.ini

Dome C.

2009/8/1 Thangappan.M <thangappan143 at gmail.com>:
> While installing mod_lcr I got the following problem.
> freeswitch at debian> load mod_lcr
> API CALL [load(mod_lcr)] output:
> -ERR [module load file routine returned an error]
>
> freeswitch at debian> 2009-08-01 18:12:15 [INFO] mod_lcr.c:522
> lcr_load_config() odbc_dsn is 192.168.1.222:freeswitch:freeswitch
> 2009-08-01 18:12:15 [INFO] mod_lcr.c:536 lcr_load_config() dsn is
> "192.168.1.222", user is "freeswitch", and password is "freeswitch"
> 2009-08-01 18:12:15 [ERR] switch_odbc.c:164 switch_odbc_handle_connect()
> STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not
> found, and no default driver specified
>
> 2009-08-01 18:12:15 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot Open ODBC
> Database!
> 2009-08-01 18:12:15 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to load lcr
> config file
> 2009-08-01 18:12:15 [CRIT] switch_loadable_module.c:839
> switch_loadable_module_load_file() Error Loading module
> /usr/local/freeswitch/mod/mod_lcr.so
> **Module load routine returned an error**
>
> What could the error?
> How can I resolve it?
>
> --
> Regards,
> Thangappan.M
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From freeswitch at peely.com  Sat Aug  1 03:36:29 2009
From: freeswitch at peely.com (peely)
Date: Sat, 1 Aug 2009 03:36:29 -0700 (PDT)
Subject: [Freeswitch-users] LUA: Independent control of each call leg.
In-Reply-To: <87f2f3b90907311153n23a6a9e6h838206f281bc7d07@mail.gmail.com>
References: <24744087.post@talk.nabble.com>
	<87f2f3b90907311153n23a6a9e6h838206f281bc7d07@mail.gmail.com>
Message-ID: <24767998.post@talk.nabble.com>


Hi,

Thanks for your response. It's a real shame I can't get the async behavior I
want from Freeswitch/LUA as this is exactly the kind of abstraction layer I
hoped for in a SIP Application Server.

Most of the apps I want to develop would be served using the LUA environment
as-is but if a few scenarios I want to be able to perform a small amount of
activity whilst both legs are in a connected state, a prime example is a
b-leg "whisper" where you are still playing "ringing" to the a-party while
the b-party answers and hears a message just before connection to the
a-party.

Regards,


Neil.



mercutioviz wrote:
> 
> The level of control you need really isn't served by doing scripting from
> the dialplan. I highly recommend using ESL and the event socket. It will
> mean a bit of a paradigm shift in your coding, but with that shift comes a
> lot of power and control over what you can do with the calls - really
> limited only by your imagination.
> 
> -MC
> 
> On Fri, Jul 31, 2009 at 8:01 AM, peely <freeswitch at peely.com> wrote:
> 
>>
>> Hi,
>>
>> I'm trying to develop an application using lua and need to control the
>> inbound and outbound legs independently, even when they are switched
>> together.
>>
>> I can initiate the outbound session but I can't seem to bridge without
>> losing control of the script.
>>
>> Does anyone know a way I can allow ingress to egress calling whilst still
>> maintaining script control mid-call? I also need to ingress to hear
>> provisional speech during outbound connect. I've looked at conferencing
>> but
>> there seems to be quite a lot of automated messaging.
>>
>>
>>
>> Thanks,
>>
>>
>> Neil.
>>
> 
> 
> 

-- 
View this message in context: http://www.nabble.com/LUA%3A-Independent-control-of-each-call-leg.-tp24744087p24767998.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From krice at freeswitch.org  Sat Aug  1 04:52:16 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 01 Aug 2009 06:52:16 -0500
Subject: [Freeswitch-users] Connect to PostgreSQL database
In-Reply-To: <7aa29e790907312331v56770633m188a88aef5d4a9cc@mail.gmail.com>
Message-ID: <C69996A0.5A439%krice@freeswitch.org>

You need to look at using ODBC... That is what pretty much everything
uses... Connect FS to ODBC and ODBC to pgsql...



From: "Thangappan.M" <thangappan143 at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Sat, 1 Aug 2009 12:01:36 +0530
To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Connect to PostgreSQL database

Dear all,

?I installed postgresql database in my machine. So now I need to connect the
database from freeswitch.When I searched about the site.They told , to load
the mod_lcr module.

I followed the following steps.
? * Edit the modules.conf file and uncommented the applications/mod_lcr line
?* make mod_lcr-install
?* edit /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml and
uncomment the mod_lcr line.
?* Reload the freeswitch


Where I made a mistake?
Tell the steps to connect the postreSQL database.

-- 
Regards,
Thangappan.M


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From brian at freeswitch.org  Sat Aug  1 09:04:54 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 1 Aug 2009 11:04:54 -0500
Subject: [Freeswitch-users] LUA: Independent control of each call leg.
In-Reply-To: <24767998.post@talk.nabble.com>
References: <24744087.post@talk.nabble.com>
	<87f2f3b90907311153n23a6a9e6h838206f281bc7d07@mail.gmail.com>
	<24767998.post@talk.nabble.com>
Message-ID: <BCC8B127-B1A6-4AC6-B729-D9E389DBC0C9@freeswitch.org>

You can do all this via ESL-Lua cd libs/esl; make luamod

/b

On Aug 1, 2009, at 5:36 AM, peely wrote:

>
> Hi,
>
> Thanks for your response. It's a real shame I can't get the async  
> behavior I
> want from Freeswitch/LUA as this is exactly the kind of abstraction  
> layer I
> hoped for in a SIP Application Server.
>
> Most of the apps I want to develop would be served using the LUA  
> environment
> as-is but if a few scenarios I want to be able to perform a small  
> amount of
> activity whilst both legs are in a connected state, a prime example  
> is a
> b-leg "whisper" where you are still playing "ringing" to the a-party  
> while
> the b-party answers and hears a message just before connection to the
> a-party.
>
> Regards,
>
>
> Neil.



From gshfreesw at gmail.com  Sat Aug  1 09:35:09 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Sat, 1 Aug 2009 12:35:09 -0400
Subject: [Freeswitch-users] Connect to PostgreSQL database
In-Reply-To: <C69996A0.5A439%krice@freeswitch.org>
References: <7aa29e790907312331v56770633m188a88aef5d4a9cc@mail.gmail.com>
	<C69996A0.5A439%krice@freeswitch.org>
Message-ID: <5070fcbd0908010935n2d481397h210ce38057f03933@mail.gmail.com>

Thangappa,

Are you using Perl Event Socket? In that use, you can use Perl DBI to
connect to any DB instead of using the FS's ODBC module.

On Sat, Aug 1, 2009 at 7:52 AM, Ken Rice <krice at freeswitch.org> wrote:

>  You need to look at using ODBC... That is what pretty much everything
> uses... Connect FS to ODBC and ODBC to pgsql...
>
>
> ------------------------------
> *From: *"Thangappan.M" <thangappan143 at gmail.com>
> *Reply-To: *<freeswitch-users at lists.freeswitch.org>
> *Date: *Sat, 1 Aug 2009 12:01:36 +0530
> *To: *freeswitch-users <freeswitch-users at lists.freeswitch.org>
> *Subject: *[Freeswitch-users] Connect to PostgreSQL database
>
>
> Dear all,
>
>  I installed postgresql database in my machine. So now I need to connect
> the database from freeswitch.When I searched about the site.They told , to
> load the mod_lcr module.
>
> I followed the following steps.
>   * Edit the modules.conf file and uncommented the applications/mod_lcr
> line
>  * make mod_lcr-install
>  * edit /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml and
> uncomment the mod_lcr line.
>  * Reload the freeswitch
>
>
> Where I made a mistake?
> Tell the steps to connect the postreSQL database.
>
> --
> Regards,
> Thangappan.M
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Sat Aug  1 09:43:28 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 01 Aug 2009 11:43:28 -0500
Subject: [Freeswitch-users] Connect to PostgreSQL database
In-Reply-To: <5070fcbd0908010935n2d481397h210ce38057f03933@mail.gmail.com>
Message-ID: <C699DAE0.5A45A%krice@freeswitch.org>

He specifically said from FreeSWITCH... The ONLY way to connect FreeSWITCH
directory to any database (this does not includes external scripts running
as a separate process using ESL) is to either a) write your own custom
module to do so or b) use ODBC on the modules that support that now



From: Shameem Shiek <gshfreesw at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Sat, 1 Aug 2009 12:35:09 -0400
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Connect to PostgreSQL database

Thangappa,

Are you using Perl Event Socket? In that use, you can use Perl DBI to
connect to any DB instead of using the FS's ODBC module.

On Sat, Aug 1, 2009 at 7:52 AM, Ken Rice <krice at freeswitch.org> wrote:
> You need to look at using ODBC... That is what pretty much everything uses...
> Connect FS to ODBC and ODBC to pgsql...
> 
> 
> 
> From: "Thangappan.M" <thangappan143 at gmail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 1 Aug 2009 12:01:36 +0530
> To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Connect to PostgreSQL database
> 
> 
> Dear all,
> 
> ?I installed postgresql database in my machine. So now I need to connect the
> database from freeswitch.When I searched about the site.They told , to load
> the mod_lcr module.
> 
> I followed the following steps.
> ? * Edit the modules.conf file and uncommented the applications/mod_lcr line
> ?* make mod_lcr-install
> ?* edit /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml and
> uncomment the mod_lcr line.
> ?* Reload the freeswitch
> 
> 
> Where I made a mistake?
> Tell the steps to connect the postreSQL database.
> 
> -- 
> Regards,
> Thangappan.M
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From darren at dmmhosting.co.uk  Sat Aug  1 03:53:06 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Sat, 1 Aug 2009 11:53:06 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <54141a3f-017f-4469-965a-1717526c5b2f@dmmhosting.co.uk>

I am considering using freeswitch and would like to know if this is possible.
 
The provider I use has a host that sits behind an OpenSER proxy. The hostname cannot get resolved by DNS on the internet.
 
Using freeswitch, at the moment, I am getting a DNS failure message for the host.
 
Is there a way of registering to this host and making calls through it by making all traffic go through the outbound proxy?
 
TIA
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4292 (20090730) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From gmaruzz at celliax.org  Sat Aug  1 11:34:34 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 1 Aug 2009 20:34:34 +0200
Subject: [Freeswitch-users] if using centos you should read this
In-Reply-To: <7b197bef0907302321p6aea0f2fhc542551b70392d6e@mail.gmail.com>
References: <6C1D282197D14BB9935770FD34B8F42A@noblesys.com> 
	<92D4AB13-73E0-4989-AC2F-A6703354D14B@freeswitch.org>
	<b16156850907302157p7898e599m8ae98c2b8f133d49@mail.gmail.com> 
	<87f2f3b90907302221u67ce4e32mf2e8faa8a4a13771@mail.gmail.com> 
	<b16156850907302236i696d3941i6f2a24ca59d9ff85@mail.gmail.com> 
	<7b197bef0907302321p6aea0f2fhc542551b70392d6e@mail.gmail.com>
Message-ID: <7b197bef0908011134u4ceeb820k13fdf6a222374235@mail.gmail.com>

http://linux.slashdot.org/story/09/08/01/1443221/CentOS-Administrator-Reappears

str8edge sends word that Lance Davis, the CentOS project administrator
who had mysteriously gone absent, has now returned and is working with
the development team to get things back on track. From their
announcement: "The CentOS Development team had a routine meeting today
with Lance Davis in attendance. During the meeting a majority of
issues were resolved immediately and a working agreement was reached
with deadlines for remaining unresolved issues. There should be no
impact to any CentOS users going forward. The CentOS project is now in
control of the CentOS.org and CentOS.info domains and owns all
trademarks, materials, and artwork in the CentOS distributions. We
look forward to working with Lance to quickly complete all the agreed
upon issues. More information will follow soon."

On Fri, Jul 31, 2009 at 8:21 AM, Giovanni Maruzzelli<gmaruzz at celliax.org> wrote:
> :-)!
>
>
> On Fri, Jul 31, 2009 at 7:36 AM, Muhammad
> Shahzad<shaheryarkh at googlemail.com> wrote:
>> Please read my email as,
>>
>>> CentOS has been a trusted platfrom for me from last 3+ years. I have
>>> developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS
>>> boxes, and 27 out of 49 Asterisk box are still running on CentOS in
>>> production environment. I really wish and hope this great project continues.
>>>
>>> I don't know any of its developers personally but i am quite sure they
>>> will resolve their differences professionally and put this project back on
>>> track.
>>
>> This damn Google Spell made meaning of my entire post the possite. ;-(
>>
>> Thank you.
>>
>>
>> On Fri, Jul 31, 2009 at 11:21 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>>>
>>>
>>> On Thu, Jul 30, 2009 at 9:57 PM, Muhammad Shahzad
>>> <shaheryarkh at googlemail.com> wrote:
>>>>
>>>> CentOS has been a trusted platfrom for me from last 3+ years. I have
>>>> developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS
>>>> boxes, and 27 out of 49 Asterisk box are still ruining on CentOS in
>>>> production environment. I really wish and hope this great project continues.
>>>>
>>>> I don't know any of its developers personally but i am quite sure they
>>>> will resolve their differences professionally and put this project back on
>>>> track.
>>>
>>> The guys doing the work have vowed to continue the project. The only real
>>> issues are who controls the centos.org domain name and how to handle
>>> donations to the project. CentOS isn't going anywhere but forward.
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From d at unwire.it  Sat Aug  1 11:57:35 2009
From: d at unwire.it (Darin Weeks)
Date: Sat, 1 Aug 2009 11:57:35 -0700
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <54141a3f-017f-4469-965a-1717526c5b2f@dmmhosting.co.uk>
References: <54141a3f-017f-4469-965a-1717526c5b2f@dmmhosting.co.uk>
Message-ID: <989132e70908011157m67680efbm427282aaab87255a@mail.gmail.com>

Have you looked over various example configurations on the wiki?  see: SIP
Provider Examples

I'm a bit confused by your questions... to make voip calls work, you
basically need to register/authenticate with proxies/gateways to send and
receive calls through them.  Sounds like you need to find out from your
provider's documentation as to how they expect you to connect with them.
Why are you trying to connect to the host behind the proxy rather than the
proxy?


On Sat, Aug 1, 2009 at 3:53 AM, Darren Williams <darren at dmmhosting.co.uk>wrote:

>  I am considering using freeswitch and would like to know if this is
> possible.
>
>
>
> The provider I use has a host that sits behind an OpenSER proxy. The
> hostname cannot get resolved by DNS on the internet.
>
>
>
> Using freeswitch, at the moment, I am getting a DNS failure message for the
> host.
>
>
>
> Is there a way of registering to this host and making calls through it by
> making all traffic go through the outbound proxy?
>
>
>
> TIA
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4292 (20090730) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( (
broadband internet for west hollywood
   d at unwire.it        http://unwire.it
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From pjintheusa at gmail.com  Sat Aug  1 12:36:09 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 1 Aug 2009 15:36:09 -0400
Subject: [Freeswitch-users] LUA: Independent control of each call leg.
In-Reply-To: <24744087.post@talk.nabble.com>
References: <24744087.post@talk.nabble.com>
Message-ID: <367751820908011236p5db2fcfbu859bdbef60d6de7e@mail.gmail.com>

 >>a prime example is a b-leg "whisper" where you are still playing
"ringing" to the a-party while
>>the b-party answers and hears a message just before connection to the
>>a-party.

You should be able to do this particular function using group_confirm. See
this page in the wiki.

http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm

In your case the test.js would contain the whisper. Note that you can send
args to this script.


On Fri, Jul 31, 2009 at 11:01 AM, peely <freeswitch at peely.com> wrote:

>
> Hi,
>
> I'm trying to develop an application using lua and need to control the
> inbound and outbound legs independently, even when they are switched
> together.
>
> I can initiate the outbound session but I can't seem to bridge without
> losing control of the script.
>
> For example, if I use:
>
>
>        local api = freeswitch.API();
>        inSession = session;
>        inSession:answer();
>        inSession:setAutoHangup(false);
>
>
>        egSession = freeswitch.Session("sofia/default/mynum at mydomain.com");
>        egSession:setAutoHangup(false);
>
>        if egSession:ready() then
>                api:execute("uuid_bridge",inSession.uuid .. " " ..
> egSession.uuid);
>        end
>
>        while egSession:ready() do
>                inSession:sleep(1000);
>        end
>
> Then I lose the script entirely, and if I use:
>
>        inSession:execute("bridge", "sofia/default/mynum at mydomain.com")
>
> Then I lose the ability to control the call whilst the outbound is in
> progress.
>
> Does anyone know a way I can allow ingress to egress calling whilst still
> maintaining script control mid-call? I also need to ingress to hear
> provisional speech during outbound connect. I've looked at conferencing but
> there seems to be quite a lot of automated messaging.
>
>
>
> Thanks,
>
>
> Neil.
>
> --
> View this message in context:
> http://www.nabble.com/LUA%3A-Independent-control-of-each-call-leg.-tp24744087p24744087.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Aug  2 01:33:32 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 Aug 2009 03:33:32 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <989132e70908011157m67680efbm427282aaab87255a@mail.gmail.com>
References: <54141a3f-017f-4469-965a-1717526c5b2f@dmmhosting.co.uk>
	<989132e70908011157m67680efbm427282aaab87255a@mail.gmail.com>
Message-ID: <65978701-BC97-4C7F-9BC5-EC66F769F3ED@freeswitch.org>

Fill out proxy... with the fake hostname... then fill out register- 
proxy and/or outbound-proxy.

/b

On Aug 1, 2009, at 1:57 PM, Darin Weeks wrote:

> Have you looked over various example configurations on the wiki?   
> see: SIP Provider Examples
>
> I'm a bit confused by your questions... to make voip calls work, you  
> basically need to register/authenticate with proxies/gateways to  
> send and receive calls through them.  Sounds like you need to find  
> out from your provider's documentation as to how they expect you to  
> connect with them.  Why are you trying to connect to the host behind  
> the proxy rather than the proxy?
>
>
> On Sat, Aug 1, 2009 at 3:53 AM, Darren Williams <darren at dmmhosting.co.uk 
> > wrote:
> I am considering using freeswitch and would like to know if this is  
> possible.
>
>
> The provider I use has a host that sits behind an OpenSER proxy. The  
> hostname cannot get resolved by DNS on the internet.
>
>
> Using freeswitch, at the moment, I am getting a DNS failure message  
> for the host.
>
>
> Is there a way of registering to this host and making calls through  
> it by making all traffic go through the outbound proxy?
>
>
> TIA
>

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From markmorreny at gmail.com  Sun Aug  2 02:04:18 2009
From: markmorreny at gmail.com (mark morreny)
Date: Sun, 2 Aug 2009 17:04:18 +0800
Subject: [Freeswitch-users] H248 support
Message-ID: <20ad6b920908020204p6a6bec6dt32f17638d778a4e0@mail.gmail.com>

Hi,

Someone told me to check here.  I am looking for a H248 supported gateway.
Does freeswitch support H248 or is there anyway to make it supportable in
freeSWITCH?

Thanks,
Mark
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From brian at freeswitch.org  Sun Aug  2 02:09:06 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 Aug 2009 04:09:06 -0500
Subject: [Freeswitch-users] H248 support
In-Reply-To: <20ad6b920908020204p6a6bec6dt32f17638d778a4e0@mail.gmail.com>
References: <20ad6b920908020204p6a6bec6dt32f17638d778a4e0@mail.gmail.com>
Message-ID: <D489C23D-0858-4221-A37B-AB5AFF821B4C@freeswitch.org>

Not at this time.. you could try contacting consulting at freeswitch.org  
to see what it might take to gain this support by funding it.

/b

On Aug 2, 2009, at 4:04 AM, mark morreny wrote:

> Hi,
>
> Someone told me to check here.  I am looking for a H248 supported  
> gateway.  Does freeswitch support H248 or is there anyway to make it  
> supportable in freeSWITCH?
>
> Thanks,
> Mark



From darren at dmmhosting.co.uk  Sun Aug  2 03:31:17 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Sun, 2 Aug 2009 11:31:17 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <52d777d0-f93e-4e41-9293-d6c403f60080@dmmhosting.co.uk>

?and/or outbound proxy?, I do not seem to be able to find this parameter.
 
It is probably my terminology which is confusing too and what exactly I want to do. All I know is that on my Thomson ST2030. If I enter:
 
Registrar Server Address: bmnha-01.bt.com
Proxy Server Address: bmnha-01.bt.com
Outbound Proxy Server:www.bbvservice-560129.bt.com
 
Everything works fine, I would just like to be able to transfer this config to freeswitch.
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 02 August 2009 09:34
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
Fill out proxy... with the fake hostname... then fill out register-proxy and/or outbound-proxy.
 
/b
 
On Aug 1, 2009, at 1:57 PM, Darin Weeks wrote:



Have you looked over various example configurations on the wiki?  see: SIP Provider Examples

I'm a bit confused by your questions... to make voip calls work, you basically need to register/authenticate with proxies/gateways to send and receive calls through them.  Sounds like you need to find out from your provider's documentation as to how they expect you to connect with them.  Why are you trying to connect to the host behind the proxy rather than the proxy?


On Sat, Aug 1, 2009 at 3:53 AM, Darren Williams <darren at dmmhosting.co.uk> wrote:
I am considering using freeswitch and would like to know if this is possible.
 
The provider I use has a host that sits behind an OpenSER proxy. The hostname cannot get resolved by DNS on the internet.
 
Using freeswitch, at the moment, I am getting a DNS failure message for the host.
 
Is there a way of registering to this host and making calls through it by making all traffic go through the outbound proxy?
 
TIA
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4297 (20090801) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4297 (20090801) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From rdenert at tng.de  Sun Aug  2 05:18:50 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Sun, 2 Aug 2009 14:18:50 +0200 (CEST)
Subject: [Freeswitch-users] Language of the speech
In-Reply-To: <6CE276B1-75B1-4470-BDD0-0E51721D3658@jerris.com>
Message-ID: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>

Hello,

sorry but I don't understand your answer.

BR

----- Urspr?ngliche Mail -----
Von: "Michael Jerris" <mike at jerris.com>
An: freeswitch-users at lists.freeswitch.org
Gesendet: Freitag, 31. Juli 2009 21:34:16 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: Re: [Freeswitch-users] Language of the speech

we don;t have any german sound files?

On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:

> Hi again,
>
> does anybody know why my freeswitch doesn't play German speech-files?
>
> Here is my construct:
>
> I'm generating a random number in my lua script:
> rand = math.random(11, 1000);
>
> The I want that the freeswitch says the "random number":
> session:execute("say", "de name_spelled iterated " ..test_nummer);
>
> It works fine in English but I don't bring the freeswitch to say the  
> number in german. I installed the freeswitch-lang- 
> de_1.0.3-1_i386.deb, change the line in freeswitch.xml:
> <section name="phrases" description="Speech Phrase Management">
>    <macros>
>      <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>    </macros>
> </section>
>
> Here is an extraction from the fs_cli:
> 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152  
> switch_core_standard_on_execute() sofia/external/anonymous at anonymous.invalid 
>  Execute set(default_language=de)
> 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous at anonymous.invalid 
>  SET [default_language]=[de]
>
> But I don't here any German speech only the English on
>
> Does anybody of you have an idea? Thanks a lot again.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
-
Register: Amtsgericht Kiel, HRB 6596 KI
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Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
-
This e-mail may contain confidential and/or privileged information. If
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unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.


From mike at jerris.com  Sun Aug  2 11:06:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 2 Aug 2009 14:06:58 -0400
Subject: [Freeswitch-users] Language of the speech
In-Reply-To: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>
References: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>
Message-ID: <A8ED0B8A-A1AD-4240-82AB-F5F125E2ACAA@jerris.com>

how is going to play files that we don't have?

On Aug 2, 2009, at 8:18 AM, Rudolf Denert wrote:

> Hello,
>
> sorry but I don't understand your answer.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Michael Jerris" <mike at jerris.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Freitag, 31. Juli 2009 21:34:16 GMT +01:00 Amsterdam/ 
> Berlin/Bern/Rom/Stockholm/Wien
> Betreff: Re: [Freeswitch-users] Language of the speech
>
> we don;t have any german sound files?
>
> On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:
>
>> Hi again,
>>
>> does anybody know why my freeswitch doesn't play German speech-files?
>>
>> Here is my construct:
>>
>> I'm generating a random number in my lua script:
>> rand = math.random(11, 1000);
>>
>> The I want that the freeswitch says the "random number":
>> session:execute("say", "de name_spelled iterated " ..test_nummer);
>>
>> It works fine in English but I don't bring the freeswitch to say the
>> number in german. I installed the freeswitch-lang-
>> de_1.0.3-1_i386.deb, change the line in freeswitch.xml:
>> <section name="phrases" description="Speech Phrase Management">
>>   <macros>
>>     <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>>   </macros>
>> </section>
>>
>> Here is an extraction from the fs_cli:
>> 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152
>> switch_core_standard_on_execute() sofia/external/anonymous at anonymous.invalid
>> Execute set(default_language=de)
>> 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous at anonymous.invalid
>> SET [default_language]=[de]
>>
>> But I don't here any German speech only the English on
>>
>> Does anybody of you have an idea? Thanks a lot again.
>



From wiltingtree at gmail.com  Sun Aug  2 12:36:07 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sun, 2 Aug 2009 15:36:07 -0400
Subject: [Freeswitch-users] Bridging a call to an extension on another PBX.
Message-ID: <eb7797d70908021236x18eabbb4t6882553da99642a0@mail.gmail.com>

Hello,
I'm trying to conference-in a call from FreeSWITCH to an extension on
another PBX using sip.

According to the documentation, I think it should look like this:

       conference abc at default dial
{sip_auth_username=myuser,sip_auth_password=mypassword}sofia/external/
101 at 1.2.3.4

where 1.2.3.4 is the ip address of the remote pbx, and 101 is the extension.

I've tried adding a gateway for it in the sip profiles, and then doing this:

      conference abc at default dial sofia/mygateway/701

Both of these methods give me a result of:

      Call Requested: result: [DESTINATION_OUT_OF_ORDER]

I set-up my soft phone to register to the same ip address with the same
credentials, and it allows me to call the extension properly.

Can somebody please tell me what I'm doing wrong?

Thanks,
Adam
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From nik.middleton at noblesolutions.co.uk  Sun Aug  2 12:38:55 2009
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Sun, 2 Aug 2009 20:38:55 +0100
Subject: [Freeswitch-users] Outbound socket question
Message-ID: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>

Hi Guys,

 

I'm using an outbound socket to control calls, and it works a charm.
However, what I'd like to do is send a custom event regarding the call
on hang-up.  The way I see things happening at the moment, and I could
be wrong, is that the socket is closed when a hang-up occurs, so am I
taking a chance trying to send the event then? (try to sneak out the
event before socket closure happens)  The other option is of course to
open an inbound socket and send the event, but I'd rather not do that if
possible.

 

Regards

  

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From jmesquita at gmail.com  Sun Aug  2 13:21:54 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 2 Aug 2009 17:21:54 -0300
Subject: [Freeswitch-users] Bridging a call to an extension on another
	PBX.
In-Reply-To: <eb7797d70908021236x18eabbb4t6882553da99642a0@mail.gmail.com>
References: <eb7797d70908021236x18eabbb4t6882553da99642a0@mail.gmail.com>
Message-ID: <5a8712120908021321n750e549cl3e2f994813cc3b2d@mail.gmail.com>

Adam,

Pastebin the logs. Also, a sip dump of both situations can really help.

To enable sip traces on FreeSWITCH all you have to do is type on the CLI:

sofia profile <profile> siptrace on/off


jmesquita

On Sun, Aug 2, 2009 at 4:36 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Hello,
> I'm trying to conference-in a call from FreeSWITCH to an extension on
> another PBX using sip.
>
> According to the documentation, I think it should look like this:
>
>        conference abc at default dial
> {sip_auth_username=myuser,sip_auth_password=mypassword}sofia/external/
> 101 at 1.2.3.4
>
> where 1.2.3.4 is the ip address of the remote pbx, and 101 is the
> extension.
>
> I've tried adding a gateway for it in the sip profiles, and then doing
> this:
>
>       conference abc at default dial sofia/mygateway/701
>
> Both of these methods give me a result of:
>
>       Call Requested: result: [DESTINATION_OUT_OF_ORDER]
>
> I set-up my soft phone to register to the same ip address with the same
> credentials, and it allows me to call the extension properly.
>
> Can somebody please tell me what I'm doing wrong?
>
> Thanks,
> Adam
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From merul at mac.com  Sun Aug  2 14:34:08 2009
From: merul at mac.com (Merul Patel)
Date: Sun, 02 Aug 2009 22:34:08 +0100
Subject: [Freeswitch-users] Configuring Sangoma U100
Message-ID: <27E91460-3D50-4DF4-AF1D-95D97634112C@mac.com>

I'm new to FS, and experimenting with it on a constrained environment  
(PCEngines ALIX board running Voyage Linux 0.62).

So far, FS has compiled fine, and I can register multiple softphones  
and make calls between them, but I'm lost at how to configure a  
Sangoma U100 so I can make and receive calls over an analogue line.

I've installed the wanpipe drivers (3.5.4) from Sangoma, and the  
wanrouter utility detects the USB device, and I've compiled it to  
support the TDM API.

FS was compiled with the Openzap module - as best as I can tell.

I thought that I would be able to use the wancfg_tdmapi utility to  
configure the /etc/wanpipe/wanpipe1.conf and then use the generated  
configuration file as the basis for configuring autoload_configs/ 
openzap.conf.xml.

However, the wancfg utility doesn't generate the wanpipe1.conf, and  
I'm stumped.

Any pointers would be much appreciated.

Best regards,

Merul
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From msc at freeswitch.org  Sun Aug  2 15:35:01 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 2 Aug 2009 15:35:01 -0700
Subject: [Freeswitch-users] Outbound socket question
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
Message-ID: <87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>

On Sun, Aug 2, 2009 at 12:38 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

>  Hi Guys,
>
>
>
> I?m using an outbound socket to control calls, and it works a charm.
> However, what I?d like to do is send a custom event regarding the call on
> hang-up.  The way I see things happening at the moment, and I could be
> wrong, is that the socket is closed when a hang-up occurs, so am I taking a
> chance trying to send the event then? (try to sneak out the event before
> socket closure happens)  The other option is of course to open an inbound
> socket and send the event, but I?d rather not do that if possible.
>
Nik,

Perhaps the "linger" event socket command will do what you need? Check out
this commit:
http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009391.html

Let me know if it works for you and I'll be sure to get it documented
properly. If you get it working I'd love to see a code snippet so we can
wikify this knowledge. :)

Thanks,
MC
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From mike at jerris.com  Sun Aug  2 15:42:57 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 2 Aug 2009 18:42:57 -0400
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <27E91460-3D50-4DF4-AF1D-95D97634112C@mac.com>
References: <27E91460-3D50-4DF4-AF1D-95D97634112C@mac.com>
Message-ID: <4241B9E6-0F1F-4F9E-A1DF-CC0A1E7BE8F9@jerris.com>


On Aug 2, 2009, at 5:34 PM, Merul Patel wrote:

> I'm new to FS, and experimenting with it on a constrained  
> environment (PCEngines ALIX board running Voyage Linux 0.62).
>
> So far, FS has compiled fine, and I can register multiple softphones  
> and make calls between them, but I'm lost at how to configure a  
> Sangoma U100 so I can make and receive calls over an analogue line.
>
> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the  
> wanrouter utility detects the USB device, and I've compiled it to  
> support the TDM API.
>
> FS was compiled with the Openzap module - as best as I can tell.
>
> I thought that I would be able to use the wancfg_tdmapi utility to  
> configure the /etc/wanpipe/wanpipe1.conf and then use the generated  
> configuration file as the basis for configuring autoload_configs/ 
> openzap.conf.xml.

try wancfg_fs

> However, the wancfg utility doesn't generate the wanpipe1.conf, and  
> I'm stumped.
>
> Any pointers would be much appreciated.
>
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From brian at freeswitch.org  Sun Aug  2 16:10:48 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 Aug 2009 18:10:48 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <52d777d0-f93e-4e41-9293-d6c403f60080@dmmhosting.co.uk>
References: <52d777d0-f93e-4e41-9293-d6c403f60080@dmmhosting.co.uk>
Message-ID: <1305B47F-256B-494A-B3D4-E1850E134A2C@freeswitch.org>

set the proxy, register-proxy to bmnha-01.bt.com and the outbound- 
proxy to www.bbvservice-560129.bt.com, I regard this type of config B  
R O K EN and the provider shouldn't be doing this with DNS names that  
do not exist in my opinion... if they are doing this for security sake  
they should just setup VPN or direct access... this type of setup must  
makes my skin crawl.

/b

On Aug 2, 2009, at 5:31 AM, Darren Williams wrote:

> ?and/or outbound proxy?, I do not seem to be able to find this  
> parameter.
>
> It is probably my terminology which is confusing too and what  
> exactly I want to do. All I know is that on my Thomson ST2030. If I  
> enter:
>
> Registrar Server Address: bmnha-01.bt.com
> Proxy Server Address: bmnha-01.bt.com
> Outbound Proxy Server:www.bbvservice-560129.bt.com
>
> Everything works fine, I would just like to be able to transfer this  
> config to freeswitch.

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From mrene_lists at avgs.ca  Sun Aug  2 18:11:11 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 2 Aug 2009 21:11:11 -0400
Subject: [Freeswitch-users] Language of the speech
In-Reply-To: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>
References: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>
Message-ID: <B4C913FC-6332-489C-B0A1-887CBEECF96D@avgs.ca>

Da gibst keine eigentliches DE audio-Datei, nur text fuer text-to- 
speech, deshalb funktionniert es nicht.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 2-Aug-09 um 8:18 AM schrieb Rudolf Denert:

> Hello,
>
> sorry but I don't understand your answer.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Michael Jerris" <mike at jerris.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Freitag, 31. Juli 2009 21:34:16 GMT +01:00 Amsterdam/ 
> Berlin/Bern/Rom/Stockholm/Wien
> Betreff: Re: [Freeswitch-users] Language of the speech
>
> we don;t have any german sound files?
>
> On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:
>
>> Hi again,
>>
>> does anybody know why my freeswitch doesn't play German speech-files?
>>
>> Here is my construct:
>>
>> I'm generating a random number in my lua script:
>> rand = math.random(11, 1000);
>>
>> The I want that the freeswitch says the "random number":
>> session:execute("say", "de name_spelled iterated " ..test_nummer);
>>
>> It works fine in English but I don't bring the freeswitch to say the
>> number in german. I installed the freeswitch-lang-
>> de_1.0.3-1_i386.deb, change the line in freeswitch.xml:
>> <section name="phrases" description="Speech Phrase Management">
>>   <macros>
>>     <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>>   </macros>
>> </section>
>>
>> Here is an extraction from the fs_cli:
>> 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152
>> switch_core_standard_on_execute() sofia/external/anonymous at anonymous.invalid
>> Execute set(default_language=de)
>> 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous at anonymous.invalid
>> SET [default_language]=[de]
>>
>> But I don't here any German speech only the English on
>>
>> Does anybody of you have an idea? Thanks a lot again.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -- 
> -- 
> Rudolf Denert, Technical Support
> TNG AG, NGN
> Projensdorfer Str. 324, D-24106 Kiel, Germany
> phone: +49 431 7097-10, fax: +49 431 7097-555
> mailto: rdenert at tng.de http://www.tng.de
> -
> Register: Amtsgericht Kiel, HRB 6596 KI
> Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
> Executive board (Vorstand): Dr. Sven Willert (Vorsitz),  Dr. Thomas
> Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg
> Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
> -
> This e-mail may contain confidential and/or privileged information. If
> you are not the intended recipient (or have received this e-mail in
> error) please notify the sender immediately and destroy this e-mail.  
> Any
> unauthorized copying, disclosure or distribution of the material in  
> this
> e-mail is strictly forbidden.
>
> Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
> Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
> E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
> Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
> die unbefugte Weitergabe dieser Mail ist nicht gestattet.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From thangappan143 at gmail.com  Sun Aug  2 22:09:23 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 3 Aug 2009 10:39:23 +0530
Subject: [Freeswitch-users] Problem in spidermonkey_odbc
Message-ID: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>

Dear all,

 I am not yet installed odbc in my machine for accessing the psql database.
I have got the following error while tried to load the mod_lcr command.

2009-08-03 15:45:47 [ERR] switch_odbc.c:164 switch_odbc_handle_connect()
STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not
found, and no default driver specified

2009-08-03 15:45:47 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot Open ODBC
Database!
2009-08-03 15:45:47 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to load lcr
config file
2009-08-03 15:45:47 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_lcr.so
**Module load routine returned an error**

I have set the correct driver informations in the odbc.ini(/etc and /home)
and odbcinst.ini(/etc).
I have create the symbolic links in the /usr/loca/freeswitch/etc/

Where I made a problem?

After some time I found that there is no spidermonket_odbc.so file.This has
been found while executing the freeswitch command. The error is,

2009-08-03 15:26:06 [ERR] mod_spidermonkey.c:931 sm_load_file() Error
Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
**/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: cannot open shared
object file: No such file or directory**

So please help me?
I am doing this for more than 8 hours.





-- 
Regards,
Thangappan.M
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From mattdfong at gmail.com  Sun Aug  2 22:22:47 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Sun, 2 Aug 2009 22:22:47 -0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
Message-ID: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>

A few users of mine have been getting hung-up on after leg b of the bridge
hangsups. I looked in the logs and they are being hungup with
an EXCHANGE_ROUTING_ERROR hangup cause. The problem is
only occurring intermittently and only when both leg a and leg b are both
passed thru external gateways (the problem does *not* exist if leg a is a
softphone connected directly to FS).
I'm wondering if there are any known incompatibilities with equipment out
there that would cause this error. Otherwise, I assume the best way to
diagnosis this further is to ngrep, which I'll do, but thought I'd ask here
first. Thanks.

--matt
hello hunter - hosted predictive dialer & voice broadcasting
http://www.hellohunter.com
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From mike at jerris.com  Sun Aug  2 23:12:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Aug 2009 01:12:51 -0500
Subject: [Freeswitch-users] Problem in spidermonkey_odbc
In-Reply-To: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
References: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
Message-ID: <D1107F74-5E72-485C-89EF-ED66A3A1AFEC@jerris.com>

You need to first install unixodbc and it's related devel packages and  
then run configure and re build freeswitch.

On Aug 3, 2009, at 12:09 AM, "Thangappan.M" <thangappan143 at gmail.com>  
wrote:

> Dear all,
>
>  I am not yet installed odbc in my machine for accessing the psql  
> database. I have got the following error while tried to load the  
> mod_lcr command.
>
> 2009-08-03 15:45:47 [ERR] switch_odbc.c:164  
> switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC] 
> [Driver Manager]Data source name not found, and no default driver  
> specified
>
> 2009-08-03 15:45:47 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot  
> Open ODBC Database!
> 2009-08-03 15:45:47 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to  
> load lcr config file
> 2009-08-03 15:45:47 [CRIT] switch_loadable_module.c:839  
> switch_loadable_module_load_file() Error Loading module /usr/local/ 
> freeswitch/mod/mod_lcr.so
> **Module load routine returned an error**
>
> I have set the correct driver informations in the odbc.ini(/etc and / 
> home) and odbcinst.ini(/etc).
> I have create the symbolic links in the /usr/loca/freeswitch/etc/
>
> Where I made a problem?
>
> After some time I found that there is no spidermonket_odbc.so  
> file.This has been found while executing the freeswitch command. The  
> error is,
>
> 2009-08-03 15:26:06 [ERR] mod_spidermonkey.c:931 sm_load_file()  
> Error Loading module /usr/local/freeswitch/mod/ 
> mod_spidermonkey_odbc.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: cannot open  
> shared object file: No such file or directory**
>
> So please help me?
> I am doing this for more than 8 hours.
>
>
>
>
>
> -- 
> Regards,
> Thangappan.M
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mrene_lists at avgs.ca  Sun Aug  2 23:20:35 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 3 Aug 2009 02:20:35 -0400
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
Message-ID: <177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>

Hi,

Digging a bit in mod_sofia releaved that it can be caused by a SIP  
code 482 (loop detected), 483 (too many hops) or 484 (address  
incomplete).

Do a SIP trace to sched more light on what's happening.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 3-Aug-09 um 1:22 AM schrieb Matthew Fong:

> EXCHANGE_ROUTING_ERROR



From thangappan143 at gmail.com  Mon Aug  3 00:07:29 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 3 Aug 2009 12:37:29 +0530
Subject: [Freeswitch-users] Problem in spidermonkey_odbc
In-Reply-To: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
References: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
Message-ID: <7aa29e790908030007l1478b8b1mfa460a556349342f@mail.gmail.com>

I have installed unixodbc at first.But I am getting the same error.

On Mon, Aug 3, 2009 at 10:39 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Dear all,
>
>  I am not yet installed odbc in my machine for accessing the psql database.
> I have got the following error while tried to load the mod_lcr command.
>
> 2009-08-03 15:45:47 [ERR] switch_odbc.c:164 switch_odbc_handle_connect()
> STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not
> found, and no default driver specified
>
> 2009-08-03 15:45:47 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot Open ODBC
> Database!
> 2009-08-03 15:45:47 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to load lcr
> config file
> 2009-08-03 15:45:47 [CRIT] switch_loadable_module.c:839
> switch_loadable_module_load_file() Error Loading module
> /usr/local/freeswitch/mod/mod_lcr.so
> **Module load routine returned an error**
>
> I have set the correct driver informations in the odbc.ini(/etc and /home)
> and odbcinst.ini(/etc).
> I have create the symbolic links in the /usr/loca/freeswitch/etc/
>
> Where I made a problem?
>
> After some time I found that there is no spidermonket_odbc.so file.This has
> been found while executing the freeswitch command. The error is,
>
> 2009-08-03 15:26:06 [ERR] mod_spidermonkey.c:931 sm_load_file() Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: cannot open shared
> object file: No such file or directory**
>
> So please help me?
> I am doing this for more than 8 hours.
>
>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From thangappan143 at gmail.com  Mon Aug  3 00:12:14 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 3 Aug 2009 12:42:14 +0530
Subject: [Freeswitch-users] Need Help In IVR
Message-ID: <7aa29e790908030012rcce847fu8320d833d2c85530@mail.gmail.com>

Dear all,

I am in the process of implementing IVR in Perl using outbound socket.

In the case of the XML macro I can easily specify the timeout,inter digit
timeout value as
 <menu name="demo_ivr"
          greet-long="phrase:demo_ivr_main_menu"
          greet-short="phrase:demo_ivr_main_menu_short"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout ="10000"
          inter-digit-timeout="2000"
          max-failures="3"
          max-timeouts="3"
          digit-len="4">
      <entry action="menu-exec-app" digits="1" param="bridge
sofia/$${domain}/888 at conference.freeswitch.org"/>
    </menu>

Is there any way for Perl to configure this values. Where are the variables
resides?
I am struggling to implement this?




-- 
Regards,
Thangappan.M
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From rdenert at tng.de  Mon Aug  3 00:57:37 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Mon, 3 Aug 2009 09:57:37 +0200 (CEST)
Subject: [Freeswitch-users] Language of the speech
In-Reply-To: <8128254.210221249286232840.JavaMail.root@zimbra.tng.de>
Message-ID: <14701697.210241249286257372.JavaMail.root@zimbra.tng.de>

Hallo,

ich habe also keine Chance mir eine Zahl vorlesen zu lassen, welche per Zufall generiere? :-/ Ich frage nur deshalb, da ich in einem Subordner diverse Soundfiles von Zahlen habe, welche ich von einem Debianpaket entpackt habe.

Gru? aus Kiel

Translation:

Hello,

there is no chance to read an random generated number from the freeswitch? I?m asking again because I have several german soundfiles of numbers which I extracted form a debianpacket. 

BR from Kiel

----- Urspr?ngliche Mail -----
Von: "Mathieu Rene" <mrene_lists at avgs.ca>
An: freeswitch-users at lists.freeswitch.org
Gesendet: Montag, 3. August 2009 03:11:11 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: Re: [Freeswitch-users] Language of the speech

Da gibst keine eigentliches DE audio-Datei, nur text fuer text-to- 
speech, deshalb funktionniert es nicht.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 2-Aug-09 um 8:18 AM schrieb Rudolf Denert:

> Hello,
>
> sorry but I don't understand your answer.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Michael Jerris" <mike at jerris.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Freitag, 31. Juli 2009 21:34:16 GMT +01:00 Amsterdam/ 
> Berlin/Bern/Rom/Stockholm/Wien
> Betreff: Re: [Freeswitch-users] Language of the speech
>
> we don;t have any german sound files?
>
> On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:
>
>> Hi again,
>>
>> does anybody know why my freeswitch doesn't play German speech-files?
>>
>> Here is my construct:
>>
>> I'm generating a random number in my lua script:
>> rand = math.random(11, 1000);
>>
>> The I want that the freeswitch says the "random number":
>> session:execute("say", "de name_spelled iterated " ..test_nummer);
>>
>> It works fine in English but I don't bring the freeswitch to say the
>> number in german. I installed the freeswitch-lang-
>> de_1.0.3-1_i386.deb, change the line in freeswitch.xml:
>> <section name="phrases" description="Speech Phrase Management">
>>   <macros>
>>     <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>>   </macros>
>> </section>
>>
>> Here is an extraction from the fs_cli:
>> 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152
>> switch_core_standard_on_execute() sofia/external/anonymous at anonymous.invalid
>> Execute set(default_language=de)
>> 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous at anonymous.invalid
>> SET [default_language]=[de]
>>
>> But I don't here any German speech only the English on
>>
>> Does anybody of you have an idea? Thanks a lot again.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -- 
> -- 
> Rudolf Denert, Technical Support
> TNG AG, NGN
> Projensdorfer Str. 324, D-24106 Kiel, Germany
> phone: +49 431 7097-10, fax: +49 431 7097-555
> mailto: rdenert at tng.de http://www.tng.de
> -
> Register: Amtsgericht Kiel, HRB 6596 KI
> Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
> Executive board (Vorstand): Dr. Sven Willert (Vorsitz),  Dr. Thomas
> Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg
> Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
> -
> This e-mail may contain confidential and/or privileged information. If
> you are not the intended recipient (or have received this e-mail in
> error) please notify the sender immediately and destroy this e-mail.  
> Any
> unauthorized copying, disclosure or distribution of the material in  
> this
> e-mail is strictly forbidden.
>
> Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
> Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
> E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
> Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
> die unbefugte Weitergabe dieser Mail ist nicht gestattet.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
-
Register: Amtsgericht Kiel, HRB 6596 KI
Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
Executive board (Vorstand): Dr. Sven Willert (Vorsitz),  Dr. Thomas
Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg
Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
-
This e-mail may contain confidential and/or privileged information. If
you are not the intended recipient (or have received this e-mail in
error) please notify the sender immediately and destroy this e-mail. Any
unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.


From merul at mac.com  Mon Aug  3 01:12:54 2009
From: merul at mac.com (Merul Patel)
Date: Mon, 03 Aug 2009 09:12:54 +0100
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <mailman.801.1249261906.2787.freeswitch-users@lists.freeswitch.org>
References: <mailman.801.1249261906.2787.freeswitch-users@lists.freeswitch.org>
Message-ID: <97532242-8290-47A9-900C-49E7567F4A4E@mac.com>

>> I'm new to FS, and experimenting with it on a constrained  
>> environment (PCEngines ALIX board running Voyage Linux 0.62).
>>
>> So far, FS has compiled fine, and I can register multiple  
>> softphones and make calls between them, but I'm lost at how to  
>> configure a Sangoma U100 so I can make and receive calls over an  
>> analogue line.
>>
>> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the  
>> wanrouter utility detects the USB device, and I've compiled it to  
>> support the TDM API.
>>
>> FS was compiled with the Openzap module - as best as I can tell.
>>
>> I thought that I would be able to use the wancfg_tdmapi utility to  
>> configure the /etc/wanpipe/wanpipe1.conf and then use the generated  
>> configuration file as the basis for configuring autoload_configs/ 
>> openzap.conf.xml.
>
> try wancfg_fs

Thanks for the suggestion Michael, but the same result occurs as when  
I try wancfg_tdmapi, ie:

"No Sangoma voice compatible cards found/configured"

>
>> However, the wancfg utility doesn't generate the wanpipe1.conf, and  
>> I'm stumped.
>>
>> Any pointers would be much appreciated.
>>



From darren at dmmhosting.co.uk  Mon Aug  3 02:12:24 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Mon, 3 Aug 2009 10:12:24 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <2d4a177f-88f4-44e9-94d9-dae6235f718f@dmmhosting.co.uk>

Brian, this ?broken? business explains a lot, I just assumed this was a normal practise. This ?outbound-proxy? parameter, I don?t see any reference to this anywhere.
 
This just causes Registration Failed with status DNS Error [503]
 
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 03 August 2009 00:11
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
set the proxy, register-proxy to bmnha-01.bt.com and the outbound-proxy to www.bbvservice-560129.bt.com, I regard this type of config B R O K EN and the provider shouldn't be doing this with DNS names that do not exist in my opinion... if they are doing this for security sake they should just setup VPN or direct access... this type of setup must makes my skin crawl.
 
/b
 
On Aug 2, 2009, at 5:31 AM, Darren Williams wrote:



?and/or outbound proxy?, I do not seem to be able to find this parameter.
 
It is probably my terminology which is confusing too and what exactly I want to do. All I know is that on my Thomson ST2030. If I enter:
 
Registrar Server Address: bmnha-01.bt.com
Proxy Server Address: bmnha-01.bt.com
Outbound Proxy Server:www.bbvservice-560129.bt.com
 
Everything works fine, I would just like to be able to transfer this config to freeswitch.
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

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From raffaele.p.guidi at gmail.com  Mon Aug  3 04:43:01 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Mon, 3 Aug 2009 13:43:01 +0200
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
Message-ID: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>

I found the answer by myself while I had finished writing the e-mail. The
correct call url is loopback/<extension> (in this case the command
is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
future reference (can't find any example of that anywhere). Is the project
wiki accesible for anyone to contribute or do I have to ask for an
authorization?

Regards,
   Raffaele

*********** ORIGINAL QUESTION *************

Hi, I'm trying to call an extension wich is not associated to a user from
the ESL (or the CLI as well) using the "ORIGINATE" command.
Now, while

originate user/1001 1000


works perfectly

with:

originate user/fakecall 1000


I have an error:

2009-08-03 13:21:19.406250 [ERR] switch_ivr_originate.c:1494 Cannot create
outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]


...this is not surprising ("fakecall" of course is not an user), but I
cannot figure out what is the correct CALL URL for this extension. Same
error is reported using sofia/internal/fakecall. It seems that the
EXECUTE_EXTENSION method could do for the magic (it works when issued from
an other extension in the dialplan) but it is not available from the CLI nor
the event socket (I'm using a binary version for windows - freeswitch.msi -
dated jul, 11th 2009).

PS: Calling "fakecall" from a registered phone (or from portaudio) works as
expected.
PPS: the configuration of fakecall in dialplan/default.xml
...
<context name="default">
   <extension name="fakecall">
<condition field="destination_number" expression="^fakecall$">
 <action application="lua" data="lua/helloworld.lua test $1"/>
</condition>
 </extension>
....
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From chad at apartmentlines.com  Mon Aug  3 03:46:10 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Mon, 3 Aug 2009 06:46:10 -0400
Subject: [Freeswitch-users] arriving today for ClueCon
Message-ID: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>

i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this  
afternoon?  anybody need help with setup?


From juanbackson at gmail.com  Mon Aug  3 04:45:21 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Mon, 3 Aug 2009 19:45:21 +0800
Subject: [Freeswitch-users] Question about dynamic registration
Message-ID: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>

Hi,

Other than curl, is there anyway to do dynamic registration?
It there anyway to embed a script in freeswitch to do the authorization?

Thanks,
Anne
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From brian at freeswitch.org  Mon Aug  3 05:40:48 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 07:40:48 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <2d4a177f-88f4-44e9-94d9-dae6235f718f@dmmhosting.co.uk>
References: <2d4a177f-88f4-44e9-94d9-dae6235f718f@dmmhosting.co.uk>
Message-ID: <184C69C8-EBEA-487D-A921-B35C478F7607@freeswitch.org>

You must be on SVN trunk.

/b

On Aug 3, 2009, at 4:12 AM, Darren Williams wrote:

> Brian, this ?broken? business explains a lot, I just assumed this  
> was a normal practise. This ?outbound-proxy? parameter, I don?t see  
> any reference to this anywhere.
>
> This just causes Registration Failed with status DNS Error [503]
>

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From brian at freeswitch.org  Mon Aug  3 05:41:41 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 07:41:41 -0500
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
Message-ID: <6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>

Just look for large groups of people with laptops.  I'm sure you can't  
miss us.

/b

On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:

> i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
> afternoon?  anybody need help with setup?



From brian at freeswitch.org  Mon Aug  3 05:42:33 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 07:42:33 -0500
Subject: [Freeswitch-users] Question about dynamic registration
In-Reply-To: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>
References: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>
Message-ID: <9FE0F6B8-C4BF-4820-8CC2-6825C5EE8422@freeswitch.org>

You could build your own module to do it how ever you please.  But  
forking a script every time to auth is not very scalable.

/b

On Aug 3, 2009, at 6:45 AM, Juan Backson wrote:

> Hi,
>
> Other than curl, is there anyway to do dynamic registration?
> It there anyway to embed a script in freeswitch to do the  
> authorization?
>
> Thanks,
> Anne
> ______



From asannucci at gmail.com  Mon Aug  3 05:54:49 2009
From: asannucci at gmail.com (bakko)
Date: Mon, 3 Aug 2009 14:54:49 +0200
Subject: [Freeswitch-users] Problem in spidermonkey_odbc
In-Reply-To: <7aa29e790908030007l1478b8b1mfa460a556349342f@mail.gmail.com>
References: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
	<7aa29e790908030007l1478b8b1mfa460a556349342f@mail.gmail.com>
Message-ID: <CC687DA31ED44F2B8541E3878F54CF7F@voztovoice>

If your linux distribution is Centos you have to install  unixODBC-devel postgresql-odbc unixODBC then compile freeswitch.

Look at this wiki page:

http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc

BR
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From msc at freeswitch.org  Mon Aug  3 05:59:34 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Aug 2009 05:59:34 -0700
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
In-Reply-To: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
References: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
Message-ID: <87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>

On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> I found the answer by myself while I had finished writing the e-mail. The
> correct call url is loopback/<extension> (in this case the command
> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
> future reference (can't find any example of that anywhere). Is the project
> wiki accesible for anyone to contribute or do I have to ask for an
> authorization?
>

All you need to do is sign up for a free account on the wiki and you can
start editing. It's a community resource and all FS users are invited to add
their respective knowledge.

As for not finding what you were looking for, does this page not have it?
http://wiki.freeswitch.org/wiki/Loopback

If not then please feel free to add to this page whatever your specific
scenario entails and give some examples.

-MC
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From a.afzali2003 at gmail.com  Mon Aug  3 06:38:03 2009
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 3 Aug 2009 17:08:03 +0330
Subject: [Freeswitch-users] Missing mod_curl
Message-ID: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>

Hi,

I'll appreciate if somebody tell me where has gone the mod_curl ? I just
need to use it for http method calls.

Regards,
-- afshin
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From jgonzalez at sqli.com  Mon Aug  3 06:41:09 2009
From: jgonzalez at sqli.com (julien)
Date: Mon, 03 Aug 2009 15:41:09 +0200
Subject: [Freeswitch-users] Authentication problem when calling softphones
	from ipphones
Message-ID: <4A76E8F5.9070804@sqli.com>

Hello everyone,
I'm using a SIP trunk to link my PBX and FS.
My problem is when I try to call a softphone on FS from my ipphone, I've 
the following error on FS during Authentication :

2009-08-03 14:58:27.123817 [DEBUG] sofia.c:4554 IP [PBX IP@] Rejected by 
acl "domains". Falling back to Digest auth.
2009-08-03 14:58:27.235492 [DEBUG] sofia.c:4554 IP [PBX IP@] Rejected by 
acl "domains". Falling back to Digest auth.
2009-08-03 14:58:27.235492 [WARNING] sofia_reg.c:1755 Can't find user 
[@[FS IP@]]
You must define a domain called '[FS IP@]' in your directory and add a 
user with the id="" attribute
and you must configure your device to use the proper domain in it's 
authentication credentials.

I defined my gateway to the PBX this way :

<include>
  <gateway name="pbxlyon">
  <param name="username" value="pbxlyon"/>
  <param name="realm" value="[PBX IP@]"/>
  <param name="password" value="pbxlyon"/>
  <param name="register" value="false"/>
  <param name="register-transport" value="udp"/>
  <param name="retry_seconds" value="300"/>
  </gateway>
</include>

I don't want the PBX to try to authenticate because I can't define a 
username nor a password for the authentication.
Thank for you time.
Best regards,
Julien GONZALEZ.







From mike at jerris.com  Mon Aug  3 06:59:35 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Aug 2009 08:59:35 -0500
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <97532242-8290-47A9-900C-49E7567F4A4E@mac.com>
References: <mailman.801.1249261906.2787.freeswitch-users@lists.freeswitch.org>
	<97532242-8290-47A9-900C-49E7567F4A4E@mac.com>
Message-ID: <FCA3464E-2A85-40F3-93AA-F8154470B399@jerris.com>

What is the output of wantouter hwprobe?

On Aug 3, 2009, at 3:12 AM, Merul Patel <merul at mac.com> wrote:

>>> I'm new to FS, and experimenting with it on a constrained
>>> environment (PCEngines ALIX board running Voyage Linux 0.62).
>>>
>>> So far, FS has compiled fine, and I can register multiple
>>> softphones and make calls between them, but I'm lost at how to
>>> configure a Sangoma U100 so I can make and receive calls over an
>>> analogue line.
>>>
>>> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the
>>> wanrouter utility detects the USB device, and I've compiled it to
>>> support the TDM API.
>>>
>>> FS was compiled with the Openzap module - as best as I can tell.
>>>
>>> I thought that I would be able to use the wancfg_tdmapi utility to
>>> configure the /etc/wanpipe/wanpipe1.conf and then use the generated
>>> configuration file as the basis for configuring autoload_configs/
>>> openzap.conf.xml.
>>
>> try wancfg_fs
>
> Thanks for the suggestion Michael, but the same result occurs as when
> I try wancfg_tdmapi, ie:
>
> "No Sangoma voice compatible cards found/configured"
>
>>
>>> However, the wancfg utility doesn't generate the wanpipe1.conf, and
>>> I'm stumped.
>>>
>>> Any pointers would be much appreciated.
>>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From mike at jerris.com  Mon Aug  3 07:03:08 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Aug 2009 09:03:08 -0500
Subject: [Freeswitch-users] Missing mod_curl
In-Reply-To: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
References: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
Message-ID: <396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>

It is still there.

On Aug 3, 2009, at 8:38 AM, afshin afzali <a.afzali2003 at gmail.com>  
wrote:

> Hi,
>
> I'll appreciate if somebody tell me where has gone the mod_curl ? I  
> just need to use it for http method calls.
>
> Regards,
> -- afshin
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From hoaianh at gmx.de  Mon Aug  3 07:30:44 2009
From: hoaianh at gmx.de (Ngo-Vi Hoai-Anh)
Date: Mon, 03 Aug 2009 16:30:44 +0200
Subject: [Freeswitch-users] telnet to event socket
Message-ID: <4A76F494.5020101@gmx.de>

Hi,

I'm taking a close look at event socket on FS 1.0.3. Configuration is 
the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and 
fsconsole.pl work but I was not able to telnet to port 8021. As I've 
done that I received somewhat like:
#>auth/request

I typed in: auth ClueCon

After some seconds I've got the message 'connection close by foreign host'

Any ideas?

Thank you
Hoai-Anh


From vkozak at abisoft.spb.ru  Mon Aug  3 07:37:43 2009
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Mon, 3 Aug 2009 18:37:43 +0400
Subject: [Freeswitch-users] Fw: FreeSwitch doesn't play music on hold
	forbriged channel
Message-ID: <C6AE09778C1441F6842140EFFC55DF20@abisoft.biz>

No. I didn't. Moreover, I tried to set it explicitly using api uuid_setvar.
  ----- Original Message ----- 
  From: Kozak Vladimir 
  To: ?????? ?????? 
  Sent: Monday, August 03, 2009 5:28 PM
  Subject: Fw: [Freeswitch-users] FreeSwitch doesn't play music on hold forbriged channel



  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, July 31, 2009 5:22 PM
  Subject: Re: [Freeswitch-users] FreeSwitch doesn't play music on hold forbriged channel


  I don't see the variable hold_music ... did you remove it? 


  /b


  On Jul 31, 2009, at 5:24 AM, Kozak Vladimir wrote:



    The scenario is the following:
    FS User A dial an extension
    Extention opens outbound socket channel to my application
    My application bridges the call to FS User B
    The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id
    The application sends hold to the bridged channel using SendMsg with Other-leg-unique-id
    User B is placed on hold but no music on hold is played to the caller (User A)


    I have outbound socket channel and the following sequence of commands/event:
    listening on [any] 8084 ...
    connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250] 34000
    connect





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From rdenert at tng.de  Mon Aug  3 07:49:19 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Mon, 3 Aug 2009 16:49:19 +0200 (CEST)
Subject: [Freeswitch-users] telnet to event socket
In-Reply-To: <4A76F494.5020101@gmx.de>
Message-ID: <2076715.215161249310959264.JavaMail.root@zimbra.tng.de>

Hi,

you only have to write "auth <password>" and hit enter twice. The default password is something like ClueCon.

BR

----- Urspr?ngliche Mail -----
Von: "Ngo-Vi Hoai-Anh" <hoaianh at gmx.de>
An: freeswitch-users at lists.freeswitch.org
Gesendet: Montag, 3. August 2009 16:30:44 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: [Freeswitch-users] telnet to event socket

Hi,

I'm taking a close look at event socket on FS 1.0.3. Configuration is 
the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and 
fsconsole.pl work but I was not able to telnet to port 8021. As I've 
done that I received somewhat like:
#>auth/request

I typed in: auth ClueCon

After some seconds I've got the message 'connection close by foreign host'

Any ideas?

Thank you
Hoai-Anh

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
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From dujinfang at gmail.com  Mon Aug  3 07:51:06 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 3 Aug 2009 22:51:06 +0800
Subject: [Freeswitch-users] telnet to event socket
In-Reply-To: <4A76F494.5020101@gmx.de>
References: <4A76F494.5020101@gmx.de>
Message-ID: <645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>


On Aug 3, 2009, at 10:30 PM, Ngo-Vi Hoai-Anh wrote:
> Hi,
>
> I'm taking a close look at event socket on FS 1.0.3. Configuration is
> the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and
> fsconsole.pl work but I was not able to telnet to port 8021. As I've
> done that I received somewhat like:
> #>auth/request
>
> I typed in: auth ClueCon
>
followed by two Enters (\n\n).

> After some seconds I've got the message 'connection close by foreign  
> host'
>
> Any ideas?
>
> Thank you
> Hoai-Anh
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From patj at linklocal.net  Mon Aug  3 08:31:11 2009
From: patj at linklocal.net (Pat Jensen)
Date: Mon, 3 Aug 2009 08:31:11 -0700
Subject: [Freeswitch-users] Authentication problem when calling
 softphones	from ipphones
In-Reply-To: <4A76E8F5.9070804@sqli.com>
References: <4A76E8F5.9070804@sqli.com>
Message-ID: <0F432CFDE6E44442BB35719DF1034F6D5F011ACFB7@ws2008.linklocal.net>

Julien,

Place an ACL entry with the IP address for your PBX in the following file:

/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml

 <list name="domains" default="deny">
<node type="allow" cidr="123.12.12.1/32"/>   <!-- your pbx ip -->
</list>

This should allow unauthenticated invites from your PBX to hit FS.  Hope this helps.

Pat
________________________________________
From: julien [jgonzalez at sqli.com]
Sent: Monday, August 03, 2009 6:41 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Authentication problem when calling softphones      from ipphones

Hello everyone,
I'm using a SIP trunk to link my PBX and FS.
My problem is when I try to call a softphone on FS from my ipphone, I've
the following error on FS during Authentication :

2009-08-03 14:58:27.123817 [DEBUG] sofia.c:4554 IP [PBX IP@] Rejected by
acl "domains". Falling back to Digest auth.
2009-08-03 14:58:27.235492 [DEBUG] sofia.c:4554 IP [PBX IP@] Rejected by
acl "domains". Falling back to Digest auth.
2009-08-03 14:58:27.235492 [WARNING] sofia_reg.c:1755 Can't find user
[@[FS IP@]]
You must define a domain called '[FS IP@]' in your directory and add a
user with the id="" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.

I defined my gateway to the PBX this way :

<include>
  <gateway name="pbxlyon">
  <param name="username" value="pbxlyon"/>
  <param name="realm" value="[PBX IP@]"/>
  <param name="password" value="pbxlyon"/>
  <param name="register" value="false"/>
  <param name="register-transport" value="udp"/>
  <param name="retry_seconds" value="300"/>
  </gateway>
</include>

I don't want the PBX to try to authenticate because I can't define a
username nor a password for the authentication.
Thank for you time.
Best regards,
Julien GONZALEZ.









From a.afzali2003 at gmail.com  Mon Aug  3 09:01:34 2009
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 3 Aug 2009 20:31:34 +0430
Subject: [Freeswitch-users] Missing mod_curl
In-Reply-To: <396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>
References: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
	<396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>
Message-ID: <debff5a80908030901w4f88767aw4f5e39d34adcaf9a@mail.gmail.com>

I've gotten 1.0.4pre9 , but i can not see it :(
-- afshin


On Mon, Aug 3, 2009 at 6:33 PM, Michael Jerris <mike at jerris.com> wrote:

> It is still there.
>
> On Aug 3, 2009, at 8:38 AM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
>
> > Hi,
> >
> > I'll appreciate if somebody tell me where has gone the mod_curl ? I
> > just need to use it for http method calls.
> >
> > Regards,
> > -- afshin
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Mon Aug  3 09:12:17 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 4 Aug 2009 00:12:17 +0800
Subject: [Freeswitch-users] Missing mod_curl
In-Reply-To: <debff5a80908030901w4f88767aw4f5e39d34adcaf9a@mail.gmail.com>
References: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
	<396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>
	<debff5a80908030901w4f88767aw4f5e39d34adcaf9a@mail.gmail.com>
Message-ID: <A948E259-A60C-484C-9113-E79A5F0E4C64@gmail.com>


On Aug 4, 2009, at 12:01 AM, afshin afzali wrote:
> I've gotten 1.0.4pre9 , but i can not see it :(
> -- afshin
>

In trunk.

>
> On Mon, Aug 3, 2009 at 6:33 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> It is still there.
>
> On Aug 3, 2009, at 8:38 AM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
>
> > Hi,
> >
> > I'll appreciate if somebody tell me where has gone the mod_curl ? I
> > just need to use it for http method calls.
> >
> > Regards,
> > -- afshin
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From a.afzali2003 at gmail.com  Mon Aug  3 09:19:17 2009
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 3 Aug 2009 20:49:17 +0430
Subject: [Freeswitch-users] Missing mod_curl
In-Reply-To: <A948E259-A60C-484C-9113-E79A5F0E4C64@gmail.com>
References: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
	<396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>
	<debff5a80908030901w4f88767aw4f5e39d34adcaf9a@mail.gmail.com>
	<A948E259-A60C-484C-9113-E79A5F0E4C64@gmail.com>
Message-ID: <debff5a80908030919k7f182f3aifef607846dc9fb88@mail.gmail.com>

Thanks

On Mon, Aug 3, 2009 at 8:42 PM, Seven Du <dujinfang at gmail.com> wrote:

>
> On Aug 4, 2009, at 12:01 AM, afshin afzali wrote:
> > I've gotten 1.0.4pre9 , but i can not see it :(
> > -- afshin
> >
>
> In trunk.
>
> >
> > On Mon, Aug 3, 2009 at 6:33 PM, Michael Jerris <mike at jerris.com>
> > wrote:
> > It is still there.
> >
> > On Aug 3, 2009, at 8:38 AM, afshin afzali <a.afzali2003 at gmail.com>
> > wrote:
> >
> > > Hi,
> > >
> > > I'll appreciate if somebody tell me where has gone the mod_curl ? I
> > > just need to use it for http method calls.
> > >
> > > Regards,
> > > -- afshin
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From darren at dmmhosting.co.uk  Mon Aug  3 10:25:37 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Mon, 3 Aug 2009 18:25:37 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <718419f1-d02d-4623-bd5a-7852608df505@dmmhosting.co.uk>

Definitely the same reply:
 
FreeSWITCH Version 1.0.trunk (14457)
 
[ERR] sofia_reg.c:1460 05061292117 Registration Failed with status DNS Error [503]. failure #1 and so on.
 
The complete config is:
 
<include>
  <gateway name="05061292117">
  <param name="username" value="05061292117"/>
  <param name="realm" value="bmnha-01.bt.com"/>
  <param name="from-user" value="05061292117"/>
  <param name="from-domain" value="bmnha-01.bt.com"/>
  <param name="password" value="XXXXXXXXXXXXXXX"/>
  <param name="proxy" value="bmnha-01.bt.com"/>
  <param name="register-proxy" value="bmnha-01.bt.com"/>
  <param name="outbound-proxy" value="www.bbvservice-560129.bt.com"/>
  <param name="register" value="true"/>
  <param name="register-transport" value="udp"/>
  <param name="retry-seconds" value="10"/>
  </gateway>
</include>
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 03 August 2009 13:41
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
You must be on SVN trunk.
 
/b
 
On Aug 3, 2009, at 4:12 AM, Darren Williams wrote:



Brian, this ?broken? business explains a lot, I just assumed this was a normal practise. This ?outbound-proxy? parameter, I don?t see any reference to this anywhere.
 
This just causes Registration Failed with status DNS Error [503]
 
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From merul at mac.com  Mon Aug  3 10:56:43 2009
From: merul at mac.com (Merul Patel)
Date: Mon, 03 Aug 2009 18:56:43 +0100
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <mailman.816.1249310297.2787.freeswitch-users@lists.freeswitch.org>
References: <mailman.816.1249310297.2787.freeswitch-users@lists.freeswitch.org>
Message-ID: <EE4A46F0-42E4-4305-A2DC-EFC4F7401DE1@mac.com>

> What is the output of wantouter hwprobe?

voyage:~# wanrouter hwprobe

-------------------------------
| Wanpipe Hardware Probe Info |
-------------------------------
1 . U100       : BUSID=1-1 : V=00

Card Cnt: U100=1

voyage:~# dahdi_hardware
usb:001/002          wanpipe-     10c4:8461 Sangoma WANPIPE USB-FXO  
Device


>
> On Aug 3, 2009, at 3:12 AM, Merul Patel <merul at mac.com> wrote:
>
>>>> I'm new to FS, and experimenting with it on a constrained
>>>> environment (PCEngines ALIX board running Voyage Linux 0.62).
>>>>
>>>> So far, FS has compiled fine, and I can register multiple
>>>> softphones and make calls between them, but I'm lost at how to
>>>> configure a Sangoma U100 so I can make and receive calls over an
>>>> analogue line.
>>>>
>>>> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the
>>>> wanrouter utility detects the USB device, and I've compiled it to
>>>> support the TDM API.
>>>>
>>>> FS was compiled with the Openzap module - as best as I can tell.
>>>>
>>>> I thought that I would be able to use the wancfg_tdmapi utility to
>>>> configure the /etc/wanpipe/wanpipe1.conf and then use the generated
>>>> configuration file as the basis for configuring autoload_configs/
>>>> openzap.conf.xml.
>>>
>>> try wancfg_fs
>>
>> Thanks for the suggestion Michael, but the same result occurs as when
>> I try wancfg_tdmapi, ie:
>>
>> "No Sangoma voice compatible cards found/configured"
>>
>>>
>>>> However, the wancfg utility doesn't generate the wanpipe1.conf, and
>>>> I'm stumped.
>>>>
>>>> Any pointers would be much appreciated.
>>>>



From dave at 3c.co.uk  Mon Aug  3 10:57:14 2009
From: dave at 3c.co.uk (David Knell)
Date: Mon, 3 Aug 2009 18:57:14 +0100
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
Message-ID: <01A26C17-A286-4A71-8B9C-8E558E963501@3c.co.uk>



-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



On 3 Aug 2009, at 13:41, Brian West <brian at freeswitch.org> wrote:

> Just look for large groups of people with laptops.  I'm sure you can't
> miss us.
>
> /b
>
> On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:
>
>> i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
>> afternoon?  anybody need help with setup?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mike at jerris.com  Mon Aug  3 11:14:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Aug 2009 13:14:27 -0500
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <EE4A46F0-42E4-4305-A2DC-EFC4F7401DE1@mac.com>
References: <mailman.816.1249310297.2787.freeswitch-users@lists.freeswitch.org>
	<EE4A46F0-42E4-4305-A2DC-EFC4F7401DE1@mac.com>
Message-ID: <019E4A58-9EC6-4198-A041-DFE2C1219E14@jerris.com>

I have yet to configure one of these cards for freeswitch so it's  
possible it's not in the config util yet.  I suggest contacting  
sangoma support for assistance

On Aug 3, 2009, at 12:56 PM, Merul Patel <merul at mac.com> wrote:

>> What is the output of wantouter hwprobe?
>
> voyage:~# wanrouter hwprobe
>
> -------------------------------
> | Wanpipe Hardware Probe Info |
> -------------------------------
> 1 . U100       : BUSID=1-1 : V=00
>
> Card Cnt: U100=1
>
> voyage:~# dahdi_hardware
> usb:001/002          wanpipe-     10c4:8461 Sangoma WANPIPE USB-FXO
> Device
>
>
>>
>> On Aug 3, 2009, at 3:12 AM, Merul Patel <merul at mac.com> wrote:
>>
>>>>> I'm new to FS, and experimenting with it on a constrained
>>>>> environment (PCEngines ALIX board running Voyage Linux 0.62).
>>>>>
>>>>> So far, FS has compiled fine, and I can register multiple
>>>>> softphones and make calls between them, but I'm lost at how to
>>>>> configure a Sangoma U100 so I can make and receive calls over an
>>>>> analogue line.
>>>>>
>>>>> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the
>>>>> wanrouter utility detects the USB device, and I've compiled it to
>>>>> support the TDM API.
>>>>>
>>>>> FS was compiled with the Openzap module - as best as I can tell.
>>>>>
>>>>> I thought that I would be able to use the wancfg_tdmapi utility to
>>>>> configure the /etc/wanpipe/wanpipe1.conf and then use the  
>>>>> generated
>>>>> configuration file as the basis for configuring autoload_configs/
>>>>> openzap.conf.xml.
>>>>
>>>> try wancfg_fs
>>>
>>> Thanks for the suggestion Michael, but the same result occurs as  
>>> when
>>> I try wancfg_tdmapi, ie:
>>>
>>> "No Sangoma voice compatible cards found/configured"
>>>
>>>>
>>>>> However, the wancfg utility doesn't generate the wanpipe1.conf,  
>>>>> and
>>>>> I'm stumped.
>>>>>
>>>>> Any pointers would be much appreciated.
>>>>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From msc at freeswitch.org  Mon Aug  3 11:13:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Aug 2009 11:13:51 -0700
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <01A26C17-A286-4A71-8B9C-8E558E963501@3c.co.uk>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
	<01A26C17-A286-4A71-8B9C-8E558E963501@3c.co.uk>
Message-ID: <87f2f3b90908031113u441f1091h63289b79f12ddc84@mail.gmail.com>

Excellent! Many of us are at the hotel already, getting everything set up.
-MC

On Mon, Aug 3, 2009 at 10:57 AM, David Knell <dave at 3c.co.uk> wrote:

>
>
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
>
> On 3 Aug 2009, at 13:41, Brian West <brian at freeswitch.org> wrote:
>
> > Just look for large groups of people with laptops.  I'm sure you can't
> > miss us.
> >
> > /b
> >
> > On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:
> >
> >> i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
> >> afternoon?  anybody need help with setup?
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Aug  3 11:16:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Aug 2009 11:16:33 -0700
Subject: [Freeswitch-users] telnet to event socket
In-Reply-To: <645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>
References: <4A76F494.5020101@gmx.de>
	<645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>
Message-ID: <87f2f3b90908031116i2ab7d7b5o38fda7ce8d9b36b5@mail.gmail.com>

On Mon, Aug 3, 2009 at 7:51 AM, Seven Du <dujinfang at gmail.com> wrote:

>
> On Aug 3, 2009, at 10:30 PM, Ngo-Vi Hoai-Anh wrote:
> > Hi,
> >
> > I'm taking a close look at event socket on FS 1.0.3. Configuration is
> > the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and
> > fsconsole.pl work but I was not able to telnet to port 8021. As I've
> > done that I received somewhat like:
> > #>auth/request
> >
> > I typed in: auth ClueCon
> >
> followed by two Enters (\n\n).
>
> > After some seconds I've got the message 'connection close by foreign
> > host'
> >
>

Don't forget that you can change the password by modifying the value in
freeswitch/conf/autoload_configs/event_socket.conf.xml"

<param name="password" value="ClueCon"/>

-MC
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From testa at voicetechnology.com.br  Mon Aug  3 11:17:27 2009
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Mon, 3 Aug 2009 15:17:27 -0300
Subject: [Freeswitch-users] FS beats Aculab Prosody S on subjective test
	on lay users for conference quality
In-Reply-To: <4A71AA86.90707@coppice.org>
References: <9cb0e15e0907290753q6ec3d45x680f5020a07699ed@mail.gmail.com>
	<1248886530.3818.2.camel@dk-d820> <4A70F593.5080204@coppice.org>
	<1248958990.4428.15.camel@dk-d820> <4A71AA86.90707@coppice.org>
Message-ID: <9cb0e15e0908031117p79355d51qf49821277d009f7d@mail.gmail.com>

Hi, a bit late on answering some of the questions, but, here we go.On
Aculab, all codecs were G711u. The same codec we have on FS:

freeswitch at conference> show channels
API CALL [show(channels)] output:
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure
c9cafeb8-803a-11de-8ceb-cb8648fd1ccf,inbound,2009-08-03
11:34:44,1249310084,sofia/internal/1000 at 192.168.0.40,CS_EXECUTE,Teste
Testa,1000,192.168.0.165,3200,conference,3200-192.168.0.40 at ultrawideband
,XML,default,L16,8000,PCMU,8000,
cc5edb90-803a-11de-8ceb-cb8648fd1ccf,inbound,2009-08-03
11:34:49,1249310089,sofia/internal/1000 at 192.168.0.40
,CS_EXECUTE,F.G.Testa,1000,192.168.0.249,3200,conference,3200-192.168.0.40 at ultrawideband
,XML,default,L16,8000,PCMU,8000,

2 total.

freeswitch at conference> conference list
API CALL [conference(list)] output:
Conference 3200-192.168.0.40 (2 members)
2;sofia/internal/1000 at 192.168.0.40
;cc5edb90-803a-11de-8ceb-cb8648fd1ccf;F.G.Testa;1000;hear|speak;0;0;300
1;sofia/internal/1000 at 192.168.0.40;c9cafeb8-803a-11de-8ceb-cb8648fd1ccf;Teste
Testa;1000;hear|speak|talking|floor;0;0;300

I think this answers some questions from Michael.
A packet dump I don't have right now.

Fernando G. Testa


On Thu, Jul 30, 2009 at 11:13 AM, Steve Underwood <steveu at coppice.org>wrote:

> David Knell wrote:
> > On Thu, 2009-07-30 at 09:21 +0800, Steve Underwood wrote:
> >
> >>
> >> High quality conferencing is a difficult task, and still a research
> >> topic. No two conferencing systems perform alike. The interesting thing
> >> about this and other reports is that the conferencing in Freeswitch is
> >> not very clever right now, yet people are already saying it beats
> >> various other offerings, including long time commercial offerings.
> >>
> >
> > It may well be that a simplistic implementation (noise gate, add them
> > all up) is all that's required for dealing with small groups or, more
> > generally, groups of any size which have a small number of active
> > speakers at any one time: it's predictable and unlikely to introduce
> > unpleasant side effects.
> >
> This is one of those situations where when you've experienced something
> better you make that your baseline for acceptability. I would consider a
> noise gate horribly crude, and VAD as the minimum for acceptable
> performance. If you've only used a noise gate you get used to it. If
> you're not sufficiently versed in the art you may well think nothing
> better is even possible.
>
> The fact that even the simple scheme, with noise gating, in Freeswitch
> is getting high praise, is pretty damning of mature commercial products.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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From pgrondin at ip5.com  Mon Aug  3 11:53:11 2009
From: pgrondin at ip5.com (Patrick Grondin)
Date: Mon, 3 Aug 2009 14:53:11 -0400
Subject: [Freeswitch-users] Using tone_detect application
In-Reply-To: <6C525B211BA8F44CAC96A4F68333A9D119326E0813@VMBX107.ihostexchange.net>
References: <6C525B211BA8F44CAC96A4F68333A9D119326E0813@VMBX107.ihostexchange.net>
Message-ID: <6C525B211BA8F44CAC96A4F68333A9D119326E10A6@VMBX107.ihostexchange.net>

Hi,

What's the correct way to detect a number changed SIT tone in Freeswitch ?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Patrick Grondin
Sent: July-28-09 3:44 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Using tone_detect application

Hi,

I'm doing some tests between 2 FS to understand how the tone_detect application works.

I'm trying to detect a SIT tone, but I can't seem to detect the 3 tones. I only get the first activated tone.

If I have all tones activated - - - > I detect only the first tone of my wav file.
If I have 2nd segment tones and up activated - - - > I detect only the second tone of my wav file.
If I have the 3rd segment tones activated - - - > I detect only the third tone of my wav file.

I see that tone_detect can detect all 3 tones, but never at the same time. Does anyone have an idea of what I could be doing wrong ? Thanks !

I'm using FreeSWITCH Version 1.0.trunk (14397).

My dialplan looks like this :

    <extension name="incoming">
      <condition field="destination_number" expression="^(514[0-9][0-9][0-9][0-9][0-9][0-9][0-9])$">
        <action application="answer"/>
        <action application="tone_detect" data="SIT1LO 913.8 w +25000 set SIT1LO=true"/>   <!--1st segment low -->
        <action application="tone_detect" data="SIT1HI 985.2 w +25000 set SIT1HI=true"/>   <!--1st segment high -->
        <action application="tone_detect" data="SIT2LO 1370.6 w +25000 set SIT2LO=true"/>   <!--2nd segment low -->
        <action application="tone_detect" data="SIT2HI 1428.5 w +25000 set SIT2HI=true"/>   <!--2nd segment high -->
        <action application="tone_detect" data="SIT3LO 1776.7 w +25000 set SIT3LO=true"/>  <!--3rd segment low -->
        <action application="tone_detect" data="SITITU1 950 w +25000 set SITITU1=true"/>   <!--2nd segment low -->
        <action application="tone_detect" data="SITITU2 1400 w +25000 set SITITU2=true"/>   <!--2nd segment high -->
        <action application="tone_detect" data="SITITU3 1800 w +25000 set SITITU3=true"/>  <!--3rd segment low -->
        <action application="bridge" data="sofia/internal/$1 at 192.168.100.9"/>
      </condition>
    </extension>
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From raffaele.p.guidi at gmail.com  Mon Aug  3 12:13:14 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Mon, 3 Aug 2009 21:13:14 +0200
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
In-Reply-To: <87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>
References: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
	<87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>
Message-ID: <b0a8792f0908031213r37c10f80m9d076f195b1ba83f@mail.gmail.com>

Yeah, it's there but, as you can see it from the google queries below it's
not easy to find that page - unless you know what exactly you are looking
for and search for "loopback". A simple example (i.e. in Sofia Syntax) would
fill the gap (I'll be happy to do it ASAP)

http://www.google.com/search?hl=en&safe=off&q=freeswitch+call+url&aq=f&oq=&aqi=

http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=<http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=>
http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+esl&aq=f&oq=&aqi=

Also, this email now shows up first in googling for "non user extension
freeswitch" (which were the keywords I was looking for). I think this has
been useful! ;)

Many thanks and regards,
   Raffaele

On Mon, Aug 3, 2009 at 14:59, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> I found the answer by myself while I had finished writing the e-mail. The
>> correct call url is loopback/<extension> (in this case the command
>> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
>> future reference (can't find any example of that anywhere). Is the project
>> wiki accesible for anyone to contribute or do I have to ask for an
>> authorization?
>>
>
> All you need to do is sign up for a free account on the wiki and you can
> start editing. It's a community resource and all FS users are invited to add
> their respective knowledge.
>
> As for not finding what you were looking for, does this page not have it?
> http://wiki.freeswitch.org/wiki/Loopback
>
> If not then please feel free to add to this page whatever your specific
> scenario entails and give some examples.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Prometheus001 at gmx.net  Mon Aug  3 12:40:36 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 03 Aug 2009 21:40:36 +0200
Subject: [Freeswitch-users] TDM API: CMD: 18 : Operation not supported
Message-ID: <4A773D34.300@gmx.net>

Hello,

I setup libpri and a sangoma card A108DE, but I cannot dial out.
At startup I receive on the D channel
    TDM API: CMD: 18 : Operation not supported

When dialling Libpri debug shows that the numbering plan is fine and
that it accepts the screened number, but then it finally hangs up with:
       Ext: 1  Cause: Info. element nonexist or not implemented (99),
class = Protocol Error (e.g. unknown message) (6) ]

Does lipri sent any incorrect message here? Protocol is EuroISDN
(Q.931/Q.921).
Anybody has discovered this already?

I am on trunk 14419.

See debug and configs below.

Best regards
Peter

Starting FS:
2009-08-03 21:37:18.264829 [DEBUG] zap_io.c:2281 span 1 [d-channel]=[1:16]
TDM API: CMD: 18
: Operation not supported
2009-08-03 21:37:18.264915 [INFO] ozmod_wanpipe.c:287 configuring device
s1c16 as OpenZAP device 1:16 fd:55 DTMF: none
2009-08-03 21:37:18.264929 [DEBUG] zap_io.c:2281 span 1
[b-channel]=[1:17-31]

Dialling:
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 Handling message
for SAPI/TEI=0/0
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- ACKing all
packets from 3 to (but not including) 4
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Since there was
nothing left, stopping T200 counter
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Stopping T203
counter since we got an ACK
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Nothing left,
starting T203 counter
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Protocol
Discriminator: Q.931 (8)  len=14
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Call Ref: len= 2
(reference 5/0x5) (Terminator)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Message type:
STATUS (125)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < [08 04 82 e3 98 74]
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Cause (len= 6) [
Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Public
network serving the local user (2)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 <                 
Ext: 1  Cause: Info. element nonexist or not implemented (99), class =
Protocol Error (e.g. unknown message) (6) ]
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 <             
Cause data 1: 98 (152, Non-Locking Shift To Codeset 0 IE)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 <             
Cause data 2: 74 (116, Redirecting Number IE)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < [14 01 01]
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Call State (len=
3) [ Ext: 0  Coding: CCITT (ITU) standard (0)  Call state: Call
Initiated (1)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Processing IE 8
(cs0, Cause)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Processing IE
20 (cs0, Call State)
2009-08-03 21:16:19.441416 [ERR] ozmod_libpri.c:88 Received unsolicited
status: Info. element nonexist or not implemented

openzap.conf
[span wanpipe PRI_1]
number => 1
trunk_type => E1
b-channel => 1:1-15
d-channel => 1:16
b-channel => 1:17-31

openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
   <settings>
     <param name="debug" value="10"/>
   </settings>
   <libpri_spans>
     <span name="PRI_1">
        <param name="node" value="cpe"/>
        <param name="dp" value="national"/>
        <param name="l1" value="alaw"/>
        <param name="switch" value="euroisdn"/>
        <param name="dialplan" value="XML"/>
        <param name="context" value="public"/>
      </span>
   </libpri_spans>




From brian at freeswitch.org  Mon Aug  3 14:24:11 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 16:24:11 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <718419f1-d02d-4623-bd5a-7852608df505@dmmhosting.co.uk>
References: <718419f1-d02d-4623-bd5a-7852608df505@dmmhosting.co.uk>
Message-ID: <C925BB2D-18A5-4741-95A3-F82D9B1F546C@freeswitch.org>

which of the hostnames is fake?

/b

On Aug 3, 2009, at 12:25 PM, Darren Williams wrote:

> Definitely the same reply:
>

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From darren at dmmhosting.co.uk  Mon Aug  3 14:39:00 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Mon, 3 Aug 2009 22:39:00 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <c857d204-9c47-4185-ba83-56737ac47462@dmmhosting.co.uk>

The bmnha-01.bt.com is the one that doesn?t resolve
www.bbvservice-560129.bt.com resolves to 62.239.15.140
 
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 03 August 2009 22:24
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
which of the hostnames is fake?
 
/b
 
On Aug 3, 2009, at 12:25 PM, Darren Williams wrote:



Definitely the same reply:
 
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From dule.maillist at gmail.com  Mon Aug  3 15:06:46 2009
From: dule.maillist at gmail.com (Dan Le)
Date: Mon, 3 Aug 2009 18:06:46 -0400
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
In-Reply-To: <b0a8792f0908031213r37c10f80m9d076f195b1ba83f@mail.gmail.com>
References: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
	<87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>
	<b0a8792f0908031213r37c10f80m9d076f195b1ba83f@mail.gmail.com>
Message-ID: <914fc92a0908031506t285d2719need5f8937c0bb2a6@mail.gmail.com>

Technically, loopback is not meant to specifically allow you to call
non-user extensions, it simply allows you to hit the dialplan.
For example,

originate sofia/gw/gwname/fakecall 1000

(where gwname is the gateway routing your calls out, and 'fakecall' the
non-user extension)

will also allow you to dial to non-user extensions. This information would
be on any of the wiki pages detailing the originate command.

Dan


On Mon, Aug 3, 2009 at 3:13 PM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> Yeah, it's there but, as you can see it from the google queries below it's
> not easy to find that page - unless you know what exactly you are looking
> for and search for "loopback". A simple example (i.e. in Sofia Syntax) would
> fill the gap (I'll be happy to do it ASAP)
>
>
> http://www.google.com/search?hl=en&safe=off&q=freeswitch+call+url&aq=f&oq=&aqi=
>
>
> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=
> <http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=>
> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+esl&aq=f&oq=&aqi=
>
> Also, this email now shows up first in googling for "non user extension
> freeswitch" (which were the keywords I was looking for). I think this has
> been useful! ;)
>
> Many thanks and regards,
>    Raffaele
>
> On Mon, Aug 3, 2009 at 14:59, Michael Collins <msc at freeswitch.org> wrote:
>
>>
>>
>> On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
>> raffaele.p.guidi at gmail.com> wrote:
>>
>>> I found the answer by myself while I had finished writing the e-mail. The
>>> correct call url is loopback/<extension> (in this case the command
>>> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
>>> future reference (can't find any example of that anywhere). Is the project
>>> wiki accesible for anyone to contribute or do I have to ask for an
>>> authorization?
>>>
>>
>> All you need to do is sign up for a free account on the wiki and you can
>> start editing. It's a community resource and all FS users are invited to add
>> their respective knowledge.
>>
>> As for not finding what you were looking for, does this page not have it?
>> http://wiki.freeswitch.org/wiki/Loopback
>>
>> If not then please feel free to add to this page whatever your specific
>> scenario entails and give some examples.
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From raffaele.p.guidi at gmail.com  Mon Aug  3 15:38:43 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Tue, 4 Aug 2009 00:38:43 +0200
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
In-Reply-To: <914fc92a0908031506t285d2719need5f8937c0bb2a6@mail.gmail.com>
References: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
	<87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>
	<b0a8792f0908031213r37c10f80m9d076f195b1ba83f@mail.gmail.com>
	<914fc92a0908031506t285d2719need5f8937c0bb2a6@mail.gmail.com>
Message-ID: <b0a8792f0908031538g20abf2e0yba8dab8ac1283305@mail.gmail.com>

uhm... does it?

freeswitch at W2GZ8VNR01> originate sofia/gateway/callwithus.com/fakecall 1001
....
API CALL [originate(sofia/gateway/callwithus.com/fakecall 1001)] output:
-ERR CALL_REJECTED


Anyway, I clearly understand that loopback allows to "hit the dialplan",
maybe I couldn't find it before because of my poor english (but googling for
"freeswitch hit the dialplan" doesn't help either). Really, I simply had
problems finding it - the information was there (is it a matter of SEO -
Search Engine Optimization ;)?

Regards,
    Raffaele

On Tue, Aug 4, 2009 at 00:06, Dan Le <dule.maillist at gmail.com> wrote:

> Technically, loopback is not meant to specifically allow you to call
> non-user extensions, it simply allows you to hit the dialplan.
> For example,
>
> originate sofia/gw/gwname/fakecall 1000
>
> (where gwname is the gateway routing your calls out, and 'fakecall' the
> non-user extension)
>
> will also allow you to dial to non-user extensions. This information would
> be on any of the wiki pages detailing the originate command.
>
> Dan
>
>
> On Mon, Aug 3, 2009 at 3:13 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> Yeah, it's there but, as you can see it from the google queries below it's
>> not easy to find that page - unless you know what exactly you are looking
>> for and search for "loopback". A simple example (i.e. in Sofia Syntax) would
>> fill the gap (I'll be happy to do it ASAP)
>>
>>
>> http://www.google.com/search?hl=en&safe=off&q=freeswitch+call+url&aq=f&oq=&aqi=
>>
>>
>> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=
>> <http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=>
>> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+esl&aq=f&oq=&aqi=
>>
>> Also, this email now shows up first in googling for "non user extension
>> freeswitch" (which were the keywords I was looking for). I think this has
>> been useful! ;)
>>
>> Many thanks and regards,
>>    Raffaele
>>
>> On Mon, Aug 3, 2009 at 14:59, Michael Collins <msc at freeswitch.org> wrote:
>>
>>>
>>>
>>> On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
>>> raffaele.p.guidi at gmail.com> wrote:
>>>
>>>> I found the answer by myself while I had finished writing the e-mail.
>>>> The correct call url is loopback/<extension> (in this case the command
>>>> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
>>>> future reference (can't find any example of that anywhere). Is the project
>>>> wiki accesible for anyone to contribute or do I have to ask for an
>>>> authorization?
>>>>
>>>
>>> All you need to do is sign up for a free account on the wiki and you can
>>> start editing. It's a community resource and all FS users are invited to add
>>> their respective knowledge.
>>>
>>> As for not finding what you were looking for, does this page not have it?
>>> http://wiki.freeswitch.org/wiki/Loopback
>>>
>>> If not then please feel free to add to this page whatever your specific
>>> scenario entails and give some examples.
>>>
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Aug  3 20:36:19 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 22:36:19 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <c857d204-9c47-4185-ba83-56737ac47462@dmmhosting.co.uk>
References: <c857d204-9c47-4185-ba83-56737ac47462@dmmhosting.co.uk>
Message-ID: <A48DE4E2-997A-4E49-AB47-A12D5E9ABFF7@freeswitch.org>

Put the real hostname in register-proxy and outbound-proxy and the  
proxy needs to hold the fake one.

/b

On Aug 3, 2009, at 4:39 PM, Darren Williams wrote:

> The bmnha-01.bt.com is the one that doesn?t resolve
> www.bbvservice-560129.bt.com resolves to 62.239.15.140
>

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From thangappan143 at gmail.com  Mon Aug  3 22:47:03 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Tue, 4 Aug 2009 11:17:03 +0530
Subject: [Freeswitch-users] Fwd: Need Help In IVR
In-Reply-To: <7aa29e790908030012rcce847fu8320d833d2c85530@mail.gmail.com>
References: <7aa29e790908030012rcce847fu8320d833d2c85530@mail.gmail.com>
Message-ID: <7aa29e790908032247h2a93e838m1ac5fd3f8a3946c7@mail.gmail.com>

Can you please help me?


---------- Forwarded message ----------
From: Thangappan.M <thangappan143 at gmail.com>
Date: Mon, Aug 3, 2009 at 12:42 PM
Subject: Need Help In IVR
To: freeswitch-users <freeswitch-users at lists.freeswitch.org>


Dear all,

I am in the process of implementing IVR in Perl using outbound socket.

In the case of the XML macro I can easily specify the timeout,inter digit
timeout value as
 <menu name="demo_ivr"
          greet-long="phrase:demo_ivr_main_menu"
          greet-short="phrase:demo_ivr_main_menu_short"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout ="10000"
          inter-digit-timeout="2000"
          max-failures="3"
          max-timeouts="3"
          digit-len="4">
      <entry action="menu-exec-app" digits="1" param="bridge
sofia/$${domain}/888 at conference.freeswitch.org"/>
    </menu>

Is there any way for Perl to configure this values. Where are the variables
resides?
I am struggling to implement this?




-- 
Regards,
Thangappan.M



-- 
Regards,
Thangappan.M
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From dujinfang at gmail.com  Tue Aug  4 00:17:32 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 4 Aug 2009 15:17:32 +0800
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH Story
Message-ID: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>

Hello All -   In the spirit of ClueCon (which we are missing this year, but
hopefully not next), we wanted to document our "FreeSWITCH Story".  We've
posted it to the wiki(
http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com) and
it is copied below.

Thank you all and enjoy a good conference!

Seven Du (seven)
Jonathan Palley (jpalley_idapted)
Idapted Ltd.


*How FreeSWITCH has created hundreds of job opportunities and changed
lives. *

We want to share our experience working with FreeSWITCH.  FreeSWITCH has
been a key enabler of our business.  We hope this story can be a small way
to say a very big THANK YOU ALL.

"Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
really allowed us to do just that.

What We Do:
We are not a telephony business; we are an educational technology and
service business. In Asia (China, in our case) students must pass English
examinations to study or work abroad and gain new experiences.  However,
there is limited access to native English speakers and the access students
can gain is typically very expensive.  At the same time, in the U.S., there
are many professionals looking for work-at-home opportunities - people who
need jobs and would create great teachers.  Through our technology and
content we empower these people to be effective English teachers.  Does it
work?  Yes.  The majority of our students are getting test scores that many
failed for years to get.  Just hours ago one student called one of our sales
agents crying with joy.  And for our teachers, they are now working in an
industry that was previously unavailable to those living in the U.S.
http://www.idapted.com

Why FreeSWITCH Enables This:
FreeSWITCH has been a key enabler of our business.  Recording calls,
controlling routing, integrating with various web-based interfaces, enabling
multiple endpoints - these are all key features of what we must do.  Most
importantly, setting up various servers and routes to mitigate cross-Pacific
and country-specific network challenges is key.  Doing what we are doing
with commercial solutions would have made the business unworkable.

Our Experiences with FreeSWITCH:
We started using FreeSWITCH as our VoIP Platform in April 2008, after
receiving unsatisfactory results with other open source solutions.  It took
one day of reading through the FreeSWITCH source code to know, "this is it.
 This is the VoIP platform we build our business on".  It took a few days of
working with the extremely competent and focused community to re-affirm this
commitment.

Our Setup:
Our teachers use a custom software that integrates a VoIP client with our
web based platform. Students connect to our teachers "on-demand".  Simply
put, on a web-based comet interface the student enters a phone number (or a
skype name or a gtalk account) and our platform bridges the best available
trainer and the student.  At the same time a web-based interface is being
updated.

The challenge for us is the connection between teachers and students over a
cross-continent network. For example, we experienced problems earlier this
year when a Asis-Pacific communication fiber broken... So, we've learned to
setup multi servers in multiple datacenters for redundancy.

We run multi instances of FreeSWITCH so we can always use the cutting edge
and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
only loads mod_dingaling. Here is one beauty of FS: We just had to create
different conf dirs (/usr/local/freeswitch, /usr/local/skype,
/usr/local/gtalk etc). This allows us to run the same code base over
different configurations, and call skype and gtalk accounts just like a
normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
abnormally or crashes, we can easily change to another FS-skype server (we
run other servers located in various places in China and HK for
redundancy).

FS --|
     |---PSTN gateways
     |--- FS-skype
     |--- FS-gtalk
     |--- FS-skype2
     |--- more ...



COMMUNITY:

The community's commitment cannot be undervalued.  The insightful, modular
design of FreeSWITCH allows anyone to contribute, whereever their skills
lie.  It also allows us to easily make modifications to the underlying code
to suit our specific use-cases  We want to highlight a few key people and
modules in the FS ecosystem:

mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
clients.  PSTN is zero-conf for the user and mitigates troubles with the end
users network/microphone, etc (which is significant with our user base).
 However, cheap providers fail randomly and FreeSWITCH's ability to control
routing, use multiple endpoints all while clearly seeing what is going on is
key.
Most importantly, anthm and the core team have been super helpful in getting
SIP to work with us.  Back in the pre 1.0 days anthm made significant
changes to mod-sofia to enable clients behind nats without STUN.  Its
important to point out that he didn't just make the changes -he forced us to
really make a compelling case as to why the changes were important for
FreeSWITCH.  This is a good thing.

skype (mod_skypiax): Due to the facts that users prefer skype, we configured
skypiax. It was unstable at the beginning and that's one of the reason we
started running that separate FS instance.  To be fair, it has caused a lot
of trouble - but we know this, its new software that takes a big risk and
implements a complex hack.  What is important is that the author of
skypiax(Giovanni Maruzzelli) has been a huge help. He's been very active
fixing bugs and logging in to our box to help trouble shoot. We owe him a
*big* thanks.

To make Skypiax more useful, we also created some patches including the ANY
and RR interfaces for sequential and round robin line hunting, some bug
fixes and other features like continue-load-on-fail and auto-skype-user
which haven't been merged into trunk yet. Thanks a community that gives us a
platform where we can all benefit and contribute.

erlang (mod_erlang_events): Another key enabler of the next release of our
system is the erlang interface.  We have a complex realtime queue routing
system has it handles input not just from freeswitch, but numerous other web
interfaces and sockets.  Erlang was the perfect technology to implement this
in and luckily an Erlang module for FreeSWITCH was already written.
Beautiful.

THE MORAL OF THE STORY:
FreeSWITCH is a great piece of software that has enabled new technologies
and business models.  The design has allowed (and the core team has
nurtured) a vibrant and exciting community that has made the software even
better.  Every day we go to work excited to push the boundaries of what can
be done with telephony technology and are confident this is the platform of
the future.

Thank you all.


Sincerely,

Du Jinfang (Seven) - Technical Operations/VoIP Manager
Jonathan Palley - CTO
Idapted Ltd.
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From hoaianh at gmx.de  Tue Aug  4 00:39:24 2009
From: hoaianh at gmx.de (Ngo-Vi Hoai-Anh)
Date: Tue, 04 Aug 2009 09:39:24 +0200
Subject: [Freeswitch-users] telnet to event socket
In-Reply-To: <645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>
References: <4A76F494.5020101@gmx.de>
	<645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>
Message-ID: <4A77E5AC.9050501@gmx.de>

Hi,

Thank you. It works now. The clue is to hit 'Enter' twice.

Seven Du schrieb:
> On Aug 3, 2009, at 10:30 PM, Ngo-Vi Hoai-Anh wrote:
>   
>> Hi,
>>
>> I'm taking a close look at event socket on FS 1.0.3. Configuration is
>> the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and
>> fsconsole.pl work but I was not able to telnet to port 8021. As I've
>> done that I received somewhat like:
>> #>auth/request
>>
>> I typed in: auth ClueCon
>>
>>     
> followed by two Enters (\n\n).
>
>   
>> After some seconds I've got the message 'connection close by foreign  
>> host'
>>
>> Any ideas?
>>
>> Thank you
>> Hoai-Anh
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From digitaldaz100 at gmail.com  Tue Aug  4 02:10:10 2009
From: digitaldaz100 at gmail.com (Darren Williams)
Date: Tue, 4 Aug 2009 10:10:10 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <26f14763-6f40-4468-91f3-e60a46892183@dmmhosting.co.uk>

This produces:
 
send 639 bytes to udp/[62.239.15.140]:5060 at 09:06:29.640488:
   ------------------------------------------------------------------------
   REGISTER sip:bmnha-01.bt.com SIP/2.0
   Via: SIP/2.0/UDP 91.121.159.57:5080;rport;branch=z9hG4bKQFFy1Ug5DcU0S
   Max-Forwards: 70
   From: <sip:05061292117 at bmnha-01.bt.com;transport=udp>;tag=NKaDDXrtme3Sr
   To: <sip:05061292117 at bmnha-01.bt.com;transport=udp>
   Call-ID: 88de957f-0907-4488-9b19-043ff13391f4
   CSeq: 118570058 REGISTER
   Contact: <sip:gw+05061292117 at 91.121.159.57:5080;transport=udp>
   Expires: 3600
   User-Agent: THOMSON ST2030 hw5 fw1.56 00-1F-9F-16-4E-99
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Content-Length: 0
 
   ------------------------------------------------------------------------
recv 726 bytes from udp/[62.239.15.140]:5060 at 09:06:29.656368:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 91.121.159.57:5080;received=91.121.159.57;branch=z9hG4bKQFFy1Ug5DcU0S;rport=5080
   From: <sip:05061292117 at bmnha-01.bt.com;transport=udp>;tag=NKaDDXrtme3Sr
   To: <sip:05061292117 at bmnha-01.bt.com;transport=udp>;tag=SD567ec99-8a92eaf49d827b084b51e364ec68ae70.bc35
   Call-ID: 88de957f-0907-4488-9b19-043ff13391f4
   CSeq: 118570058 REGISTER
   WWW-Authenticate: Digest realm="bmnha-01.bt.com", nonce="4a77fb4b6fe33ffccf77e6bdd6f7a7a397cdfd17", qop="auth"
   Server: Sip EXpress router (0.9.6 (sparc/solaris))
   Content-Length: 0
   Warning: 392 sip:5060 "Noisy feedback tells:  pid=9150 req_src_ip=172.20.92.61 req_src_port=5060 in_uri=sip:bmnha-01.bt.com out_uri=sip:bmnha-01.bt.com via_cnt==1"
 
   ------------------------------------------------------------------------
2009-08-04 11:06:29.655585 [ERR] sofia_reg.c:1460 05061292117 Registration Failed with status Operation has no matching challenge  [904]. failure #1 and so on.
 
Should my side not have responded at this point with another REGISTER with auth information in?
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 04 August 2009 04:36
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
Put the real hostname in register-proxy and outbound-proxy and the proxy needs to hold the fake one.
 
/b
 
On Aug 3, 2009, at 4:39 PM, Darren Williams wrote:



The bmnha-01.bt.com is the one that doesn?t resolve
www.bbvservice-560129.bt.com resolves to 62.239.15.140
 
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From markmorreny at gmail.com  Tue Aug  4 04:08:36 2009
From: markmorreny at gmail.com (mark morreny)
Date: Tue, 4 Aug 2009 19:08:36 +0800
Subject: [Freeswitch-users] event socket vs erlang
Message-ID: <20ad6b920908040408x48802b84mbf83e20a2cb5d2f@mail.gmail.com>

Hi,

I have seen people using both event socket and erlang to control freeSWITCH
externally.

What is the pros and cons of using event socket vs erlang?

Thanks,
Mark
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From rdenert at tng.de  Tue Aug  4 04:14:43 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Tue, 4 Aug 2009 13:14:43 +0200 (CEST)
Subject: [Freeswitch-users] Module in Lua not working
In-Reply-To: <19314518.221201249383746271.JavaMail.root@zimbra.tng.de>
Message-ID: <7203453.221221249384483704.JavaMail.root@zimbra.tng.de>

Hello again!

I need some help again, because I have little trouble with a few modules. The first one is luasocket, the seconde one is luasql. I always get the error:

error loading module 'socket' from file '/usr/local/lib/lua/5.1/socket/core.so': /usr/local/lib/lua/5.1/socket/core.so: undefined symbol: lua_getmetatable

and

error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': /usr/local/lib/lua/5.1/luasql/mysql.so: undefined symbol: lua_pushlstring

in the fs_cli

This looks like a conflict with an older version of these modules. Is this right? If yes, what should i update?

Thanks again.

BR

-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
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die unbefugte Weitergabe dieser Mail ist nicht gestattet.


From gmaruzz at celliax.org  Tue Aug  4 06:35:29 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 4 Aug 2009 15:35:29 +0200
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
Message-ID: <7b197bef0908040635k330c73cayc354748f40a3bf19@mail.gmail.com>

Cool!



On Tue, Aug 4, 2009 at 9:17 AM, Seven Du<dujinfang at gmail.com> wrote:
> Hello All -
> ?? In the spirit of ClueCon (which we are missing this year, but hopefully
> not next), we wanted to document our "FreeSWITCH Story". ?We've posted it to
> the
> wiki(http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
> and it is copied below.
> Thank you all and enjoy a good conference!
> Seven Du (seven)
> Jonathan Palley (jpalley_idapted)
> Idapted Ltd.
>
> How FreeSWITCH has created hundreds of job opportunities and changed lives.
> We want to share our experience working with FreeSWITCH. ?FreeSWITCH has
> been a key enabler of our business. ?We hope this story can be a small way
> to say a very big THANK YOU ALL.
> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
> really allowed us to do just that.
> What We Do:
> We are not a telephony business; we are an educational technology and
> service business. In Asia (China, in our case) students must pass English
> examinations to study or work abroad and gain new experiences. ?However,
> there is limited access to native English speakers and the access students
> can gain is typically very expensive. ?At the same time, in the U.S., there
> are many professionals looking for work-at-home opportunities - people who
> need jobs and would create great teachers. ?Through our technology and
> content we empower these people to be effective English teachers. ?Does it
> work? ?Yes. ?The majority of our students are getting test scores that many
> failed for years to get. ?Just hours ago one student called one of our sales
> agents crying with joy. ?And for our teachers, they are now working in an
> industry that was previously unavailable to those living in the U.S.
> ?http://www.idapted.com
> Why FreeSWITCH Enables This:
> FreeSWITCH has been a key enabler of our business. ?Recording calls,
> controlling routing, integrating with various web-based interfaces, enabling
> multiple endpoints - these are all key features of what we must do. ?Most
> importantly, setting up various servers and routes to mitigate cross-Pacific
> and country-specific network challenges is key. ?Doing what we are doing
> with commercial solutions would have made the business unworkable.
> Our Experiences with FreeSWITCH:
> We started using FreeSWITCH as our VoIP Platform in April 2008, after
> receiving unsatisfactory results with other open source solutions. ?It took
> one day of reading through the FreeSWITCH source code to know, "this is it.
> ?This is the VoIP platform we build our business on". ?It took a few days of
> working with the extremely competent and focused community to re-affirm this
> commitment.
> Our Setup:
> Our teachers use a custom software that integrates a VoIP client with our
> web based platform. Students connect to our teachers "on-demand". ?Simply
> put, on a web-based comet interface the student enters a phone number (or a
> skype name or a gtalk account) and our platform bridges the best available
> trainer and the student. ?At the same time a web-based interface is being
> updated.
> The challenge for us is the connection between teachers and students over a
> cross-continent network. For example, we experienced problems earlier this
> year when a Asis-Pacific communication fiber broken... So, we've learned to
> setup multi servers in multiple datacenters for redundancy.
>
> We run multi instances of FreeSWITCH so we can always use the cutting edge
> and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
> connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
> only loads mod_dingaling. Here is one beauty of FS: We just had to create
> different conf dirs (/usr/local/freeswitch, /usr/local/skype,
> /usr/local/gtalk etc). This allows us to run the same code base over
> different configurations, and call skype and gtalk accounts just like a
> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
> abnormally or crashes, we can easily change to another FS-skype server (we
> run other servers located in various places in China and HK for
> redundancy).
> FS --|
> ?? ? |---PSTN gateways
> ?? ? |--- FS-skype
> ?? ? |--- FS-gtalk
> ?? ? |--- FS-skype2
> ?? ? |--- more ...
>
>
> COMMUNITY:
> The community's commitment cannot be undervalued. ?The insightful, modular
> design of FreeSWITCH allows anyone to contribute, whereever their skills
> lie. ?It also allows us to easily make modifications to the underlying code
> to suit our specific use-cases ?We want to highlight a few key people and
> modules in the FS ecosystem:
> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
> clients. ?PSTN is zero-conf for the user and mitigates troubles with the end
> users network/microphone, etc (which is significant with our user base).
> ?However, cheap providers fail randomly and FreeSWITCH's ability to control
> routing, use multiple endpoints all while clearly seeing what is going on is
> key.
> Most importantly, anthm and the core team have been super helpful in getting
> SIP to work with us. ?Back in the pre 1.0 days anthm made significant
> changes to mod-sofia to enable clients behind nats without STUN. ?Its
> important to point out that he didn't just make the changes -he forced us to
> really make a compelling case as to why the changes were important for
> FreeSWITCH. ?This is a good thing.
> skype (mod_skypiax): Due to the facts that users prefer skype, we configured
> skypiax. It was unstable at the beginning and that's one of the reason we
> started running that separate FS instance. ?To be fair, it has caused a lot
> of trouble - but we know this, its new software that takes a big risk and
> implements a complex hack. ?What is important is that the author of
> skypiax(Giovanni Maruzzelli) has been a huge help. He's been very active
> fixing bugs and logging in to our box to help trouble shoot. We owe him a
> *big* thanks.
> To make Skypiax more useful, we also created some patches including the ANY
> and RR interfaces for sequential and round robin line hunting, some bug
> fixes and other features like continue-load-on-fail and auto-skype-user
> which haven't been merged into trunk yet. Thanks a community that gives us a
> platform where we can all benefit and contribute.
> erlang (mod_erlang_events): Another key enabler of the next release of our
> system is the erlang interface. ?We have a complex realtime queue routing
> system has it handles input not just from freeswitch, but numerous other web
> interfaces and sockets. ?Erlang was the perfect technology to implement this
> in and luckily an Erlang module for FreeSWITCH was already written.
> Beautiful.
> THE MORAL OF THE STORY:
> FreeSWITCH is a great piece of software that has enabled new technologies
> and business models. ?The design has allowed (and the core team has
> nurtured) a vibrant and exciting community that has made the software even
> better. ?Every day we go to work excited to push the boundaries of what can
> be done with telephony technology and are confident this is the platform of
> the future.
> Thank you all.
>
> Sincerely,
> Du Jinfang (Seven) - Technical Operations/VoIP Manager
> Jonathan Palley - CTO
> Idapted Ltd.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From andrew at hijacked.us  Tue Aug  4 06:49:27 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 4 Aug 2009 09:49:27 -0400
Subject: [Freeswitch-users] event socket vs erlang
In-Reply-To: <20ad6b920908040408x48802b84mbf83e20a2cb5d2f@mail.gmail.com>
References: <20ad6b920908040408x48802b84mbf83e20a2cb5d2f@mail.gmail.com>
Message-ID: <20090804134926.GA27629@hijacked.us>

On Tue, Aug 04, 2009 at 07:08:36PM +0800, mark morreny wrote:
> Hi,
> 
> I have seen people using both event socket and erlang to control freeSWITCH
> externally.
> 
> What is the pros and cons of using event socket vs erlang?
>

It depends on if you want to use erlang or not. The erlang module
provides most of the event socket functionality plus a couple extras
(dynamic XML bindings ala xml_curl, intelligent message delivery, etc).
If you're not already planning to use erlang, it's probably better to
dig out the relevant event socket library module for your langague
instead.

Andrew - author of the erlang module


From jmesquita at gmail.com  Tue Aug  4 07:04:03 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 4 Aug 2009 11:04:03 -0300
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
Message-ID: <5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>

If this is good for me to hear, I would imagine to the core team.

Despite of this not being a group support meeting, I have to say that: Thank
you for sharing, Seven.

jmesquita

On Tue, Aug 4, 2009 at 4:17 AM, Seven Du <dujinfang at gmail.com> wrote:

> Hello All -   In the spirit of ClueCon (which we are missing this year,
> but hopefully not next), we wanted to document our "FreeSWITCH Story".
>  We've posted it to the wiki(
> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com) and
> it is copied below.
>
> Thank you all and enjoy a good conference!
>
> Seven Du (seven)
> Jonathan Palley (jpalley_idapted)
> Idapted Ltd.
>
>
> *How FreeSWITCH has created hundreds of job opportunities and changed
> lives. *
>
> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
> been a key enabler of our business.  We hope this story can be a small way
> to say a very big THANK YOU ALL.
>
> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
> really allowed us to do just that.
>
> What We Do:
> We are not a telephony business; we are an educational technology and
> service business. In Asia (China, in our case) students must pass English
> examinations to study or work abroad and gain new experiences.  However,
> there is limited access to native English speakers and the access students
> can gain is typically very expensive.  At the same time, in the U.S., there
> are many professionals looking for work-at-home opportunities - people who
> need jobs and would create great teachers.  Through our technology and
> content we empower these people to be effective English teachers.  Does it
> work?  Yes.  The majority of our students are getting test scores that many
> failed for years to get.  Just hours ago one student called one of our sales
> agents crying with joy.  And for our teachers, they are now working in an
> industry that was previously unavailable to those living in the U.S.
> http://www.idapted.com
>
> Why FreeSWITCH Enables This:
> FreeSWITCH has been a key enabler of our business.  Recording calls,
> controlling routing, integrating with various web-based interfaces, enabling
> multiple endpoints - these are all key features of what we must do.  Most
> importantly, setting up various servers and routes to mitigate cross-Pacific
> and country-specific network challenges is key.  Doing what we are doing
> with commercial solutions would have made the business unworkable.
>
> Our Experiences with FreeSWITCH:
> We started using FreeSWITCH as our VoIP Platform in April 2008, after
> receiving unsatisfactory results with other open source solutions.  It took
> one day of reading through the FreeSWITCH source code to know, "this is it.
>  This is the VoIP platform we build our business on".  It took a few days of
> working with the extremely competent and focused community to re-affirm this
> commitment.
>
> Our Setup:
> Our teachers use a custom software that integrates a VoIP client with our
> web based platform. Students connect to our teachers "on-demand".  Simply
> put, on a web-based comet interface the student enters a phone number (or a
> skype name or a gtalk account) and our platform bridges the best available
> trainer and the student.  At the same time a web-based interface is being
> updated.
>
> The challenge for us is the connection between teachers and students over a
> cross-continent network. For example, we experienced problems earlier this
> year when a Asis-Pacific communication fiber broken... So, we've learned to
> setup multi servers in multiple datacenters for redundancy.
>
> We run multi instances of FreeSWITCH so we can always use the cutting edge
> and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
> connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
> only loads mod_dingaling. Here is one beauty of FS: We just had to create
> different conf dirs (/usr/local/freeswitch, /usr/local/skype,
> /usr/local/gtalk etc). This allows us to run the same code base over
> different configurations, and call skype and gtalk accounts just like a
> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
> abnormally or crashes, we can easily change to another FS-skype server (we
> run other servers located in various places in China and HK for
> redundancy).
>
> FS --|
>      |---PSTN gateways
>      |--- FS-skype
>      |--- FS-gtalk
>      |--- FS-skype2
>      |--- more ...
>
>
>
> COMMUNITY:
>
> The community's commitment cannot be undervalued.  The insightful, modular
> design of FreeSWITCH allows anyone to contribute, whereever their skills
> lie.  It also allows us to easily make modifications to the underlying code
> to suit our specific use-cases  We want to highlight a few key people and
> modules in the FS ecosystem:
>
> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
> clients.  PSTN is zero-conf for the user and mitigates troubles with the end
> users network/microphone, etc (which is significant with our user base).
>  However, cheap providers fail randomly and FreeSWITCH's ability to control
> routing, use multiple endpoints all while clearly seeing what is going on is
> key.
> Most importantly, anthm and the core team have been super helpful in
> getting SIP to work with us.  Back in the pre 1.0 days anthm made
> significant changes to mod-sofia to enable clients behind nats without STUN.
>  Its important to point out that he didn't just make the changes -he forced
> us to really make a compelling case as to why the changes were important for
> FreeSWITCH.  This is a good thing.
>
> skype (mod_skypiax): Due to the facts that users prefer skype, we
> configured skypiax. It was unstable at the beginning and that's one of the
> reason we started running that separate FS instance.  To be fair, it has
> caused a lot of trouble - but we know this, its new software that takes a
> big risk and implements a complex hack.  What is important is that the
> author of skypiax(Giovanni Maruzzelli) has been a huge help. He's been very
> active fixing bugs and logging in to our box to help trouble shoot. We owe
> him a *big* thanks.
>
> To make Skypiax more useful, we also created some patches including the ANY
> and RR interfaces for sequential and round robin line hunting, some bug
> fixes and other features like continue-load-on-fail and auto-skype-user
> which haven't been merged into trunk yet. Thanks a community that gives us a
> platform where we can all benefit and contribute.
>
> erlang (mod_erlang_events): Another key enabler of the next release of our
> system is the erlang interface.  We have a complex realtime queue routing
> system has it handles input not just from freeswitch, but numerous other web
> interfaces and sockets.  Erlang was the perfect technology to implement this
> in and luckily an Erlang module for FreeSWITCH was already written.
> Beautiful.
>
> THE MORAL OF THE STORY:
> FreeSWITCH is a great piece of software that has enabled new technologies
> and business models.  The design has allowed (and the core team has
> nurtured) a vibrant and exciting community that has made the software even
> better.  Every day we go to work excited to push the boundaries of what can
> be done with telephony technology and are confident this is the platform of
> the future.
>
> Thank you all.
>
>
> Sincerely,
>
> Du Jinfang (Seven) - Technical Operations/VoIP Manager
> Jonathan Palley - CTO
> Idapted Ltd.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rdenert at tng.de  Tue Aug  4 08:41:18 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Tue, 4 Aug 2009 17:41:18 +0200 (CEST)
Subject: [Freeswitch-users] Module in Lua not working
In-Reply-To: <7203453.221221249384483704.JavaMail.root@zimbra.tng.de>
Message-ID: <17585545.224751249400478914.JavaMail.root@zimbra.tng.de>

Hello,

there is no more problem. I installed 1.0.4 and everthing is fine. :-)

BR

----- Urspr?ngliche Mail -----
Von: "Rudolf Denert" <rdenert at tng.de>
An: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
Gesendet: Dienstag, 4. August 2009 13:14:43 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: [Freeswitch-users] Module in Lua not working

Hello again!

I need some help again, because I have little trouble with a few modules. The first one is luasocket, the seconde one is luasql. I always get the error:

error loading module 'socket' from file '/usr/local/lib/lua/5.1/socket/core.so': /usr/local/lib/lua/5.1/socket/core.so: undefined symbol: lua_getmetatable

and

error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': /usr/local/lib/lua/5.1/luasql/mysql.so: undefined symbol: lua_pushlstring

in the fs_cli

This looks like a conflict with an older version of these modules. Is this right? If yes, what should i update?

Thanks again.

BR

-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
-
Register: Amtsgericht Kiel, HRB 6596 KI
Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
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-
This e-mail may contain confidential and/or privileged information. If
you are not the intended recipient (or have received this e-mail in
error) please notify the sender immediately and destroy this e-mail. Any
unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
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Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
-
Register: Amtsgericht Kiel, HRB 6596 KI
Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
Executive board (Vorstand): Dr. Sven Willert (Vorsitz),  Dr. Thomas
Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg
Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
-
This e-mail may contain confidential and/or privileged information. If
you are not the intended recipient (or have received this e-mail in
error) please notify the sender immediately and destroy this e-mail. Any
unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.


From matt at hellohunter.com  Tue Aug  4 11:50:36 2009
From: matt at hellohunter.com (Matt Hunter)
Date: Tue, 4 Aug 2009 11:50:36 -0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
Message-ID: <4256bf830908041150x5a450797j62389d4513997f9c@mail.gmail.com>

Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the
logs below, but I am still at a loss at being able to identify the error or
reproduce it consistently. The below log indicates to me that my FS server
is initiating sending 2 BYE message to my DID provider (didforsale.com). Is
there a way I can look further inside FreeSWITCH to see why it is sending
this BYE packet?

sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
Route: <sip:65.217.40.210:5060;lr>
Route: <sip:63.110.102.238;lr>
Max-Forwards: 70
From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
To: <sip:+1909635XXXX at 199.173.100.144:5060
;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
CSeq: 118584736 BYE
Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0


sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
Route: <sip:65.217.40.210:5060;lr>
Route: <sip:63.110.102.238;lr>
Max-Forwards: 70
From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
To: <sip:+1909635XXXX at 199.173.100.144:5060
;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
CSeq: 118584736 BYE
Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0



On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> Digging a bit in mod_sofia releaved that it can be caused by a SIP
> code 482 (loop detected), 483 (too many hops) or 484 (address
> incomplete).
>
> Do a SIP trace to sched more light on what's happening.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> Am 3-Aug-09 um 1:22 AM schrieb Matthew Fong:
>
> > EXCHANGE_ROUTING_ERROR
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mattdfong at gmail.com  Tue Aug  4 11:51:03 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Tue, 4 Aug 2009 11:51:03 -0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
Message-ID: <4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>

Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the
logs below, but I am still at a loss at being able to identify the error or
reproduce it consistently. The below log indicates to me that my FS server
is initiating sending 2 BYE message to my DID provider (didforsale.com). Is
there a way I can look further inside FreeSWITCH to see why it is sending
this BYE packet?

sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
Route: <sip:65.217.40.210:5060;lr>
Route: <sip:63.110.102.238;lr>
Max-Forwards: 70
From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
To: <sip:+1909635XXXX at 199.173.100.144:5060
;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
CSeq: 118584736 BYE
Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0


sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
Route: <sip:65.217.40.210:5060;lr>
Route: <sip:63.110.102.238;lr>
Max-Forwards: 70
From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
To: <sip:+1909635XXXX at 199.173.100.144:5060
;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
CSeq: 118584736 BYE
Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0

On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> Digging a bit in mod_sofia releaved that it can be caused by a SIP
> code 482 (loop detected), 483 (too many hops) or 484 (address
> incomplete).
>
> Do a SIP trace to sched more light on what's happening.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> Am 3-Aug-09 um 1:22 AM schrieb Matthew Fong:
>
> > EXCHANGE_ROUTING_ERROR
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gregt at cgicommunications.com  Tue Aug  4 12:12:25 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Tue, 4 Aug 2009 15:12:25 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>
Message-ID: <E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>

Hi, does anyone have an example of a simple PHP socket script that  
will listen and spawn of a process that handles the incoming call?

I understand the inbound socket and code such as this
http://wiki.freeswitch.org/wiki/PHP_Event_Socket
that will let me initiate operations. It's the constantly running php  
socket program that I can't get my head around, and how it will spawn  
another php script that will be able to do things like answer the  
session, get dtmf, etc.
--
Greg


On Aug 2, 2009, at 6:35 PM, Michael Collins wrote:

>
>
> On Sun, Aug 2, 2009 at 12:38 PM, Nik Middleton <nik.middleton at noblesolutions.co.uk 
> > wrote:
> Hi Guys,
>
>
> I?m using an outbound socket to control calls, and it works a  
> charm.  However, what I?d like to do is send a custom event  
> regarding the call on hang-up.  The way I see things happening at  
> the moment, and I could be wrong, is that the socket is closed when  
> a hang-up occurs, so am I taking a chance trying to send the event  
> then? (try to sneak out the event before socket closure happens)   
> The other option is of course to open an inbound socket and send the  
> event, but I?d rather not do that if possible.
>
> Nik,
>
> Perhaps the "linger" event socket command will do what you need?  
> Check out this commit:
> http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009391.html
>
> Let me know if it works for you and I'll be sure to get it  
> documented properly. If you get it working I'd love to see a code  
> snippet so we can wikify this knowledge. :)
>
> Thanks,
> MC
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From william.suffill at gmail.com  Tue Aug  4 16:04:08 2009
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 4 Aug 2009 19:04:08 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local> 
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com> 
	<E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>
Message-ID: <6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>

I wrote some notes on this but have yet to wiki it.

example of an outbound socket connection where the call is answered, a
variable is set then perhaps play one of the pre-installed files and
hangup.
ivrd

fs_ivrd comes with freeswitch. It being a small daemon just invokes
the script defined in a variable and passes data from it via STDIN/OUT

Since this is an outbound socket connections it needs to be defined in
the dialplan.



Ex:

<extension name="outbound-socket">

    <condition field="destination_number" expression="^55(522)$">

        <action application="set"
data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>

        <action application="socket" data="127.0.0.1:8084 async full"/>

    </condition>

  </extension>



The above dialplan sample would invoke ivr-demo.php when 55522 is
called as long as fs_ivrd is running. To start fs_ivrd:

/usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 8004



It takes 2 arguments -h for hostname and -p for port.



PHP Code

#!/usr/bin/php -q

<?php



    // set a couple of things so we dont kill the system

    ob_implicit_flush(true);

    set_time_limit(30);





    // Open stdin so we can read the AGI data in

    $in = fopen("php://stdin", "r");

    // Connect

    echo "connect\n\n";

    // Answer

    echo "sendmsg\n";

    echo "call-command: execute\n";

    echo "execute-app-name: answer\n\n";



    // Play a prompt

    echo "sendmsg\n";

    echo "call-command: execute\n";

    echo "execute-app-name: playback\n";

    echo "execute-app-arg:
/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n";



    // Wait

    sleep(5);



   // Hangup

   echo "sendmsg\n";

   echo "call-command: hangup\n\n";



fclose($in);



?>



ivrd will call this script for each call. All itdoes is answer the
channel tell FreeSWITCH to play the ?welcome to freeswitch? prompt.
Since the script is now controlling all call flow I needed to add a
wait or it would send the hangup immediately before the prompt was
played.

Some improvements possible but that's 1 way to do it. It would be
possible to do the socket directly in PHP but fs_ivrd is a nice option
too.

-- W


From diego.viola at gmail.com  Tue Aug  4 17:54:34 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 20:54:34 -0400
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
Message-ID: <86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>

Very cool, and yes, FreeSWITCH does rock =D

Both the software and the community ;)

2009/8/4 Jo?o Mesquita <jmesquita at gmail.com>

> If this is good for me to hear, I would imagine to the core team.
>
> Despite of this not being a group support meeting, I have to say that:
> Thank you for sharing, Seven.
>
> jmesquita
>
> On Tue, Aug 4, 2009 at 4:17 AM, Seven Du <dujinfang at gmail.com> wrote:
>
>> Hello All -   In the spirit of ClueCon (which we are missing this year,
>> but hopefully not next), we wanted to document our "FreeSWITCH Story".
>>  We've posted it to the wiki(
>> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
>> and it is copied below.
>>
>> Thank you all and enjoy a good conference!
>>
>> Seven Du (seven)
>> Jonathan Palley (jpalley_idapted)
>> Idapted Ltd.
>>
>>
>> *How FreeSWITCH has created hundreds of job opportunities and changed
>> lives. *
>>
>> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
>> been a key enabler of our business.  We hope this story can be a small way
>> to say a very big THANK YOU ALL.
>>
>> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
>> really allowed us to do just that.
>>
>> What We Do:
>> We are not a telephony business; we are an educational technology and
>> service business. In Asia (China, in our case) students must pass English
>> examinations to study or work abroad and gain new experiences.  However,
>> there is limited access to native English speakers and the access students
>> can gain is typically very expensive.  At the same time, in the U.S., there
>> are many professionals looking for work-at-home opportunities - people who
>> need jobs and would create great teachers.  Through our technology and
>> content we empower these people to be effective English teachers.  Does it
>> work?  Yes.  The majority of our students are getting test scores that many
>> failed for years to get.  Just hours ago one student called one of our sales
>> agents crying with joy.  And for our teachers, they are now working in an
>> industry that was previously unavailable to those living in the U.S.
>> http://www.idapted.com
>>
>> Why FreeSWITCH Enables This:
>> FreeSWITCH has been a key enabler of our business.  Recording calls,
>> controlling routing, integrating with various web-based interfaces, enabling
>> multiple endpoints - these are all key features of what we must do.  Most
>> importantly, setting up various servers and routes to mitigate cross-Pacific
>> and country-specific network challenges is key.  Doing what we are doing
>> with commercial solutions would have made the business unworkable.
>>
>> Our Experiences with FreeSWITCH:
>> We started using FreeSWITCH as our VoIP Platform in April 2008, after
>> receiving unsatisfactory results with other open source solutions.  It took
>> one day of reading through the FreeSWITCH source code to know, "this is it.
>>  This is the VoIP platform we build our business on".  It took a few days of
>> working with the extremely competent and focused community to re-affirm this
>> commitment.
>>
>> Our Setup:
>> Our teachers use a custom software that integrates a VoIP client with our
>> web based platform. Students connect to our teachers "on-demand".  Simply
>> put, on a web-based comet interface the student enters a phone number (or a
>> skype name or a gtalk account) and our platform bridges the best available
>> trainer and the student.  At the same time a web-based interface is being
>> updated.
>>
>> The challenge for us is the connection between teachers and students over
>> a cross-continent network. For example, we experienced problems earlier this
>> year when a Asis-Pacific communication fiber broken... So, we've learned to
>> setup multi servers in multiple datacenters for redundancy.
>>
>> We run multi instances of FreeSWITCH so we can always use the cutting edge
>> and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
>> connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
>> only loads mod_dingaling. Here is one beauty of FS: We just had to create
>> different conf dirs (/usr/local/freeswitch, /usr/local/skype,
>> /usr/local/gtalk etc). This allows us to run the same code base over
>> different configurations, and call skype and gtalk accounts just like a
>> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
>> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
>> abnormally or crashes, we can easily change to another FS-skype server (we
>> run other servers located in various places in China and HK for
>> redundancy).
>>
>> FS --|
>>      |---PSTN gateways
>>      |--- FS-skype
>>      |--- FS-gtalk
>>      |--- FS-skype2
>>      |--- more ...
>>
>>
>>
>> COMMUNITY:
>>
>> The community's commitment cannot be undervalued.  The insightful, modular
>> design of FreeSWITCH allows anyone to contribute, whereever their skills
>> lie.  It also allows us to easily make modifications to the underlying code
>> to suit our specific use-cases  We want to highlight a few key people and
>> modules in the FS ecosystem:
>>
>> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
>> clients.  PSTN is zero-conf for the user and mitigates troubles with the end
>> users network/microphone, etc (which is significant with our user base).
>>  However, cheap providers fail randomly and FreeSWITCH's ability to control
>> routing, use multiple endpoints all while clearly seeing what is going on is
>> key.
>> Most importantly, anthm and the core team have been super helpful in
>> getting SIP to work with us.  Back in the pre 1.0 days anthm made
>> significant changes to mod-sofia to enable clients behind nats without STUN.
>>  Its important to point out that he didn't just make the changes -he forced
>> us to really make a compelling case as to why the changes were important for
>> FreeSWITCH.  This is a good thing.
>>
>> skype (mod_skypiax): Due to the facts that users prefer skype, we
>> configured skypiax. It was unstable at the beginning and that's one of the
>> reason we started running that separate FS instance.  To be fair, it has
>> caused a lot of trouble - but we know this, its new software that takes a
>> big risk and implements a complex hack.  What is important is that the
>> author of skypiax(Giovanni Maruzzelli) has been a huge help. He's been very
>> active fixing bugs and logging in to our box to help trouble shoot. We owe
>> him a *big* thanks.
>>
>> To make Skypiax more useful, we also created some patches including the
>> ANY and RR interfaces for sequential and round robin line hunting, some bug
>> fixes and other features like continue-load-on-fail and auto-skype-user
>> which haven't been merged into trunk yet. Thanks a community that gives us a
>> platform where we can all benefit and contribute.
>>
>> erlang (mod_erlang_events): Another key enabler of the next release of our
>> system is the erlang interface.  We have a complex realtime queue routing
>> system has it handles input not just from freeswitch, but numerous other web
>> interfaces and sockets.  Erlang was the perfect technology to implement this
>> in and luckily an Erlang module for FreeSWITCH was already written.
>> Beautiful.
>>
>> THE MORAL OF THE STORY:
>> FreeSWITCH is a great piece of software that has enabled new technologies
>> and business models.  The design has allowed (and the core team has
>> nurtured) a vibrant and exciting community that has made the software even
>> better.  Every day we go to work excited to push the boundaries of what can
>> be done with telephony technology and are confident this is the platform of
>> the future.
>>
>> Thank you all.
>>
>>
>> Sincerely,
>>
>> Du Jinfang (Seven) - Technical Operations/VoIP Manager
>> Jonathan Palley - CTO
>> Idapted Ltd.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Tue Aug  4 17:55:22 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 20:55:22 -0400
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
Message-ID: <86a32abc0908041755v7400796dwb21a13d039635539@mail.gmail.com>

Cool, I wish I could be there, next year =D

Any pics or videos of ClueCon? :)

On Mon, Aug 3, 2009 at 8:41 AM, Brian West <brian at freeswitch.org> wrote:

> Just look for large groups of people with laptops.  I'm sure you can't
> miss us.
>
> /b
>
> On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:
>
> > i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
> > afternoon?  anybody need help with setup?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Tue Aug  4 17:56:27 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 20:56:27 -0400
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
	<86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>
Message-ID: <86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>

Maybe you can link your testimonial or put it here also? :D

http://wiki.freeswitch.org/wiki/Testimonials

On Tue, Aug 4, 2009 at 8:54 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Very cool, and yes, FreeSWITCH does rock =D
>
> Both the software and the community ;)
>
> 2009/8/4 Jo?o Mesquita <jmesquita at gmail.com>
>
> If this is good for me to hear, I would imagine to the core team.
>>
>> Despite of this not being a group support meeting, I have to say that:
>> Thank you for sharing, Seven.
>>
>> jmesquita
>>
>> On Tue, Aug 4, 2009 at 4:17 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>>>  Hello All -   In the spirit of ClueCon (which we are missing this year,
>>> but hopefully not next), we wanted to document our "FreeSWITCH Story".
>>>  We've posted it to the wiki(
>>> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
>>> and it is copied below.
>>>
>>> Thank you all and enjoy a good conference!
>>>
>>> Seven Du (seven)
>>> Jonathan Palley (jpalley_idapted)
>>> Idapted Ltd.
>>>
>>>
>>> *How FreeSWITCH has created hundreds of job opportunities and changed
>>> lives. *
>>>
>>> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
>>> been a key enabler of our business.  We hope this story can be a small way
>>> to say a very big THANK YOU ALL.
>>>
>>> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
>>> really allowed us to do just that.
>>>
>>> What We Do:
>>> We are not a telephony business; we are an educational technology and
>>> service business. In Asia (China, in our case) students must pass English
>>> examinations to study or work abroad and gain new experiences.  However,
>>> there is limited access to native English speakers and the access students
>>> can gain is typically very expensive.  At the same time, in the U.S., there
>>> are many professionals looking for work-at-home opportunities - people who
>>> need jobs and would create great teachers.  Through our technology and
>>> content we empower these people to be effective English teachers.  Does it
>>> work?  Yes.  The majority of our students are getting test scores that many
>>> failed for years to get.  Just hours ago one student called one of our sales
>>> agents crying with joy.  And for our teachers, they are now working in an
>>> industry that was previously unavailable to those living in the U.S.
>>> http://www.idapted.com
>>>
>>> Why FreeSWITCH Enables This:
>>> FreeSWITCH has been a key enabler of our business.  Recording calls,
>>> controlling routing, integrating with various web-based interfaces, enabling
>>> multiple endpoints - these are all key features of what we must do.  Most
>>> importantly, setting up various servers and routes to mitigate cross-Pacific
>>> and country-specific network challenges is key.  Doing what we are doing
>>> with commercial solutions would have made the business unworkable.
>>>
>>> Our Experiences with FreeSWITCH:
>>> We started using FreeSWITCH as our VoIP Platform in April 2008, after
>>> receiving unsatisfactory results with other open source solutions.  It took
>>> one day of reading through the FreeSWITCH source code to know, "this is it.
>>>  This is the VoIP platform we build our business on".  It took a few days of
>>> working with the extremely competent and focused community to re-affirm this
>>> commitment.
>>>
>>> Our Setup:
>>> Our teachers use a custom software that integrates a VoIP client with our
>>> web based platform. Students connect to our teachers "on-demand".  Simply
>>> put, on a web-based comet interface the student enters a phone number (or a
>>> skype name or a gtalk account) and our platform bridges the best available
>>> trainer and the student.  At the same time a web-based interface is being
>>> updated.
>>>
>>> The challenge for us is the connection between teachers and students over
>>> a cross-continent network. For example, we experienced problems earlier this
>>> year when a Asis-Pacific communication fiber broken... So, we've learned to
>>> setup multi servers in multiple datacenters for redundancy.
>>>
>>> We run multi instances of FreeSWITCH so we can always use the cutting
>>> edge and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS)
>>> instance connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and
>>> FS-gtalk only loads mod_dingaling. Here is one beauty of FS: We just had to
>>> create different conf dirs (/usr/local/freeswitch, /usr/local/skype,
>>> /usr/local/gtalk etc). This allows us to run the same code base over
>>> different configurations, and call skype and gtalk accounts just like a
>>> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
>>> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
>>> abnormally or crashes, we can easily change to another FS-skype server (we
>>> run other servers located in various places in China and HK for
>>> redundancy).
>>>
>>> FS --|
>>>      |---PSTN gateways
>>>      |--- FS-skype
>>>      |--- FS-gtalk
>>>      |--- FS-skype2
>>>      |--- more ...
>>>
>>>
>>>
>>> COMMUNITY:
>>>
>>> The community's commitment cannot be undervalued.  The insightful,
>>> modular design of FreeSWITCH allows anyone to contribute, whereever their
>>> skills lie.  It also allows us to easily make modifications to the
>>> underlying code to suit our specific use-cases  We want to highlight a few
>>> key people and modules in the FS ecosystem:
>>>
>>> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
>>> clients.  PSTN is zero-conf for the user and mitigates troubles with the end
>>> users network/microphone, etc (which is significant with our user base).
>>>  However, cheap providers fail randomly and FreeSWITCH's ability to control
>>> routing, use multiple endpoints all while clearly seeing what is going on is
>>> key.
>>> Most importantly, anthm and the core team have been super helpful in
>>> getting SIP to work with us.  Back in the pre 1.0 days anthm made
>>> significant changes to mod-sofia to enable clients behind nats without STUN.
>>>  Its important to point out that he didn't just make the changes -he forced
>>> us to really make a compelling case as to why the changes were important for
>>> FreeSWITCH.  This is a good thing.
>>>
>>> skype (mod_skypiax): Due to the facts that users prefer skype, we
>>> configured skypiax. It was unstable at the beginning and that's one of the
>>> reason we started running that separate FS instance.  To be fair, it has
>>> caused a lot of trouble - but we know this, its new software that takes a
>>> big risk and implements a complex hack.  What is important is that the
>>> author of skypiax(Giovanni Maruzzelli) has been a huge help. He's been very
>>> active fixing bugs and logging in to our box to help trouble shoot. We owe
>>> him a *big* thanks.
>>>
>>> To make Skypiax more useful, we also created some patches including the
>>> ANY and RR interfaces for sequential and round robin line hunting, some bug
>>> fixes and other features like continue-load-on-fail and auto-skype-user
>>> which haven't been merged into trunk yet. Thanks a community that gives us a
>>> platform where we can all benefit and contribute.
>>>
>>> erlang (mod_erlang_events): Another key enabler of the next release of
>>> our system is the erlang interface.  We have a complex realtime queue
>>> routing system has it handles input not just from freeswitch, but numerous
>>> other web interfaces and sockets.  Erlang was the perfect technology to
>>> implement this in and luckily an Erlang module for FreeSWITCH was already
>>> written. Beautiful.
>>>
>>> THE MORAL OF THE STORY:
>>> FreeSWITCH is a great piece of software that has enabled new technologies
>>> and business models.  The design has allowed (and the core team has
>>> nurtured) a vibrant and exciting community that has made the software even
>>> better.  Every day we go to work excited to push the boundaries of what can
>>> be done with telephony technology and are confident this is the platform of
>>> the future.
>>>
>>> Thank you all.
>>>
>>>
>>> Sincerely,
>>>
>>> Du Jinfang (Seven) - Technical Operations/VoIP Manager
>>> Jonathan Palley - CTO
>>> Idapted Ltd.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From gabe at gundy.org  Tue Aug  4 18:53:23 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 4 Aug 2009 19:53:23 -0600
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <86a32abc0908041755v7400796dwb21a13d039635539@mail.gmail.com>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
	<86a32abc0908041755v7400796dwb21a13d039635539@mail.gmail.com>
Message-ID: <903da5680908041853y7f828878pc539470b328066dc@mail.gmail.com>

On Tue, Aug 4, 2009 at 6:55 PM, Diego Viola<diego.viola at gmail.com> wrote:
> Cool, I wish I could be there, next year =D

Tomorrow is my wedding anniversary 8/5.  If ClueCon keeps getting
scheduled on that date, I don't know if I'll ever get to go :(


> Any pics or videos of ClueCon? :)

Yes, bring on the vids.

Best,
Gabe


From diego.viola at gmail.com  Tue Aug  4 18:56:27 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 21:56:27 -0400
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <903da5680908041853y7f828878pc539470b328066dc@mail.gmail.com>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
	<86a32abc0908041755v7400796dwb21a13d039635539@mail.gmail.com>
	<903da5680908041853y7f828878pc539470b328066dc@mail.gmail.com>
Message-ID: <86a32abc0908041856u4d4a5c4byff3c9f73be0e5031@mail.gmail.com>

Looking forward to this talk =D

"FreeSWITCH: Learning to Think Fourth Dimensionally"

;)

On Tue, Aug 4, 2009 at 9:53 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Tue, Aug 4, 2009 at 6:55 PM, Diego Viola<diego.viola at gmail.com> wrote:
> > Cool, I wish I could be there, next year =D
>
> Tomorrow is my wedding anniversary 8/5.  If ClueCon keeps getting
> scheduled on that date, I don't know if I'll ever get to go :(
>
>
> > Any pics or videos of ClueCon? :)
>
> Yes, bring on the vids.
>
> Best,
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Tue Aug  4 19:22:07 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 22:22:07 -0400
Subject: [Freeswitch-users] Looking for some FreeSWITCH job
Message-ID: <86a32abc0908041922j5dac756g49b2d87db74479b1@mail.gmail.com>

Hi,

I'm currently looking for some FS jobs, I really need one, I'm currently
unemployed and looking for some serious FreeSWITCH jobs.

Anyone?

P.S: I also do FS and web development with any language, PHP, Ruby, etc.
anything really.

Thanks,

Diego
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From jpalley at idapted.com  Tue Aug  4 21:33:32 2009
From: jpalley at idapted.com (Jonathan Palley)
Date: Wed, 5 Aug 2009 12:33:32 +0800
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com> 
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com> 
	<86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com> 
	<86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>
Message-ID: <2d8777c00908042133o6e2ffd33g37da13392ac6ccb@mail.gmail.com>

Diego -   Already done.  See the bottom of the page (we linked to another
page because of its length)!

:)
Jonathan Palley
Idapted Ltd.

On Wed, Aug 5, 2009 at 8:56 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Maybe you can link your testimonial or put it here also? :D
>
> http://wiki.freeswitch.org/wiki/Testimonials
>
>
> On Tue, Aug 4, 2009 at 8:54 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Very cool, and yes, FreeSWITCH does rock =D
>>
>> Both the software and the community ;)
>>
>> 2009/8/4 Jo?o Mesquita <jmesquita at gmail.com>
>>
>> If this is good for me to hear, I would imagine to the core team.
>>>
>>> Despite of this not being a group support meeting, I have to say that:
>>> Thank you for sharing, Seven.
>>>
>>> jmesquita
>>>
>>> On Tue, Aug 4, 2009 at 4:17 AM, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>>>  Hello All -   In the spirit of ClueCon (which we are missing this
>>>> year, but hopefully not next), we wanted to document our "FreeSWITCH Story".
>>>>  We've posted it to the wiki(
>>>> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
>>>> and it is copied below.
>>>>
>>>> Thank you all and enjoy a good conference!
>>>>
>>>> Seven Du (seven)
>>>> Jonathan Palley (jpalley_idapted)
>>>> Idapted Ltd.
>>>>
>>>>
>>>> *How FreeSWITCH has created hundreds of job opportunities and changed
>>>> lives. *
>>>>
>>>> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
>>>> been a key enabler of our business.  We hope this story can be a small way
>>>> to say a very big THANK YOU ALL.
>>>>
>>>> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH
>>>> has really allowed us to do just that.
>>>>
>>>> What We Do:
>>>> We are not a telephony business; we are an educational technology and
>>>> service business. In Asia (China, in our case) students must pass English
>>>> examinations to study or work abroad and gain new experiences.  However,
>>>> there is limited access to native English speakers and the access students
>>>> can gain is typically very expensive.  At the same time, in the U.S., there
>>>> are many professionals looking for work-at-home opportunities - people who
>>>> need jobs and would create great teachers.  Through our technology and
>>>> content we empower these people to be effective English teachers.  Does it
>>>> work?  Yes.  The majority of our students are getting test scores that many
>>>> failed for years to get.  Just hours ago one student called one of our sales
>>>> agents crying with joy.  And for our teachers, they are now working in an
>>>> industry that was previously unavailable to those living in the U.S.
>>>> http://www.idapted.com
>>>>
>>>> Why FreeSWITCH Enables This:
>>>> FreeSWITCH has been a key enabler of our business.  Recording calls,
>>>> controlling routing, integrating with various web-based interfaces, enabling
>>>> multiple endpoints - these are all key features of what we must do.  Most
>>>> importantly, setting up various servers and routes to mitigate cross-Pacific
>>>> and country-specific network challenges is key.  Doing what we are doing
>>>> with commercial solutions would have made the business unworkable.
>>>>
>>>> Our Experiences with FreeSWITCH:
>>>> We started using FreeSWITCH as our VoIP Platform in April 2008, after
>>>> receiving unsatisfactory results with other open source solutions.  It took
>>>> one day of reading through the FreeSWITCH source code to know, "this is it.
>>>>  This is the VoIP platform we build our business on".  It took a few days of
>>>> working with the extremely competent and focused community to re-affirm this
>>>> commitment.
>>>>
>>>> Our Setup:
>>>> Our teachers use a custom software that integrates a VoIP client with
>>>> our web based platform. Students connect to our teachers "on-demand".
>>>>  Simply put, on a web-based comet interface the student enters a phone
>>>> number (or a skype name or a gtalk account) and our platform bridges the
>>>> best available trainer and the student.  At the same time a web-based
>>>> interface is being updated.
>>>>
>>>> The challenge for us is the connection between teachers and students
>>>> over a cross-continent network. For example, we experienced problems earlier
>>>> this year when a Asis-Pacific communication fiber broken... So, we've
>>>> learned to setup multi servers in multiple datacenters for redundancy.
>>>>
>>>> We run multi instances of FreeSWITCH so we can always use the cutting
>>>> edge and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS)
>>>> instance connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and
>>>> FS-gtalk only loads mod_dingaling. Here is one beauty of FS: We just had to
>>>> create different conf dirs (/usr/local/freeswitch, /usr/local/skype,
>>>> /usr/local/gtalk etc). This allows us to run the same code base over
>>>> different configurations, and call skype and gtalk accounts just like a
>>>> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
>>>> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
>>>> abnormally or crashes, we can easily change to another FS-skype server (we
>>>> run other servers located in various places in China and HK for
>>>> redundancy).
>>>>
>>>> FS --|
>>>>      |---PSTN gateways
>>>>      |--- FS-skype
>>>>      |--- FS-gtalk
>>>>      |--- FS-skype2
>>>>      |--- more ...
>>>>
>>>>
>>>>
>>>> COMMUNITY:
>>>>
>>>> The community's commitment cannot be undervalued.  The insightful,
>>>> modular design of FreeSWITCH allows anyone to contribute, whereever their
>>>> skills lie.  It also allows us to easily make modifications to the
>>>> underlying code to suit our specific use-cases  We want to highlight a few
>>>> key people and modules in the FS ecosystem:
>>>>
>>>> mod_sofia: SIP is how we connect to our PSTN gateways and to our
>>>> teachers clients.  PSTN is zero-conf for the user and mitigates troubles
>>>> with the end users network/microphone, etc (which is significant with our
>>>> user base).  However, cheap providers fail randomly and FreeSWITCH's ability
>>>> to control routing, use multiple endpoints all while clearly seeing what is
>>>> going on is key.
>>>> Most importantly, anthm and the core team have been super helpful in
>>>> getting SIP to work with us.  Back in the pre 1.0 days anthm made
>>>> significant changes to mod-sofia to enable clients behind nats without STUN.
>>>>  Its important to point out that he didn't just make the changes -he forced
>>>> us to really make a compelling case as to why the changes were important for
>>>> FreeSWITCH.  This is a good thing.
>>>>
>>>> skype (mod_skypiax): Due to the facts that users prefer skype, we
>>>> configured skypiax. It was unstable at the beginning and that's one of the
>>>> reason we started running that separate FS instance.  To be fair, it has
>>>> caused a lot of trouble - but we know this, its new software that takes a
>>>> big risk and implements a complex hack.  What is important is that the
>>>> author of skypiax(Giovanni Maruzzelli) has been a huge help. He's been very
>>>> active fixing bugs and logging in to our box to help trouble shoot. We owe
>>>> him a *big* thanks.
>>>>
>>>> To make Skypiax more useful, we also created some patches including the
>>>> ANY and RR interfaces for sequential and round robin line hunting, some bug
>>>> fixes and other features like continue-load-on-fail and auto-skype-user
>>>> which haven't been merged into trunk yet. Thanks a community that gives us a
>>>> platform where we can all benefit and contribute.
>>>>
>>>> erlang (mod_erlang_events): Another key enabler of the next release of
>>>> our system is the erlang interface.  We have a complex realtime queue
>>>> routing system has it handles input not just from freeswitch, but numerous
>>>> other web interfaces and sockets.  Erlang was the perfect technology to
>>>> implement this in and luckily an Erlang module for FreeSWITCH was already
>>>> written. Beautiful.
>>>>
>>>> THE MORAL OF THE STORY:
>>>> FreeSWITCH is a great piece of software that has enabled new
>>>> technologies and business models.  The design has allowed (and the core team
>>>> has nurtured) a vibrant and exciting community that has made the software
>>>> even better.  Every day we go to work excited to push the boundaries of what
>>>> can be done with telephony technology and are confident this is the platform
>>>> of the future.
>>>>
>>>> Thank you all.
>>>>
>>>>
>>>> Sincerely,
>>>>
>>>> Du Jinfang (Seven) - Technical Operations/VoIP Manager
>>>> Jonathan Palley - CTO
>>>> Idapted Ltd.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Tue Aug  4 21:41:20 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 5 Aug 2009 12:41:20 +0800
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <2d8777c00908042133o6e2ffd33g37da13392ac6ccb@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
	<86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>
	<86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>
	<2d8777c00908042133o6e2ffd33g37da13392ac6ccb@mail.gmail.com>
Message-ID: <23f91030908042141h9f79d61w86599122677670@mail.gmail.com>

And I  added this on the wiki page:
mod_conference and mod_fifo: We also use FreeSWITCH in our office
environment as a PBX for call center and customer service connected with
VoIP and PSTN(openzap) gateways. It is integrated into our CRM system
naturally and just made sales process, business logic and world wide
conference much more simpler and easier.

:)

7

2009/8/5 Jonathan Palley <jpalley at idapted.com>

> Diego -   Already done.  See the bottom of the page (we linked to another
> page because of its length)!
>
> :)
> Jonathan Palley
> Idapted Ltd.
>
>
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From diego.viola at gmail.com  Tue Aug  4 22:11:24 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 5 Aug 2009 01:11:24 -0400
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <23f91030908042141h9f79d61w86599122677670@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
	<86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>
	<86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>
	<2d8777c00908042133o6e2ffd33g37da13392ac6ccb@mail.gmail.com>
	<23f91030908042141h9f79d61w86599122677670@mail.gmail.com>
Message-ID: <86a32abc0908042211g2e023d1fgd6f8eb25630561d9@mail.gmail.com>

Very cool, thanks guys, you make FreeSWITCH even better =D

Your story rocks!

On Wed, Aug 5, 2009 at 12:41 AM, Seven Du<dujinfang at gmail.com> wrote:
> And I ?added this on the wiki page:
> mod_conference and mod_fifo: We also use FreeSWITCH in our office
> environment as a PBX for call center and customer service connected with
> VoIP and PSTN(openzap) gateways. It is integrated into our CRM system
> naturally and just made sales process, business logic and world wide
> conference much more simpler and easier.
> :)
> 7
>
> 2009/8/5 Jonathan Palley <jpalley at idapted.com>
>>
>> Diego -
>> ??Already done. ?See the bottom of the page (we linked to another page
>> because of?its?length)!
>> :)
>> Jonathan?Palley
>> Idapted?Ltd.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From velu.technical at gmail.com  Tue Aug  4 23:14:55 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Wed, 5 Aug 2009 11:44:55 +0530
Subject: [Freeswitch-users] execute function in ESL.pm module is not working
Message-ID: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>

Dear All,
   I registered alarm signal in my Perl server program.
   If ALARM signal occurred I execute the following statement in signal
handler.
    "$conn->execute("playback",$sound_path."voicemail/vm-goodbye.wav")"
  The above statement didn't play that wave file. But before generating the
ALARM signal it worked.

What is the problem?

Please help me in this problem....

Also Is there any idea to do timeout for DTMF digits?

Thanks...

Regards,
K.Velusamy
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From brad.tuan at gmail.com  Wed Aug  5 03:26:01 2009
From: brad.tuan at gmail.com (Brad Tuan)
Date: Wed, 5 Aug 2009 18:26:01 +0800
Subject: [Freeswitch-users] How to change the contact when fs sending
	REGISTER??
Message-ID: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>

As title ,I know how to do when sending INVITE

but how to do it when fs sending REGISTER??

For example , when gateway registering , the contact is
gw+abcd at XXX.XXX.XXX.XXX ,

how to change it to *abcd at XXX.XXX.XXX.XXX??*
**
*Please help*
**
**
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From gregt at cgicommunications.com  Wed Aug  5 05:54:04 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Wed, 5 Aug 2009 08:54:04 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>
	<E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>
	<6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>
Message-ID: <FE05AA09-CA4B-40E0-BE43-626667FFEDA9@cgicommunications.com>

Thanks so much, William. This gives me a great start.
--
Greg



On Aug 4, 2009, at 7:04 PM, William Suffill wrote:

> I wrote some notes on this but have yet to wiki it.
>
> example of an outbound socket connection where the call is answered, a
> variable is set then perhaps play one of the pre-installed files and
> hangup.
> ivrd
>
> fs_ivrd comes with freeswitch. It being a small daemon just invokes
> the script defined in a variable and passes data from it via STDIN/OUT
>
> Since this is an outbound socket connections it needs to be defined in
> the dialplan.
>
>
>
> Ex:
>
> <extension name="outbound-socket">
>
>    <condition field="destination_number" expression="^55(522)$">
>
>        <action application="set"
> data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
>
>        <action application="socket" data="127.0.0.1:8084 async full"/>
>
>    </condition>
>
>  </extension>
>
>
>
> The above dialplan sample would invoke ivr-demo.php when 55522 is
> called as long as fs_ivrd is running. To start fs_ivrd:
>
> /usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 8004
>
>
>
> It takes 2 arguments -h for hostname and -p for port.
>
>
>
> PHP Code
>
> #!/usr/bin/php -q
>
> <?php
>
>
>
>    // set a couple of things so we dont kill the system
>
>    ob_implicit_flush(true);
>
>    set_time_limit(30);
>
>
>
>
>
>    // Open stdin so we can read the AGI data in
>
>    $in = fopen("php://stdin", "r");
>
>    // Connect
>
>    echo "connect\n\n";
>
>    // Answer
>
>    echo "sendmsg\n";
>
>    echo "call-command: execute\n";
>
>    echo "execute-app-name: answer\n\n";
>
>
>
>    // Play a prompt
>
>    echo "sendmsg\n";
>
>    echo "call-command: execute\n";
>
>    echo "execute-app-name: playback\n";
>
>    echo "execute-app-arg:
> /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr- 
> welcome_to_freeswitch.wav\n\n";
>
>
>
>    // Wait
>
>    sleep(5);
>
>
>
>   // Hangup
>
>   echo "sendmsg\n";
>
>   echo "call-command: hangup\n\n";
>
>
>
> fclose($in);
>
>
>
> ?>
>
>
>
> ivrd will call this script for each call. All itdoes is answer the
> channel tell FreeSWITCH to play the ?welcome to freeswitch? prompt.
> Since the script is now controlling all call flow I needed to add a
> wait or it would send the hangup immediately before the prompt was
> played.
>
> Some improvements possible but that's 1 way to do it. It would be
> possible to do the socket directly in PHP but fs_ivrd is a nice option
> too.
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at dialyourleads.com  Mon Aug  3 12:23:04 2009
From: mike at dialyourleads.com (Michael Frager)
Date: Mon, 3 Aug 2009 15:23:04 -0400
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
Message-ID: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>

Hello,

I'm in the process of moving my VOIP application from Asterisk to
FreeSWITCH.

I was wondering if it is possible to emulate the call announcement feature
that is available on Asterisk.

On Asterisk it looks like this, with the "A(...)" parameter:

Dial(SIP/15555551212|180|A(connecttone1))

Note that this announcement is only played for the called party, the calling
party does NOT hear the tone.

I'm guessing this can be done with FreeSWITCH.

Does anyone know how I might accomplish this?


Thanks in advance,

-Mike Fragre
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From david.nembrot at sogeti.com  Wed Aug  5 01:18:30 2009
From: david.nembrot at sogeti.com (David Nembrot)
Date: Wed,  5 Aug 2009 10:18:30 +0200
Subject: [Freeswitch-users] Qualify IM comm. across two distinct SIP domains
Message-ID: <20090805101830.bn9e3vzjms08oskg@mail.sogeti.com>

Hi everybody,

? ?I've just configured two Freeswitch servers (FS#1 and FS#2) to?
enable SIP communications between their two distinct SIP domains. The?
fact is that the IP telephony is up & running across these two?
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory?
way, my guess is that I'm missing something in the config files in?
order to enable IM services throughout the two domains.. Since they?
are in different networks, it seems reasonable for example to force?
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my?
question:

? ?How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

? ?or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.
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   Hi everybody,

   I've just configured two Freeswitch servers (FS#1 and FS#2) to  
enable SIP communications between their two distinct SIP domains. The  
fact is that the IP telephony is up & running across these two  
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory  
way, my guess is that I'm missing something in the config files in  
order to enable IM services throughout the two domains.. Since they  
are in different networks, it seems reasonable for example to force  
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my  
question:

   How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

   or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.


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From enno.egbert at googlemail.com  Wed Aug  5 01:52:20 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Wed, 5 Aug 2009 01:52:20 -0700 (PDT)
Subject: [Freeswitch-users]  TLS/SRTP with Innovaphone IP200
Message-ID: <24823167.post@talk.nabble.com>


Hello,

i have a problem using a innovaphone ip200 with freeswitch and tls/srtp. The
freeswitch certificate is in the trust list of the phone and it works with
tls for incomming calls. But outgoing calls were rejected to the mailbox.
The freeswitch configuration is ok, because it works with a snom 320.

Who can help me to confugure the IP200?

Thanks

NOx
-- 
View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24823167.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From ovh at 5ca.com  Wed Aug  5 04:22:10 2009
From: ovh at 5ca.com (Otto)
Date: Wed, 5 Aug 2009 08:22:10 -0300
Subject: [Freeswitch-users] Looking for some FreeSWITCH job
Message-ID: <6B7F51E2C75FB1458C416CFEAD0E822F5D9E5B@ml.5CA.INT>

Diego,

 

You can contact me.

 

Br.

 

Otto van Haaren

www.5ca.com <http://www.5ca.com/> 

ovh at 5ca.com

 

________________________________

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Diego Viola
Sent: dinsdag 4 augustus 2009 23:22
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Looking for some FreeSWITCH job

 

Hi,

I'm currently looking for some FS jobs, I really need one, I'm currently
unemployed and looking for some serious FreeSWITCH jobs.

Anyone?

P.S: I also do FS and web development with any language, PHP, Ruby, etc.
anything really.

Thanks,

Diego

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From david.nembrot at sogeti.com  Wed Aug  5 05:32:10 2009
From: david.nembrot at sogeti.com (David Nembrot)
Date: Wed,  5 Aug 2009 14:32:10 +0200
Subject: [Freeswitch-users] Qualify IM comm. across two distinct SIP domains
Message-ID: <20090805143210.um73qlhsgck80kko@mail.sogeti.com>

Hi everybody,

? ?I've just configured two Freeswitch servers (FS#1 and FS#2) to?
enable SIP communications between their two distinct SIP domains. The?
fact is that the IP telephony is up & running across these two?
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory?
way, my guess is that I'm missing something in the config files in?
order to enable IM services throughout the two domains.. Since they?
are in different networks, it seems reasonable for example to force?
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my?
question:

? ?How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

? ?or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.
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-------------- next part --------------
Hi everybody,

? ?I've just configured two Freeswitch servers (FS#1 and FS#2) to?
enable SIP communications between their two distinct SIP domains. The?
fact is that the IP telephony is up & running across these two?
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory?
way, my guess is that I'm missing something in the config files in?
order to enable IM services throughout the two domains.. Since they?
are in different networks, it seems reasonable for example to force?
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my?
question:

? ?How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

? ?or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.
-------------- next part --------------
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-------------- next part --------------


   Hi everybody,

   I've just configured two Freeswitch servers (FS#1 and FS#2) to  
enable SIP communications between their two distinct SIP domains. The  
fact is that the IP telephony is up & running across these two  
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory  
way, my guess is that I'm missing something in the config files in  
order to enable IM services throughout the two domains.. Since they  
are in different networks, it seems reasonable for example to force  
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my  
question:

   How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

   or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.


-------------- next part --------------
[Pi??ce jointe retir??e??: type d'origine de la pi??ce jointe: "text/html", nom: "Version_HTML_du_message"]

From dujinfang at gmail.com  Wed Aug  5 06:15:32 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 5 Aug 2009 21:15:32 +0800
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
In-Reply-To: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
References: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
Message-ID: <EC77D164-BAAD-48FA-A3C4-6BFE05CC2E8F@gmail.com>

I think you can check the loopback endpoint or inline dialplan.

On Aug 4, 2009, at 3:23 AM, Michael Frager wrote:
> Hello,
>
> I'm in the process of moving my VOIP application from Asterisk to  
> FreeSWITCH.
>
> I was wondering if it is possible to emulate the call announcement  
> feature that is available on Asterisk.
>
> On Asterisk it looks like this, with the "A(...)" parameter:
>
> Dial(SIP/15555551212|180|A(connecttone1))
>
> Note that this announcement is only played for the called party, the  
> calling party does NOT hear the tone.
>
> I'm guessing this can be done with FreeSWITCH.
>
> Does anyone know how I might accomplish this?
>
>
> Thanks in advance,
>
> -Mike Fragre
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From pjintheusa at gmail.com  Wed Aug  5 06:20:16 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 5 Aug 2009 09:20:16 -0400
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
In-Reply-To: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
References: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
Message-ID: <367751820908050620t40566a84y8422a132142e3c34@mail.gmail.com>

Mike,

I am not familiar with Asterisk so I am not 100% sure this is what you are
looking for, but check out
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation

It may contain what you need

Phillip Jones


On Mon, Aug 3, 2009 at 3:23 PM, Michael Frager <mike at dialyourleads.com>wrote:

> Hello,
>
> I'm in the process of moving my VOIP application from Asterisk to
> FreeSWITCH.
>
> I was wondering if it is possible to emulate the call announcement feature
> that is available on Asterisk.
>
> On Asterisk it looks like this, with the "A(...)" parameter:
>
> Dial(SIP/15555551212|180|A(connecttone1))
>
> Note that this announcement is only played for the called party, the
> calling party does NOT hear the tone.
>
> I'm guessing this can be done with FreeSWITCH.
>
> Does anyone know how I might accomplish this?
>
>
> Thanks in advance,
>
> -Mike Fragre
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Aug  5 06:22:52 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 08:22:52 -0500
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <24823167.post@talk.nabble.com>
References: <24823167.post@talk.nabble.com>
Message-ID: <DF06C882-055F-488D-B860-46643AF9B22B@freeswitch.org>

Can you ship me a phone to test with?  That's usually the missing  
element when testing this stuff is I just can't afford to buy every  
phone to test with.

/b

On Aug 5, 2009, at 3:52 AM, NOx-WHV wrote:

>
> Hello,
>
> i have a problem using a innovaphone ip200 with freeswitch and tls/ 
> srtp. The
> freeswitch certificate is in the trust list of the phone and it  
> works with
> tls for incomming calls. But outgoing calls were rejected to the  
> mailbox.
> The freeswitch configuration is ok, because it works with a snom 320.
>
> Who can help me to confugure the IP200?
>
> Thanks
>
> NOx
> -- 



From brian at freeswitch.org  Wed Aug  5 06:23:48 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 08:23:48 -0500
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
In-Reply-To: <367751820908050620t40566a84y8422a132142e3c34@mail.gmail.com>
References: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
	<367751820908050620t40566a84y8422a132142e3c34@mail.gmail.com>
Message-ID: <488F9396-69AF-4A50-943D-DD554F946741@freeswitch.org>

That is exactly what he's lookin for.

/b

On Aug 5, 2009, at 8:20 AM, Phillip Jones wrote:

> Mike,
>
> I am not familiar with Asterisk so I am not 100% sure this is what  
> you are looking for, but check out http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
>
> It may contain what you need
>
> Phillip Jones

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From msc at freeswitch.org  Wed Aug  5 06:40:52 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 08:40:52 -0500
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
In-Reply-To: <488F9396-69AF-4A50-943D-DD554F946741@freeswitch.org>
References: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
	<367751820908050620t40566a84y8422a132142e3c34@mail.gmail.com>
	<488F9396-69AF-4A50-943D-DD554F946741@freeswitch.org>
Message-ID: <87f2f3b90908050640x5ec18d4ag417f38e31108b301@mail.gmail.com>

Could someone please add this to the FreeSWITCH wiki's Rosetta Stone page?
Thanks!-MC

On Wed, Aug 5, 2009 at 8:23 AM, Brian West <brian at freeswitch.org> wrote:

> That is exactly what he's lookin for.
> /b
>
> On Aug 5, 2009, at 8:20 AM, Phillip Jones wrote:
>
> Mike,
>
> I am not familiar with Asterisk so I am not 100% sure this is what you are
> looking for, but check out
> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
>
> It may contain what you need
>
> Phillip Jones
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From max.bridgewater at gmail.com  Wed Aug  5 06:56:39 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Wed, 5 Aug 2009 09:56:39 -0400
Subject: [Freeswitch-users] How to distinguish between the different type of
	call rejections from Javascript?
Message-ID: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>

Hi,

Say i originate a call to a mobile phone and the call fails. There are many
possible reasons: congestion, user busy, call rejected by user, etc. Is
there a way i can get the failure code from Javascript?
Thanks,
Max.
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From dujinfang at gmail.com  Wed Aug  5 07:02:50 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 5 Aug 2009 22:02:50 +0800
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
Message-ID: <D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>

variable_originate_disposition

On Aug 5, 2009, at 9:56 PM, Max Bridgewater wrote:
> Hi,
>
> Say i originate a call to a mobile phone and the call fails. There  
> are many possible reasons: congestion, user busy, call rejected by  
> user, etc. Is there a way i can get the failure code from Javascript?
> Thanks,
> Max.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jim at evolutiontel.net  Wed Aug  5 07:07:50 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Thu, 6 Aug 2009 00:07:50 +1000
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <24823167.post@talk.nabble.com>
References: <24823167.post@talk.nabble.com>
Message-ID: <ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>

Hi NOx,

Can you clarify the direction of the calls.  When you say outgoing do
you mean a call is terminating to the ip200?

I have been down a similar path while testing Eyebeam.  If the
terminating phone sets an option to only accept secure calls and FS
does not send Secure Descriptions in the INVITE, Eyebeam would respond
with 415 response code and the call would fail.  Depending on your
diaplan this could send your call to voicemail.

To fix it I added the following code to dialplan.
<action application="set" data="continue_on_fail=79"/>
<bridge blah blah blah>
<action application="set" data="bypass_media=false"/>
<action application="set" data="proxy_media=true"/>
<action application="set" data="ringback=$${uk-ring}"/>
<action application="pre_answer"/>
<action application="export" data="sip_secure_media=true"/>
<bridge blah blah blah>

The continue on fail captures the 415 response code forces the call to
continue to the next bridge while sip_secure_media forces the second
invite to include security descriptors.  The rest was required because
I did not want to proxy media if the call was not secure, obviously if
the call is secure on a point to point basis FS will have to proxy the
media and this was the only way I could find for it to work.

Hope this helps.

Regards,


On Wed, Aug 5, 2009 at 6:52 PM, NOx-WHV<enno.egbert at googlemail.com> wrote:
>
> Hello,
>
> i have a problem using a innovaphone ip200 with freeswitch and tls/srtp. The
> freeswitch certificate is in the trust list of the phone and it works with
> tls for incomming calls. But outgoing calls were rejected to the mailbox.
> The freeswitch configuration is ok, because it works with a snom 320.
>
> Who can help me to confugure the IP200?
>
> Thanks
>
> NOx
> --
> View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24823167.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From gregt at cgicommunications.com  Wed Aug  5 07:40:31 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Wed, 5 Aug 2009 10:40:31 -0400
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
Message-ID: <63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>

Trying to make phpmod and it fails with this:

/usr/bin/ld: cannot find -laspell
collect2: ld returned 1 exit status
make[1]: *** [ESL.so] Error 1

I do have php-devel on this Centos 5.2 machine. Any ideas?
--
Greg
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From brian at freeswitch.org  Wed Aug  5 07:46:51 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 09:46:51 -0500
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
Message-ID: <2B197B9C-9854-4888-8726-0402C2F0689A@freeswitch.org>

install aspell-devel

also Please don't hijack threads... please click new message and start  
a new thread.

Thanks,
Brian

On Aug 5, 2009, at 9:40 AM, Greg Thoen wrote:

> Trying to make phpmod and it fails with this:
>
> /usr/bin/ld: cannot find -laspell
> collect2: ld returned 1 exit status
> make[1]: *** [ESL.so] Error 1
>
> I do have php-devel on this Centos 5.2 machine. Any ideas?
> --
> Greg

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From william.suffill at gmail.com  Wed Aug  5 07:47:38 2009
From: william.suffill at gmail.com (William Suffill)
Date: Wed, 5 Aug 2009 10:47:38 -0400
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> 
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> 
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca> 
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com> 
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
Message-ID: <6b65470d0908050747r46da5b14q6ac3f955ad110e71@mail.gmail.com>

You don't have the dev library for aspell

yum install aspell-devel

should do the tick.

-- W


From msc at freeswitch.org  Wed Aug  5 07:54:45 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 5 Aug 2009 09:54:45 -0500
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>
References: <24823167.post@talk.nabble.com>
	<ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>
Message-ID: <95500CDE-C8A9-4F37-958D-43F27B5A6B3F@freeswitch.org>

Jim,

Just curious - could you document this use case on the wiki? Maybe you  
could create a page describing the setup and then link to it from the  
TLS page.

Thanks,
MC

Sent from my iPhone

On Aug 5, 2009, at 9:07 AM, Jim Burke <jim at evolutiontel.net> wrote:

> Hi NOx,
>
> Can you clarify the direction of the calls.  When you say outgoing do
> you mean a call is terminating to the ip200?
>
> I have been down a similar path while testing Eyebeam.  If the
> terminating phone sets an option to only accept secure calls and FS
> does not send Secure Descriptions in the INVITE, Eyebeam would respond
> with 415 response code and the call would fail.  Depending on your
> diaplan this could send your call to voicemail.
>
> To fix it I added the following code to dialplan.
> <action application="set" data="continue_on_fail=79"/>
> <bridge blah blah blah>
> <action application="set" data="bypass_media=false"/>
> <action application="set" data="proxy_media=true"/>
> <action application="set" data="ringback=$${uk-ring}"/>
> <action application="pre_answer"/>
> <action application="export" data="sip_secure_media=true"/>
> <bridge blah blah blah>
>
> The continue on fail captures the 415 response code forces the call to
> continue to the next bridge while sip_secure_media forces the second
> invite to include security descriptors.  The rest was required because
> I did not want to proxy media if the call was not secure, obviously if
> the call is secure on a point to point basis FS will have to proxy the
> media and this was the only way I could find for it to work.
>
> Hope this helps.
>
> Regards,
>
>
> On Wed, Aug 5, 2009 at 6:52 PM, NOx-WHV<enno.egbert at googlemail.com>  
> wrote:
>>
>> Hello,
>>
>> i have a problem using a innovaphone ip200 with freeswitch and tls/ 
>> srtp. The
>> freeswitch certificate is in the trust list of the phone and it  
>> works with
>> tls for incomming calls. But outgoing calls were rejected to the  
>> mailbox.
>> The freeswitch configuration is ok, because it works with a snom 320.
>>
>> Who can help me to confugure the IP200?
>>
>> Thanks
>>
>> NOx
>> --
>> View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24823167.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From helmut.kuper at ewetel.de  Wed Aug  5 07:56:30 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 05 Aug 2009 16:56:30 +0200
Subject: [Freeswitch-users] pocketsphinx
In-Reply-To: <87f2f3b90907311145p6c8d5907mc4545c710e9605fb@mail.gmail.com>
References: <4A5739DE.1080800@ewetel.de> <4A72EF33.4070404@ewetel.de>
	<87f2f3b90907311145p6c8d5907mc4545c710e9605fb@mail.gmail.com>
Message-ID: <4A799D9E.2020103@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

today I put my documentation on FS wiki describing the steps to get a
8kHz sample rate acoustic model basing on voxforge's data for german
language. It's not complete, yet.

You can found it here: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx

regards
Helmut


On 31.07.2009 20:45, Michael Collins wrote:
> Helmut,
> 
> Your hard work is appreciated. Like Brian said, we'd all be interested
> in knowing more. Please feel free to put this on the wiki or see me off
> list and we'll discuss further how to document it for the good of the FS
> community.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFKeZ2e4tZeNddg3dwRAlwuAKCXK6b/f3J7tRmcev0/EPAUFGZBbgCfXMQW
B8MAREKeR82dTFnYyFeutig=
=1/Jr
-----END PGP SIGNATURE-----


From woodydickson at gmail.com  Wed Aug  5 08:05:20 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 5 Aug 2009 23:05:20 +0800
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
Message-ID: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>

Hi,

I want to implement a module where freeSWITCH would try to bridge to an
extension and if the bridging operation fails, my module can use the hangup
code to determine the next cause of action.

With switch_caller_extension_add_application(session, extension, "bridge",
"sofia/gateway/mygw/1232323);, if there is an error ( 503 received for
instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or the
module's APP) and go on to the next action.  Is there anyway to control it
so that freeSWITCH would remain to be within the module's APP funtion and
continue executing the code after switch_call_extension_add_application,
when let's say a 4XX or 5XX or CANCEL ( from originator) is received?

I have tried it and found that if the bridging is successful, freeSWITCH
would continue executing the code after
switch_caller_extension_add_application, but if an error is received, then
it would just move on to the next action.

Does anyone know how to deal with this problem?

Woody
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From gregt at cgicommunications.com  Wed Aug  5 08:21:41 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Wed, 5 Aug 2009 11:21:41 -0400
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <2B197B9C-9854-4888-8726-0402C2F0689A@freeswitch.org>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
	<2B197B9C-9854-4888-8726-0402C2F0689A@freeswitch.org>
Message-ID: <BFC0A2A3-F0D5-40DC-8E1B-AA2FB2FFB372@cgicommunications.com>

Oops, what thread did I hijack? I just sent a new email message to freeswitch-users at lists.freeswitch.org
What should I have done?
--
Greg


On Aug 5, 2009, at 10:46 AM, Brian West wrote:

> install aspell-devel
>
> also Please don't hijack threads... please click new message and  
> start a new thread.
>
> Thanks,
> Brian
>
> On Aug 5, 2009, at 9:40 AM, Greg Thoen wrote:
>
>> Trying to make phpmod and it fails with this:
>>
>> /usr/bin/ld: cannot find -laspell
>> collect2: ld returned 1 exit status
>> make[1]: *** [ESL.so] Error 1
>>
>> I do have php-devel on this Centos 5.2 machine. Any ideas?
>> --
>> Greg
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Wed Aug  5 08:39:31 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 10:39:31 -0500
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <BFC0A2A3-F0D5-40DC-8E1B-AA2FB2FFB372@cgicommunications.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
	<2B197B9C-9854-4888-8726-0402C2F0689A@freeswitch.org>
	<BFC0A2A3-F0D5-40DC-8E1B-AA2FB2FFB372@cgicommunications.com>
Message-ID: <44BA0B61-3971-479B-816C-5501035DB28D@freeswitch.org>

Re: [Freeswitch-users] ESL Wrapper



It happens when you click reply... change the subject and the body...  
thats how you hijack a thread.  :)  Happens to the best of us.



/b





On Aug 5, 2009, at 10:21 AM, Greg Thoen wrote:

> Oops, what thread did I hijack? I just sent a new email message to freeswitch-users at lists.freeswitch.org
> What should I have done?
> --
> Greg
>
>
> On Aug 5, 2009, at 10:46 AM, Brian West wrote:
>
>> install aspell-devel
>>
>> also Please don't hijack threads... please click new message and  
>> start a new thread.
>>
>> Thanks,
>> Brian
>>
>> On Aug 5, 2009, at 9:40 AM, Greg Thoen wrote:
>>
>>> Trying to make phpmod and it fails with this:
>>>
>>> /usr/bin/ld: cannot find -laspell
>>> collect2: ld returned 1 exit status
>>> make[1]: *** [ESL.so] Error 1
>>>
>>> I do have php-devel on this Centos 5.2 machine. Any ideas?
>>> --
>>> Greg
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jmesquita at gmail.com  Wed Aug  5 09:15:30 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 5 Aug 2009 13:15:30 -0300
Subject: [Freeswitch-users] How to change the contact when fs sending
	REGISTER??
In-Reply-To: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>
References: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>
Message-ID: <5a8712120908050915k2f414a37vd651f9323114cdbf@mail.gmail.com>

Add the following line to the gw definition:

<param name="extension-in-contact" value="true"/>

jmesquita

On Wed, Aug 5, 2009 at 7:26 AM, Brad Tuan <brad.tuan at gmail.com> wrote:

> As title ,I know how to do when sending INVITE
>
> but how to do it when fs sending REGISTER??
>
> For example , when gateway registering , the contact is
> gw+abcd at XXX.XXX.XXX.XXX ,
>
> how to change it to *abcd at XXX.XXX.XXX.XXX??*
> **
> *Please help*
> **
> **
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at gmail.com  Wed Aug  5 09:23:26 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 5 Aug 2009 13:23:26 -0300
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
Message-ID: <5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>

My guess is that you will receive a message here:

switch_status_t channel_receive_message(switch_core_session_t *session,
switch_core_session_message_t *msg)

The problem here is that you don't have the exact SIP code but there is a
clear relationship between the codes and the messages you receive on the
channel, so I am guessing that is all the same.

Hope this helps.

jmesquita

On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> I want to implement a module where freeSWITCH would try to bridge to an
> extension and if the bridging operation fails, my module can use the hangup
> code to determine the next cause of action.
>
> With switch_caller_extension_add_application(session, extension, "bridge",
> "sofia/gateway/mygw/1232323);, if there is an error ( 503 received for
> instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or the
> module's APP) and go on to the next action.  Is there anyway to control it
> so that freeSWITCH would remain to be within the module's APP funtion and
> continue executing the code after switch_call_extension_add_application,
> when let's say a 4XX or 5XX or CANCEL ( from originator) is received?
>
> I have tried it and found that if the bridging is successful, freeSWITCH
> would continue executing the code after
> switch_caller_extension_add_application, but if an error is received, then
> it would just move on to the next action.
>
> Does anyone know how to deal with this problem?
>
> Woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Wed Aug  5 09:34:50 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 5 Aug 2009 11:34:50 -0500
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
Message-ID: <E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>

The hangup cause will be in the originate_disposition channel variable  
on the A-leg.

sip_term_status will contain the sip code and  
proto_specific_hangup_cause will contain sip:<code>.


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 5-Aug-09 um 11:23 AM schrieb Jo?o Mesquita:

> My guess is that you will receive a message here:
>
> switch_status_t channel_receive_message(switch_core_session_t  
> *session, switch_core_session_message_t *msg)
>
> The problem here is that you don't have the exact SIP code but there  
> is a clear relationship between the codes and the messages you  
> receive on the channel, so I am guessing that is all the same.
>
> Hope this helps.
>
> jmesquita
>
> On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson  
> <woodydickson at gmail.com> wrote:
> Hi,
>
> I want to implement a module where freeSWITCH would try to bridge to  
> an extension and if the bridging operation fails, my module can use  
> the hangup code to determine the next cause of action.
>
> With switch_caller_extension_add_application(session, extension,  
> "bridge", "sofia/gateway/mygw/1232323);, if there is an error ( 503  
> received for instance ) in the outgoing INVITE, freeSWITCH would  
> leave my module ( or the module's APP) and go on to the next  
> action.  Is there anyway to control it so that freeSWITCH would  
> remain to be within the module's APP funtion and continue executing  
> the code after switch_call_extension_add_application, when let's say  
> a 4XX or 5XX or CANCEL ( from originator) is received?
>
> I have tried it and found that if the bridging is successful,  
> freeSWITCH would continue executing the code after  
> switch_caller_extension_add_application, but if an error is  
> received, then it would just move on to the next action.
>
> Does anyone know how to deal with this problem?
>
> Woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mayamatakeshi at gmail.com  Wed Aug  5 09:35:38 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 6 Aug 2009 01:35:38 +0900
Subject: [Freeswitch-users] Monitoring On-Hold/Off-Hold
Message-ID: <15b9404e0908050935m62a6581bxec3f3055e013753f@mail.gmail.com>

Hello,
I'm using mod_event_socket to monitor FS.
I'm using "events plain ALL' and I get lots of channel events. But
curiously, when some channel puts the call on-hold/off-hold, I don't
get any notification. Is it possible to get these events? Am I missing
some setting?

regards,
takeshi


From gregt at cgicommunications.com  Wed Aug  5 10:31:23 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Wed, 5 Aug 2009 13:31:23 -0400
Subject: [Freeswitch-users] Best practice for inbound calls with scripting
Message-ID: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>

Hi. I am setting up a large inbound only system with multiple DIDs  
coming in; each call is processed fairly intensively with db lookups,  
wav files played, pocketsphinx is used, wav files recorded, etc.

Before I get too far down one path, I was wondering if anyone had any  
insight into the best, most scaleable way to do this out of the  
several methods I can do:

1	Call comes in
	dialplan calls specific javascript based on DID
	javascript uses ODBC to pull info from local mysql db
	call is handled in javascript

2	Call comes in
	dialplan calls specific javascript based on DID
	javascript uses CURL to get info from local mysql db
	call is handled in javascript

3	Call comes in
	php socket is listening for call
	php script runs, pulling info from mysql db
	call is handled in php using esl.php

4	Call comes in
	dialplan calls specific lua script  based on DID
	lua uses luasql.mysql to get info from local mysql db
	call is handled in lua using lua api

5	Call comes in
	xml_curl is used for dynamic dialplan
	js called, continues like #1

I know that they will all do essentially the same thing. But once I go  
down the path, I don't want to find out that the way I chose chokes  
with 50 simultaneous inbound calls on different DIDs. Any comments  
would be appreciated. Thanks.
--
Greg


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From jmesquita at gmail.com  Wed Aug  5 10:55:37 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 5 Aug 2009 14:55:37 -0300
Subject: [Freeswitch-users] Monitoring On-Hold/Off-Hold
In-Reply-To: <15b9404e0908050935m62a6581bxec3f3055e013753f@mail.gmail.com>
References: <15b9404e0908050935m62a6581bxec3f3055e013753f@mail.gmail.com>
Message-ID: <5a8712120908051055k95f01d4x61ef7b7dff80298a@mail.gmail.com>

I only see one way out of this. If you manage presence, an event like the
following is sent:

Event-Name: PRESENCE_IN
Core-UUID: 189b12c0-7fb0-11de-b0bc-37eec03ad00f
FreeSWITCH-Hostname: cl-t146-421cl
FreeSWITCH-IPv4: XXXXXX
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-05%2013%3A42%3A24
Event-Date-GMT: Wed,%2005%20Aug%202009%2017%3A42%3A24%20GMT
Event-Date-Timestamp: 1249494144628132
Event-Calling-File: switch_channel.c
Event-Calling-Function: switch_channel_presence
Event-Calling-Line-Number: 472
Channel-State: CS_HIBERNATE
Channel-State-Number: 8
Channel-Name: XXXXX
Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f
Call-Direction: inbound
Presence-Call-Direction: inbound
Answer-State: answered
Caller-Username: 1000
Caller-Dialplan: XML
Caller-Caller-ID-Name: Mesquita
Caller-Caller-ID-Number: 1000
Caller-Network-Addr: XXXXX
Caller-Destination-Number: 1005
Caller-Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f
Caller-Source: mod_sofia
Caller-Context: XXXXX
Caller-Channel-Name: XXXXX
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249494132128119
Caller-Channel-Created-Time: 1249494132128119
Caller-Channel-Answered-Time: 1249494139500129
Caller-Channel-Progress-Time: 1249494132368119
Caller-Channel-Progress-Media-Time: 0
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Other-Leg-Username: 1000
Other-Leg-Dialplan: XML
Other-Leg-Caller-ID-Name: Joao%20Mesquita
Other-Leg-Caller-ID-Number: 1000
Other-Leg-Network-Addr: 190.2.41.65
Other-Leg-Destination-Number:
sip%3A1005%40192.168.0.106%3A4559%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1005%2540190.2.41.65%253A4559
Other-Leg-Unique-ID: 4e7622ac-81e7-11de-b0bc-37eec03ad00f
Other-Leg-Source: mod_sofia
Other-Leg-Context: XXXXX
Other-Leg-Channel-Name: XXXXXX
Other-Leg-Screen-Bit: true
Other-Leg-Privacy-Hide-Name: false
Other-Leg-Privacy-Hide-Number: false
proto: src/switch_channel.c
login: src/switch_channel.c
from: XXXXXX
rpid: unknown
status: hold
event_type: presence
alt_event_type: dialog
event_count: 3

Content-Length: 543
Content-Type: text/event-plain

Other than that, I think it can be patched. I will take a look at it.

Guys, should this be patched on the state machine itself or on the mod_sofia
channel_receive_message?

jmesquita


On Wed, Aug 5, 2009 at 1:35 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> Hello,
> I'm using mod_event_socket to monitor FS.
> I'm using "events plain ALL' and I get lots of channel events. But
> curiously, when some channel puts the call on-hold/off-hold, I don't
> get any notification. Is it possible to get these events? Am I missing
> some setting?
>
> regards,
> takeshi
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dule.maillist at gmail.com  Wed Aug  5 10:59:38 2009
From: dule.maillist at gmail.com (Dan Le)
Date: Wed, 5 Aug 2009 13:59:38 -0400
Subject: [Freeswitch-users] Best practice for inbound calls with
	scripting
In-Reply-To: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
References: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
Message-ID: <914fc92a0908051059na35f58ar3f102ef4d80e0027@mail.gmail.com>

One common recommendation is to use lua over js, since it's lighter-weight,
using less resources.
Dan

On Wed, Aug 5, 2009 at 1:31 PM, Greg Thoen <gregt at cgicommunications.com>wrote:

> Hi. I am setting up a large inbound only system with multiple DIDs coming
> in; each call is processed fairly intensively with db lookups, wav files
> played, pocketsphinx is used, wav files recorded, etc.
> Before I get too far down one path, I was wondering if anyone had any
> insight into the best, most scaleable way to do this out of the several
> methods I can do:
> 1 Call comes in
> dialplan calls specific javascript based on DID
> javascript uses ODBC to pull info from local mysql db
> call is handled in javascript
>
> 2 Call comes in
> dialplan calls specific javascript based on DID
> javascript uses CURL to get info from local mysql db
> call is handled in javascript
>
> 3 Call comes in
> php socket is listening for call
> php script runs, pulling info from mysql db
> call is handled in php using esl.php
>
> 4 Call comes in
> dialplan calls specific lua script  based on DID
> lua uses luasql.mysql to get info from local mysql db
> call is handled in lua using lua api
>
> 5 Call comes in
> xml_curl is used for dynamic dialplan
> js called, continues like #1
>
> I know that they will all do
> essentially the same thing. But once I go down the path, I don't want to find out that the way I chose chokes with 50 simultaneous inbound calls on different DIDs. Any comments would be appreciated. Thanks.
>
> --
>
> Greg
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From technical at ttnc.co.uk  Wed Aug  5 11:20:46 2009
From: technical at ttnc.co.uk (TTNC - Adnan Barakat)
Date: Wed, 05 Aug 2009 19:20:46 +0100
Subject: [Freeswitch-users] Best practice for inbound calls with
	scripting
In-Reply-To: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
References: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
Message-ID: <4A79CD7E.6010806@ttnc.co.uk>

Greg Thoen wrote:
> 4 Call comes in
> dialplan calls specific lua script  based on DID
> lua uses luasql.mysql to get info from local mysql db
> call is handled in lua using lua api
This is the option we chose in our set-up (although mysql is on a remote 
server), currently we have over 25k DIDs running on FreeSWITCH via this 
set-up and everything is working wonderfully.

We were using option 5 previously, but found 4 to be much better in general.

Hope this helps.


Adnan


From edpimentl at gmail.com  Wed Aug  5 11:48:16 2009
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 5 Aug 2009 14:48:16 -0400
Subject: [Freeswitch-users] Best practice for inbound calls with
	scripting
In-Reply-To: <914fc92a0908051059na35f58ar3f102ef4d80e0027@mail.gmail.com>
References: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
	<914fc92a0908051059na35f58ar3f102ef4d80e0027@mail.gmail.com>
Message-ID: <9dc4a1670908051148g38958616j57c2a7eebf9f607f@mail.gmail.com>

Here are other scenarios to add

Call request comes in via SMS
Call request comes in via XMPP
Call request comes in via EMAIL
Call request comes in via TWITTER/FACEBOOK/WAVE/

Best regards,
-E
Gpro.ws
edpimentl [SKype ]



On Wed, Aug 5, 2009 at 1:59 PM, Dan Le <dule.maillist at gmail.com> wrote:

> One common recommendation is to use lua over js, since it's lighter-weight,
> using less resources.
> Dan
>
> On Wed, Aug 5, 2009 at 1:31 PM, Greg Thoen <gregt at cgicommunications.com>wrote:
>
>> Hi. I am setting up a large inbound only system with multiple DIDs coming
>> in; each call is processed fairly intensively with db lookups, wav files
>> played, pocketsphinx is used, wav files recorded, etc.
>> Before I get too far down one path, I was wondering if anyone had any
>> insight into the best, most scaleable way to do this out of the several
>> methods I can do:
>> 1 Call comes in
>>  dialplan calls specific javascript based on DID
>> javascript uses ODBC to pull info from local mysql db
>> call is handled in javascript
>>
>> 2 Call comes in
>> dialplan calls specific javascript based on DID
>> javascript uses CURL to get info from local mysql db
>>  call is handled in javascript
>>
>> 3 Call comes in
>> php socket is listening for call
>>  php script runs, pulling info from mysql db
>> call is handled in php using esl.php
>>
>> 4 Call comes in
>>  dialplan calls specific lua script  based on DID
>> lua uses luasql.mysql to get info from local mysql db
>> call is handled in lua using lua api
>>
>> 5 Call comes in
>> xml_curl is used for dynamic dialplan
>> js called, continues like #1
>>
>> I know that they will all do
>> essentially the same thing. But once I go down the path, I don't want to find out that the way I chose chokes with 50 simultaneous inbound calls on different DIDs. Any comments would be appreciated. Thanks.
>>
>> --
>>
>> Greg
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kjoseph.us at gmail.com  Wed Aug  5 13:24:50 2009
From: kjoseph.us at gmail.com (Joseph Khoury)
Date: Wed, 5 Aug 2009 13:24:50 -0700
Subject: [Freeswitch-users] Looking for some FreeSWITCH job
In-Reply-To: <6B7F51E2C75FB1458C416CFEAD0E822F5D9E5B@ml.5CA.INT>
References: <6B7F51E2C75FB1458C416CFEAD0E822F5D9E5B@ml.5CA.INT>
Message-ID: <7d1481c30908051324j5b806cb6y527162037470fead@mail.gmail.com>

Diego,
Contact me also.

Joseph

www.alosmart.com
(myname) at alosmart.com

On Wed, Aug 5, 2009 at 4:22 AM, Otto <ovh at 5ca.com> wrote:

>  Diego,
>
>
>
> You can contact me.
>
>
>
> Br.
>
>
>
> Otto van Haaren
>
> www.5ca.com
>
> ovh at 5ca.com
>
>
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
> *Sent:* dinsdag 4 augustus 2009 23:22
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Looking for some FreeSWITCH job
>
>
>
> Hi,
>
> I'm currently looking for some FS jobs, I really need one, I'm currently
> unemployed and looking for some serious FreeSWITCH jobs.
>
> Anyone?
>
> P.S: I also do FS and web development with any language, PHP, Ruby, etc.
> anything really.
>
> Thanks,
>
> Diego
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tomabroad at gmail.com  Wed Aug  5 11:46:37 2009
From: tomabroad at gmail.com (tom)
Date: Wed, 5 Aug 2009 14:46:37 -0400
Subject: [Freeswitch-users] q: fresh installation: Couldnt open
	/usr/local/freeswitch/conf/freeswitch.xml
Message-ID: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>

hi just installed freeswitch via svn.
- bootstrap
- configure
- make install
- ./freeswitch

gives me:
acerdebian:/usr/local/freeswitch/bin# ./freeswitch
Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch
-waste.
auto-adjusting stack size for optimal performance....
2009-08-05 14:41:16.908170 [INFO] switch_event.c:565 Activate Eventing
Engine.
2009-08-05 14:41:16.910182 [DEBUG] switch_event.c:553 Create event dispatch
thread 0
2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open
/usr/local/freeswitch/conf/freeswitch.xml (No such file or directory)
Cannot Initialize [Cannot Open log directory or XML Root!]


bump - help

thx
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From mattdfong at gmail.com  Wed Aug  5 13:52:13 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Wed, 5 Aug 2009 13:52:13 -0700
Subject: [Freeswitch-users] q: fresh installation: Couldnt open
	/usr/local/freeswitch/conf/freeswitch.xml
In-Reply-To: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
References: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
Message-ID: <4256bf830908051352t26753d57nbeb6ba27f2a5f77a@mail.gmail.com>

Does the file exist at /usr/local/freeswitch/conf/freeswitch.xml? does the
user you are executing freeswitch as have permission to read the file?
--matt
hello hunter - hosted predictive dialer & voice broadcasting
http://www.hellohunter.com

On Wed, Aug 5, 2009 at 11:46 AM, tom <tomabroad at gmail.com> wrote:

> hi just installed freeswitch via svn.
> - bootstrap
> - configure
> - make install
> - ./freeswitch
>
> gives me:
> acerdebian:/usr/local/freeswitch/bin# ./freeswitch
> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
> ./freeswitch -waste.
> auto-adjusting stack size for optimal performance....
> 2009-08-05 14:41:16.908170 [INFO] switch_event.c:565 Activate Eventing
> Engine.
> 2009-08-05 14:41:16.910182 [DEBUG] switch_event.c:553 Create event dispatch
> thread 0
> 2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt
> open /usr/local/freeswitch/conf/freeswitch.xml (No such file or directory)

Cannot Initialize [Cannot Open log directory or XML Root!]
>
>
> bump - help
>
> thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sprice at gmail.com  Wed Aug  5 13:53:16 2009
From: sprice at gmail.com (SP)
Date: Wed, 5 Aug 2009 15:53:16 -0500
Subject: [Freeswitch-users] q: fresh installation: Couldnt open
	/usr/local/freeswitch/conf/freeswitch.xml
In-Reply-To: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
References: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
Message-ID: <7e2ac3270908051353m33c3f158r4bd7ec2d415f7a7a@mail.gmail.com>

2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open
/usr/local/freeswitch/conf/freeswitch.xml *(No such file or directory)*

On Wed, Aug 5, 2009 at 13:46, tom<tomabroad at gmail.com> wrote:
> hi just installed freeswitch via svn.
> - bootstrap
> - configure
> - make install
> - ./freeswitch
>
> gives me:
> acerdebian:/usr/local/freeswitch/bin# ./freeswitch
> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
./freeswitch
> -waste.
> auto-adjusting stack size for optimal performance....
> 2009-08-05 14:41:16.908170 [INFO] switch_event.c:565 Activate Eventing
> Engine.
> 2009-08-05 14:41:16.910182 [DEBUG] switch_event.c:553 Create event
dispatch
> thread 0
> 2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open
> /usr/local/freeswitch/conf/freeswitch.xml (No such file or directory)
> Cannot Initialize [Cannot Open log directory or XML Root!]
>
>
> bump - help
>
> thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Shannon
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From raffaele.p.guidi at gmail.com  Wed Aug  5 13:54:46 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Wed, 5 Aug 2009 22:54:46 +0200
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
Message-ID: <b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>

interesting! what values can contain "variable_originate_disposition"? And
can I set them manually in a script to reject a call simulating user busy or
call rejected? A lua example?
Thanks,
   Raffaele

On Wed, Aug 5, 2009 at 16:02, Seven Du <dujinfang at gmail.com> wrote:

> variable_originate_disposition
>
> On Aug 5, 2009, at 9:56 PM, Max Bridgewater wrote:
> > Hi,
> >
> > Say i originate a call to a mobile phone and the call fails. There
> > are many possible reasons: congestion, user busy, call rejected by
> > user, etc. Is there a way i can get the failure code from Javascript?
> > Thanks,
> > Max.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Aug  5 13:59:07 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 15:59:07 -0500
Subject: [Freeswitch-users] How to change the contact when fs sending
	REGISTER??
In-Reply-To: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>
References: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>
Message-ID: <978FA06A-8591-44DF-BC08-5CAC2668F6ED@freeswitch.org>

My first question is why would you have to change it?  :P

/b

On Aug 5, 2009, at 5:26 AM, Brad Tuan wrote:

> As title ,I know how to do when sending INVITE
>
> but how to do it when fs sending REGISTER??
>
> For example , when gateway registering , the contact is gw+abcd at XXX.XXX.XXX.XXX 
>  ,
>
> how to change it to abcd at XXX.XXX.XXX.XXX??
>
> Please help
>

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From gmaruzz at celliax.org  Wed Aug  5 14:00:17 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 5 Aug 2009 23:00:17 +0200
Subject: [Freeswitch-users] freepbx for freeswitch
Message-ID: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>

Yay!

http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-future

Darren Schreiber has made the announcement and is doinng a
presentation of FreePBX V3 right now at www.cluecon.com.


From tomabroad at gmail.com  Wed Aug  5 14:09:37 2009
From: tomabroad at gmail.com (tom)
Date: Wed, 5 Aug 2009 17:09:37 -0400
Subject: [Freeswitch-users] q: fresh installation: Couldnt open
	/usr/local/freeswitch/conf/freeswitch.xml
In-Reply-To: <7e2ac3270908051353m33c3f158r4bd7ec2d415f7a7a@mail.gmail.com>
References: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
	<7e2ac3270908051353m33c3f158r4bd7ec2d415f7a7a@mail.gmail.com>
Message-ID: <6f7c60c40908051409h57277282m6f6e94af47b6d75d@mail.gmail.com>

solved via "make samples"

thx tom
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From nik.middleton at noblesolutions.co.uk  Wed Aug  5 14:24:47 2009
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Wed, 5 Aug 2009 22:24:47 +0100
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
Message-ID: <A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>

I'd heard rumours that this was going to happen and it's great news and
good news for FS as well.  With a user friendly front end, FS is sure to
fly.  I have no doubt that this will be the first of many.

Regards,

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Giovanni Maruzzelli
Sent: 05 August 2009 22:00
To: freeswitch-users at lists.freeswitch.org;
freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-users] freepbx for freeswitch

Yay!

http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
ure

Darren Schreiber has made the announcement and is doinng a
presentation of FreePBX V3 right now at www.cluecon.com.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From msc at freeswitch.org  Wed Aug  5 14:35:30 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 16:35:30 -0500
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
Message-ID: <87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>

Of course, don't forget this:


For those who want a commercial solution built upon FreeSWITCH: You've got
it!
-MC

On Wed, Aug 5, 2009 at 4:24 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

> I'd heard rumours that this was going to happen and it's great news and
> good news for FS as well.  With a user friendly front end, FS is sure to
> fly.  I have no doubt that this will be the first of many.
>
> Regards,
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Giovanni Maruzzelli
> Sent: 05 August 2009 22:00
> To: freeswitch-users at lists.freeswitch.org;
> freeswitch-dev at lists.freeswitch.org
> Subject: [Freeswitch-users] freepbx for freeswitch
>
> Yay!
>
> http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
> ure
>
> Darren Schreiber has made the announcement and is doinng a
> presentation of FreePBX V3 right now at www.cluecon.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Wed Aug  5 14:49:49 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 5 Aug 2009 17:49:49 -0400
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
	<87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>
Message-ID: <86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>

Yay for FreePBX =D

Does Cudatel uses FreeSWITCH as the engine?

On Wed, Aug 5, 2009 at 5:35 PM, Michael Collins<msc at freeswitch.org> wrote:
> Of course, don't forget this:
> 
>
> For those who want a commercial solution built upon FreeSWITCH: You've got
> it!
> -MC
>
> On Wed, Aug 5, 2009 at 4:24 PM, Nik Middleton
> <nik.middleton at noblesolutions.co.uk> wrote:
>>
>> I'd heard rumours that this was going to happen and it's great news and
>> good news for FS as well. ?With a user friendly front end, FS is sure to
>> fly. ?I have no doubt that this will be the first of many.
>>
>> Regards,
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Giovanni Maruzzelli
>> Sent: 05 August 2009 22:00
>> To: freeswitch-users at lists.freeswitch.org;
>> freeswitch-dev at lists.freeswitch.org
>> Subject: [Freeswitch-users] freepbx for freeswitch
>>
>> Yay!
>>
>> http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
>> ure
>>
>> Darren Schreiber has made the announcement and is doinng a
>> presentation of FreePBX V3 right now at www.cluecon.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From sprice at gmail.com  Wed Aug  5 14:59:31 2009
From: sprice at gmail.com (SP)
Date: Wed, 5 Aug 2009 16:59:31 -0500
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
	<87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>
	<86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>
Message-ID: <7e2ac3270908051459l16db1152k8eb6734add1957a3@mail.gmail.com>

would it be pimped if it didn't?

On Wed, Aug 5, 2009 at 16:49, Diego Viola<diego.viola at gmail.com> wrote:
> Yay for FreePBX =D
>
> Does Cudatel uses FreeSWITCH as the engine?
>
> On Wed, Aug 5, 2009 at 5:35 PM, Michael Collins<msc at freeswitch.org> wrote:
>> Of course, don't forget this:
>> 
>>
>> For those who want a commercial solution built upon FreeSWITCH: You've got
>> it!
>> -MC
>>
>> On Wed, Aug 5, 2009 at 4:24 PM, Nik Middleton
>> <nik.middleton at noblesolutions.co.uk> wrote:
>>>
>>> I'd heard rumours that this was going to happen and it's great news and
>>> good news for FS as well. ?With a user friendly front end, FS is sure to
>>> fly. ?I have no doubt that this will be the first of many.
>>>
>>> Regards,
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Giovanni Maruzzelli
>>> Sent: 05 August 2009 22:00
>>> To: freeswitch-users at lists.freeswitch.org;
>>> freeswitch-dev at lists.freeswitch.org
>>> Subject: [Freeswitch-users] freepbx for freeswitch
>>>
>>> Yay!
>>>
>>> http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
>>> ure
>>>
>>> Darren Schreiber has made the announcement and is doinng a
>>> presentation of FreePBX V3 right now at www.cluecon.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shannon


From msc at freeswitch.org  Wed Aug  5 15:00:06 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 17:00:06 -0500
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
	<87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>
	<86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>
Message-ID: <87f2f3b90908051500u59e5f6ack4402818e226df7a7@mail.gmail.com>

On Wed, Aug 5, 2009 at 4:49 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Yay for FreePBX =D
>
> Does Cudatel uses FreeSWITCH as the engine?


Yes it does!
-MC
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From msc at freeswitch.org  Wed Aug  5 15:02:28 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 17:02:28 -0500
Subject: [Freeswitch-users] pocketsphinx
In-Reply-To: <4A799D9E.2020103@ewetel.de>
References: <4A5739DE.1080800@ewetel.de> <4A72EF33.4070404@ewetel.de>
	<87f2f3b90907311145p6c8d5907mc4545c710e9605fb@mail.gmail.com>
	<4A799D9E.2020103@ewetel.de>
Message-ID: <87f2f3b90908051502o3943cbcdy35c91cd11174d743@mail.gmail.com>

On Wed, Aug 5, 2009 at 9:56 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Michael,
>
> today I put my documentation on FS wiki describing the steps to get a
> 8kHz sample rate acoustic model basing on voxforge's data for german
> language. It's not complete, yet.
>
> You can found it here: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx
>
> regards
> Helmut
>

Thanks!
I'd like to ask the community members who are interested in ASR and
PocketSphinx to please review Helmut's page and add to it as well as offer
feedback.
-MC
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From msc at freeswitch.org  Wed Aug  5 15:04:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 17:04:54 -0500
Subject: [Freeswitch-users] Fwd: Need Help In IVR
In-Reply-To: <7aa29e790908032247h2a93e838m1ac5fd3f8a3946c7@mail.gmail.com>
References: <7aa29e790908030012rcce847fu8320d833d2c85530@mail.gmail.com>
	<7aa29e790908032247h2a93e838m1ac5fd3f8a3946c7@mail.gmail.com>
Message-ID: <87f2f3b90908051504i70dd6e6ag7bc5fe3fa0200dd2@mail.gmail.com>

Could you give us an update on what you have so far? How about you put your
dialplan and perl script into a pastebin so that we can get a good frame of
reference?
Thanks,
MC

On Tue, Aug 4, 2009 at 12:47 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Can you please help me?
>
>
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From msc at freeswitch.org  Wed Aug  5 15:16:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 17:16:41 -0500
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
Message-ID: <87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>

On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> interesting! what values can contain "variable_originate_disposition"? And
> can I set them manually in a script to reject a call simulating user busy or
> call rejected? A lua example?
> Thanks,
>    Raffaele
>

Start here:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup

And note the link to the hangup causes. As far as Lua, I'm not sure there's
a good reason to do it there. Could you give us pseudo code example of what
you're thinking of doing?
-MC
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From raffaele.p.guidi at gmail.com  Wed Aug  5 16:19:10 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Thu, 6 Aug 2009 01:19:10 +0200
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
Message-ID: <b0a8792f0908051619s66caac7na886720273396bdd@mail.gmail.com>

Actually I was reading that page, right now. I wrote a small lua script that
simulates a call with random wait time before answering, randomly not
answering at all and saying things for a random times once answered. This
would be useful for testing purposes simulating load, letting call center
operators try scripts against "fake" numbers with a "realistic" behaviour
and eventually to test and debug an automated dialer.
The script is almost ready (can contribute it should you find it useful), I
used it today to simulate load on my windows laptop with 50 concurrent calls
and peaks of 20/30 simultaneous calls connected (cpu was below 3%). I only
miss some use cases such as some CALL_REJECTED, USER_BUSY, NO_ANSWER

On Thu, Aug 6, 2009 at 00:16, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> interesting! what values can contain "variable_originate_disposition"? And
>> can I set them manually in a script to reject a call simulating user busy or
>> call rejected? A lua example?
>> Thanks,
>>    Raffaele
>>
>
> Start here:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>
> And note the link to the hangup causes. As far as Lua, I'm not sure there's
> a good reason to do it there. Could you give us pseudo code example of what
> you're thinking of doing?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Aug  5 16:23:22 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 18:23:22 -0500
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
Message-ID: <121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>

The bigger problem is some end points won't hang up the call in a  
consistent manner... some phones say user_rejected or user_busy when  
you reject the call with the reject button.

/b

On Aug 5, 2009, at 5:16 PM, Michael Collins wrote:

>
>
> On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <raffaele.p.guidi at gmail.com 
> > wrote:
> interesting! what values can contain  
> "variable_originate_disposition"? And can I set them manually in a  
> script to reject a call simulating user busy or call rejected? A lua  
> example?
>
> Thanks,
>    Raffaele
>
> Start here:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>
> And note the link to the hangup causes. As far as Lua, I'm not sure  
> there's a good reason to do it there. Could you give us pseudo code  
> example of what you're thinking of doing?
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nicolas at medularis.com  Wed Aug  5 16:41:47 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 5 Aug 2009 19:41:47 -0400
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
Message-ID: <1b46b4e80908051641y170b8d07g98ac29f69250f2ce@mail.gmail.com>

I'm bridging 2 calls in a javascript file, I originate the first call and
then execute a bridge with an origination string for the second call. If I
hangup the first call while trying to make the second call, I get this on
the console:

2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL

But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
NORMAL_CLEARING. And the variable_originate_disposition has a value of
"failure". Where can I get the detail of knowing the call/bridge failed
because of 'ORIGINATOR_CANCEL' as reported through the console?

Thanks!

Nicolas


 the value of variable_originate_disposition at the events level and when I
have an origination failure due to 'ORIGINATOR_CANCEL

On Wed, Aug 5, 2009 at 6:16 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> interesting! what values can contain "variable_originate_disposition"? And
>> can I set them manually in a script to reject a call simulating user busy or
>> call rejected? A lua example?
>> Thanks,
>>    Raffaele
>>
>
> Start here:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>
> And note the link to the hangup causes. As far as Lua, I'm not sure there's
> a good reason to do it there. Could you give us pseudo code example of what
> you're thinking of doing?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From raffaele.p.guidi at gmail.com  Wed Aug  5 16:44:11 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Thu, 6 Aug 2009 01:44:11 +0200
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
	<121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
Message-ID: <b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>

Well, I would randomly insert all of those cases to make it more
realistic... only thing I cannot manage to issue USER_BUSY from lua (and
neither from the dialplan, actually).

<anti-action application="respond" data="407" /> (407 or 486 or whatever...)


doesn't behave as I expected and neither

<action application="hangup" data="407" /> (407 or 486 or USER_BUSY or
whatever...)


and I cannot find a a session:reject(<hangupcause>) method in lua.

Can you give me a hint?

On Thu, Aug 6, 2009 at 01:23, Brian West <brian at freeswitch.org> wrote:

> The bigger problem is some end points won't hang up the call in a
> consistent manner... some phones say user_rejected or user_busy when you
> reject the call with the reject button.
> /b
>
> On Aug 5, 2009, at 5:16 PM, Michael Collins wrote:
>
>
>
> On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> interesting! what values can contain "variable_originate_disposition"? And
>> can I set them manually in a script to reject a call simulating user busy or
>> call rejected? A lua example?
>> Thanks,
>>    Raffaele
>>
>
> Start here:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>
> And note the link to the hangup causes. As far as Lua, I'm not sure there's
> a good reason to do it there. Could you give us pseudo code example of what
> you're thinking of doing?
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From woodydickson at gmail.com  Wed Aug  5 17:20:51 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Thu, 6 Aug 2009 08:20:51 +0800
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
	<E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
Message-ID: <a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>

Hi,

The problem is that I need freeswitch to continue executing the code after
switch_status_t channel_receive_message even when it gets error SIP code
from the destination.  Is that possible?

I know if I set up another action after my module in the dialplan.xml, I can
catch that.

But I would like the code to execute within the route that I have.  Is that
doable?

Woody

On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> The hangup cause will be in the originate_disposition channel
> variable on the A-leg.
> sip_term_status will contain the sip code and proto_specific_hangup_cause
> will contain sip:<code>.
>
>
>   Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
>  Am 5-Aug-09 um 11:23 AM schrieb Jo?o Mesquita:
>
> My guess is that you will receive a message here:
>
> switch_status_t channel_receive_message(switch_core_session_t *session,
> switch_core_session_message_t *msg)
>
> The problem here is that you don't have the exact SIP code but there is a
> clear relationship between the codes and the messages you receive on the
> channel, so I am guessing that is all the same.
>
> Hope this helps.
>
> jmesquita
>
> On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson <woodydickson at gmail.com>wrote:
>
>> Hi,
>>
>> I want to implement a module where freeSWITCH would try to bridge to an
>> extension and if the bridging operation fails, my module can use the hangup
>> code to determine the next cause of action.
>>
>> With switch_caller_extension_add_application(session, extension, "bridge",
>> "sofia/gateway/mygw/1232323);, if there is an error ( 503 received for
>> instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or the
>> module's APP) and go on to the next action.  Is there anyway to control it
>> so that freeSWITCH would remain to be within the module's APP funtion and
>> continue executing the code after switch_call_extension_add_application,
>> when let's say a 4XX or 5XX or CANCEL ( from originator) is received?
>>
>> I have tried it and found that if the bridging is successful, freeSWITCH
>> would continue executing the code after
>> switch_caller_extension_add_application, but if an error is received, then
>> it would just move on to the next action.
>>
>> Does anyone know how to deal with this problem?
>>
>> Woody
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Wed Aug  5 17:36:56 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 5 Aug 2009 19:36:56 -0500
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
	<E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
	<a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>
Message-ID: <9521EE86-37FF-4989-8F58-F61E5110E5C5@avgs.ca>

Hi,

You can set the "continue_on_fail" variable to true (or to the hangup  
causes you want it to ignore) and it'll keep executing whats queued.
For receive_message, unless you hook the session thats being created  
as a B-leg, you won't get anything relevant.
Also set hangup_after_bridge=true if you want to stop failing over  
when it worked.

Im curious, what are you coding? you can transfer the call in the  
dialplan without having to do all this manual queuing in C, thats why  
the routing state and dialplan modules exist. If you need to pull data  
from somewhere you can fill in channel variables that you can  
reference in the dialplan.

/*!
   \brief Transfer an existing session to another location
   \param session the session to transfer
   \param extension the new extension
   \param dialplan the new dialplan (OPTIONAL, may be NULL)
   \param context the new context (OPTIONAL, may be NULL)
*/
SWITCH_DECLARE(switch_status_t) switch_ivr_session_transfer(_In_  
switch_core_session_t *session, const char *extension, const char  
*dialplan,
															const char *context);



Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 5-Aug-09 um 7:20 PM schrieb Woody Dickson:

> Hi,
>
> The problem is that I need freeswitch to continue executing the code  
> after switch_status_t channel_receive_message even when it gets  
> error SIP code from the destination.  Is that possible?
>
> I know if I set up another action after my module in the  
> dialplan.xml, I can catch that.
>
> But I would like the code to execute within the route that I have.   
> Is that doable?
>
> Woody
>
> On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> The hangup cause will be in the originate_disposition channel  
> variable on the A-leg.
>
> sip_term_status will contain the sip code and  
> proto_specific_hangup_cause will contain sip:<code>.
>
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> Am 5-Aug-09 um 11:23 AM schrieb Jo?o Mesquita:
>
>> My guess is that you will receive a message here:
>>
>> switch_status_t channel_receive_message(switch_core_session_t  
>> *session, switch_core_session_message_t *msg)
>>
>> The problem here is that you don't have the exact SIP code but  
>> there is a clear relationship between the codes and the messages  
>> you receive on the channel, so I am guessing that is all the same.
>>
>> Hope this helps.
>>
>> jmesquita
>>
>> On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson <woodydickson at gmail.com 
>> > wrote:
>> Hi,
>>
>> I want to implement a module where freeSWITCH would try to bridge  
>> to an extension and if the bridging operation fails, my module can  
>> use the hangup code to determine the next cause of action.
>>
>> With switch_caller_extension_add_application(session, extension,  
>> "bridge", "sofia/gateway/mygw/1232323);, if there is an error ( 503  
>> received for instance ) in the outgoing INVITE, freeSWITCH would  
>> leave my module ( or the module's APP) and go on to the next  
>> action.  Is there anyway to control it so that freeSWITCH would  
>> remain to be within the module's APP funtion and continue executing  
>> the code after switch_call_extension_add_application, when let's  
>> say a 4XX or 5XX or CANCEL ( from originator) is received?
>>
>> I have tried it and found that if the bridging is successful,  
>> freeSWITCH would continue executing the code after  
>> switch_caller_extension_add_application, but if an error is  
>> received, then it would just move on to the next action.
>>
>> Does anyone know how to deal with this problem?
>>
>> Woody
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From vladrodionov at gmail.com  Wed Aug  5 16:30:09 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Wed, 5 Aug 2009 16:30:09 -0700
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to configure?)
Message-ID: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>

Hi, everybody

This is a newbie question: Suppose I have XX (variable dynamic number) DIDs
assigned to one sip trunk (from VOIP provider ABC ). All calls coming from
VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is it
possible in FS? If yes, how everything should be configuered? Dialplan, sip
gateway? One more question: suppose it is doeable as I hope then how can I
get in my script CalleeID (not a CallerID)? Basicaly,

I want to acomplish the following:

1. Avoid re-configuring FS every time I got new bunch of DIDs
assigned/released from/to my Voip provider.
2. Have a way of extracting CalleeID in my script.

TIA,

Vladimir Rodionov
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From dujinfang at gmail.com  Wed Aug  5 18:26:59 2009
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 6 Aug 2009 09:26:59 +0800
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
References: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
Message-ID: <23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>

mod_easyroute?

2009/8/6 Vladimir Rodionov <vladrodionov at gmail.com>

> Hi, everybody
>
> This is a newbie question: Suppose I have XX (variable dynamic number) DIDs
> assigned to one sip trunk (from VOIP provider ABC ). All calls coming from
> VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is it
> possible in FS? If yes, how everything should be configuered? Dialplan, sip
> gateway? One more question: suppose it is doeable as I hope then how can I
> get in my script CalleeID (not a CallerID)? Basicaly,
>
> I want to acomplish the following:
>
> 1. Avoid re-configuring FS every time I got new bunch of DIDs
> assigned/released from/to my Voip provider.
> 2. Have a way of extracting CalleeID in my script.
>
> TIA,
>
> Vladimir Rodionov
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vladrodionov at gmail.com  Wed Aug  5 18:57:41 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Wed, 5 Aug 2009 18:57:41 -0700
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>
References: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
	<23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>
Message-ID: <3c233920908051857t5037d169r6bc4555b53cd015c@mail.gmail.com>

No, it is more like static routing. I need my *script program* be invoked
when somebody dial in. That is it. One script for all inbound DIDs. Suppose
I have thousand of them.  I think I know how to accomplish this but I am not
sure yet.

in my dialplan I need to define:

<!-- Launch a JavaScript application if dialed in-->
   <extension name="ProviderABC">
    <condition field="source" expression="mod_sofia"/>
    *<condition field="destination_number" expression="^1NXXNXXXXXX$">*
     <action application="javascript"
data="/usr/local/freeswitch/scripts/myapp.js"/>
    </condition>
   </extension>


In provider configuration:

<gateway name="voicepulse">
           <!--/// account username *required* ///-->
           <param name="username" value="your-username"/>
           <!--/// auth realm: *optional* same as gateway name, if blank ///-->
           <param name="realm" value="nyc.voicepulse.com"/>
           <!--/// account password *required* ///-->
           <param name="password" value="your-password"/>
           <!--/// extension for inbound calls: *optional* same as
username, if blank ///-->
*           * *<param name="extension" value="1NXXNXXXXXX"/> *
           <!--/// proxy host: *optional* same as realm, if blank ///-->
           <param name="proxy" value="nyc.voicepulse.com"/>
           <!--/// expire in seconds: *optional* 3600, if blank ///-->
           <param name="expire-seconds" value="600"/>

	   <param name="register" value="true"/>
         </gateway>


Something like this, yes? I can use regular expressions in
destination_number?

Q: There is object Session in JavaScript, Lua. Is Session.destination ==
destination_number from incoming call? It is not clear for me from what I
have read so far.

TIA,

-Vladimir Rodionov

On Wed, Aug 5, 2009 at 6:26 PM, Seven Du <dujinfang at gmail.com> wrote:

> mod_easyroute?
>
> 2009/8/6 Vladimir Rodionov <vladrodionov at gmail.com>
>
>> Hi, everybody
>>
>> This is a newbie question: Suppose I have XX (variable dynamic number)
>> DIDs assigned to one sip trunk (from VOIP provider ABC ). All calls coming
>> from VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is
>> it possible in FS? If yes, how everything should be configuered? Dialplan,
>> sip gateway? One more question: suppose it is doeable as I hope then how can
>> I get in my script CalleeID (not a CallerID)? Basicaly,
>>
>> I want to acomplish the following:
>>
>> 1. Avoid re-configuring FS every time I got new bunch of DIDs
>> assigned/released from/to my Voip provider.
>> 2. Have a way of extracting CalleeID in my script.
>>
>> TIA,
>>
>> Vladimir Rodionov
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Aug  5 19:03:26 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 21:03:26 -0500
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
	<121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
	<b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>
Message-ID: <87f2f3b90908051903y16395a62uf23feec9b3fdceab@mail.gmail.com>

On Wed, Aug 5, 2009 at 6:44 PM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> Well, I would randomly insert all of those cases to make it more
> realistic... only thing I cannot manage to issue USER_BUSY from lua (and
> neither from the dialplan, actually).
>
> <anti-action application="respond" data="407" /> (407 or 486 or
> whatever...)
>
>
> doesn't behave as I expected and neither
>
> <action application="hangup" data="407" /> (407 or 486 or USER_BUSY or
> whatever...)
>
>
> and I cannot find a a session:reject(<hangupcause>) method in lua.
>
> Can you give me a hint?
>

You can execute pretty much any dialplan app with the session:execute
command:
http://wiki.freeswitch.org/wiki/Lua#session:execute

Try something like:
session:execute("hangup","USER_BUSY");

-MC
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From msc at freeswitch.org  Wed Aug  5 20:40:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 22:40:31 -0500
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <3c233920908051857t5037d169r6bc4555b53cd015c@mail.gmail.com>
References: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
	<23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>
	<3c233920908051857t5037d169r6bc4555b53cd015c@mail.gmail.com>
Message-ID: <87f2f3b90908052040s21545229qba24937bd2b14540@mail.gmail.com>

On Wed, Aug 5, 2009 at 8:57 PM, Vladimir Rodionov <vladrodionov at gmail.com>wrote:

> No, it is more like static routing. I need my *script program* be invoked
> when somebody dial in. That is it. One script for all inbound DIDs. Suppose
> I have thousand of them.  I think I know how to accomplish this but I am not
> sure yet.
>

Have the external profile be used only for provider ABC, or define a new
profile. Then in the profile have the calls go to a specific context. You
could have something like this in the sip profile definition:

<param name="context" value="abc_calls"/>

Then create a dialplan context called "abc_calls" that handles all inbound
calls. Create a file in conf/dialplan/ called abc_calls.xml:

<include>
  <context name="abc_calls">
    <extension name="abc_calls">
      <condition field="destination_number" expression="^(.*)$">
        <action application="lua" data="myscript.lua"/>
      </condition>
    </extension>
  </context>
</include>

Essentially you're just creating a SIP profile and a dialplan context that
are servicing your VoIP provider. You can add other profiles/contexts for
other providers if need be.

Let us know how it goes...
-MC
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From pete at privateconnect.com  Wed Aug  5 20:45:30 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Wed, 05 Aug 2009 20:45:30 -0700
Subject: [Freeswitch-users]
 =?utf-8?q?Multiple_DIDs_per_SIP_trunk_=28how_t?=
 =?utf-8?q?o_configure=3F=29?=
Message-ID: <20090805204530.2ad02225396a31c9de30536f2e338977.43419788fb.wbe@email04.secureserver.net>

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From woodydickson at gmail.com  Wed Aug  5 20:54:28 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Thu, 6 Aug 2009 11:54:28 +0800
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <9521EE86-37FF-4989-8F58-F61E5110E5C5@avgs.ca>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
	<E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
	<a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>
	<9521EE86-37FF-4989-8F58-F61E5110E5C5@avgs.ca>
Message-ID: <a44688a0908052054j405c1707j382b3d6ccac8f8a9@mail.gmail.com>

Hi,

In my module, I will collect a list of available failover route that I can
use to failover to whenever a particular error is received.  However, these
available routes has different condition and the condition changes every
half a minute.  Therefore, I need to catch the hangup cause after bridge,
and then figure out the next workable available route based on the latest
condition setting.

It seems like this is only prossible to be done within a C module.  Any
suggestion will be greatly appreciated.

Woody

On Thu, Aug 6, 2009 at 8:36 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
> You can set the "continue_on_fail" variable to true (or to the hangup
> causes you want it to ignore) and it'll keep executing whats queued. For
> receive_message, unless you hook the session thats being created as a B-leg,
> you won't get anything relevant.
> Also set hangup_after_bridge=true if you want to stop failing over when it
> worked.
>
> Im curious, what are you coding? you can transfer the call in the dialplan
> without having to do all this manual queuing in C, thats why the routing
> state and dialplan modules exist. If you need to pull data from somewhere
> you can fill in channel variables that you can reference in the dialplan.
>
> /*!
>   \brief Transfer an existing session to another location
>   \param session the session to transfer
>   \param extension the new extension
>   \param dialplan the new dialplan (OPTIONAL, may be NULL)
>   \param context the new context (OPTIONAL, may be NULL)
> */
> SWITCH_DECLARE(switch_status_t) switch_ivr_session_transfer(_In_
> switch_core_session_t *session, const char *extension, const char *dialplan,
> const char *context);
>
>
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> Am 5-Aug-09 um 7:20 PM schrieb Woody Dickson:
>
> Hi,
>
> The problem is that I need freeswitch to continue executing the code after
> switch_status_t channel_receive_message even when it gets error SIP code
> from the destination.  Is that possible?
>
> I know if I set up another action after my module in the dialplan.xml, I
> can catch that.
>
> But I would like the code to execute within the route that I have.  Is that
> doable?
>
> Woody
>
> On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
>> The hangup cause will be in the originate_disposition channel
>> variable on the A-leg.
>> sip_term_status will contain the sip code and proto_specific_hangup_cause
>> will contain sip:<code>.
>>
>>
>>   Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>>  Am 5-Aug-09 um 11:23 AM schrieb Jo?o Mesquita:
>>
>> My guess is that you will receive a message here:
>>
>> switch_status_t channel_receive_message(switch_core_session_t *session,
>> switch_core_session_message_t *msg)
>>
>> The problem here is that you don't have the exact SIP code but there is a
>> clear relationship between the codes and the messages you receive on the
>> channel, so I am guessing that is all the same.
>>
>> Hope this helps.
>>
>> jmesquita
>>
>> On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson <woodydickson at gmail.com>wrote:
>>
>>> Hi,
>>>
>>> I want to implement a module where freeSWITCH would try to bridge to an
>>> extension and if the bridging operation fails, my module can use the hangup
>>> code to determine the next cause of action.
>>>
>>> With switch_caller_extension_add_application(session, extension,
>>> "bridge", "sofia/gateway/mygw/1232323);, if there is an error ( 503 received
>>> for instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or
>>> the module's APP) and go on to the next action.  Is there anyway to control
>>> it so that freeSWITCH would remain to be within the module's APP funtion and
>>> continue executing the code after switch_call_extension_add_application,
>>> when let's say a 4XX or 5XX or CANCEL ( from originator) is received?
>>>
>>> I have tried it and found that if the bridging is successful, freeSWITCH
>>> would continue executing the code after
>>> switch_caller_extension_add_application, but if an error is received, then
>>> it would just move on to the next action.
>>>
>>> Does anyone know how to deal with this problem?
>>>
>>> Woody
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From velu.technical at gmail.com  Wed Aug  5 21:38:25 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Thu, 6 Aug 2009 10:08:25 +0530
Subject: [Freeswitch-users] Fwd: execute function in ESL.pm module is not
	working
In-Reply-To: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>
References: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>
Message-ID: <1452e2980908052138jed5d676oe5ce73b0b4b0e37e@mail.gmail.com>

Please any one help for this problem..

---------- Forwarded message ----------
From: velusamy velu <velu.technical at gmail.com>
Date: Wed, Aug 5, 2009 at 11:44 AM
Subject: execute function in ESL.pm module is not working
To: freeswitch-users at lists.freeswitch.org


Dear All,
   I registered alarm signal in my Perl server program.
   If ALARM signal occurred I execute the following statement in signal
handler.
    "$conn->execute("playback",$sound_path."voicemail/vm-goodbye.wav")"
  The above statement didn't play that wave file. But before generating the
ALARM signal it worked.

What is the problem?

Please help me in this problem....

Also Is there any idea to do timeout for DTMF digits?

Thanks...

Regards,
K.Velusamy
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From msc at freeswitch.org  Wed Aug  5 22:54:17 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Aug 2009 00:54:17 -0500
Subject: [Freeswitch-users] Fwd: execute function in ESL.pm module is
	not working
In-Reply-To: <1452e2980908052138jed5d676oe5ce73b0b4b0e37e@mail.gmail.com>
References: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>
	<1452e2980908052138jed5d676oe5ce73b0b4b0e37e@mail.gmail.com>
Message-ID: <87f2f3b90908052254l3ecc7fa0ybc92d87c587a9b0d@mail.gmail.com>

On Wed, Aug 5, 2009 at 11:38 PM, velusamy velu <velu.technical at gmail.com>wrote:

> Please any one help for this problem..
>
>
Sorry for the delay but many of the FreeSWITCH experts are at ClueCon right
now so we'll ask for your patience... in the meantime could you pastebin
your script and your dialplan entry so that we can take a look at them?

Thanks,
MC
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From mayamatakeshi at gmail.com  Thu Aug  6 00:03:06 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 6 Aug 2009 16:03:06 +0900
Subject: [Freeswitch-users] Monitoring On-Hold/Off-Hold
In-Reply-To: <5a8712120908051055k95f01d4x61ef7b7dff80298a@mail.gmail.com>
References: <15b9404e0908050935m62a6581bxec3f3055e013753f@mail.gmail.com>
	<5a8712120908051055k95f01d4x61ef7b7dff80298a@mail.gmail.com>
Message-ID: <15b9404e0908060003s2eb17434q1de769fdde444ad2@mail.gmail.com>

2009/8/6 Jo?o Mesquita <jmesquita at gmail.com>:
> I only see one way out of this. If you manage presence, an event like the
> following is sent:
>
> Event-Name: PRESENCE_IN
> Core-UUID: 189b12c0-7fb0-11de-b0bc-37eec03ad00f
> FreeSWITCH-Hostname: cl-t146-421cl
> FreeSWITCH-IPv4: XXXXXX
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-08-05%2013%3A42%3A24
> Event-Date-GMT: Wed,%2005%20Aug%202009%2017%3A42%3A24%20GMT
> Event-Date-Timestamp: 1249494144628132
> Event-Calling-File: switch_channel.c
> Event-Calling-Function: switch_channel_presence
> Event-Calling-Line-Number: 472
> Channel-State: CS_HIBERNATE
> Channel-State-Number: 8
> Channel-Name: XXXXX
> Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f
> Call-Direction: inbound
> Presence-Call-Direction: inbound
> Answer-State: answered
> Caller-Username: 1000
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: Mesquita
> Caller-Caller-ID-Number: 1000
> Caller-Network-Addr: XXXXX
> Caller-Destination-Number: 1005
> Caller-Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f
> Caller-Source: mod_sofia
> Caller-Context: XXXXX
> Caller-Channel-Name: XXXXX
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1249494132128119
> Caller-Channel-Created-Time: 1249494132128119
> Caller-Channel-Answered-Time: 1249494139500129
> Caller-Channel-Progress-Time: 1249494132368119
> Caller-Channel-Progress-Media-Time: 0
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
> Other-Leg-Username: 1000
> Other-Leg-Dialplan: XML
> Other-Leg-Caller-ID-Name: Joao%20Mesquita
> Other-Leg-Caller-ID-Number: 1000
> Other-Leg-Network-Addr: 190.2.41.65
> Other-Leg-Destination-Number:
> sip%3A1005%40192.168.0.106%3A4559%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1005%2540190.2.41.65%253A4559
> Other-Leg-Unique-ID: 4e7622ac-81e7-11de-b0bc-37eec03ad00f
> Other-Leg-Source: mod_sofia
> Other-Leg-Context: XXXXX
> Other-Leg-Channel-Name: XXXXXX
> Other-Leg-Screen-Bit: true
> Other-Leg-Privacy-Hide-Name: false
> Other-Leg-Privacy-Hide-Number: false
> proto: src/switch_channel.c
> login: src/switch_channel.c
> from: XXXXXX
> rpid: unknown
> status: hold
> event_type: presence
> alt_event_type: dialog
> event_count: 3
>
> Content-Length: 543
> Content-Type: text/event-plain
>
> Other than that, I think it can be patched. I will take a look at it.

Thanks, that would be the best.

Just in case someone else needs this:
I have also tried to watch for
CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE with Application set to
playback and some indication of MOH in the Application-Data header.
That would work but:
- they will be fired continuously if you set hold_music=some_file
- they will not be fired if you set hold_music=silence (of course)


From enno.egbert at googlemail.com  Thu Aug  6 02:09:24 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Thu, 6 Aug 2009 02:09:24 -0700 (PDT)
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>
References: <24823167.post@talk.nabble.com>
	<ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>
Message-ID: <24841050.post@talk.nabble.com>


Hi Jim,

yes! It?s possible to call the IP200 full encrypted for example from a SNOM
or phonerlite. But when i try to call the SNOM from a innovaphone, the call
fails and i only hear the mailbox.

To modify the dialplan i am not so sure how i works. I don?t have any
experience of configure freeswitch or working with xml files. :confused:

In my dialplan i just modify a few lines. If you want, you can have a look
on the file in the attachment. 

http://www.nabble.com/file/p24841050/default.xml default.xml 

Thanks for your help. =)

NOx


Jim Burke-2 wrote:
> 
> Hi NOx,
> 
> Can you clarify the direction of the calls.  When you say outgoing do
> you mean a call is terminating to the ip200?
> 
> I have been down a similar path while testing Eyebeam.  If the
> terminating phone sets an option to only accept secure calls and FS
> does not send Secure Descriptions in the INVITE, Eyebeam would respond
> with 415 response code and the call would fail.  Depending on your
> diaplan this could send your call to voicemail.
> 
> To fix it I added the following code to dialplan.
> <action application="set" data="continue_on_fail=79"/>
> <bridge blah blah blah>
> <action application="set" data="bypass_media=false"/>
> <action application="set" data="proxy_media=true"/>
> <action application="set" data="ringback=$${uk-ring}"/>
> <action application="pre_answer"/>
> <action application="export" data="sip_secure_media=true"/>
> <bridge blah blah blah>
> 
> The continue on fail captures the 415 response code forces the call to
> continue to the next bridge while sip_secure_media forces the second
> invite to include security descriptors.  The rest was required because
> I did not want to proxy media if the call was not secure, obviously if
> the call is secure on a point to point basis FS will have to proxy the
> media and this was the only way I could find for it to work.
> 
> Hope this helps.
> 
> Regards,
> 
> 
> On Wed, Aug 5, 2009 at 6:52 PM, NOx-WHV<enno.egbert at googlemail.com> wrote:
>>
>> Hello,
>>
>> i have a problem using a innovaphone ip200 with freeswitch and tls/srtp.
>> The
>> freeswitch certificate is in the trust list of the phone and it works
>> with
>> tls for incomming calls. But outgoing calls were rejected to the mailbox.
>> The freeswitch configuration is ok, because it works with a snom 320.
>>
>> Who can help me to confugure the IP200?
>>
>> Thanks
>>
>> NOx
>> --
>> View this message in context:
>> http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24823167.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24841050.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From juanbackson at gmail.com  Thu Aug  6 02:24:03 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 6 Aug 2009 17:24:03 +0800
Subject: [Freeswitch-users] Question about dynamic registration
In-Reply-To: <9FE0F6B8-C4BF-4820-8CC2-6825C5EE8422@freeswitch.org>
References: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>
	<9FE0F6B8-C4BF-4820-8CC2-6825C5EE8422@freeswitch.org>
Message-ID: <27c25bc40908060224l38ad47fdje17065b13647905d@mail.gmail.com>

Hi,

Is there a sample module that I can take a look at on how to do that?
I don't understand how to get the registration request and how to pass back
auth result to freeswitch.

JB

On Mon, Aug 3, 2009 at 8:42 PM, Brian West <brian at freeswitch.org> wrote:

> You could build your own module to do it how ever you please.  But
> forking a script every time to auth is not very scalable.
>
> /b
>
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From markmorreny at gmail.com  Thu Aug  6 02:26:53 2009
From: markmorreny at gmail.com (mark morreny)
Date: Thu, 6 Aug 2009 17:26:53 +0800
Subject: [Freeswitch-users] question about latest version of mod_limit
Message-ID: <20ad6b920908060226t2dcf532aobab23dc0299d0f05@mail.gmail.com>

Hello,

I have the following setup in the dialplan.  Then, I fire up sipp to send
5calls/s and I expect to get limit-pass=false in most of the INFO output.
However, I am getting all "limit-pass=pass".

Does anyone know what is wrong with my dialplan?


 <context name="internal">

  <extension name="test-limit">

    <condition field="destination_number" expression="^9(.*)$">

        <action application="limit_hash_execute" data="192.168.1.102 6000
1/1 transfer to-next" />
        <action application="set" data="limit-pass=false"/>
        <action application="info" />
        <action application="hangup" />

    </condition>

  </extension>
    <extension name="to-next">
        <condition field="destination_number" expression="^to-next$">
        <action application="set" data="limit-pass=true"/>
        <action application="info" />
        <action application="hangup" />



    </condition>
    </extension>
</context>
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From enno.egbert at googlemail.com  Thu Aug  6 02:52:01 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Thu, 6 Aug 2009 02:52:01 -0700 (PDT)
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <DF06C882-055F-488D-B860-46643AF9B22B@freeswitch.org>
References: <24823167.post@talk.nabble.com>
	<DF06C882-055F-488D-B860-46643AF9B22B@freeswitch.org>
Message-ID: <24841078.post@talk.nabble.com>


Hi Brian, 

where i have to ship the phone? Maybe i can ship it to you. But i have first
to ask the owner, because i also borrow it.

Thanks for your help

NOx
 

Brian West-3 wrote:
> 
> Can you ship me a phone to test with?  That's usually the missing  
> element when testing this stuff is I just can't afford to buy every  
> phone to test with.
> 
> /b
> 
> On Aug 5, 2009, at 3:52 AM, NOx-WHV wrote:
> 
>>
>> Hello,
>>
>> i have a problem using a innovaphone ip200 with freeswitch and tls/ 
>> srtp. The
>> freeswitch certificate is in the trust list of the phone and it  
>> works with
>> tls for incomming calls. But outgoing calls were rejected to the  
>> mailbox.
>> The freeswitch configuration is ok, because it works with a snom 320.
>>
>> Who can help me to confugure the IP200?
>>
>> Thanks
>>
>> NOx
>> -- 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24841078.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From thangappan143 at gmail.com  Thu Aug  6 02:55:56 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Thu, 6 Aug 2009 15:25:56 +0530
Subject: [Freeswitch-users] IVR on Freeswitch
Message-ID: <7aa29e790908060255p36577d41hffe05ccc380c2540@mail.gmail.com>

Dear all,

   I am in the process of implementing IVR in Perl using event bound socket
on FreeSWITCH. I want to use all the functionality of IVR in my
implementation.

   I have seen the XML MACRO (default implementation) in the ivr.conf .xml
and demo/en/demo-ivr.xml .In that file I don't want to handle the
inter-digit time out and response timeout and all.I can just configure the
seconds.It will automatically works specified in the  <menu> tag.

I want to specify the menu definitions how they have specified in the XML
field in the ivr.conf.xml
Is there any way in Perl to do that?

I want to handle timeout,interdigittimeout  in Perl.Please help me?





-- 
Regards,
Thangappan.M
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From dujinfang at gmail.com  Thu Aug  6 02:58:59 2009
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 6 Aug 2009 17:58:59 +0800
Subject: [Freeswitch-users] A few questions about lua
Message-ID: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>

ALL-
I have a few questions when scripting lua. According to wiki, it is possible
to run looping forever lua scripts through start-up config or luarun.

1) Will the lua script stop when unload mod_lua? I experienced core dump
when unload mod_lua while there was a running lua script. Reported on jira.

2) How to stop a forever running lua script?  I stop it by listening a
CUSTOM event fired elsewhere. See code below. Is there any standard way like
luastop ?

3) Any way to show how many running lua scripts? luashow ?

4) It seems cannot get the lua script name in a lua script, I made a patch
to jira by assign it to the argv[0].

5) Seems that only EventConsumer("all") working.
EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work. Any
idea to this?

Thanks a lot.



code example:

con = freeswitch.EventConsumer("all");

argv[0] = "test.lua"

freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "] Starting
=====\n");

local all_events = 0


for e in (function() return con:pop(1) end) do
  -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
   all_events = all_events + 1;
freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
 event_name = e:getHeader("Event-Name") or ""
event_subclass = e:getHeader("Event-Subclass") or ""
 if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..
"]---Exiting------\n")
  break
end

end
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From dome at tel.co.th  Thu Aug  6 03:48:58 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 6 Aug 2009 17:48:58 +0700
Subject: [Freeswitch-users] Numeric Value Ranges Expressions in dialplan
Message-ID: <8ccbff060908060348l75ee1062ua6145da2d6f7c4e9@mail.gmail.com>

Dear sir,

    Is posible to check numeric range in dialplan (expression).
   example i got balance vaiable from somewhere and want to check > 0
or not before call bridge application.
   ( I don't want to call scripts)

Best regards.

Dome C.


From jason at jasonjgw.net  Thu Aug  6 04:19:04 2009
From: jason at jasonjgw.net (Jason White)
Date: Thu, 6 Aug 2009 21:19:04 +1000
Subject: [Freeswitch-users] Numeric Value Ranges Expressions in dialplan
In-Reply-To: <8ccbff060908060348l75ee1062ua6145da2d6f7c4e9@mail.gmail.com>
References: <8ccbff060908060348l75ee1062ua6145da2d6f7c4e9@mail.gmail.com>
Message-ID: <20090806111903.GB17479@jdc.jasonjgw.net>

Dome Charoenyost <dome at tel.co.th> wrote:
>     Is posible to check numeric range in dialplan (expression).
>    example i got balance vaiable from somewhere and want to check > 0
> or not before call bridge application.
>    ( I don't want to call scripts)

Can you write a regular expression to match it?

^[1-9]\d*$
for example, might be a good start to identify non-zero integers.



From saeedahmad1981 at gmail.com  Thu Aug  6 05:05:48 2009
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Thu, 6 Aug 2009 14:05:48 +0200
Subject: [Freeswitch-users] SVN error
Message-ID: <E47E8DBFC76D4E769B7BBF60561CAA27@saeedlaptop>

Hi,

While doing 'make current' or 'svn up' I am getting following errors:

svn: REPORT request failed on '/svn/!svn/vcc/default'

svn: Can't find a temporary directory: Internal error

 

- Saeed

 

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From raffaele.p.guidi at gmail.com  Thu Aug  6 05:49:19 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Thu, 6 Aug 2009 14:49:19 +0200
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <87f2f3b90908051903y16395a62uf23feec9b3fdceab@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
	<121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
	<b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>
	<87f2f3b90908051903y16395a62uf23feec9b3fdceab@mail.gmail.com>
Message-ID: <b0a8792f0908060549v40196e6fm6da7f95f47193bb4@mail.gmail.com>

Done, it (of course, thanks) worked smoothly. I've published the example on
the wiki.
http://wiki.freeswitch.org/wiki/Fakecall_responder (and linked in mod_lua
samples)

Regards,
   Raffaele

On Thu, Aug 6, 2009 at 04:03, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 6:44 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> Well, I would randomly insert all of those cases to make it more
>> realistic... only thing I cannot manage to issue USER_BUSY from lua (and
>> neither from the dialplan, actually).
>>
>> <anti-action application="respond" data="407" /> (407 or 486 or
>> whatever...)
>>
>>
>> doesn't behave as I expected and neither
>>
>> <action application="hangup" data="407" /> (407 or 486 or USER_BUSY or
>> whatever...)
>>
>>
>> and I cannot find a a session:reject(<hangupcause>) method in lua.
>>
>> Can you give me a hint?
>>
>
> You can execute pretty much any dialplan app with the session:execute
> command:
> http://wiki.freeswitch.org/wiki/Lua#session:execute
>
> Try something like:
> session:execute("hangup","USER_BUSY");
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From elihay at savion.huji.ac.il  Thu Aug  6 03:47:00 2009
From: elihay at savion.huji.ac.il (Eli Hayun)
Date: Thu, 06 Aug 2009 13:47:00 +0300
Subject: [Freeswitch-users] A few questions about lua
In-Reply-To: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
References: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
Message-ID: <1249555620.5449.1.camel@eli-desktop>

Hi
I dont know about events so much but I cannot see variable "e" is
setting

event_name = e:getHeader("Event-Name") or "" 
event_subclass = e:getHeader("Event-Subclass") or "" 

regurds
Eli

On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:

> ALL-
> 
> 
> 
> I have a few questions when scripting lua. According to wiki, it is
> possible to run looping forever lua scripts through start-up config or
> luarun. 
> 
> 
> 1) Will the lua script stop when unload mod_lua? I experienced core
> dump when unload mod_lua while there was a running lua script.
> Reported on jira.
> 
> 
> 2) How to stop a forever running lua script?  I stop it by listening a
> CUSTOM event fired elsewhere. See code below. Is there any standard
> way like luastop ?
> 
> 
> 3) Any way to show how many running lua scripts? luashow ?
> 
> 
> 4) It seems cannot get the lua script name in a lua script, I made a
> patch to jira by assign it to the argv[0].
> 
> 
> 5) Seems that only EventConsumer("all") working.
> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work.
> Any idea to this?
> 
> 
> Thanks a lot.
> 
> 
> 
> 
>                                                
> code example:
> 
> 
> con = freeswitch.EventConsumer("all");      
> 
> 
> argv[0] = "test.lua"
>                        
> freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "]
> Starting =====\n");
>                         
> local all_events = 0
> 
>                                      
> for e in (function() return con:pop(1) end) do
>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
>    all_events = all_events + 1;
> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
> 
> event_name = e:getHeader("Event-Name") or ""
> event_subclass = e:getHeader("Event-Subclass") or ""
> 
> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
>  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..
> "]---Exiting------\n")
>  break
> end
> 
> 
> end 
> 
> 
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From lakindia89 at gmail.com  Thu Aug  6 03:47:20 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 6 Aug 2009 16:17:20 +0530
Subject: [Freeswitch-users] Error while creating object
Message-ID: <7d79b3930908060347xe5be545yfeeafad761aba274@mail.gmail.com>

Hi all,
Greets.

I am in the process of controlling the freeswitch with perl.
I have read about mod_perl and I wrote some scripts to test which works
fine.
Yesterday I tried to access the digit_set function.
So I create an object for the freeswitch::DTMF.
But it reported the following error.

2009-08-06 15:53:46 [ERR] mod_perl.c:69 Perl_safe_eval() [require
'/usr/local/freeswitch/conf/test.pl';]
No matching function for overloaded 'new_DTMF' at
/usr/local/freeswitch/perl/freeswitch.pm line 197.
Compilation failed in require at (eval 2) line 1.

Here is my code.

#!/usr/bin/perl
use strict;
use freeswitch;
our $session;
$session->execute("bridge","user/1010");
my $sess=&freeswitch::DTMF::new;
return 1;

The bridge is working fine. But while creating the object it said error.

Can any one explain why this happens and how can I correct it?
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From jim.page at redmatter.com  Thu Aug  6 04:24:15 2009
From: jim.page at redmatter.com (Jim Page)
Date: Thu, 6 Aug 2009 12:24:15 +0100
Subject: [Freeswitch-users] CURL directory issue
Message-ID: <E0C2263D8B20E14390DF8AA03E1F99A0C506DCB1@sha-exch9.shared.ifeltd.com>

Afternoon All

I wonder if someone (perhaps even the illustrious intralanman) could help me out with a problem I am experiencing with a CURL directory.

In the interests of understanding how the mechanism works, I am using a super-braindead php script to return info about a specific set of users. I plan to move to something more sophisticated once the proof of concept is complete, possibly based on intralanman's scripts.

The basic problem is that all works fine (boot, register, voicemail etc), except that user's <variables> seem not to be being read correctly, eg 'toll_allow' and 'user_context'. Here's a typical user XML message I am returning:

<document type="freeswitch/xml">
  <section name="directory">
   <domain name="pbx.redmatter.com">
    <params>
      <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
    </params>

    <variables>
      <variable name="record_stereo" value="true"/>
      <variable name="default_gateway" value="$${default_provider}"/>
      <variable name="default_areacode" value="$${default_areacode}"/>
      <variable name="transfer_fallback_extension" value="operator"/>
    </variables>

    <groups>
      <group name="default">
        <users>            <user id="1009" cidr="172.30.99.0/24">
               <params>
                 <param name="password" value="$${default_password}"/>
                 <param name="vm-password" value="1009"/>
                 <param name="vm-email-all-messages" value="true"/>
                 <param name="vm-attach-file" value="true"/>
                 <param name="vm-mailto" value="1009 at redmatter.com"/>
                 <param name="vm-keep-local-after-email" value="true"/>
                 <param name="vm_message_ext" value="mp3"/>
               </params>
               <variables>
                 <variable name="toll_allow" value="domestic,international,local"/>
                 <variable name="accountcode" value="1009"/>
                 <variable name="user_context" value="default"/>
                 <variable name="effective_caller_id_name" value="Extension 1009"/>
                 <variable name="effective_caller_id_number" value="1009"/>
                 <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
                 <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
                 <variable name="callgroup" value="techsupport"/>
               </variables>
             </user>        </users>
      </group>
    </groups>
   </domain>
 </section>
</document>

I return this kind of message in all cases except the (sip_auth_method=="REGISTER") request message where I return

<document type="freeswitch/xml">
  <section name="directory">
    <domain name="pbx.redmatter.com">
      <user id="1007">
        <params>
          <param name="password" value="1234"/>
        </params>
      </user>
    </domain>
  </section>
</document>

Also it's probably worth mentioning that I have removed all trace of xml from conf/directory and I don't believe there is a conflict happening there. 

The phones register correctly. The trouble is they don't operate on the correct dialplan context (I fixed that by hardcoding the internal gateway to dialplan default), but the 'toll_allow' variable is now not working so that outbound calls fail, which is what made me think that the user variables are being ignored.

Freeswitch version is 1.0.4, built by me and running on a dell 1950 running Centos 5.3 x86_64. HTTP application is running on standard Centos 5.3 apache/php.

Any ideas gratefully and humbly received.

All the best
Jim


From kevin at johnnyvoip.com  Thu Aug  6 08:48:08 2009
From: kevin at johnnyvoip.com (Kevin Green)
Date: Thu, 6 Aug 2009 11:48:08 -0400
Subject: [Freeswitch-users] CURL directory issue
In-Reply-To: <E0C2263D8B20E14390DF8AA03E1F99A0C506DCB1@sha-exch9.shared.ifeltd.com>
References: <E0C2263D8B20E14390DF8AA03E1F99A0C506DCB1@sha-exch9.shared.ifeltd.com>
Message-ID: <ab57a3e80908060848x64bbc808r162273eb4cbcd363@mail.gmail.com>

Try returning the full information on the register. It may be that the
variables are read onto the user profile upon registration and since you are
only supplying a dumbed down version for registration the variables aren't
being read and cached.

Regards,
   Kevin Green

On Thu, Aug 6, 2009 at 7:24 AM, Jim Page <jim.page at redmatter.com> wrote:

> Afternoon All
>
> I wonder if someone (perhaps even the illustrious intralanman) could help
> me out with a problem I am experiencing with a CURL directory.
>
> In the interests of understanding how the mechanism works, I am using a
> super-braindead php script to return info about a specific set of users. I
> plan to move to something more sophisticated once the proof of concept is
> complete, possibly based on intralanman's scripts.
>
> The basic problem is that all works fine (boot, register, voicemail etc),
> except that user's <variables> seem not to be being read correctly, eg
> 'toll_allow' and 'user_context'. Here's a typical user XML message I am
> returning:
>
> <document type="freeswitch/xml">
>  <section name="directory">
>   <domain name="pbx.redmatter.com">
>    <params>
>      <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>    </params>
>
>    <variables>
>      <variable name="record_stereo" value="true"/>
>      <variable name="default_gateway" value="$${default_provider}"/>
>      <variable name="default_areacode" value="$${default_areacode}"/>
>      <variable name="transfer_fallback_extension" value="operator"/>
>    </variables>
>
>    <groups>
>      <group name="default">
>        <users>            <user id="1009" cidr="172.30.99.0/24">
>               <params>
>                 <param name="password" value="$${default_password}"/>
>                 <param name="vm-password" value="1009"/>
>                 <param name="vm-email-all-messages" value="true"/>
>                 <param name="vm-attach-file" value="true"/>
>                 <param name="vm-mailto" value="1009 at redmatter.com"/>
>                 <param name="vm-keep-local-after-email" value="true"/>
>                 <param name="vm_message_ext" value="mp3"/>
>               </params>
>               <variables>
>                 <variable name="toll_allow"
> value="domestic,international,local"/>
>                 <variable name="accountcode" value="1009"/>
>                 <variable name="user_context" value="default"/>
>                 <variable name="effective_caller_id_name" value="Extension
> 1009"/>
>                 <variable name="effective_caller_id_number" value="1009"/>
>                 <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>                 <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>                 <variable name="callgroup" value="techsupport"/>
>               </variables>
>             </user>        </users>
>      </group>
>    </groups>
>   </domain>
>  </section>
> </document>
>
> I return this kind of message in all cases except the
> (sip_auth_method=="REGISTER") request message where I return
>
> <document type="freeswitch/xml">
>  <section name="directory">
>    <domain name="pbx.redmatter.com">
>      <user id="1007">
>        <params>
>          <param name="password" value="1234"/>
>        </params>
>      </user>
>    </domain>
>  </section>
> </document>
>
> Also it's probably worth mentioning that I have removed all trace of xml
> from conf/directory and I don't believe there is a conflict happening there.
>
> The phones register correctly. The trouble is they don't operate on the
> correct dialplan context (I fixed that by hardcoding the internal gateway to
> dialplan default), but the 'toll_allow' variable is now not working so that
> outbound calls fail, which is what made me think that the user variables are
> being ignored.
>
> Freeswitch version is 1.0.4, built by me and running on a dell 1950 running
> Centos 5.3 x86_64. HTTP application is running on standard Centos 5.3
> apache/php.
>
> Any ideas gratefully and humbly received.
>
> All the best
> Jim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Thu Aug  6 08:52:07 2009
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 6 Aug 2009 23:52:07 +0800
Subject: [Freeswitch-users] A few questions about lua
In-Reply-To: <1249555620.5449.1.camel@eli-desktop>
References: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
	<1249555620.5449.1.camel@eli-desktop>
Message-ID: <EBFA6F4D-07D0-4C6D-ABDF-8A8FF3976FE2@gmail.com>

for e in (function() return con:pop(1) end) do

btw, the script works.

Thanks.
On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote:
> Hi
> I dont know about events so much but I cannot see variable "e" is  
> setting
>
> event_name = e:getHeader("Event-Name") or ""
> event_subclass = e:getHeader("Event-Subclass") or ""
>
> regurds
> Eli
>
> On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:
>> ALL-
>>
>>
>> I have a few questions when scripting lua. According to wiki, it is  
>> possible to run looping forever lua scripts through start-up config  
>> or luarun.
>>
>>
>> 1) Will the lua script stop when unload mod_lua? I experienced core  
>> dump when unload mod_lua while there was a running lua script.  
>> Reported on jira.
>>
>>
>> 2) How to stop a forever running lua script?  I stop it by  
>> listening a CUSTOM event fired elsewhere. See code below. Is there  
>> any standard way like luastop ?
>>
>>
>> 3) Any way to show how many running lua scripts? luashow ?
>>
>>
>> 4) It seems cannot get the lua script name in a lua script, I made  
>> a patch to jira by assign it to the argv[0].
>>
>>
>> 5) Seems that only EventConsumer("all") working.  
>> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to  
>> work. Any idea to this?
>>
>>
>> Thanks a lot.
>>
>>
>>
>>
>>
>> code example:
>>
>>
>> con = freeswitch.EventConsumer("all");
>>
>>
>> argv[0] = "test.lua"
>>
>> freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "]  
>> Starting =====\n");
>>
>> local all_events = 0
>>
>> for e in (function() return con:pop(1) end) do
>>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
>>    all_events = all_events + 1;
>> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
>>
>> event_name = e:getHeader("Event-Name") or ""
>> event_subclass = e:getHeader("Event-Subclass") or ""
>>
>> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
>>  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..  
>> "]---Exiting------\n")
>>  break
>> end
>>
>>
>> end
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From ivan at myrvold.org  Thu Aug  6 08:53:35 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Thu, 6 Aug 2009 17:53:35 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
Message-ID: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>

Is skypiax now working on Mac OS X in Freeswitch?

Ivan


From brian at freeswitch.org  Thu Aug  6 08:55:37 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 6 Aug 2009 10:55:37 -0500
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
Message-ID: <1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>

I'm not sure about that one.... I haven't tried lately because the API  
differs on the Mac last I looked at it.

/b

On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:

> Is skypiax now working on Mac OS X in Freeswitch?
>
> Ivan



From vladrodionov at gmail.com  Thu Aug  6 08:59:36 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Thu, 6 Aug 2009 08:59:36 -0700
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <20090805204530.2ad02225396a31c9de30536f2e338977.43419788fb.wbe@email04.secureserver.net>
References: <20090805204530.2ad02225396a31c9de30536f2e338977.43419788fb.wbe@email04.secureserver.net>
Message-ID: <3c233920908060859q1b8bf67dkbe5750591446fd6@mail.gmail.com>

Pete,

Thank you for script. I can not find find channel variables rdnis,
sip_to_user and all others which start with "sb" on wiki page

http://wiki.freeswitch.org/wiki/Channel_Variables

Are they undocumented?

-Vladimir Rodionov


On Wed, Aug 5, 2009 at 8:45 PM, Pete Mueller <pete at privateconnect.com>wrote:

> Disclaimer: I'm not familiar with all the mods of FS, There may be one that
> does this already.  There are probably many ways to do this, I am just
> offering one that works well for me.
>
> Item #1 - Findout the callee #.   "destination_number" can be set to
> several different things based on the gateway configuration (forced override
> with an extension) and may or may not start with a "+" so the example below
> may not work.  To make matters worse, different gateways set fields
> differently when they hand off the call.  The most reliable I've found is
> "rdnis" or "sip_to_user" , however if you know you are going to stay with
> one gateway, you can relay on the oddities of the way they are configured.
> I had to write something relatively generic, so I moved all processing to a
> script (see #3 below)
>
> Item #2 - Find the caller ID. This is located in "caller_id_number", but
> remember in your processing that caller ID may be "anonymous", "restricted",
> "unknown" or some other word when dealing with blocked/private numbers.  You
> cannot looks for just numbers.
>
> Item #3 - Routing.  As I mentioned I have 100s of numbers across many
> gateways, so I needed a way to route the calls to the right places AND know
> which gateway the call came in on, so I can bridge the call out the same
> gateway.  I handled this by creating a small DB table (using postgreSQL) and
> connecting using LUA and luasql.  The table has three fields: number,
> gateway, and extension to route to.  In my public.xml I list all the places
> a call can be routed to and the last entry is a unconditional transfer to
> the "switchboard" script.  The switchboard script matches "rdnis" and
> "sip_to_user" to find the callee and then performs a lookup for the
> extension to route to.
>
> If you would like a copy of my switchboard script I can provide it to you
> in a PM.
> -pete
>
>  -------- Original Message --------
> Subject: Re: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
> configure?)
> From: Vladimir Rodionov <vladrodionov at gmail.com>
> Date: Wed, August 05, 2009 6:57 pm
> To: freeswitch-users at lists.freeswitch.org
>
> No, it is more like static routing. I need my *script program* be invoked
> when somebody dial in. That is it. One script for all inbound DIDs. Suppose
> I have thousand of them.  I think I know how to accomplish this but I am not
> sure yet.
>
> in my dialplan I need to define:
>
> <!-- Launch a JavaScript application if dialed in-->
>    <extension name="ProviderABC">
>     <condition field="source" expression="mod_sofia"/>
>     *<condition field="destination_number" expression="^1NXXNXXXXXX$">*
>
>      <action application="javascript" data="/usr/local/freeswitch/scripts/myapp.js"/>
>     </condition>
>    </extension>
>
>
> In provider configuration:
>
> <gateway name="voicepulse">
>            <!--/// account username *required* ///-->
>            <param name="username" value="your-username"/>
>            <!--/// auth realm: *optional* same as gateway name, if blank ///-->
>
>            <param name="realm" value="nyc.voicepulse.com"/>
>            <!--/// account password *required* ///-->
>            <param name="password" value="your-password"/>
>
>            <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
> *           * *<param name="extension" value="1NXXNXXXXXX"/> *
>            <!--/// proxy host: *optional* same as realm, if blank ///-->
>
>            <param name="proxy" value="nyc.voicepulse.com"/>
>            <!--/// expire in seconds: *optional* 3600, if blank ///-->
>            <param name="expire-seconds" value="600"/>
>
> 	   <param name="register" value="true"/>
>          </gateway>
>
>
> Something like this, yes? I can use regular expressions in
> destination_number?
>
> Q: There is object Session in JavaScript, Lua. Is Session.destination ==
> destination_number from incoming call? It is not clear for me from what I
> have read so far.
>
> TIA,
>
> -Vladimir Rodionov
>
> On Wed, Aug 5, 2009 at 6:26 PM, Seven Du <dujinfang at gmail.com> wrote:
>
>> mod_easyroute?
>>
>> 2009/8/6 Vladimir Rodionov <vladrodionov at gmail.com>
>>
>>>  Hi, everybody
>>>
>>> This is a newbie question: Suppose I have XX (variable dynamic number)
>>> DIDs assigned to one sip trunk (from VOIP provider ABC ). All calls coming
>>> from VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is
>>> it possible in FS? If yes, how everything should be configuered? Dialplan,
>>> sip gateway? One more question: suppose it is doeable as I hope then how can
>>> I get in my script CalleeID (not a CallerID)? Basicaly,
>>>
>>> I want to acomplish the following:
>>>
>>> 1. Avoid re-configuring FS every time I got new bunch of DIDs
>>> assigned/released from/to my Voip provider.
>>> 2. Have a way of extracting CalleeID in my script.
>>>
>>> TIA,
>>>
>>> Vladimir Rodionov
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vladrodionov at gmail.com  Thu Aug  6 09:00:36 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Thu, 6 Aug 2009 09:00:36 -0700
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <87f2f3b90908052040s21545229qba24937bd2b14540@mail.gmail.com>
References: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
	<23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>
	<3c233920908051857t5037d169r6bc4555b53cd015c@mail.gmail.com>
	<87f2f3b90908052040s21545229qba24937bd2b14540@mail.gmail.com>
Message-ID: <3c233920908060900j33cf4cb3hf50ca4452d1b245e@mail.gmail.com>

Thanks, I will give it it a try and let you know.

On Wed, Aug 5, 2009 at 8:40 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 8:57 PM, Vladimir Rodionov <vladrodionov at gmail.com>wrote:
>
>> No, it is more like static routing. I need my *script program* be invoked
>> when somebody dial in. That is it. One script for all inbound DIDs. Suppose
>> I have thousand of them.  I think I know how to accomplish this but I am not
>> sure yet.
>>
>
> Have the external profile be used only for provider ABC, or define a new
> profile. Then in the profile have the calls go to a specific context. You
> could have something like this in the sip profile definition:
>
> <param name="context" value="abc_calls"/>
>
> Then create a dialplan context called "abc_calls" that handles all inbound
> calls. Create a file in conf/dialplan/ called abc_calls.xml:
>
> <include>
>   <context name="abc_calls">
>     <extension name="abc_calls">
>       <condition field="destination_number" expression="^(.*)$">
>         <action application="lua" data="myscript.lua"/>
>       </condition>
>     </extension>
>   </context>
> </include>
>
> Essentially you're just creating a SIP profile and a dialplan context that
> are servicing your VoIP provider. You can add other profiles/contexts for
> other providers if need be.
>
> Let us know how it goes...
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Thu Aug  6 09:37:25 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 6 Aug 2009 18:37:25 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org> 
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
Message-ID: <7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>

No, it needs implementation of the message pump between the module and
the Skype API.

It's probably kind of trivial, if no other problems I'm not aware of.

I do not have a Mac to implement it, tough :-(.

-giovanni





Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org> wrote:
> I'm not sure about that one.... I haven't tried lately because the API
> differs on the Mac last I looked at it.
>
> /b
>
> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>
>> Is skypiax now working on Mac OS X in Freeswitch?
>>
>> Ivan
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From nicolas at medularis.com  Thu Aug  6 09:38:34 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 6 Aug 2009 12:38:34 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
Message-ID: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>

I'm bridging 2 calls in a javascript file, I originate the first call and
then execute a bridge with an origination string for the second call. If I
hangup the first call while trying to make the second call, I get this on
the console:

2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL

But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
NORMAL_CLEARING. And the variable_originate_disposition has a value of
"failure". Where can I get the detail of the call/bridge failure due to
'ORIGINATOR_CANCEL' as reported through the console?

Thanks!

Nicolas
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From msc at freeswitch.org  Thu Aug  6 09:45:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Aug 2009 11:45:33 -0500
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <b0a8792f0908060549v40196e6fm6da7f95f47193bb4@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
	<121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
	<b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>
	<87f2f3b90908051903y16395a62uf23feec9b3fdceab@mail.gmail.com>
	<b0a8792f0908060549v40196e6fm6da7f95f47193bb4@mail.gmail.com>
Message-ID: <87f2f3b90908060945y5cb7e842le9c9ba02bc3c303b@mail.gmail.com>

On Thu, Aug 6, 2009 at 7:49 AM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> Done, it (of course, thanks) worked smoothly. I've published the example on
> the wiki.
> http://wiki.freeswitch.org/wiki/Fakecall_responder (and linked in mod_lua
> samples)
>
> Regards,
>    Raffaele
>

Thanks for paying the wiki tax! We appreciate it when folks document their
knowledge. Please let me know if you have any wiki questions in the future.
-MC
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From vkozak at abisoft.spb.ru  Thu Aug  6 10:01:09 2009
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Thu, 6 Aug 2009 21:01:09 +0400
Subject: [Freeswitch-users] FreeSwitch doesn't play music on hold forbriged
	channel
Message-ID: <EC1055E8A337451F9F5ADC7414BDFDD9@abisoft.biz>


The scenario is the following:
FS User A dial an extension
Extention opens outbound socket channel to my application
My application bridges the call to FS User B
The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id
The application sends hold to the bridged channel using SendMsg with Other-leg-unique-id
User B is placed on hold but no music on hold is played to the caller (User A)


I have outbound socket channel and the following sequence of commands/event:
listening on [any] 8084 ...
connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250] 34000
connect

myevents

SendMsg
call-command: execute
execute-app-name: bridge
execute-app-arg:user/1000 at uat.agent.starpoundtech.net

Channel-Username: 1001
Channel-Dialplan: XML
Channel-Caller-ID-Name: 1001
Channel-Caller-ID-Number: 1001
Channel-Network-Addr: 172.26.10.39
Channel-Destination-Number: 6666
Channel-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Channel-Source: mod_sofia
Channel-Context: default
Channel-Channel-Name: sofia/internal/1001%40172.26.200.250
Channel-Profile-Index: 1
Channel-Profile-Created-Time: 1249142681680114
Channel-Channel-Created-Time: 1249142681680114
Channel-Channel-Answered-Time: 0
Channel-Channel-Progress-Time: 0
Channel-Channel-Progress-Media-Time: 1249142681809352
Channel-Channel-Hangup-Time: 0
Channel-Channel-Transfer-Time: 0
Channel-Screen-Bit: true
Channel-Privacy-Hide-Name: false
Channel-Privacy-Hide-Number: false
Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/internal/1001%40172.26.200.250
Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Call-Direction: inbound
Answer-State: early
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Network-Addr: 172.26.10.39
Caller-Destination-Number: 6666
Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249142681680114
Caller-Channel-Created-Time: 1249142681680114
Caller-Channel-Answered-Time: 0
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 1249142681809352
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
variable_sip_received_ip: 172.26.10.39
variable_sip_received_port: 13488
variable_sip_via_protocol: udp
variable_sip_authorized: true
variable_sip_mailbox: 1001
variable_sip_auth_username: 1001
variable_sip_auth_realm: 172.26.200.250
variable_mailbox: 1001
variable_toll_allow: domestic,international,local
variable_accountcode: 1001
variable_user_context: default
variable_effective_caller_id_name: Extension%201001
variable_effective_caller_id_number: 1001
variable_outbound_caller_id_name: StarPound%20FreeSWITCH
variable_outbound_caller_id_number: 0000000000
variable_callgroup: techsupport
variable_sip_from_user: 1001
variable_sip_from_uri: 1001%40172.26.200.250
variable_sip_from_host: 172.26.200.250
variable_sip_from_user_stripped: 1001
variable_sip_from_tag: bd11f93c
variable_sofia_profile_name: internal
variable_sip_req_user: 6666
variable_sip_req_uri: 6666%40172.26.200.250
variable_sip_req_host: 172.26.200.250
variable_sip_to_user: 6666
variable_sip_to_uri: 6666%40172.26.200.250
variable_sip_to_host: 172.26.200.250
variable_sip_contact_user: 1001
variable_sip_contact_port: 13488
variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
variable_sip_contact_host: 172.26.10.39
variable_channel_name: sofia/internal/1001%40172.26.200.250
variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
variable_sip_via_host: 172.26.10.39
variable_sip_via_port: 13488
variable_sip_via_rport: 13488
variable_max_forwards: 70
variable_presence_id: 1001%40172.26.200.250
variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_remote_media_ip: 172.26.10.39
variable_remote_media_port: 29826
variable_read_codec: PCMU
variable_read_rate: 8000
variable_write_codec: PCMU
variable_write_rate: 8000
variable_use_profile: nat
variable_record_stereo: true
variable_transfer_fallback_extension: operator
variable_numbering_plan: US
variable_default_areacode: 918
variable_default_gateway: example.com
variable_user_name: default
variable_domain_name: 172.26.200.250
variable_current_application_data: 172.26.200.251%3A8084%20async%20full
variable_current_application: socket
variable_socket_host: 172.26.200.251
variable_local_media_ip: 172.26.200.250
variable_local_media_port: 29370
variable_endpoint_disposition: EARLY%20MEDIA
variable_sip_nat_detected: true
Content-Type: command/reply
Reply-Text: %2BOK%0A
Socket-Mode: async
Control: full

Content-Type: command/reply
Reply-Text: +OK Events Enabled

Content-Type: command/reply
Reply-Text: +OK

Content-Length: 1541
Content-Type: text/event-plain

Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/internal/1001%40172.26.200.250
Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Call-Direction: inbound
Answer-State: early
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Network-Addr: 172.26.10.39
Caller-Destination-Number: 6666
Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249142681680114
Caller-Channel-Created-Time: 1249142681680114
Caller-Channel-Answered-Time: 0
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 1249142681809352
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Application: bridge
Application-Data: user/1000%40uat.agent.starpoundtech.net
Event-Name: CHANNEL_EXECUTE
Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
FreeSWITCH-Hostname: centos4-4-vm
FreeSWITCH-IPv4: 172.26.200.250
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-01%2020%3A04%3A51
Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A51%20GMT
Event-Date-Timestamp: 1249142691754598
Event-Calling-File: switch_core_session.c
Event-Calling-Function: switch_core_session_exec
Event-Calling-Line-Number: 1333

Content-Length: 5242
Content-Type: text/event-plain

Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/internal/1001%40172.26.200.250
Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Call-Direction: inbound
Answer-State: answered
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Network-Addr: 172.26.10.39
Caller-Destination-Number: 6666
Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249142681680114
Caller-Channel-Created-Time: 1249142681680114
Caller-Channel-Answered-Time: 1249142692414509
Caller-Channel-Progress-Time: 1249142691898434
Caller-Channel-Progress-Media-Time: 1249142681809352
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Other-Leg-Username: 1001
Other-Leg-Dialplan: XML
Other-Leg-Caller-ID-Name: Extension%201001
Other-Leg-Caller-ID-Number: 1001
Other-Leg-Network-Addr: 172.26.10.39
Other-Leg-Destination-Number: sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
Other-Leg-Source: mod_sofia
Other-Leg-Context: default
Other-Leg-Channel-Name: sofia/internal/sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
Other-Leg-Screen-Bit: true
Other-Leg-Privacy-Hide-Name: false
Other-Leg-Privacy-Hide-Number: false
variable_sip_received_ip: 172.26.10.39
variable_sip_received_port: 13488
variable_sip_via_protocol: udp
variable_sip_authorized: true
variable_sip_mailbox: 1001
variable_sip_auth_username: 1001
variable_sip_auth_realm: 172.26.200.250
variable_mailbox: 1001
variable_toll_allow: domestic,international,local
variable_accountcode: 1001
variable_user_context: default
variable_effective_caller_id_name: Extension%201001
variable_effective_caller_id_number: 1001
variable_outbound_caller_id_name: StarPound%20FreeSWITCH
variable_outbound_caller_id_number: 0000000000
variable_callgroup: techsupport
variable_sip_from_user: 1001
variable_sip_from_uri: 1001%40172.26.200.250
variable_sip_from_host: 172.26.200.250
variable_sip_from_user_stripped: 1001
variable_sip_from_tag: bd11f93c
variable_sofia_profile_name: internal
variable_sip_req_user: 6666
variable_sip_req_uri: 6666%40172.26.200.250
variable_sip_req_host: 172.26.200.250
variable_sip_to_user: 6666
variable_sip_to_uri: 6666%40172.26.200.250
variable_sip_to_host: 172.26.200.250
variable_sip_contact_user: 1001
variable_sip_contact_port: 13488
variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
variable_sip_contact_host: 172.26.10.39
variable_channel_name: sofia/internal/1001%40172.26.200.250
variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
variable_sip_via_host: 172.26.10.39
variable_sip_via_port: 13488
variable_sip_via_rport: 13488
variable_max_forwards: 70
variable_presence_id: 1001%40172.26.200.250
variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_remote_media_ip: 172.26.10.39
variable_remote_media_port: 29826
variable_read_codec: PCMU
variable_read_rate: 8000
variable_write_codec: PCMU
variable_write_rate: 8000
variable_use_profile: nat
variable_record_stereo: true
variable_transfer_fallback_extension: operator
variable_numbering_plan: US
variable_default_areacode: 918
variable_default_gateway: example.com
variable_user_name: default
variable_domain_name: 172.26.200.250
variable_socket_host: 172.26.200.251
variable_local_media_ip: 172.26.200.250
variable_local_media_port: 29370
variable_endpoint_disposition: EARLY%20MEDIA
variable_current_application_data: user/1000%40uat.agent.starpoundtech.net
variable_current_application: bridge
variable_dialed_user: 1000
variable_dialed_domain: uat.agent.starpoundtech.net
variable_originate_disposition: failure
variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
variable_sip_redirect_contact_user_0: 1000
variable_sip_redirect_contact_host_0: 172.26.10.39
variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_sip_nat_detected: true
Event-Name: CHANNEL_ANSWER
Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
FreeSWITCH-Hostname: centos4-4-vm
FreeSWITCH-IPv4: 172.26.200.250
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-01%2020%3A04%3A52
Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
Event-Date-Timestamp: 1249142692414509
Event-Calling-File: switch_channel.c
Event-Calling-Function: switch_channel_perform_mark_answered
Event-Calling-Line-Number: 1776

Content-Length: 5233
Content-Type: text/event-plain

Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/internal/1001%40172.26.200.250
Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Call-Direction: inbound
Answer-State: answered
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Network-Addr: 172.26.10.39
Caller-Destination-Number: 6666
Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249142681680114
Caller-Channel-Created-Time: 1249142681680114
Caller-Channel-Answered-Time: 1249142692414509
Caller-Channel-Progress-Time: 1249142691898434
Caller-Channel-Progress-Media-Time: 1249142681809352
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Other-Leg-Username: 1001
Other-Leg-Dialplan: XML
Other-Leg-Caller-ID-Name: Extension%201001
Other-Leg-Caller-ID-Number: 1001
Other-Leg-Network-Addr: 172.26.10.39
Other-Leg-Destination-Number: sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
Other-Leg-Source: mod_sofia
Other-Leg-Context: default
Other-Leg-Channel-Name: sofia/internal/sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
Other-Leg-Screen-Bit: true
Other-Leg-Privacy-Hide-Name: false
Other-Leg-Privacy-Hide-Number: false
variable_sip_received_ip: 172.26.10.39
variable_sip_received_port: 13488
variable_sip_via_protocol: udp
variable_sip_authorized: true
variable_sip_mailbox: 1001
variable_sip_auth_username: 1001
variable_sip_auth_realm: 172.26.200.250
variable_mailbox: 1001
variable_toll_allow: domestic,international,local
variable_accountcode: 1001
variable_user_context: default
variable_effective_caller_id_name: Extension%201001
variable_effective_caller_id_number: 1001
variable_outbound_caller_id_name: StarPound%20FreeSWITCH
variable_outbound_caller_id_number: 0000000000
variable_callgroup: techsupport
variable_sip_from_user: 1001
variable_sip_from_uri: 1001%40172.26.200.250
variable_sip_from_host: 172.26.200.250
variable_sip_from_user_stripped: 1001
variable_sip_from_tag: bd11f93c
variable_sofia_profile_name: internal
variable_sip_req_user: 6666
variable_sip_req_uri: 6666%40172.26.200.250
variable_sip_req_host: 172.26.200.250
variable_sip_to_user: 6666
variable_sip_to_uri: 6666%40172.26.200.250
variable_sip_to_host: 172.26.200.250
variable_sip_contact_user: 1001
variable_sip_contact_port: 13488
variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
variable_sip_contact_host: 172.26.10.39
variable_channel_name: sofia/internal/1001%40172.26.200.250
variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
variable_sip_via_host: 172.26.10.39
variable_sip_via_port: 13488
variable_sip_via_rport: 13488
variable_max_forwards: 70
variable_presence_id: 1001%40172.26.200.250
variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_remote_media_ip: 172.26.10.39
variable_remote_media_port: 29826
variable_read_codec: PCMU
variable_read_rate: 8000
variable_write_codec: PCMU
variable_write_rate: 8000
variable_use_profile: nat
variable_record_stereo: true
variable_transfer_fallback_extension: operator
variable_numbering_plan: US
variable_default_areacode: 918
variable_default_gateway: example.com
variable_user_name: default
variable_domain_name: 172.26.200.250
variable_socket_host: 172.26.200.251
variable_local_media_ip: 172.26.200.250
variable_local_media_port: 29370
variable_current_application_data: user/1000%40uat.agent.starpoundtech.net
variable_current_application: bridge
variable_dialed_user: 1000
variable_dialed_domain: uat.agent.starpoundtech.net
variable_sip_redirect_contact_user_0: 1000
variable_sip_redirect_contact_host_0: 172.26.10.39
variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_sip_nat_detected: true
variable_endpoint_disposition: ANSWER
variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
variable_originate_disposition: SUCCESS
Event-Name: CHANNEL_BRIDGE
Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
FreeSWITCH-Hostname: centos4-4-vm
FreeSWITCH-IPv4: 172.26.200.250
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-01%2020%3A04%3A52
Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
Event-Date-Timestamp: 1249142692414509
Event-Calling-File: switch_ivr_bridge.c
Event-Calling-Function: switch_ivr_multi_threaded_bridge
Event-Calling-Line-Number: 828

SendMsg 94b59a38-57c4-4703-9c6e-9985d832d119
call-command: execute
execute-app-name: hold

Content-Type: command/reply
Reply-Text: +OK




- I don't see the variable hold_music ... did you remove it? 

- I didn't. Moreover, I tried to set it explicitly using api uuid_setvar.
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From msc at freeswitch.org  Thu Aug  6 10:30:09 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Aug 2009 12:30:09 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 Release Announcement
Message-ID: <87f2f3b90908061030m1afd5adajcc90e31b6a10b5f6@mail.gmail.com>

We are happy to announce the official release of FreeSWITCH 1.0.4! Please
visit this link <http://digg.com/d3zt5X> to Digg and read the story, and
then spread the word!

Thanks for being such a great community!
-Michael
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From mattdfong at gmail.com  Thu Aug  6 11:25:32 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Thu, 6 Aug 2009 11:25:32 -0700
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
Message-ID: <4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>

Hi Nicolas,
do you have a copy of the .js code you can paste. I would guess tho, that
ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to
false. Just a guess tho.

Hangup causes can be found here:
http://wiki.freeswitch.org/wiki/Hangup_causes

<http://wiki.freeswitch.org/wiki/Hangup_causes>--matt
hello hunter - hosted predictive dialer & voice broadcasting
http://www.hellohunter.com


On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

> I'm bridging 2 calls in a javascript file, I originate the first call and
> then execute a bridge with an origination string for the second call. If I
> hangup the first call while trying to make the second call, I get this on
> the console:
>
> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/005622170039 [KILL]
> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/005622170039 [BREAK]
> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
> Cause: ORIGINATOR_CANCEL
>
> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
> NORMAL_CLEARING. And the variable_originate_disposition has a value of
> "failure". Where can I get the detail of the call/bridge failure due to
> 'ORIGINATOR_CANCEL' as reported through the console?
>
> Thanks!
>
> Nicolas
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nicolas at medularis.com  Thu Aug  6 12:45:01 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 6 Aug 2009 15:45:01 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
	<4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
Message-ID: <1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>

Hi Matt,

Actually I'm explicitly setting hangup_after_bridge to true, think setting
it to false would help? I'm going to try that.

Here's the JS code:
(Note: session.getVariable() doesn't work, FS complains saying it is not a
function, also tried self.session.getVariable() - that's what the wiki says
- and FS complains that self does not exist)

----------------
var uuid = argv[0]; // Call identifier
var dialstr1 = argv[1]; // Dial string obtained from previous call to LCR
var dialstr2 = argv[2]; // Dial string obtained from previous call to LCR
var greeting_snd = "/var/audio/alert.wav";

console_log("notice", "*********** STARTING C2C Call ***********\n");
timeout = 30;

console_log("notice", "*********** DIALING "+dialstr1+" ***********\n");

//var stUsRing = session.getVariable("us-ring");  // This doesn't work,
self.session.getVariable doesn't work either
var stUsRing = "%(2000,4000,440,480)";

// Create new_session
new_session = new Session(originate_str1);
console_log("notice", "*********** Leg1: " + new_session.cause + "
***********\n");

if (new_session.ready()) {
        // log to the console
        console_log("notice", "*********** Leg1 ("+dialstr1+") CONNECTED!
***********\n");
        console_log("notice", "*********** Playing greeting sound:
"+greeting_snd+" ***********\n");

        new_session.execute("sleep", 100);
        new_session.execute("playback", greeting_snd);

        // Originate second call and bridge
    originate_str2 =
"{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;

        // Create new_session
        new_session.execute("bridge", originate_str2);
        console_log("notice", "*********** Leg2: " + new_session.cause + "
***********\n");

        if (new_session.ready()) {
                console_log("notice", "*********** Leg2 ("+dialstr2+")
CONNECTED! ***********\n");
        }
}

exit();
----------------

Thanks!


Nicolas


On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> Hi Nicolas,
> do you have a copy of the .js code you can paste. I would guess tho, that
> ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to
> false. Just a guess tho.
>
> Hangup causes can be found here:
> http://wiki.freeswitch.org/wiki/Hangup_causes
>
> <http://wiki.freeswitch.org/wiki/Hangup_causes> --matt
> hello hunter - hosted predictive dialer & voice broadcasting
> http://www.hellohunter.com
>
>
> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> I'm bridging 2 calls in a javascript file, I originate the first call and
>> then execute a bridge with an origination string for the second call. If I
>> hangup the first call while trying to make the second call, I get this on
>> the console:
>>
>> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
>> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
>> sofia/external/005622170039 [KILL]
>> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/005622170039 [BREAK]
>> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
>> Cause: ORIGINATOR_CANCEL
>>
>> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
>> NORMAL_CLEARING. And the variable_originate_disposition has a value of
>> "failure". Where can I get the detail of the call/bridge failure due to
>> 'ORIGINATOR_CANCEL' as reported through the console?
>>
>> Thanks!
>>
>> Nicolas
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jim.page at redmatter.com  Thu Aug  6 14:55:08 2009
From: jim.page at redmatter.com (Jim Page)
Date: Thu, 6 Aug 2009 22:55:08 +0100
Subject: [Freeswitch-users] CURL directory issue
In-Reply-To: <ab57a3e80908060848x64bbc808r162273eb4cbcd363@mail.gmail.com>
References: <E0C2263D8B20E14390DF8AA03E1F99A0C506DCB1@sha-exch9.shared.ifeltd.com>
	<ab57a3e80908060848x64bbc808r162273eb4cbcd363@mail.gmail.com>
Message-ID: <C45A122C-C1C0-4267-9028-C192CED52ED4@redmatter.com>

Spot on. Many thanks!
Jim

Sent from my iPhone

On 6 Aug 2009, at 18:02, "Kevin Green" <kevin at johnnyvoip.com<mailto:kevin at johnnyvoip.com>> wrote:

Try returning the full information on the register. It may be that the variables are read onto the user profile upon registration and since you are only supplying a dumbed down version for registration the variables aren't being read and cached.

Regards,
   Kevin Green

On Thu, Aug 6, 2009 at 7:24 AM, Jim Page <<mailto:jim.page at redmatter.com>jim.page at redmatter.com<mailto:jim.page at redmatter.com>> wrote:
Afternoon All

I wonder if someone (perhaps even the illustrious intralanman) could help me out with a problem I am experiencing with a CURL directory.

In the interests of understanding how the mechanism works, I am using a super-braindead php script to return info about a specific set of users. I plan to move to something more sophisticated once the proof of concept is complete, possibly based on intralanman's scripts.

The basic problem is that all works fine (boot, register, voicemail etc), except that user's <variables> seem not to be being read correctly, eg 'toll_allow' and 'user_context'. Here's a typical user XML message I am returning:

<document type="freeswitch/xml">
 <section name="directory">
  <domain name="<http://pbx.redmatter.com>pbx.redmatter.com<http://pbx.redmatter.com>">
   <params>
     <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
   </params>

   <variables>
     <variable name="record_stereo" value="true"/>
     <variable name="default_gateway" value="$${default_provider}"/>
     <variable name="default_areacode" value="$${default_areacode}"/>
     <variable name="transfer_fallback_extension" value="operator"/>
   </variables>

   <groups>
     <group name="default">
       <users>            <user id="1009" cidr="172.30.99.0/24<http://172.30.99.0/24>">
              <params>
                <param name="password" value="$${default_password}"/>
                <param name="vm-password" value="1009"/>
                <param name="vm-email-all-messages" value="true"/>
                <param name="vm-attach-file" value="true"/>
                <param name="vm-mailto" value="<mailto:1009 at redmatter.com>1009 at redmatter.com<mailto:1009 at redmatter.com>"/>
                <param name="vm-keep-local-after-email" value="true"/>
                <param name="vm_message_ext" value="mp3"/>
              </params>
              <variables>
                <variable name="toll_allow" value="domestic,international,local"/>
                <variable name="accountcode" value="1009"/>
                <variable name="user_context" value="default"/>
                <variable name="effective_caller_id_name" value="Extension 1009"/>
                <variable name="effective_caller_id_number" value="1009"/>
                <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
                <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
                <variable name="callgroup" value="techsupport"/>
              </variables>
            </user>        </users>
     </group>
   </groups>
  </domain>
 </section>
</document>

I return this kind of message in all cases except the (sip_auth_method=="REGISTER") request message where I return

<document type="freeswitch/xml">
 <section name="directory">
   <domain name="<http://pbx.redmatter.com>pbx.redmatter.com<http://pbx.redmatter.com>">
     <user id="1007">
       <params>
         <param name="password" value="1234"/>
       </params>
     </user>
   </domain>
 </section>
</document>

Also it's probably worth mentioning that I have removed all trace of xml from conf/directory and I don't believe there is a conflict happening there.

The phones register correctly. The trouble is they don't operate on the correct dialplan context (I fixed that by hardcoding the internal gateway to dialplan default), but the 'toll_allow' variable is now not working so that outbound calls fail, which is what made me think that the user variables are being ignored.

Freeswitch version is 1.0.4, built by me and running on a dell 1950 running Centos 5.3 x86_64. HTTP application is running on standard Centos 5.3 apache/php.

Any ideas gratefully and humbly received.

All the best
Jim

_______________________________________________
FreeSWITCH-users mailing list
<mailto:FreeSWITCH-users at lists.freeswitch.org>FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://www.freeswitch.org>http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From Nick.Lemberger at lkfd.net  Thu Aug  6 16:19:03 2009
From: Nick.Lemberger at lkfd.net (Nick Lemberger)
Date: Thu, 06 Aug 2009 18:19:03 -0500
Subject: [Freeswitch-users] Lua Script Return Value & mod_xmlrpc
Message-ID: <4A7B1EC4.2C9A.00FE.0@lkfd.net>

Is it possible to have a LUA script return something to the client when accessed via the XML RPC gateway & luarun?

ie: access the url: http://FSip:8080/api/luarun?myscript.lua and have the script return a value?

-Nick



From pete at privateconnect.com  Thu Aug  6 16:35:30 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Thu, 06 Aug 2009 16:35:30 -0700
Subject: [Freeswitch-users]
	=?utf-8?q?Lua_Script_Return_Value_=26_mod=5Fxm?= =?utf-8?q?lrpc?=
Message-ID: <20090806163530.2ad02225396a31c9de30536f2e338977.1282ed1d44.wbe@email04.secureserver.net>

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From raffaele.p.guidi at gmail.com  Thu Aug  6 17:23:59 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Fri, 7 Aug 2009 02:23:59 +0200
Subject: [Freeswitch-users] A few questions about lua
In-Reply-To: <EBFA6F4D-07D0-4C6D-ABDF-8A8FF3976FE2@gmail.com>
References: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
	<1249555620.5449.1.camel@eli-desktop>
	<EBFA6F4D-07D0-4C6D-ABDF-8A8FF3976FE2@gmail.com>
Message-ID: <b0a8792f0908061723q4fe2a24cobea858addbf4e07@mail.gmail.com>

>> 5) Seems that only EventConsumer("all") working.
EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work. Any
idea to this?

isn't it CHANNEL_HAN*G*UP? Is the G missing only in the email or in the
code, too?

On Thu, Aug 6, 2009 at 17:52, Seven Du <dujinfang at gmail.com> wrote:

> for e in (function() return con:pop(1) end) do
>
> btw, the script works.
>
> Thanks.
> On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote:
> > Hi
> > I dont know about events so much but I cannot see variable "e" is
> > setting
> >
> > event_name = e:getHeader("Event-Name") or ""
> > event_subclass = e:getHeader("Event-Subclass") or ""
> >
> > regurds
> > Eli
> >
> > On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:
> >> ALL-
> >>
> >>
> >> I have a few questions when scripting lua. According to wiki, it is
> >> possible to run looping forever lua scripts through start-up config
> >> or luarun.
> >>
> >>
> >> 1) Will the lua script stop when unload mod_lua? I experienced core
> >> dump when unload mod_lua while there was a running lua script.
> >> Reported on jira.
> >>
> >>
> >> 2) How to stop a forever running lua script?  I stop it by
> >> listening a CUSTOM event fired elsewhere. See code below. Is there
> >> any standard way like luastop ?
> >>
> >>
> >> 3) Any way to show how many running lua scripts? luashow ?
> >>
> >>
> >> 4) It seems cannot get the lua script name in a lua script, I made
> >> a patch to jira by assign it to the argv[0].
> >>
> >>
> >> 5) Seems that only EventConsumer("all") working.
> >> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to
> >> work. Any idea to this?
> >>
> >>
> >> Thanks a lot.
> >>
> >>
> >>
> >>
> >>
> >> code example:
> >>
> >>
> >> con = freeswitch.EventConsumer("all");
> >>
> >>
> >> argv[0] = "test.lua"
> >>
> >> freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "]
> >> Starting =====\n");
> >>
> >> local all_events = 0
> >>
> >> for e in (function() return con:pop(1) end) do
> >>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
> >>    all_events = all_events + 1;
> >> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
> >>
> >> event_name = e:getHeader("Event-Name") or ""
> >> event_subclass = e:getHeader("Event-Subclass") or ""
> >>
> >> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
> >>  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..
> >> "]---Exiting------\n")
> >>  break
> >> end
> >>
> >>
> >> end
> >>
> >>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vladrodionov at gmail.com  Thu Aug  6 17:55:55 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Thu, 6 Aug 2009 17:55:55 -0700
Subject: [Freeswitch-users] Lua on Windows and additional modules
Message-ID: <3c233920908061755h5d17aa0as3fb24743215a8298@mail.gmail.com>

Good evening,
This is newbie question.

The FreeSWITCH lua module does not support sockets and sql out of box that
is why
I just installed LuaBinaries (including socket, sql modules). My dev
environment is Win XP not Linux/Unix.

I am trying to understand what will happen when lua_module get this:

require "socket" or
require "luasql.mysql"
?

How does lua_module look up additional lua modules on Windows platform?

Do I have to set some env variables?

TIA
-Vladimir Rodionov
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From dujinfang at gmail.com  Thu Aug  6 20:45:16 2009
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 7 Aug 2009 11:45:16 +0800
Subject: [Freeswitch-users] A few questions about lua
In-Reply-To: <b0a8792f0908061723q4fe2a24cobea858addbf4e07@mail.gmail.com>
References: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
	<1249555620.5449.1.camel@eli-desktop>
	<EBFA6F4D-07D0-4C6D-ABDF-8A8FF3976FE2@gmail.com>
	<b0a8792f0908061723q4fe2a24cobea858addbf4e07@mail.gmail.com>
Message-ID: <23f91030908062045s6ebc35efne6f5ce999f085506@mail.gmail.com>

Sorry it's a typo. I read the code, it works not like in event socket. So,
only works with one event.
either
EventConsumer("all")
or
EventConsumer("CUSTOM", "lua::stop");


Thank you.

2009/8/7 Raffaele P. Guidi <raffaele.p.guidi at gmail.com>

> >> 5) Seems that only EventConsumer("all") working.
> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work. Any
> idea to this?
>
> isn't it CHANNEL_HAN*G*UP? Is the G missing only in the email or in the
> code, too?
>
> On Thu, Aug 6, 2009 at 17:52, Seven Du <dujinfang at gmail.com> wrote:
>
>> for e in (function() return con:pop(1) end) do
>>
>> btw, the script works.
>>
>> Thanks.
>> On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote:
>> > Hi
>> > I dont know about events so much but I cannot see variable "e" is
>> > setting
>> >
>> > event_name = e:getHeader("Event-Name") or ""
>> > event_subclass = e:getHeader("Event-Subclass") or ""
>> >
>> > regurds
>> > Eli
>> >
>> > On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:
>> >> ALL-
>> >>
>> >>
>> >> I have a few questions when scripting lua. According to wiki, it is
>> >> possible to run looping forever lua scripts through start-up config
>> >> or luarun.
>> >>
>> >>
>> >> 1) Will the lua script stop when unload mod_lua? I experienced core
>> >> dump when unload mod_lua while there was a running lua script.
>> >> Reported on jira.
>> >>
>> >>
>> >> 2) How to stop a forever running lua script?  I stop it by
>> >> listening a CUSTOM event fired elsewhere. See code below. Is there
>> >> any standard way like luastop ?
>> >>
>> >>
>> >> 3) Any way to show how many running lua scripts? luashow ?
>> >>
>> >>
>> >> 4) It seems cannot get the lua script name in a lua script, I made
>> >> a patch to jira by assign it to the argv[0].
>> >>
>> >>
>> >> 5) Seems that only EventConsumer("all") working.
>> >> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to
>> >> work. Any idea to this?
>> >>
>> >>
>> >> Thanks a lot.
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> code example:
>> >>
>> >>
>> >> con = freeswitch.EventConsumer("all");
>> >>
>> >>
>> >> argv[0] = "test.lua"
>> >>
>> >> freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "]
>> >> Starting =====\n");
>> >>
>> >> local all_events = 0
>> >>
>> >> for e in (function() return con:pop(1) end) do
>> >>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
>> >>    all_events = all_events + 1;
>> >> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
>> >>
>> >> event_name = e:getHeader("Event-Name") or ""
>> >> event_subclass = e:getHeader("Event-Subclass") or ""
>> >>
>> >> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
>> >>  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..
>> >> "]---Exiting------\n")
>> >>  break
>> >> end
>> >>
>> >>
>> >> end
>> >>
>> >>
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Thu Aug  6 21:07:00 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Fri, 7 Aug 2009 11:07:00 +0700
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 Release Announcement
In-Reply-To: <87f2f3b90908061030m1afd5adajcc90e31b6a10b5f6@mail.gmail.com>
References: <87f2f3b90908061030m1afd5adajcc90e31b6a10b5f6@mail.gmail.com>
Message-ID: <8ccbff060908062107p1f21fd8dp8c631d7a627520f8@mail.gmail.com>

Good News..

2009/8/7 Michael Collins <msc at freeswitch.org>:
> We are happy to announce the official release of FreeSWITCH 1.0.4! Please
> visit this link to Digg and read the story, and then spread the word!
>
> Thanks for being such a great community!
> -Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From pete at privateconnect.com  Thu Aug  6 21:59:34 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Thu, 06 Aug 2009 21:59:34 -0700
Subject: [Freeswitch-users] Lua on Windows and additional modules
Message-ID: <20090806215934.2ad02225396a31c9de30536f2e338977.122e91f14a.wbe@email04.secureserver.net>

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From velu.technical at gmail.com  Thu Aug  6 22:06:21 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Fri, 7 Aug 2009 10:36:21 +0530
Subject: [Freeswitch-users] Fwd: execute function in ESL.pm module is
	not working
In-Reply-To: <87f2f3b90908052254l3ecc7fa0ybc92d87c587a9b0d@mail.gmail.com>
References: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>
	<1452e2980908052138jed5d676oe5ce73b0b4b0e37e@mail.gmail.com>
	<87f2f3b90908052254l3ecc7fa0ybc92d87c587a9b0d@mail.gmail.com>
Message-ID: <1452e2980908062206s3b33f867if569a10fff078673@mail.gmail.com>

Dear Expert,
   Thanks for you reply....

My Perl Script is,
use strict;
use warnings;

#---------------------------------------------------------------------------
# Event socket library.
# Socket programming
# printing the data structures
# Using posix parametered functions.
#---------------------------------------------------------------------------
use lib('/root/freeswitch-1.0.3/libs/esl/perl/');
require ESL;
use IO::Socket::INET;
use Data::Dumper qw(Dumper);
use POSIX;
use Config::IniFiles;


# Global variables to store the socket connection and eneterd DTM digits.
my ($conn,$digit);
$digit='';

#Registering the ALARM signal.
$SIG{ALRM}=\&sub_alr;

# When alarm signal occurs call the play_digit function
sub sub_alr {
        print "IN Sigalarm---\n";
        &play_digit;
        return ;
}       # ----------  end of subroutine sub_alr  ----------



# Play the voice files for menu.
sub play(){
                $conn->execute("playback","ivr/ivr-please.wav");
                $conn->execute("playback","ivr/ivr-enter_ext.wav");
}


sub play_digit {
        print "In Play Digit....\n";
        my      ( $par1 )       = $digit; #$digit is global variable

        print "Eneterd Digits=",$digit,"\n";
        ################################################################
        # Here what is my problem the execute function is not working  #
        ################################################################
        $conn->execute("phrase", "spell,$par1");
        return ;
}       # ----------  end of subroutine play_digit  ----------


#---------------------------------------------------------------------------
# IP address and port of the server.
# Sound path file.
#---------------------------------------------------------------------------
my $ip = "192.168.1.222"; my $port = '5057';
my $sound_path = "/usr/local/freeswitch/sounds/en/us/callie/";


# Creating a socket
my $sock = new IO::Socket::INET (
        LocalHost => $ip,
        LocalPort => $port,
        Proto     => 'tcp',
        Listen    => 1,
        Reuse     => 1
);
# Checking the error.
die "Cannot create a socket:$!\n" unless $sock;


for(;;){

        my $new_socket = $sock->accept();
        print "Current Process Id:".POSIX::getpid()."\n";
        my $pid = fork();

        if($pid){
                close($new_socket);
                next;
        }

        print "Child Process Id:".POSIX::getpid()."\n";

        my $fd = fileno($new_socket);
        print "File Number:$fd\n";

        # Create a conenction with Event socket library.
        $conn = new ESL::ESLconnection($fd);

        # Getting the connection informations and values of the variables.
        my $info = $conn->getInfo();

        # Getting the caller id and print the statement.
        my $caller_id =$info->getHeader("caller-caller-id-number");
        printf  "Connected from %s\n", $caller_id;

        # Receive the events from only in this switch.
        $conn->sendRecv("myevents");

        # Answer the call.
        $conn->execute("answer");

        # playback the welcome message.
        $conn->setEventLock("true");

$conn->execute("playback",$sound_path."ivr/ivr-welcome_to_freeswitch.wav");
        $conn->execute("sleep", "1000");

        &play;


        alarm(10);
        while($conn->connected()){

                # Receive the event
                my $event = $conn->recvEvent();

                # Check the event is received
                if($event){
                        # Get the event name  and print it.
                        my $name = $event->getHeader("event-name");
                        print "EVENT:[$name]\n";

                        # If the event name is DTMF then print the  enterted
digit.
                        if($name eq 'DTMF'){
                                my $digi = $event->getHeader("dtmf-digit");
                                # Here concatenate the eneterd digits
                                $digit.=$digi;
                        }
                }
        }
        # Kill the child process.
        print "Disconnected:$caller_id\n"; kill 9,POSIX::getpid();
}


My dial plan is,
<!-- Testing IVR -->
    <extension name="Test">
      <condition field="destination_number" expression="^(200)$">
            <action application="socket" data="192.168.1.222:5057 async
full"/>
      </condition>
    </extension>

The output of the Script is,

Current Process Id:2906
Child Process Id:2908
File Number:4
Connected from 1000
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_ANSWER]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[DTMF]
EVENT:[DTMF]
EVENT:[DTMF]
EVENT:[DTMF]
IN Sigalarm---
In Play Digit....
Eneterd Digits=7485
Disconnected:1000

When alarm signal generated, it prints digits but it won't execute the
"execute" function..

Please  any one give suggestions where I made wrong...

Thanks...

Regards,
Velusamy.


On Thu, Aug 6, 2009 at 11:24 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 11:38 PM, velusamy velu <velu.technical at gmail.com>wrote:
>
>> Please any one help for this problem..
>>
>>
> Sorry for the delay but many of the FreeSWITCH experts are at ClueCon right
> now so we'll ask for your patience... in the meantime could you pastebin
> your script and your dialplan entry so that we can take a look at them?
>
> Thanks,
> MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ryder86 at googlemail.com  Fri Aug  7 00:36:16 2009
From: ryder86 at googlemail.com (Artem Vasiliev)
Date: Fri, 7 Aug 2009 11:36:16 +0400
Subject: [Freeswitch-users] Softphone control
Message-ID: <cb6be45b0908070036x5063615ev93e2526427905a23@mail.gmail.com>

Hi

I have FreeSwitch and external application, which communicates to it via
event socket - listens for events for certain number and gives some
commands.
Is it possible for this application to control client softphones, for
example, make them answer or hold, using the event socket or other
FreeSwitch capabilities?
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From dujinfang at gmail.com  Fri Aug  7 01:10:17 2009
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 7 Aug 2009 16:10:17 +0800
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <cb6be45b0908070036x5063615ev93e2526427905a23@mail.gmail.com>
References: <cb6be45b0908070036x5063615ev93e2526427905a23@mail.gmail.com>
Message-ID: <23f91030908070110j199fc1fcpcce4685318a1e8c9@mail.gmail.com>

You can run FreeSWITCH as a softphone and control it.
http://wiki.freeswitch.org/wiki/Freeswitch_softphone

2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>

> Hi
>
> I have FreeSwitch and external application, which communicates to it via
> event socket - listens for events for certain number and gives some
> commands.
> Is it possible for this application to control client softphones, for
> example, make them answer or hold, using the event socket or other
> FreeSwitch capabilities?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ryder86 at googlemail.com  Fri Aug  7 04:02:04 2009
From: ryder86 at googlemail.com (Artem Vasiliev)
Date: Fri, 7 Aug 2009 15:02:04 +0400
Subject: [Freeswitch-users] Softphone control
Message-ID: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>

No, I don't want to make softphone from FreeSwitch

I have FS and several users with eyeBeam softphones. I need to control
those eyeBeams

>You can run FreeSWITCH as a softphone and control it.
>http://wiki.freeswitch.org/wiki/Freeswitch_softphone

>2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>

>> Hi
>>
>> I have FreeSwitch and external application, which communicates to it via
>> event socket - listens for events for certain number and gives some
>> commands.
>> Is it possible for this application to control client softphones, for
>> example, make them answer or hold, using the event socket or other
>> FreeSwitch capabilities?
>>


From merul at mac.com  Fri Aug  7 04:17:12 2009
From: merul at mac.com (Merul Patel)
Date: Fri, 07 Aug 2009 12:17:12 +0100
Subject: [Freeswitch-users] /etc/openzap/tones.conf for UK
Message-ID: <599DBC67-02FE-4F9E-9F87-6D1749B81B11@mac.com>

Where can I find a sample tones.conf file for the UK? Am trying to  
configure a USBFXO device for outbound calls.

Thanks in advance,

Merul


From kevin at johnnyvoip.com  Fri Aug  7 08:00:39 2009
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 7 Aug 2009 11:00:39 -0400
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
References: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
Message-ID: <ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>

>From what I am aware you can't use FreeSWITCH to control a softphone
directly though you can make it do things that will have a similar end
result. You could set eyeBeam to auto-answer calls if you want them to
answer right away or orginiate a call that is auto-answered but not bridge
the call until a user on the eyeBeam presses a digit or a socket control
tells it to connect the two ends. You can also use FreeSWITCH to place the
line on hold using event sockets, this will place it on hold in the server
and not directly like placing it on hold in eyeBeam (i.e. the hold button in
eyeBeam likely wont show it as being on hold).

Beyond that if you want to directly control the clients you would need to
look at getting an API access into the eyeBeam client.

I hope this will help.

Regards,
   Kevin Green


On Fri, Aug 7, 2009 at 7:02 AM, Artem Vasiliev <ryder86 at googlemail.com>wrote:

> No, I don't want to make softphone from FreeSwitch
>
> I have FS and several users with eyeBeam softphones. I need to control
> those eyeBeams
>
> >You can run FreeSWITCH as a softphone and control it.
> >http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>
> >2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>
>
> >> Hi
> >>
> >> I have FreeSwitch and external application, which communicates to it via
> >> event socket - listens for events for certain number and gives some
> >> commands.
> >> Is it possible for this application to control client softphones, for
> >> example, make them answer or hold, using the event socket or other
> >> FreeSwitch capabilities?
> >>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From asannucci at gmail.com  Fri Aug  7 08:21:51 2009
From: asannucci at gmail.com (bakko)
Date: Fri, 7 Aug 2009 17:21:51 +0200
Subject: [Freeswitch-users] Spanish Prompts
Message-ID: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>

I'd like to begin record spanish prompts for FS.

Do you know any software/hardware to make it?

Thank you

BR



From nicolas at medularis.com  Fri Aug  7 09:43:29 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 7 Aug 2009 12:43:29 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
	<4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
	<1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>
Message-ID: <1b46b4e80908070943j6e55c125xde064029513b099c@mail.gmail.com>

I changed the script to set hangup_after_bridge to false, but still the same
thing happens, I get this on the console:

2009-08-07 12:27:44.229091 [NOTICE] sofia.c:322 Hangup
sofia/external/00569xxxxxxx [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2009-08-07 12:27:44.229091 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/00569xxxxxxx [KILL]
2009-08-07 12:27:44.229091 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/00569xxxxxxx [BREAK]
2009-08-07 12:27:44.231471 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-07 12:27:44.231471 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-07 12:27:44.231471 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-07 12:27:44.231471 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-07 12:27:44.231471 [DEBUG] switch_core_state_machine.c:398
(sofia/external/00569xxxxxxx) Running State Change CS_HANGUP
2009-08-07 12:27:44.231471 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL
2009-08-07 12:27:44.231471 [NOTICE] c2c.js:1 *********** Leg2:
NORMAL_CLEARING ***********


The second to last line comes from the script, and prints the hangup_cause
of he session, instead of getting ORIGINATOR_CANCEL, I'm getting
NORMAL_CLEARING. Where is the ORIGINATOR_CANCEL value set?


Thanks!

Nicolas

On Thu, Aug 6, 2009 at 3:45 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi Matt,
>
> Actually I'm explicitly setting hangup_after_bridge to true, think setting
> it to false would help? I'm going to try that.
>
> Here's the JS code:
> (Note: session.getVariable() doesn't work, FS complains saying it is not a
> function, also tried self.session.getVariable() - that's what the wiki says
> - and FS complains that self does not exist)
>
> ----------------
> var uuid = argv[0]; // Call identifier
> var dialstr1 = argv[1]; // Dial string obtained from previous call to LCR
> var dialstr2 = argv[2]; // Dial string obtained from previous call to LCR
> var greeting_snd = "/var/audio/alert.wav";
>
> console_log("notice", "*********** STARTING C2C Call ***********\n");
> timeout = 30;
>
> console_log("notice", "*********** DIALING "+dialstr1+" ***********\n");
>
> //var stUsRing = session.getVariable("us-ring");  // This doesn't work,
> self.session.getVariable doesn't work either
> var stUsRing = "%(2000,4000,440,480)";
>
> // Create new_session
> new_session = new Session(originate_str1);
> console_log("notice", "*********** Leg1: " + new_session.cause + "
> ***********\n");
>
> if (new_session.ready()) {
>         // log to the console
>         console_log("notice", "*********** Leg1 ("+dialstr1+") CONNECTED!
> ***********\n");
>         console_log("notice", "*********** Playing greeting sound:
> "+greeting_snd+" ***********\n");
>
>         new_session.execute("sleep", 100);
>         new_session.execute("playback", greeting_snd);
>
>         // Originate second call and bridge
>     originate_str2 =
> "{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;
>
>         // Create new_session
>         new_session.execute("bridge", originate_str2);
>         console_log("notice", "*********** Leg2: " + new_session.cause + "
> ***********\n");
>
>         if (new_session.ready()) {
>                 console_log("notice", "*********** Leg2 ("+dialstr2+")
> CONNECTED! ***********\n");
>         }
> }
>
> exit();
> ----------------
>
> Thanks!
>
>
> Nicolas
>
>
>
> On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong <mattdfong at gmail.com> wrote:
>
>> Hi Nicolas,
>> do you have a copy of the .js code you can paste. I would guess tho, that
>> ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to
>> false. Just a guess tho.
>>
>> Hangup causes can be found here:
>> http://wiki.freeswitch.org/wiki/Hangup_causes
>>
>> <http://wiki.freeswitch.org/wiki/Hangup_causes> --matt
>> hello hunter - hosted predictive dialer & voice broadcasting
>> http://www.hellohunter.com
>>
>>
>> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>> I'm bridging 2 calls in a javascript file, I originate the first call and
>>> then execute a bridge with an origination string for the second call. If I
>>> hangup the first call while trying to make the second call, I get this on
>>> the console:
>>>
>>> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
>>> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
>>> sofia/external/005622170039 [KILL]
>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/005622170039 [BREAK]
>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
>>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>>> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
>>> Cause: ORIGINATOR_CANCEL
>>>
>>> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
>>> NORMAL_CLEARING. And the variable_originate_disposition has a value of
>>> "failure". Where can I get the detail of the call/bridge failure due to
>>> 'ORIGINATOR_CANCEL' as reported through the console?
>>>
>>> Thanks!
>>>
>>> Nicolas
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From pjintheusa at gmail.com  Fri Aug  7 10:28:23 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 7 Aug 2009 13:28:23 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <1b46b4e80908070943j6e55c125xde064029513b099c@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
	<4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
	<1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>
	<1b46b4e80908070943j6e55c125xde064029513b099c@mail.gmail.com>
Message-ID: <367751820908071028q7075b710hb0d6eed8c1d4dc54@mail.gmail.com>

What does

bridge_hangup_cause

give you?

On Fri, Aug 7, 2009 at 12:43 PM, Nicolas Brenner <nicolas at medularis.com> wrote:
>
> I changed the script to set hangup_after_bridge to false, but still the same thing happens, I get this on the console:
>
> 2009-08-07 12:27:44.229091 [NOTICE] sofia.c:322 Hangup sofia/external/00569xxxxxxx [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> 2009-08-07 12:27:44.229091 [DEBUG] switch_channel.c:1683 Send signal sofia/external/00569xxxxxxx [KILL]
> 2009-08-07 12:27:44.229091 [DEBUG] switch_core_session.c:932 Send signal sofia/external/00569xxxxxxx [BREAK]
> 2009-08-07 12:27:44.231471 [NOTICE] switch_ivr_originate.c:1994 Hangup sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_channel.c:1683 Send signal sofia/external/005622170039 [KILL]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_core_session.c:932 Send signal sofia/external/005622170039 [BREAK]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_ivr_originate.c:2134 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_core_state_machine.c:398 (sofia/external/00569xxxxxxx) Running State Change CS_HANGUP
> 2009-08-07 12:27:44.231471 [INFO] mod_dptools.c:2092 Originate Failed.? Cause: ORIGINATOR_CANCEL
> 2009-08-07 12:27:44.231471 [NOTICE] c2c.js:1 *********** Leg2: NORMAL_CLEARING ***********
>
>
> The second to last line comes from the script, and prints the hangup_cause of he session, instead of getting ORIGINATOR_CANCEL, I'm getting NORMAL_CLEARING. Where is the ORIGINATOR_CANCEL value set?
>
>
> Thanks!
>
> Nicolas
>
> On Thu, Aug 6, 2009 at 3:45 PM, Nicolas Brenner <nicolas at medularis.com> wrote:
>>
>> Hi Matt,
>>
>> Actually I'm explicitly setting hangup_after_bridge to true, think setting it to false would help? I'm going to try that.
>>
>> Here's the JS code:
>> (Note: session.getVariable() doesn't work, FS complains saying it is not a function, also tried self.session.getVariable() - that's what the wiki says - and FS complains that self does not exist)
>>
>> ----------------
>> var uuid = argv[0]; // Call identifier
>> var dialstr1 = argv[1]; // Dial string obtained from previous call to LCR
>> var dialstr2 = argv[2]; // Dial string obtained from previous call to LCR
>> var greeting_snd = "/var/audio/alert.wav";
>>
>> console_log("notice", "*********** STARTING C2C Call ***********\n");
>> timeout = 30;
>>
>> console_log("notice", "*********** DIALING "+dialstr1+" ***********\n");
>>
>> //var stUsRing = session.getVariable("us-ring");? // This doesn't work, self.session.getVariable doesn't work either
>> var stUsRing = "%(2000,4000,440,480)";
>>
>> // Create new_session
>> new_session = new Session(originate_str1);
>> console_log("notice", "*********** Leg1: " + new_session.cause + " ***********\n");
>>
>> if (new_session.ready()) {
>> ??????? // log to the console
>> ??????? console_log("notice", "*********** Leg1 ("+dialstr1+") CONNECTED! ***********\n");
>> ??????? console_log("notice", "*********** Playing greeting sound: "+greeting_snd+" ***********\n");
>>
>> ??????? new_session.execute("sleep", 100);
>> ??????? new_session.execute("playback", greeting_snd);
>>
>> ??????? // Originate second call and bridge
>> ??? originate_str2 = "{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;
>>
>> ??????? // Create new_session
>> ??????? new_session.execute("bridge", originate_str2);
>> ??????? console_log("notice", "*********** Leg2: " + new_session.cause + " ***********\n");
>>
>> ??????? if (new_session.ready()) {
>> ??????????????? console_log("notice", "*********** Leg2 ("+dialstr2+") CONNECTED! ***********\n");
>> ??????? }
>> }
>>
>> exit();
>> ----------------
>>
>> Thanks!
>>
>>
>> Nicolas
>>
>>
>> On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong <mattdfong at gmail.com> wrote:
>>>
>>> Hi Nicolas,
>>> do you have a copy of the .js code you can paste. I would guess tho, that ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to false. Just a guess tho.
>>> Hangup causes can be found here:
>>> http://wiki.freeswitch.org/wiki/Hangup_causes
>>> --matt
>>> hello hunter - hosted predictive dialer & voice broadcasting
>>> http://www.hellohunter.com
>>>
>>> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com> wrote:
>>>>
>>>> I'm bridging 2 calls in a javascript file, I originate the first call and then execute a bridge with an origination string for the second call. If I hangup the first call while trying to make the second call, I get this on the console:
>>>>
>>>> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal sofia/external/005622170039 [KILL]
>>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal sofia/external/005622170039 [BREAK]
>>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>>>> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.? Cause: ORIGINATOR_CANCEL
>>>>
>>>> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see NORMAL_CLEARING. And the variable_originate_disposition has a value of "failure". Where can I get the detail of the call/bridge failure due to 'ORIGINATOR_CANCEL' as reported through the console?
>>>>
>>>> Thanks!
>>>>
>>>> Nicolas
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From raffaele.p.guidi at gmail.com  Fri Aug  7 11:50:57 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Fri, 7 Aug 2009 20:50:57 +0200
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>
References: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
	<ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>
Message-ID: <b0a8792f0908071150x68f3e3b5v596bad90bc56d1ad@mail.gmail.com>

Maybe Artem is interested in CTI (computer telephony integration) -
click2dial, opening a url (or statrting a program) on incoming call...?

On Fri, Aug 7, 2009 at 17:00, Kevin Green <kevin at johnnyvoip.com> wrote:

> From what I am aware you can't use FreeSWITCH to control a softphone
> directly though you can make it do things that will have a similar end
> result. You could set eyeBeam to auto-answer calls if you want them to
> answer right away or orginiate a call that is auto-answered but not bridge
> the call until a user on the eyeBeam presses a digit or a socket control
> tells it to connect the two ends. You can also use FreeSWITCH to place the
> line on hold using event sockets, this will place it on hold in the server
> and not directly like placing it on hold in eyeBeam (i.e. the hold button in
> eyeBeam likely wont show it as being on hold).
>
> Beyond that if you want to directly control the clients you would need to
> look at getting an API access into the eyeBeam client.
>
> I hope this will help.
>
> Regards,
>    Kevin Green
>
>
>
> On Fri, Aug 7, 2009 at 7:02 AM, Artem Vasiliev <ryder86 at googlemail.com>wrote:
>
>> No, I don't want to make softphone from FreeSwitch
>>
>> I have FS and several users with eyeBeam softphones. I need to control
>> those eyeBeams
>>
>> >You can run FreeSWITCH as a softphone and control it.
>> >http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>>
>> >2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>
>>
>> >> Hi
>> >>
>> >> I have FreeSwitch and external application, which communicates to it
>> via
>> >> event socket - listens for events for certain number and gives some
>> >> commands.
>> >> Is it possible for this application to control client softphones, for
>> >> example, make them answer or hold, using the event socket or other
>> >> FreeSwitch capabilities?
>> >>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lfurrea at gmail.com  Fri Aug  7 11:57:55 2009
From: lfurrea at gmail.com (Luis F Urrea)
Date: Fri, 7 Aug 2009 12:57:55 -0600
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
Message-ID: <f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>

On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
edition .

Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for FS
playback.

Here's a guide that has been put together for reference on what to record.

http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml

Regards,



On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:

> I'd like to begin record spanish prompts for FS.
>
> Do you know any software/hardware to make it?
>
> Thank you
>
> BR
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Fri Aug  7 12:10:35 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 7 Aug 2009 15:10:35 -0400
Subject: [Freeswitch-users] Calling multiple destinations with fail over
Message-ID: <367751820908071210y233741fatad5cd451dd64df7@mail.gmail.com>

Hi there,

I am trying to implement a scenario where I can terminate calls to
multiple destinations AND have termination carrier fail over.
Currently I can see how to do one or the other. But not both.

Multiple destinations is easy:

 <action application="bridge"
data="[origination_caller_id_number=1234]sofia/SIP_PROVIDER_1/6095551234,[origination_caller_id_number=55555]sofia/SIP_PROVIDER_1/7325551234"/>

Failover appears to use the same mechanism:

<action application="bridge"
data="sofia/gateway/primary/blah|sofia/gateway/secondary/blah"/>

I can not get my head around how use these together, such that:


- 6095551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 6095551234  through SIP_PROVIDER_2
- Called party does not answer
7325551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 7325551234 through SIP_PROVIDER_2
- Called party answers


This must be a fairly comment requirement so any ideas on what I might
be missing would be very welcome.

Thanks


Phillip Jones


From jmesquita at gmail.com  Fri Aug  7 13:50:32 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 7 Aug 2009 17:50:32 -0300
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <b0a8792f0908071150x68f3e3b5v596bad90bc56d1ad@mail.gmail.com>
References: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
	<ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>
	<b0a8792f0908071150x68f3e3b5v596bad90bc56d1ad@mail.gmail.com>
Message-ID: <5a8712120908071350v2eac4613j7fd53e5158680742@mail.gmail.com>

Stay tuned on fsgui. It will get there really soon.

jmesquita

On Fri, Aug 7, 2009 at 3:50 PM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> Maybe Artem is interested in CTI (computer telephony integration) -
> click2dial, opening a url (or statrting a program) on incoming call...?
>
>
> On Fri, Aug 7, 2009 at 17:00, Kevin Green <kevin at johnnyvoip.com> wrote:
>
>> From what I am aware you can't use FreeSWITCH to control a softphone
>> directly though you can make it do things that will have a similar end
>> result. You could set eyeBeam to auto-answer calls if you want them to
>> answer right away or orginiate a call that is auto-answered but not bridge
>> the call until a user on the eyeBeam presses a digit or a socket control
>> tells it to connect the two ends. You can also use FreeSWITCH to place the
>> line on hold using event sockets, this will place it on hold in the server
>> and not directly like placing it on hold in eyeBeam (i.e. the hold button in
>> eyeBeam likely wont show it as being on hold).
>>
>> Beyond that if you want to directly control the clients you would need to
>> look at getting an API access into the eyeBeam client.
>>
>> I hope this will help.
>>
>> Regards,
>>     Kevin Green
>>
>>
>>
>> On Fri, Aug 7, 2009 at 7:02 AM, Artem Vasiliev <ryder86 at googlemail.com>wrote:
>>
>>> No, I don't want to make softphone from FreeSwitch
>>>
>>> I have FS and several users with eyeBeam softphones. I need to control
>>> those eyeBeams
>>>
>>> >You can run FreeSWITCH as a softphone and control it.
>>> >http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>>>
>>> >2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>
>>>
>>> >> Hi
>>> >>
>>> >> I have FreeSwitch and external application, which communicates to it
>>> via
>>> >> event socket - listens for events for certain number and gives some
>>> >> commands.
>>> >> Is it possible for this application to control client softphones, for
>>> >> example, make them answer or hold, using the event socket or other
>>> >> FreeSwitch capabilities?
>>> >>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nicolas at medularis.com  Fri Aug  7 14:20:29 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 7 Aug 2009 17:20:29 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <367751820908071028q7075b710hb0d6eed8c1d4dc54@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
	<4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
	<1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>
	<1b46b4e80908070943j6e55c125xde064029513b099c@mail.gmail.com>
	<367751820908071028q7075b710hb0d6eed8c1d4dc54@mail.gmail.com>
Message-ID: <1b46b4e80908071420v42c3f73wfeddacb8af3d7067@mail.gmail.com>

That variable is not available, it is not included with the
CHANNEL_HANGUP_COMPLETE event info. However I discovered that when the
bridge does not work, there are two CHANNEL_HANGUP_COMPLETE events, one for
each leg, nevertheless for some reason the daemon I have watching the events
misses the second leg event, so I was only seeing the result of the first
leg hangup, which is NORMAL_CLEARING, and the second event's hangup_cause is
ORIGINATOR_CANCEL. I don't know why my daemon is missing the event though.
I'll have to dig into this further.


On Fri, Aug 7, 2009 at 1:28 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> What does
>
> bridge_hangup_cause
>
> give you?
>
> On Fri, Aug 7, 2009 at 12:43 PM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
> >
> > I changed the script to set hangup_after_bridge to false, but still the
> same thing happens, I get this on the console:
> >
> > 2009-08-07 12:27:44.229091 [NOTICE] sofia.c:322 Hangup
> sofia/external/00569xxxxxxx [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> > 2009-08-07 12:27:44.229091 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/00569xxxxxxx [KILL]
> > 2009-08-07 12:27:44.229091 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/00569xxxxxxx [BREAK]
> > 2009-08-07 12:27:44.231471 [NOTICE] switch_ivr_originate.c:1994 Hangup
> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> > 2009-08-07 12:27:44.231471 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/005622170039 [KILL]
> > 2009-08-07 12:27:44.231471 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/005622170039 [BREAK]
> > 2009-08-07 12:27:44.231471 [DEBUG] switch_ivr_originate.c:2134 Originate
> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> > 2009-08-07 12:27:44.231471 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/00569xxxxxxx) Running State Change CS_HANGUP
> > 2009-08-07 12:27:44.231471 [INFO] mod_dptools.c:2092 Originate Failed.
> Cause: ORIGINATOR_CANCEL
> > 2009-08-07 12:27:44.231471 [NOTICE] c2c.js:1 *********** Leg2:
> NORMAL_CLEARING ***********
> >
> >
> > The second to last line comes from the script, and prints the
> hangup_cause of he session, instead of getting ORIGINATOR_CANCEL, I'm
> getting NORMAL_CLEARING. Where is the ORIGINATOR_CANCEL value set?
> >
> >
> > Thanks!
> >
> > Nicolas
> >
> > On Thu, Aug 6, 2009 at 3:45 PM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
> >>
> >> Hi Matt,
> >>
> >> Actually I'm explicitly setting hangup_after_bridge to true, think
> setting it to false would help? I'm going to try that.
> >>
> >> Here's the JS code:
> >> (Note: session.getVariable() doesn't work, FS complains saying it is not
> a function, also tried self.session.getVariable() - that's what the wiki
> says - and FS complains that self does not exist)
> >>
> >> ----------------
> >> var uuid = argv[0]; // Call identifier
> >> var dialstr1 = argv[1]; // Dial string obtained from previous call to
> LCR
> >> var dialstr2 = argv[2]; // Dial string obtained from previous call to
> LCR
> >> var greeting_snd = "/var/audio/alert.wav";
> >>
> >> console_log("notice", "*********** STARTING C2C Call ***********\n");
> >> timeout = 30;
> >>
> >> console_log("notice", "*********** DIALING "+dialstr1+" ***********\n");
> >>
> >> //var stUsRing = session.getVariable("us-ring");  // This doesn't work,
> self.session.getVariable doesn't work either
> >> var stUsRing = "%(2000,4000,440,480)";
> >>
> >> // Create new_session
> >> new_session = new Session(originate_str1);
> >> console_log("notice", "*********** Leg1: " + new_session.cause + "
> ***********\n");
> >>
> >> if (new_session.ready()) {
> >>         // log to the console
> >>         console_log("notice", "*********** Leg1 ("+dialstr1+")
> CONNECTED! ***********\n");
> >>         console_log("notice", "*********** Playing greeting sound:
> "+greeting_snd+" ***********\n");
> >>
> >>         new_session.execute("sleep", 100);
> >>         new_session.execute("playback", greeting_snd);
> >>
> >>         // Originate second call and bridge
> >>     originate_str2 =
> "{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;
> >>
> >>         // Create new_session
> >>         new_session.execute("bridge", originate_str2);
> >>         console_log("notice", "*********** Leg2: " + new_session.cause +
> " ***********\n");
> >>
> >>         if (new_session.ready()) {
> >>                 console_log("notice", "*********** Leg2 ("+dialstr2+")
> CONNECTED! ***********\n");
> >>         }
> >> }
> >>
> >> exit();
> >> ----------------
> >>
> >> Thanks!
> >>
> >>
> >> Nicolas
> >>
> >>
> >> On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong <mattdfong at gmail.com>
> wrote:
> >>>
> >>> Hi Nicolas,
> >>> do you have a copy of the .js code you can paste. I would guess tho,
> that ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge
> to false. Just a guess tho.
> >>> Hangup causes can be found here:
> >>> http://wiki.freeswitch.org/wiki/Hangup_causes
> >>> --matt
> >>> hello hunter - hosted predictive dialer & voice broadcasting
> >>> http://www.hellohunter.com
> >>>
> >>> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
> >>>>
> >>>> I'm bridging 2 calls in a javascript file, I originate the first call
> and then execute a bridge with an origination string for the second call. If
> I hangup the first call while trying to make the second call, I get this on
> the console:
> >>>>
> >>>> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> >>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/005622170039 [KILL]
> >>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/005622170039 [BREAK]
> >>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134
> Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> >>>> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
> Cause: ORIGINATOR_CANCEL
> >>>>
> >>>> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I
> see NORMAL_CLEARING. And the variable_originate_disposition has a value of
> "failure". Where can I get the detail of the call/bridge failure due to
> 'ORIGINATOR_CANCEL' as reported through the console?
> >>>>
> >>>> Thanks!
> >>>>
> >>>> Nicolas
> >>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nicolas at medularis.com  Fri Aug  7 15:10:25 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 7 Aug 2009 18:10:25 -0400
Subject: [Freeswitch-users] Error trying to use PHP ESL
Message-ID: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>

Hi,

I'm trying to get started with the ESL using PHP. I compiled the ESL, then
phpmod according to the wiki instructions, but then when I try the examples
in the libs/esl/php dir, they fail saying:

PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
/usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132

Checking ESL.php on line 132, I see there are several different declarations
for the function __construct() with different parameters each, which makes
sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
higher than that or something? What could be the problem?

Thanks!

Nicolas
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From dave at 3c.co.uk  Fri Aug  7 15:54:08 2009
From: dave at 3c.co.uk (David Knell)
Date: Fri, 07 Aug 2009 17:54:08 -0500
Subject: [Freeswitch-users] Cluecon 2009
Message-ID: <1249685648.16901.34.camel@dk-d820>

Just a quick note to say thanks to Cluecon's organisers for putting
together such a useful, informative and packed three days.  I've come
away with a head full of ideas, a bunch of new contacts and a collection
of things to do; I'd thoroughly recommend that anyone interested in IP
telephony blocks out the first week of August 2010, right now..!

Cheers --

Dave

-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From gcd at i.ph  Fri Aug  7 16:44:23 2009
From: gcd at i.ph (Nandy Dagondon)
Date: Sat, 8 Aug 2009 07:44:23 +0800
Subject: [Freeswitch-users] /etc/openzap/tones.conf for UK
In-Reply-To: <599DBC67-02FE-4F9E-9F87-6D1749B81B11@mac.com>
References: <599DBC67-02FE-4F9E-9F87-6D1749B81B11@mac.com>
Message-ID: <7d0bfd8c0908071644n2933a91y996745e9f894cf90@mail.gmail.com>

you can create your tones.conf using call progress tones  found at
http://www.3amsystems.com/wireline/tone-search.htm


On Fri, Aug 7, 2009 at 7:17 PM, Merul Patel <merul at mac.com> wrote:

> Where can I find a sample tones.conf file for the UK? Am trying to
> configure a USBFXO device for outbound calls.
>
> Thanks in advance,
>
> Merul
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Fri Aug  7 16:44:41 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 7 Aug 2009 19:44:41 -0400
Subject: [Freeswitch-users] State of originated call
Message-ID: <c55d46950908071644m7604eb57jf5018c2b378d3255@mail.gmail.com>

Hi,

using javascript, i do originate the call this way:

Session s= new Session(originateStr);

>From this point, is it possible to know what states the call is going
through? In a previous message it was suggested that
variable_originate_disposition would give me the response code. Now, how to
i use this in practice in a script?
How do i for instance retrieve a 180 response code when rining is hapening
on the remote end?

Thanks,
Max.
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From nicolas at medularis.com  Fri Aug  7 17:23:51 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 7 Aug 2009 20:23:51 -0400
Subject: [Freeswitch-users] Best practices / tips for Event socket daemon
Message-ID: <1b46b4e80908071723m18bf3bcnb2505f0f760b907c@mail.gmail.com>

Hi, I built an event socket daemon that waits for certain events, when it
receives those events, it does some processing and keeps waiting for more
events. The daemon is written on PHP and uses a slightly modified version of
fs_sock.php (from contrib/intralanman/PHP/fs_sock/).

What I am doing / what I want to do: I am generating calls and bridging them
using a JS script. Then the daemon logs the info about the calls and keeps
track of their status in a database.

The problem is: the daemon is apparently missing out on some events, and I
think it is because of the processing/updating on the DB it has to do each
time it "catches" an event on the socket.

My question is: which language would you recommend for the task, and how
would you go about handling events? Should the dameon fork a process for
each event it receives so that it doesn't miss any events? should there be
more than one daemon? ... any tips and recommendations are welcome.

Thanks!

Nicolas
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From rehan at supertec.com  Fri Aug  7 19:21:36 2009
From: rehan at supertec.com (Rehan Ahmed Allahwala)
Date: Sat, 8 Aug 2009 10:21:36 +0800
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <1249685648.16901.34.camel@dk-d820>
References: <1249685648.16901.34.camel@dk-d820>
Message-ID: <865f01c80908071921q13b4f98bxea7426b720a69e83@mail.gmail.com>

Hi all, hope u got ur hands on our mobile pouches from didx.net and
got ur snaps with suzanne for our blogs and new upcoming magazine

Thanks ,

Rehan

On 8/8/09, David Knell <dave at 3c.co.uk> wrote:
> Just a quick note to say thanks to Cluecon's organisers for putting
> together such a useful, informative and packed three days.  I've come
> away with a head full of ideas, a bunch of new contacts and a collection
> of things to do; I'd thoroughly recommend that anyone interested in IP
> telephony blocks out the first week of August 2010, right now..!
>
> Cheers --
>
> Dave
>
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From vladrodionov at gmail.com  Fri Aug  7 19:42:38 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Fri, 7 Aug 2009 19:42:38 -0700
Subject: [Freeswitch-users] Best practices / tips for Event socket daemon
In-Reply-To: <1b46b4e80908071723m18bf3bcnb2505f0f760b907c@mail.gmail.com>
References: <1b46b4e80908071723m18bf3bcnb2505f0f760b907c@mail.gmail.com>
Message-ID: <3c233920908071942g522df080g63b2abc31efd6fcc@mail.gmail.com>

Forking process on every incoming event is terrible idea IMO. Threads are
more lightweight than processes.
Can you use threads in PHP? I am not familiar with PHP (Java developer
myself).  I can explain how I would implement it in Java.

There is one SocketReader thread and several Worker threads in a thread
pool.

"SocketReader" thread - reads data (events) from socket. When event arrives
SocketReader checks thread pool, get one Worker (if any) and makes it to
process event. If there no available Workers in a pool then event goes
directly to a EventQueue. When Worker finishes it checks EventQueue and if
there are no events in a queue Worker goes back to thread pool, otherwise it
process event from queue.

-Vladimir Rodionov



On Fri, Aug 7, 2009 at 5:23 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi, I built an event socket daemon that waits for certain events, when it
> receives those events, it does some processing and keeps waiting for more
> events. The daemon is written on PHP and uses a slightly modified version of
> fs_sock.php (from contrib/intralanman/PHP/fs_sock/).
>
> What I am doing / what I want to do: I am generating calls and bridging
> them using a JS script. Then the daemon logs the info about the calls and
> keeps track of their status in a database.
>
> The problem is: the daemon is apparently missing out on some events, and I
> think it is because of the processing/updating on the DB it has to do each
> time it "catches" an event on the socket.
>
> My question is: which language would you recommend for the task, and how
> would you go about handling events? Should the dameon fork a process for
> each event it receives so that it doesn't miss any events? should there be
> more than one daemon? ... any tips and recommendations are welcome.
>
> Thanks!
>
> Nicolas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From andrew at hijacked.us  Fri Aug  7 19:56:14 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Fri, 7 Aug 2009 22:56:14 -0400
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
Message-ID: <20090808025613.GA19871@hijacked.us>

On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
> Hi,
> 
> I'm trying to get started with the ESL using PHP. I compiled the ESL, then
> phpmod according to the wiki instructions, but then when I try the examples
> in the libs/esl/php dir, they fail saying:
> 
> PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
> /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132
> 
> Checking ESL.php on line 132, I see there are several different declarations
> for the function __construct() with different parameters each, which makes
> sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
> higher than that or something? What could be the problem?
>

Someone in the IRC channel mentioned this too. I looked at it briefly
and it looks like the latest 'swigall' screwed it up. The original
reporter said he'd file a jira, but you may want to check yourself and
if not make one yourself. In the meantime, the previous version of the
file was reported to work if you really need it.

Andrew


From neffs1 at gmail.com  Fri Aug  7 07:02:13 2009
From: neffs1 at gmail.com (David Kreitschmann)
Date: Fri, 7 Aug 2009 16:02:13 +0200
Subject: [Freeswitch-users] Compact, fanless appliance?
In-Reply-To: <7d0bfd8c0907051621u1c3f553fh96a9df8557952e47@mail.gmail.com>
References: <7d0bfd8c0907041941s75e541dex982fae2195858ea4@mail.gmail.com>
	<C6757EE0.5891C%krice@suspicious.org>
	<7d0bfd8c0907051621u1c3f553fh96a9df8557952e47@mail.gmail.com>
Message-ID: <149B556F-65B6-40F7-802B-92B81447C24F@gmail.com>

I am thinking about using this combination as a router, should also be  
a nice platform for freeswitch

http://www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm
http://www.cartft.com/catalog/il/1058
http://www.cartft.com/catalog/il/1087

energy efficient netbook chipset (most other boards use the desktop  
version), PCI slot, small case, fanless.
you can put the system on an usb stick and just hide it behind the  
front panel. no need for reinstall on the device itself, just put in a  
another usb stick and you're good to go.

or you put in two disks in raid1 for redundancy if you need the disk  
space.

if you don't need PCI you can use this enclosure
http://www.cartft.com/catalog/il/1081


David

Am 06.07.2009 um 01:21 schrieb Nandy Dagondon:

> ok. w/ my apologies. - nandy
>
>
> On Sun, Jul 5, 2009 at 10:49 AM, Ken Rice <krice at suspicious.org>  
> wrote:
> No need to bump these things as this is a mailing list and it annoys  
> quite a few people when you do that
>
>
> From: Nandy Dagondon <gcd at i.ph>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 5 Jul 2009 10:41:18 +0800
>
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Compact, fanless appliance?
>
> just bumping this topic.
> -nandy
>
> On Fri, May 8, 2009 at 12:44 AM, Fred-145 <codecomplete at free.fr>  
> wrote:
>
>
> Antonio Gallo wrote:
> > Alix cases are like 6/9 ? from their shop site. I think its easy  
> to find
> > someone who work with aluminium that can make for you custom boxes  
> for
> > like like 6/20 ? at pcs
>
> Unfortunately, none of the PCEngines cases (www.pcengines.ch/order1.php?c=2 
>  <http://www.pcengines.ch/order1.php?c=2> )
>
> allow for a PCI slot, either on top of the mobo, or away from it :-/
>
> I'll see if I can get those from Soekris (http://soekris.eu/shop/cases_en/ 
> )
> allow this, and if I can get a good price for a case + PSU.
>
> Thank you.
> --
> View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23430873.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From fraunhofer.lists.freeswitch-001 at traced.net  Thu Aug  6 13:28:19 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Thu, 6 Aug 2009 22:28:19 +0200
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
Message-ID: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>

Hello List, Hello *,

First of all the usual excuses: sorry for the bad english and the long
email, no native speaker and i really tried to make it shorter, but i
guess this would result in even more "check back"s than it already
does :)

we're currently running in a weird "lockup"-scenario in our loadtests.

Our setup is the following:

three freeswitch servers, let's call them A(-leg), M(aster), and
B(-leg) with the goal in mind to initiate calls on M which calls A,
play some file, bridge to B, limit call length and play (different)
prompts to A and B if they exceed that limit.

(Note that A and B work fine, regardless of the amount of load we put on them)
A and B are silly dialplan logic, accepting calls on a certain
extension after a random delay and playing moh. Before calling
playback to a localstream they call a lua script which schedules
hangup somewhere in future (which works flawlessly)

Calls are initiated on M using some hacked up loadgen-script issuing
http requests like
? originate [sofiaSyntaxToExtensionOn_A] 6000
. The 6000 extension on M has the following (xml) dialplan which
essentially does the following:
------
answer()
...playback file...
...set some callerid stuff
set bypass_media
bridge to extension 6009 on B
------
we use "execute_on_answer" on the b-leg to run a script which limits
the length of the call (doesn't matter if it's done via "export
nolocal" or "inlined" into the data part of the bridge application
"{execute_on_answer=lua ...}")

<action application="export" data="nolocal:execute_on_answer=lua
lua/schedule-hangup.lua ${uuid}" />

the lua script "schedula-hangup.lua" does essentially the following:

------
api = freeswitch.API();
local res = api:execute("sched_api", "+10 none lua
lua/c2c-hangup-timeout.lua " .. argv[1]);
------

the 10 seconds are just to speed up the time until it gets stuck.

this is where things start to go wrong. if I comment out the call to
the "schedule-hangup" script, everything works fine, even if it's
under heavy load.

c2c-hangup-timeout.lua does the following:
------------------
local sess = argv[1];
if(sess)
then
? ?freeswitch.consoleLog("INFO", "c2c-hangup-timeout.lua for uuid "
.. sess .. "\n");

? ?api = freeswitch.API();
? ?local stillValid = api:execute("uuid_getvar", sess .. "
Dummy-DoesChannelExists");
? ?if(stillValid:sub(1,4) == "-ERR")
? ?then
? ? ? ?log("session uuid " .. sess .. " disappeared (nothing bad)");
? ?else
? ? ? ?-- this is important!!! Otherwise the aleg get's just hung up!
? ? ? ?api:execute("uuid_media", sess);
? ? ? ?api:execute("uuid_transfer", sess .. " -both timeout");
? ?end
else -- /if(sess)
? ?log("called with nil session?");
end -- /if(sess)

------------------

i guess this needs some explanation:
we get the uuid of the channel as argument in argv[1]. We don't use
? local session = freeswitch.Session(uuid);
since if the channel referenced by "uuid" does not exist any longer,
freeswitch (or the lua bindings) try to interpret the uuid as an
"originate string" and can't figure out how to call that. So we use a
dummy api call to get some channel variable. If the channel does not
exist any longer (A or B already hung up), we get an error message
starting with "-ERR", otherwise the channel still exists (we get
"_unset_" as the value, if it's not set) and we continue by getting
freeswitch back in the media path (uuid_media) and then transferring
both legs to an extension called "timeout" which plays some prompt and
finally calls hangup().

If we don't do the uuid_media call, one of the legs gets hung up when
we transfer them to the extension. This looks like the following on
the console after issuing "uuid_transfer [uuid] -both timeout"
(extensions are not the same as in our loadgen example above)


--------------
2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1334 Hangup (*)
sofia/internal/1000 [CS_HIBERNATE] [BLIND_TRANSFER]
2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer
sofia/internal/1004 at 192.168.179.177:5060 to XML[timeout at default]
2009-07-23 19:57:19.865703 [INFO] mod_dialplan_xml.c:310 Processing
BFR1004->timeout in context default
API CALL [uuid_transfer(73812082-77b1-11de-b9f8-a10bb0eb9f69 -both
timeout)] output:
+OK

2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer (**)
sofia/internal/1000 to XML[timeout at default]
2009-07-23 19:57:19.865703 [NOTICE] switch_core_session.c:1084 Session
60 (sofia/internal/1000) Ended
2009-07-23 19:57:19.865703 [NOTICE] switch_core_session.c:1086 Close
Channel sofia/internal/1000 [CS_DESTROY]
-----------

note that it first does Hangup (denoted by *, no that's not an
asterisk :) on extension 1000 and then tries to Transfer (**) the hung
up channel to the dial plan. this could be the same as in an earlier
post to the list "SIP re-invite / bypass_media // Phillip Jones //
Wed, 01 Jul 2009 13:30:53 -0700)"

This is why we do not directly call sched_transfer() but call a script
in between to do the uuid_media() call. I couldn't figure out how to
call that directly from the xml dialplan and/or how to check if the
channel still exists.

so... after using uuid_media(), both legs are transferred without an
(intermediate|bogus) hangup() call.

This only works fine if we've few concurrent calls. There is no magic
borderline where it starts to refuse work.

Some of the Symptoms are: traffic decreased to zero as no new channels
are successfully brought up, some of the signaling traffic is not
ACKed or OKed, scheduled jobs are not run.

if i read the output of "show channels" correctly, they're all stuck
in different applications like hangup(), some are calling lua but most
of them are in signaling_bridge(). Freeswitch is still responding on
the console and there's almost no load on the machine (no busy polling
or some other kind of running amok).

if i kill one of them using uuid_kill() or kill all of them
using"fsctl hupall" i get "Task was executed late by 866 seconds 12379
sched_api_function (none)" messages and the usual cleanup takes place.
As a quick hack i tried to schedule a uuid_kill() call 20 seconds
after the scheduling call to the lua script but that job is not
executed either.

So what am I doing wrong? Is it some deadlock where uuid_media() and
uuid_transfer() ?are waiting for the other to finish?
Or some other silly simple thing i missed?

Thx in advance

?Benedikt.


From email.list.subscriber at gmail.com  Thu Aug  6 15:03:56 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Thu, 6 Aug 2009 18:03:56 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
Message-ID: <4a7b530a.29578c0a.53a8.0450@mx.google.com>

Hello,

Does anyone have, or know where to get, a pre-compiled copy of
FreeSwitch for Solaris 10/x86?

# uname -a
SunOS hrndvsoi-zm01 5.10 Generic_137112-07 i86pc i386 i86pc


I'm stuck trying to run make/gmake/\/opt/gnu/bin/make:

<snip>
"make" results in:
make: Fatal error: Command failed for target `all-recursive'
Current working directory /home/vmorales/freeswitch
*** Error code 1
make: Fatal error: Command failed for target `all'
</snip>


"gmake" & "/opt/gnu/bin/make" result in:

<snip>
creating libfreeswitch.la
(cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la
libfreeswitch.la)
gcc -I/home/vmorales/freeswitch/src/include
-I/home/vmorales/freeswitch/libs/libteletone/src -fPIC -Werror -g
-ggdb -DPATH_MAX=2048 -g -m32 -I/usr/sfw/include -Wall -std=c99
-pedantic -m32 -o .libs/freeswitch freeswitch-switch.o  -L/usr/sfw/lib
-lm ./.libs/libfreeswitch.so
/home/vmorales/freeswitch/libs/apr/.libs/libapr-1.a
-L/home/vmorales/freeswitch/libs/srtp /usr/sfw/lib/libstdc++.so
libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt -ldl -lnsl -lpthread
libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lcurses -lsocket
-R/home/vmorales/freeswitch-build/lib -R/usr/sfw/lib
Undefined                       first referenced
 symbol                             in file
XML_Parse                           ./.libs/libfreeswitch.so
XML_ParserCreate                    ./.libs/libfreeswitch.so
XML_ErrorString                     ./.libs/libfreeswitch.so
herror                              ./.libs/libfreeswitch.so
XML_SetUserData                     ./.libs/libfreeswitch.so
XML_ParserFree                      ./.libs/libfreeswitch.so
XML_GetErrorCode                    ./.libs/libfreeswitch.so
XML_SetCharacterDataHandler         ./.libs/libfreeswitch.so
XML_SetElementHandler               ./.libs/libfreeswitch.so
ld: fatal: Symbol referencing errors. No output written to
.libs/freeswitch
collect2: ld returned 1 exit status
gmake[2]: *** [freeswitch] Error 1
gmake[1]: *** [all-recursive] Error 1
gmake: *** [all] Error 2
</snip>


Thanks in advance for any information provided.

Vladimir



From alan at chandlerfamily.org.uk  Fri Aug  7 01:18:35 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Fri, 07 Aug 2009 09:18:35 +0100
Subject: [Freeswitch-users] New to Freeswitch - some help needed
Message-ID: <4A7BE35B.8010709@chandlerfamily.org.uk>

I apologize, as my first post to this list, that I ask a detailed set of 
questions, but I have spend some time looking at all the docs and can't 
get what I need to do completely sorted in my head.  I am definitely one 
who likes to UNDERSTAND what is happening rather than follow blank 
recipies, so please bear with me as I try understand all the details. I 
do understand about networking, NAT etc - but I am new to SIP/RTP and in 
particular what I think is a double NAT problem


Firstly - what am I trying to achieve:

I am in the UK and have a small home network behind a D-Link DIR-100 
Router/NAT/Firewall one of those machines, running Debian Lenny, acts as 
my main server for everything (and in an earlier incarnation was the 
firewall/router/nat box too - I only say this is because I had all this 
working using Asterisk a year or so ago, but with this important 
difference in configuration).  Many of the ports on the firewall are 
port forwarded to this machine. I have set Freeswitch up on this server 
to act as a small voip pbx for the home - but MORE IMPORTANTLY - to 
enable my daughter from her house to talk to us.  At my house locally I 
have a Linksys PAP2T two phone SIP box - and that is working with 
Freeswitch's default configuration (I set up to be 1000 and 1001 and 
used all the facilities).  I will later add a Linksys SPA 3102 - 
although I DO NOT intend to use its facility to bridge to the normal 
phone network.

My daughter, living in another house, also has a Nat box (unknown - its 
part of her ADSL modem/router/wireless access point) and also has a 
PAP2T which she will connect to the her network.  This will be her phone.

There is a family relation living in Australia who will load up a 
whatever softphone that we tell him to use.  I expect, but don't know, 
that he will behind a NAT box too.

Later, I have some friends in the USA that I might wish to add it too - 
especially so that we can hold some teleconferences.  They will have a 
mixture of Windows and MACs, and I will need to recommend softphone 
clients for them.

I want to set this up as a small private voice network, so anyone can 
ring anyone else.  I will add fancy facilities such as conferencing and 
voicemail later - I just want to get the basics working first.

Secondly

I installed a stun client on my home machine and ran it against 
stun.freeswitch.org.

It reported:-

Primary: Independent Mapping, Independent Filter, preserves ports, no 
hairpin

But I have no idea what this means - I can't find any clear statement 
via googling for it - how this set of answers maps to the different 
types of NAT that might be required to get this to all work.  CAN 
SOMEONE ENLIGHTEN me please.

Thirdly

I have set up a sip profile called "double nat" from the recipe in the 
wiki.  This defines the SIP port to be 5090.

However, what I DO NOT UNDERSTAND is how the PAP2T box in my daughters 
house will initiate a connection to my server.  Presumably, I have to 
port forward 5090 from the nat box to my server.  IS THAT CORRECT?

I also assume I will have to tell her to use STUN (I believe this is an 
option on the PAP2T)

Fourthly

If I understand SIP correctly, it just initiates the session and the two 
end points then communicate directly via RTP.  What I don't understand 
is how does a session transition from SIP to RTP via the connection set 
up in the the first phase (in terms of passing through the NAT boxes). 
In particular WHICH OF THE TEST RESULTS from my stun client indicate it 
will do the right thing.  (I am going to take a laptop to my daughters 
house with a stun client in to test her network this weekend).

Could someone explain please.

  Fifthly

Is there a recommended SIP softphone with all the right facilities (STUN 
support?)  that works on MAC and WINDOWS (I only use linux myself).

Apologies for the length of this.  I am eager to get the answers so I 
can use an opportunity this weekend to get it working.


-- 
Alan Chandler
http://www.chandlerfamily.org.uk



From alan at chandlerfamily.org.uk  Fri Aug  7 05:19:51 2009
From: alan at chandlerfamily.org.uk (alan at chandlerfamily.org.uk)
Date: Fri, 7 Aug 2009 13:19:51 +0100 (BST)
Subject: [Freeswitch-users] Auto Nat
Message-ID: <9f0cf614f2b2ffb9f7a2edae3bf3d2ff.squirrel@webmail.chandlerfamily.org.uk>

I sent my first e-mail to the list this morning (about 4 hours ago) but it
does not seem to have arrived back, even though I have received other,
later posts.

I have another question related to the first (about how to set everything
up in a double nat environment) - so if I see this and not the other, I
will send the first again.

I am currently running the stable version 1.0.3 of freeswitch.  The wiki
page says that auto-nat is introduced at r 13612.  Is this before or after
that revision? (I don't want to have to download and rebuild the entire
thing if I don't have to).






From alan at chandlerfamily.org.uk  Fri Aug  7 06:17:03 2009
From: alan at chandlerfamily.org.uk (alan at chandlerfamily.org.uk)
Date: Fri, 7 Aug 2009 14:17:03 +0100 (BST)
Subject: [Freeswitch-users] 1.0.4 builds 1.0.3_1 debs
Message-ID: <631613604d8c9862f1cf43f80cb99f94.squirrel@webmail.chandlerfamily.org.uk>

I just downloaded 1.0.4 and build debian packages with it, and it
delivered .deb files names as 1.0.3-1




From demuel at thephinix.org  Fri Aug  7 21:04:44 2009
From: demuel at thephinix.org (demuel at thephinix.org)
Date: Sat, 8 Aug 2009 05:04:44 +0100 (BST)
Subject: [Freeswitch-users] New to Freeswitch - some help needed
In-Reply-To: <4A7BE35B.8010709@chandlerfamily.org.uk>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
Message-ID: <1bfed71a616412ee86da3a7f18b990a0.squirrel@www.thephinix.org>

What a long detailed list of todos. You certainly can't find that kind of
answers in here.

> I apologize, as my first post to this list, that I ask a detailed set of
> questions, but I have spend some time looking at all the docs and can't
> get what I need to do completely sorted in my head.  I am definitely one
> who likes to UNDERSTAND what is happening rather than follow blank
> recipies, so please bear with me as I try understand all the details. I
> do understand about networking, NAT etc - but I am new to SIP/RTP and in
> particular what I think is a double NAT problem
>
>
> Firstly - what am I trying to achieve:
>
> I am in the UK and have a small home network behind a D-Link DIR-100
> Router/NAT/Firewall one of those machines, running Debian Lenny, acts as
> my main server for everything (and in an earlier incarnation was the
> firewall/router/nat box too - I only say this is because I had all this
> working using Asterisk a year or so ago, but with this important
> difference in configuration).  Many of the ports on the firewall are
> port forwarded to this machine. I have set Freeswitch up on this server
> to act as a small voip pbx for the home - but MORE IMPORTANTLY - to
> enable my daughter from her house to talk to us.  At my house locally I
> have a Linksys PAP2T two phone SIP box - and that is working with
> Freeswitch's default configuration (I set up to be 1000 and 1001 and
> used all the facilities).  I will later add a Linksys SPA 3102 -
> although I DO NOT intend to use its facility to bridge to the normal
> phone network.
>
> My daughter, living in another house, also has a Nat box (unknown - its
> part of her ADSL modem/router/wireless access point) and also has a
> PAP2T which she will connect to the her network.  This will be her phone.
>
> There is a family relation living in Australia who will load up a
> whatever softphone that we tell him to use.  I expect, but don't know,
> that he will behind a NAT box too.
>
> Later, I have some friends in the USA that I might wish to add it too -
> especially so that we can hold some teleconferences.  They will have a
> mixture of Windows and MACs, and I will need to recommend softphone
> clients for them.
>
> I want to set this up as a small private voice network, so anyone can
> ring anyone else.  I will add fancy facilities such as conferencing and
> voicemail later - I just want to get the basics working first.
>
> Secondly
>
> I installed a stun client on my home machine and ran it against
> stun.freeswitch.org.
>
> It reported:-
>
> Primary: Independent Mapping, Independent Filter, preserves ports, no
> hairpin
>
> But I have no idea what this means - I can't find any clear statement
> via googling for it - how this set of answers maps to the different
> types of NAT that might be required to get this to all work.  CAN
> SOMEONE ENLIGHTEN me please.
>
> Thirdly
>
> I have set up a sip profile called "double nat" from the recipe in the
> wiki.  This defines the SIP port to be 5090.
>
> However, what I DO NOT UNDERSTAND is how the PAP2T box in my daughters
> house will initiate a connection to my server.  Presumably, I have to
> port forward 5090 from the nat box to my server.  IS THAT CORRECT?
>
> I also assume I will have to tell her to use STUN (I believe this is an
> option on the PAP2T)
>
> Fourthly
>
> If I understand SIP correctly, it just initiates the session and the two
> end points then communicate directly via RTP.  What I don't understand
> is how does a session transition from SIP to RTP via the connection set
> up in the the first phase (in terms of passing through the NAT boxes).
> In particular WHICH OF THE TEST RESULTS from my stun client indicate it
> will do the right thing.  (I am going to take a laptop to my daughters
> house with a stun client in to test her network this weekend).
>
> Could someone explain please.
>
>   Fifthly
>
> Is there a recommended SIP softphone with all the right facilities (STUN
> support?)  that works on MAC and WINDOWS (I only use linux myself).
>
> Apologies for the length of this.  I am eager to get the answers so I
> can use an opportunity this weekend to get it working.
>
>
> --
> Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From jason at jasonjgw.net  Fri Aug  7 21:23:42 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 8 Aug 2009 14:23:42 +1000
Subject: [Freeswitch-users] New to Freeswitch - some help needed
In-Reply-To: <4A7BE35B.8010709@chandlerfamily.org.uk>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
Message-ID: <20090808042342.GA3558@jdc.jasonjgw.net>

Alan Chandler <alan at chandlerfamily.org.uk> wrote:
 
> I want to set this up as a small private voice network, so anyone can 
> ring anyone else.  I will add fancy facilities such as conferencing and 
> voicemail later - I just want to get the basics working first.

I have a similar arrangement operating here which involves friends and
colleagues in the U.S., as well as a local VoIP provider that gives me access
to the PSTN.

To eliminate NAT issues, we are using IPv6: each of us has an IPv6 over IPv4
tunnel configured to provide access to the IPv6 Internet. NAT and all the
problems associated with it go away.

Another option, although I don't know how well real-time communication works
in this setting, would be to create a VPN using, for example, OpenVPN so that
the clients and server all appear to be on the same lan.

Alternatively, you could play with port forwarding and FreeSWITCH settings in
an attempt to work around the nat issues - good luck!

I can't answer any questions about MacOS or Windows softphones - there are no
MacOS or Windows machines in my life.



From gcd at i.ph  Fri Aug  7 21:33:37 2009
From: gcd at i.ph (Nandy Dagondon)
Date: Sat, 8 Aug 2009 12:33:37 +0800
Subject: [Freeswitch-users] Auto Nat
In-Reply-To: <9f0cf614f2b2ffb9f7a2edae3bf3d2ff.squirrel@webmail.chandlerfamily.org.uk>
References: <9f0cf614f2b2ffb9f7a2edae3bf3d2ff.squirrel@webmail.chandlerfamily.org.uk>
Message-ID: <7d0bfd8c0908072133p7b7f68aancfabdb78d5fe7954@mail.gmail.com>

r 13612 is after 1.0.3. you better get 1.0.4 recently released.

-nandy

On Fri, Aug 7, 2009 at 8:19 PM, <alan at chandlerfamily.org.uk> wrote:

> I sent my first e-mail to the list this morning (about 4 hours ago) but it
> does not seem to have arrived back, even though I have received other,
> later posts.
>
> I have another question related to the first (about how to set everything
> up in a double nat environment) - so if I see this and not the other, I
> will send the first again.
>
> I am currently running the stable version 1.0.3 of freeswitch.  The wiki
> page says that auto-nat is introduced at r 13612.  Is this before or after
> that revision? (I don't want to have to download and rebuild the entire
> thing if I don't have to).
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Fri Aug  7 21:36:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 00:36:58 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
Message-ID: <84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>

This is not currently a supported platform, it only builds on 64 bit  
right now I think on solaris.

Mike

On Aug 6, 2009, at 6:03 PM, vmorales wrote:

> Hello,
>
> Does anyone have, or know where to get, a pre-compiled copy of
> FreeSwitch for Solaris 10/x86?



From mcampbellsmith at gmail.com  Fri Aug  7 21:39:14 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sat, 8 Aug 2009 14:39:14 +1000
Subject: [Freeswitch-users] New to Freeswitch - some help needed
In-Reply-To: <4A7BE35B.8010709@chandlerfamily.org.uk>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
Message-ID: <33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>

Hi Alan,

I hope you find your answers here as these are the sort of things that
are hard to find on the wiki, which is somewhat outdated in areas.  If
you do find your answers, please post them back here for everyone
else.

I am new to FS also, so my comments below may not be 100% correct!

1. Very similar to what I want to have setup as well.  Do you have a
static IP address at home.  If not, get a dyndns account and setup an
entry there so that your friends/family can register using your dns
name instead of ip address

2. No idea.  Maybe try another stun server?

3. Not sure if double-NAT is needed now with the newer builds of
FreeSwitch.  Download the latest 1.0.4 to be on the safeside and
compile it again!  (I have FS 1.0.4 pre9 and it works I think).  As
long as your clients can register remotely you should be okay. I think
FS can work around most home NATs.  Make sure you have auto-nat set in
your internal.xml file (I think its this one)

4. SIP is the signaling.  RTP is the payload, or voice in your case.
Any transition is done via the SIP signaling.  This is how FS can
transfer calls etc or use the media bypass mode by specifying the IP
address where the RTP should be sent, which does not have to be the
same as the signaling.  Make sure you enable tracing in the
internal.xml file so you can debug the signaling.

You don't need to take a laptop to your daughters to test this.  Use
an internet sip phone like flaphone.com, which works through your web
browser.  This will register with an external IP address exactly like
your daughters and save you time traveling.  Note that sound isn't so
clear for me using this service, but it helps with debugging.

I also would recommend a sip client on windows like Zoiper, or
CounterPath's X-Lite.. both are free.  X-Lite is well known, Zoiper
allows for multiple SIP registrations and comes in a portable version.







On Fri, Aug 7, 2009 at 6:18 PM, Alan Chandler<alan at chandlerfamily.org.uk> wrote:
> I apologize, as my first post to this list, that I ask a detailed set of
> questions, but I have spend some time looking at all the docs and can't
> get what I need to do completely sorted in my head. ?I am definitely one
> who likes to UNDERSTAND what is happening rather than follow blank
> recipies, so please bear with me as I try understand all the details. I
> do understand about networking, NAT etc - but I am new to SIP/RTP and in
> particular what I think is a double NAT problem
>
>
> Firstly - what am I trying to achieve:
>
> I am in the UK and have a small home network behind a D-Link DIR-100
> Router/NAT/Firewall one of those machines, running Debian Lenny, acts as
> my main server for everything (and in an earlier incarnation was the
> firewall/router/nat box too - I only say this is because I had all this
> working using Asterisk a year or so ago, but with this important
> difference in configuration). ?Many of the ports on the firewall are
> port forwarded to this machine. I have set Freeswitch up on this server
> to act as a small voip pbx for the home - but MORE IMPORTANTLY - to
> enable my daughter from her house to talk to us. ?At my house locally I
> have a Linksys PAP2T two phone SIP box - and that is working with
> Freeswitch's default configuration (I set up to be 1000 and 1001 and
> used all the facilities). ?I will later add a Linksys SPA 3102 -
> although I DO NOT intend to use its facility to bridge to the normal
> phone network.
>
> My daughter, living in another house, also has a Nat box (unknown - its
> part of her ADSL modem/router/wireless access point) and also has a
> PAP2T which she will connect to the her network. ?This will be her phone.
>
> There is a family relation living in Australia who will load up a
> whatever softphone that we tell him to use. ?I expect, but don't know,
> that he will behind a NAT box too.
>
> Later, I have some friends in the USA that I might wish to add it too -
> especially so that we can hold some teleconferences. ?They will have a
> mixture of Windows and MACs, and I will need to recommend softphone
> clients for them.
>
> I want to set this up as a small private voice network, so anyone can
> ring anyone else. ?I will add fancy facilities such as conferencing and
> voicemail later - I just want to get the basics working first.
>
> Secondly
>
> I installed a stun client on my home machine and ran it against
> stun.freeswitch.org.
>
> It reported:-
>
> Primary: Independent Mapping, Independent Filter, preserves ports, no
> hairpin
>
> But I have no idea what this means - I can't find any clear statement
> via googling for it - how this set of answers maps to the different
> types of NAT that might be required to get this to all work. ?CAN
> SOMEONE ENLIGHTEN me please.
>
> Thirdly
>
> I have set up a sip profile called "double nat" from the recipe in the
> wiki. ?This defines the SIP port to be 5090.
>
> However, what I DO NOT UNDERSTAND is how the PAP2T box in my daughters
> house will initiate a connection to my server. ?Presumably, I have to
> port forward 5090 from the nat box to my server. ?IS THAT CORRECT?
>
> I also assume I will have to tell her to use STUN (I believe this is an
> option on the PAP2T)
>
> Fourthly
>
> If I understand SIP correctly, it just initiates the session and the two
> end points then communicate directly via RTP. ?What I don't understand
> is how does a session transition from SIP to RTP via the connection set
> up in the the first phase (in terms of passing through the NAT boxes).
> In particular WHICH OF THE TEST RESULTS from my stun client indicate it
> will do the right thing. ?(I am going to take a laptop to my daughters
> house with a stun client in to test her network this weekend).
>
> Could someone explain please.
>
> ?Fifthly
>
> Is there a recommended SIP softphone with all the right facilities (STUN
> support?) ?that works on MAC and WINDOWS (I only use linux myself).
>
> Apologies for the length of this. ?I am eager to get the answers so I
> can use an opportunity this weekend to get it working.
>
>
> --
> Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dome at tel.co.th  Fri Aug  7 21:57:08 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 8 Aug 2009 11:57:08 +0700
Subject: [Freeswitch-users] 1.0.4 builds 1.0.3_1 debs
In-Reply-To: <631613604d8c9862f1cf43f80cb99f94.squirrel@webmail.chandlerfamily.org.uk>
References: <631613604d8c9862f1cf43f80cb99f94.squirrel@webmail.chandlerfamily.org.uk>
Message-ID: <8ccbff060908072157u19d0cb1rb48913832c74024@mail.gmail.com>

dch -i

2009/8/7  <alan at chandlerfamily.org.uk>:
> I just downloaded 1.0.4 and build debian packages with it, and it
> delivered .deb files names as 1.0.3-1
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From velu.technical at gmail.com  Fri Aug  7 22:28:27 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Sat, 8 Aug 2009 10:58:27 +0530
Subject: [Freeswitch-users] ESL for Perl
Message-ID: <1452e2980908072228u5e5fc191ufa2cdf002d903396@mail.gmail.com>

Dear all,
   When I do make for Perl ESL libraries in "esl/perl" directory I have got
the following error.

/usr/lib/gcc/i486-linux-gnu/4.1.2/../../../../lib/crt1.o: In function
`_start':
../sysdeps/i386/elf/start.S:115: undefined reference to `main'
esl_wrap.o: In function `_wrap_eslSetLogLevel':
esl_wrap.cpp:(.text+0x1c10): undefined reference to `eslSetLogLevel'
esl_wrap.o: In function `_wrap_ESLconnection_setEventLock':
esl_wrap.cpp:(.text+0x3395): undefined reference to
`ESLconnection::setEventLock(char const*)'
esl_wrap.o: In function `_wrap_ESLconnection_setBlockingExecute':
esl_wrap.cpp:(.text+0x3c65): undefined reference to
`ESLconnection::setBlockingExecute(char const*)'
esl_wrap.o: In function `_wrap_ESLconnection_execute':
esl_wrap.cpp:(.text+0x41ab): undefined reference to
`ESLconnection::execute(char const*, char const*, char const*)'
........

I have understood the when creating soft link ESL.o that error has occurred.

What is the problem?
Is there any dependency to create that link?

please help me.......

Thanks

Regards,
Velusamy
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From mike at jerris.com  Fri Aug  7 22:36:18 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 01:36:18 -0400
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <26f14763-6f40-4468-91f3-e60a46892183@dmmhosting.co.uk>
References: <26f14763-6f40-4468-91f3-e60a46892183@dmmhosting.co.uk>
Message-ID: <3A70DE01-8DEB-489B-9CAF-DA650CD3F2B1@jerris.com>

Please follow up with this on jira so we can make sure this gets  
addressed.

Mike

On Aug 4, 2009, at 5:10 AM, Darren Williams wrote:

> This produces:
>
> send 639 bytes to udp/[62.239.15.140]:5060 at 09:06:29.640488:
>     
> ------------------------------------------------------------------------
>    REGISTER sip:bmnha-01.bt.com SIP/2.0
>    Via: SIP/2.0/UDP  
> 91.121.159.57:5080;rport;branch=z9hG4bKQFFy1Ug5DcU0S
>    Max-Forwards: 70
>    From: <sip: 
> 05061292117 at bmnha-01.bt.com;transport=udp>;tag=NKaDDXrtme3Sr
>    To: <sip:05061292117 at bmnha-01.bt.com;transport=udp>
>    Call-ID: 88de957f-0907-4488-9b19-043ff13391f4
>    CSeq: 118570058 REGISTER
>    Contact: <sip:gw+05061292117 at 91.121.159.57:5080;transport=udp>
>    Expires: 3600
>    User-Agent: THOMSON ST2030 hw5 fw1.56 00-1F-9F-16-4E-99
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
>    Supported: timer, precondition, path, replaces
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
> recv 726 bytes from udp/[62.239.15.140]:5060 at 09:06:29.656368:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP  
> 91.121.159.57 
> :5080;received=91.121.159.57;branch=z9hG4bKQFFy1Ug5DcU0S;rport=5080
>    From: <sip: 
> 05061292117 at bmnha-01.bt.com;transport=udp>;tag=NKaDDXrtme3Sr
>    To: <sip: 
> 05061292117 
> @bmnha 
> -01 
> .bt 
> .com 
> ;transport=udp>;tag=SD567ec99-8a92eaf49d827b084b51e364ec68ae70.bc35
>    Call-ID: 88de957f-0907-4488-9b19-043ff13391f4
>    CSeq: 118570058 REGISTER
>    WWW-Authenticate: Digest realm="bmnha-01.bt.com",  
> nonce="4a77fb4b6fe33ffccf77e6bdd6f7a7a397cdfd17", qop="auth"
>    Server: Sip EXpress router (0.9.6 (sparc/solaris))
>    Content-Length: 0
>    Warning: 392 sip:5060 "Noisy feedback tells:  pid=9150  
> req_src_ip=172.20.92.61 req_src_port=5060 in_uri=sip:bmnha-01.bt.com  
> out_uri=sip:bmnha-01.bt.com via_cnt==1"
>
>     
> ------------------------------------------------------------------------
> 2009-08-04 11:06:29.655585 [ERR] sofia_reg.c:1460 05061292117  
> Registration Failed with status Operation has no matching challenge   
> [904]. failure #1 and so on.
>
> Should my side not have responded at this point with another  
> REGISTER with auth information in?

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From mike at jerris.com  Fri Aug  7 22:42:46 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 01:42:46 -0400
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
Message-ID: <3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>

http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP

turn the logging all the way up and see what it says.

Mike

On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:

> Hi Mathieu, thanks for the reply. I enabled sip trace logging and  
> got the logs below, but I am still at a loss at being able to  
> identify the error or reproduce it consistently. The below log  
> indicates to me that my FS server is initiating sending 2 BYE  
> message to my DID provider (didforsale.com). Is there a way I can  
> look further inside FreeSWITCH to see why it is sending this BYE  
> packet?
>
>
> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
> Route: <sip:65.217.40.210:5060;lr>
> Route: <sip:63.110.102.238;lr>
> Max-Forwards: 70
> From: <sip: 
> +1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
> To: <sip: 
> + 
> 1909635XXXX 
> @199.173.100.144:5060;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
> CSeq: 118584736 BYE
> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
> Content-Length: 0
>
>
> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
> Route: <sip:65.217.40.210:5060;lr>
> Route: <sip:63.110.102.238;lr>
> Max-Forwards: 70
> From: <sip: 
> +1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
> To: <sip: 
> + 
> 1909635XXXX 
> @199.173.100.144:5060;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
> CSeq: 118584736 BYE
> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
> Content-Length: 0
>
> On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> Hi,
>
> Digging a bit in mod_sofia releaved that it can be caused by a SIP
> code 482 (loop detected), 483 (too many hops) or 484 (address
> incomplete).
>
> Do a SIP trace to sched more light on what's happening.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca

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From mike at jerris.com  Fri Aug  7 22:46:10 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 01:46:10 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>
	<E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>
	<6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>
Message-ID: <782241E8-A1A6-4074-8D0A-4D362CF45FA5@jerris.com>

there is some other info on http://wiki.freeswitch.org/wiki/Perl_esl  
about this as well.  could somone try to put together some decent docs  
on fs_ivrd on the wiki as I keep getting asked this stuff privately as  
well.

Mike

On Aug 4, 2009, at 7:04 PM, William Suffill wrote:

> I wrote some notes on this but have yet to wiki it.
>
> example of an outbound socket connection where the call is answered, a
> variable is set then perhaps play one of the pre-installed files and
> hangup.
> ivrd
>
> fs_ivrd comes with freeswitch. It being a small daemon just invokes
> the script defined in a variable and passes data from it via STDIN/OUT
>
> Since this is an outbound socket connections it needs to be defined in
> the dialplan.
>
>
>
> Ex:
>
> <extension name="outbound-socket">
>
>    <condition field="destination_number" expression="^55(522)$">
>
>        <action application="set"
> data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
>
>        <action application="socket" data="127.0.0.1:8084 async full"/>
>
>    </condition>
>
>  </extension>
>
>
>
> The above dialplan sample would invoke ivr-demo.php when 55522 is
> called as long as fs_ivrd is running. To start fs_ivrd:
>
> /usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 8004
>
>
>
> It takes 2 arguments -h for hostname and -p for port.
>
>
>
> PHP Code
>
> #!/usr/bin/php -q
>
> <?php
>
>
>
>    // set a couple of things so we dont kill the system
>
>    ob_implicit_flush(true);
>
>    set_time_limit(30);
>
>
>
>
>
>    // Open stdin so we can read the AGI data in
>
>    $in = fopen("php://stdin", "r");
>
>    // Connect
>
>    echo "connect\n\n";
>
>    // Answer
>
>    echo "sendmsg\n";
>
>    echo "call-command: execute\n";
>
>    echo "execute-app-name: answer\n\n";
>
>
>
>    // Play a prompt
>
>    echo "sendmsg\n";
>
>    echo "call-command: execute\n";
>
>    echo "execute-app-name: playback\n";
>
>    echo "execute-app-arg:
> /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr- 
> welcome_to_freeswitch.wav\n\n";
>
>
>
>    // Wait
>
>    sleep(5);
>
>
>
>   // Hangup
>
>   echo "sendmsg\n";
>
>   echo "call-command: hangup\n\n";
>
>
>
> fclose($in);
>
>
>
> ?>
>
>
>
> ivrd will call this script for each call. All itdoes is answer the
> channel tell FreeSWITCH to play the ?welcome to freeswitch? prompt.
> Since the script is now controlling all call flow I needed to add a
> wait or it would send the hangup immediately before the prompt was
> played.
>
> Some improvements possible but that's 1 way to do it. It would be
> possible to do the socket directly in PHP but fs_ivrd is a nice option
> too.
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Fri Aug  7 22:49:01 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 01:49:01 -0400
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
Message-ID: <179DF0A9-1E96-43EB-976C-FE00365436BC@jerris.com>

Please report this error to centos bug tracker, aspell-devel should be  
a dependency of php-devel.

Mike

On Aug 5, 2009, at 10:40 AM, Greg Thoen wrote:

> Trying to make phpmod and it fails with this:
>
> /usr/bin/ld: cannot find -laspell
> collect2: ld returned 1 exit status
> make[1]: *** [ESL.so] Error 1
>
> I do have php-devel on this Centos 5.2 machine. Any ideas?
> --
> Greg
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Aug  7 22:55:00 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 01:55:00 -0400
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <a44688a0908052054j405c1707j382b3d6ccac8f8a9@mail.gmail.com>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
	<E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
	<a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>
	<9521EE86-37FF-4989-8F58-F61E5110E5C5@avgs.ca>
	<a44688a0908052054j405c1707j382b3d6ccac8f8a9@mail.gmail.com>
Message-ID: <358E66DE-C5F1-418B-806B-2441E8F42130@jerris.com>

you could probably pull off the same thing with xml_curl for dialplan  
and a simple set of bridge and then transfer in the actions.

Mike

On Aug 5, 2009, at 11:54 PM, Woody Dickson wrote:

> Hi,
>
> In my module, I will collect a list of available failover route that  
> I can use to failover to whenever a particular error is received.   
> However, these available routes has different condition and the  
> condition changes every half a minute.  Therefore, I need to catch  
> the hangup cause after bridge, and then figure out the next workable  
> available route based on the latest condition setting.
>
> It seems like this is only prossible to be done within a C module.   
> Any suggestion will be greatly appreciated.
>
> Woody



From mike at jerris.com  Fri Aug  7 22:57:21 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 01:57:21 -0400
Subject: [Freeswitch-users] Question about dynamic registration
In-Reply-To: <27c25bc40908060224l38ad47fdje17065b13647905d@mail.gmail.com>
References: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>
	<9FE0F6B8-C4BF-4820-8CC2-6825C5EE8422@freeswitch.org>
	<27c25bc40908060224l38ad47fdje17065b13647905d@mail.gmail.com>
Message-ID: <03C39E0E-2303-40EC-9B84-5D0AE499436D@jerris.com>

http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/cparker/mod_xml_radius

On Aug 6, 2009, at 5:24 AM, Juan Backson wrote:

> Hi,
>
> Is there a sample module that I can take a look at on how to do that?
> I don't understand how to get the registration request and how to  
> pass back auth result to freeswitch.
>
> JB
>
> On Mon, Aug 3, 2009 at 8:42 PM, Brian West <brian at freeswitch.org>  
> wrote:
> You could build your own module to do it how ever you please.  But
> forking a script every time to auth is not very scalable.
>
> /b
> _______________________________________________

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From mike at jerris.com  Fri Aug  7 23:11:12 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 02:11:12 -0400
Subject: [Freeswitch-users] Lua on Windows and additional modules
In-Reply-To: <3c233920908061755h5d17aa0as3fb24743215a8298@mail.gmail.com>
References: <3c233920908061755h5d17aa0as3fb24743215a8298@mail.gmail.com>
Message-ID: <0EE782AE-FC28-4029-AA01-76D87BC08069@jerris.com>

check the sample config files for options to specify these:

http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/autoload_configs/lua.conf.xml?r=10747


On Aug 6, 2009, at 8:55 PM, Vladimir Rodionov wrote:

> Good evening,
> This is newbie question.
>
> The FreeSWITCH lua module does not support sockets and sql out of  
> box that is why
> I just installed LuaBinaries (including socket, sql modules). My dev  
> environment is Win XP not Linux/Unix.
>
> I am trying to understand what will happen when lua_module get this:
>
> require "socket" or
> require "luasql.mysql"
> ?
>
> How does lua_module look up additional lua modules on Windows  
> platform?
>
> Do I have to set some env variables?
>
> TIA
> -Vladimir Rodionov



From merul at mac.com  Sat Aug  8 01:58:04 2009
From: merul at mac.com (Merul Patel)
Date: Sat, 08 Aug 2009 09:58:04 +0100
Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk
	-	Freeswitch equivalent?
Message-ID: <C5CDD462-A4BF-4316-B35C-4892C92DB9F4@mac.com>

Hi,

I came across this thread (http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/010172.html 
) from January, and I'm having a similar problem, in that from the  
moment an incoming analogue call starts ringing, it takes around 5-7  
seconds before the dialplan gets executed.

Tailing the freeswitch log (pastebin'd here: http://pastebin.com/f2a6f7945) 
, it seems to indicate that there is either a problem with an  
unhandled event 17 and/or a delay with the call being put in the  
GET_CALLERID state.

In my case, I'm more concerned about answering the call rather than  
getting the caller id, so I tried setting <param name="enable- 
callerid" value="false"/> in openzap.conf.xml, as per the earlier  
post's author but this still doesn't get the dialplan to execute  
immediately.

I'm running FS 1.0.4 built from the tarball, with zaptel 1.4.12.1 and  
wanpipe 3.5.4.18 in conjunction with a USBFXO device from Sangoma. I'm  
using the out -of-the-box configuration for FS, with the exception of  
one modified public dialplan. My openzap.conf.xml is:

<configuration name="openzap.conf" description="OpenZAP Configuration">
   <settings>
     <param name="debug" value="0"/>
     <!--<param name="hold-music" value="$${moh_uri}"/>-->
     <!--<param name="enable-analog-option" value="call-swap"/>-->
     <!--<param name="enable-analog-option" value="3-way"/>-->
   </settings>
    <pri_spans/>
   <!-- one entry here per openzap span -->
   <analog_spans>
     <span id="1">
       <!--<param name="hold-music" value="$${moh_uri}"/>-->
       <!--<param name="enable-analog-option" value="call-swap"/>-->
       <!--<param name="enable-analog-option" value="3-way"/>-->
       <param name="tonegroup" value="uk"/>
       <param name="digit-timeout" value="2000"/>
       <param name="max-digits" value="11"/>
       <param name="dialplan" value="XML"/>
       <param name="context" value="public"/>
       <param name="enable-callerid" value="true"/>
       <!-- regex to stop dialing when it matches -->
       <!--<param name="dial-regex" value="5555"/>-->
       <!-- regex to stop dialing when it does not match -->
       <!--<param name="fail-dial-regex" value="^5"/>-->
     </span>
   </analog_spans>
</configuration>

Couldn't find any jiras for this issue, so not sure whether it was  
resolved. Any pointers would be welcome.

BR

Merul


From pjintheusa at gmail.com  Sat Aug  8 03:48:51 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 03:48:51 -0700
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
In-Reply-To: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>
References: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>
Message-ID: <367751820908080348p6165b113kc3be468cb32fabb3@mail.gmail.com>

Hi there,

Not sure whether this helps but test this without set bypass_media. In
my setup I have noticed the leg A session ends when bypass_media is
true. Call/bridge continue successfully.

Phillip Jones

On Thu, Aug 6, 2009 at 1:28 PM, Benedikt
Fraunhofer<fraunhofer.lists.freeswitch-001 at traced.net> wrote:
> Hello List, Hello *,
>
> First of all the usual excuses: sorry for the bad english and the long
> email, no native speaker and i really tried to make it shorter, but i
> guess this would result in even more "check back"s than it already
> does :)
>
> we're currently running in a weird "lockup"-scenario in our loadtests.
>
> Our setup is the following:
>
> three freeswitch servers, let's call them A(-leg), M(aster), and
> B(-leg) with the goal in mind to initiate calls on M which calls A,
> play some file, bridge to B, limit call length and play (different)
> prompts to A and B if they exceed that limit.
>
> (Note that A and B work fine, regardless of the amount of load we put on them)
> A and B are silly dialplan logic, accepting calls on a certain
> extension after a random delay and playing moh. Before calling
> playback to a localstream they call a lua script which schedules
> hangup somewhere in future (which works flawlessly)
>
> Calls are initiated on M using some hacked up loadgen-script issuing
> http requests like
> ? originate [sofiaSyntaxToExtensionOn_A] 6000
> . The 6000 extension on M has the following (xml) dialplan which
> essentially does the following:
> ------
> answer()
> ...playback file...
> ...set some callerid stuff
> set bypass_media
> bridge to extension 6009 on B
> ------
> we use "execute_on_answer" on the b-leg to run a script which limits
> the length of the call (doesn't matter if it's done via "export
> nolocal" or "inlined" into the data part of the bridge application
> "{execute_on_answer=lua ...}")
>
> <action application="export" data="nolocal:execute_on_answer=lua
> lua/schedule-hangup.lua ${uuid}" />
>
> the lua script "schedula-hangup.lua" does essentially the following:
>
> ------
> api = freeswitch.API();
> local res = api:execute("sched_api", "+10 none lua
> lua/c2c-hangup-timeout.lua " .. argv[1]);
> ------
>
> the 10 seconds are just to speed up the time until it gets stuck.
>
> this is where things start to go wrong. if I comment out the call to
> the "schedule-hangup" script, everything works fine, even if it's
> under heavy load.
>
> c2c-hangup-timeout.lua does the following:
> ------------------
> local sess = argv[1];
> if(sess)
> then
> ? ?freeswitch.consoleLog("INFO", "c2c-hangup-timeout.lua for uuid "
> .. sess .. "\n");
>
> ? ?api = freeswitch.API();
> ? ?local stillValid = api:execute("uuid_getvar", sess .. "
> Dummy-DoesChannelExists");
> ? ?if(stillValid:sub(1,4) == "-ERR")
> ? ?then
> ? ? ? ?log("session uuid " .. sess .. " disappeared (nothing bad)");
> ? ?else
> ? ? ? ?-- this is important!!! Otherwise the aleg get's just hung up!
> ? ? ? ?api:execute("uuid_media", sess);
> ? ? ? ?api:execute("uuid_transfer", sess .. " -both timeout");
> ? ?end
> else -- /if(sess)
> ? ?log("called with nil session?");
> end -- /if(sess)
>
> ------------------
>
> i guess this needs some explanation:
> we get the uuid of the channel as argument in argv[1]. We don't use
> ? local session = freeswitch.Session(uuid);
> since if the channel referenced by "uuid" does not exist any longer,
> freeswitch (or the lua bindings) try to interpret the uuid as an
> "originate string" and can't figure out how to call that. So we use a
> dummy api call to get some channel variable. If the channel does not
> exist any longer (A or B already hung up), we get an error message
> starting with "-ERR", otherwise the channel still exists (we get
> "_unset_" as the value, if it's not set) and we continue by getting
> freeswitch back in the media path (uuid_media) and then transferring
> both legs to an extension called "timeout" which plays some prompt and
> finally calls hangup().
>
> If we don't do the uuid_media call, one of the legs gets hung up when
> we transfer them to the extension. This looks like the following on
> the console after issuing "uuid_transfer [uuid] -both timeout"
> (extensions are not the same as in our loadgen example above)
>
>
> --------------
> 2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1334 Hangup (*)
> sofia/internal/1000 [CS_HIBERNATE] [BLIND_TRANSFER]
> 2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer
> sofia/internal/1004 at 192.168.179.177:5060 to XML[timeout at default]
> 2009-07-23 19:57:19.865703 [INFO] mod_dialplan_xml.c:310 Processing
> BFR1004->timeout in context default
> API CALL [uuid_transfer(73812082-77b1-11de-b9f8-a10bb0eb9f69 -both
> timeout)] output:
> +OK
>
> 2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer (**)
> sofia/internal/1000 to XML[timeout at default]
> 2009-07-23 19:57:19.865703 [NOTICE] switch_core_session.c:1084 Session
> 60 (sofia/internal/1000) Ended
> 2009-07-23 19:57:19.865703 [NOTICE] switch_core_session.c:1086 Close
> Channel sofia/internal/1000 [CS_DESTROY]
> -----------
>
> note that it first does Hangup (denoted by *, no that's not an
> asterisk :) on extension 1000 and then tries to Transfer (**) the hung
> up channel to the dial plan. this could be the same as in an earlier
> post to the list "SIP re-invite / bypass_media // Phillip Jones //
> Wed, 01 Jul 2009 13:30:53 -0700)"
>
> This is why we do not directly call sched_transfer() but call a script
> in between to do the uuid_media() call. I couldn't figure out how to
> call that directly from the xml dialplan and/or how to check if the
> channel still exists.
>
> so... after using uuid_media(), both legs are transferred without an
> (intermediate|bogus) hangup() call.
>
> This only works fine if we've few concurrent calls. There is no magic
> borderline where it starts to refuse work.
>
> Some of the Symptoms are: traffic decreased to zero as no new channels
> are successfully brought up, some of the signaling traffic is not
> ACKed or OKed, scheduled jobs are not run.
>
> if i read the output of "show channels" correctly, they're all stuck
> in different applications like hangup(), some are calling lua but most
> of them are in signaling_bridge(). Freeswitch is still responding on
> the console and there's almost no load on the machine (no busy polling
> or some other kind of running amok).
>
> if i kill one of them using uuid_kill() or kill all of them
> using"fsctl hupall" i get "Task was executed late by 866 seconds 12379
> sched_api_function (none)" messages and the usual cleanup takes place.
> As a quick hack i tried to schedule a uuid_kill() call 20 seconds
> after the scheduling call to the lua script but that job is not
> executed either.
>
> So what am I doing wrong? Is it some deadlock where uuid_media() and
> uuid_transfer() ?are waiting for the other to finish?
> Or some other silly simple thing i missed?
>
> Thx in advance
>
> ?Benedikt.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From pjintheusa at gmail.com  Sat Aug  8 03:59:56 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 03:59:56 -0700
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
Message-ID: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>

Hi there,

I am trying to implement a scenario where I can terminate calls to
multiple destinations AND have termination carrier fail over.
Currently I can see how to do one or the other. But not both.

Multiple destinations is easy:

 <action application="bridge"
data="sofia/SIP_PROVIDER_1/6095551234,sofia/SIP_PROVIDER_1/7325551234"/>

Failover appears to use the same mechanism:

<action application="bridge"
data="sofia/gateway/primary/blah|sofia/gateway/secondary/blah"/>

I can not get my head around how use these together, such that:


- 6095551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 6095551234  through SIP_PROVIDER_2
- Called party does not answer
7325551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 7325551234 through SIP_PROVIDER_2
- Called party answers


This must be a fairly common requirement so any ideas on what I might
be missing would be very welcome.

Thanks


Phillip Jones


From mike at jerris.com  Sat Aug  8 04:28:22 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 07:28:22 -0400
Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk -
	Freeswitch equivalent?
In-Reply-To: <C5CDD462-A4BF-4316-B35C-4892C92DB9F4@mac.com>
References: <C5CDD462-A4BF-4316-B35C-4892C92DB9F4@mac.com>
Message-ID: <E15045EA-3517-4174-B12E-12143BFCE27B@jerris.com>

is this just the timeout waiting for your max digits?

       <param name="max-digits" value="11"/>


On Aug 8, 2009, at 4:58 AM, Merul Patel wrote:

> Hi,
>
> I came across this thread (http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/010172.html
> ) from January, and I'm having a similar problem, in that from the
> moment an incoming analogue call starts ringing, it takes around 5-7
> seconds before the dialplan gets executed.
>
> Tailing the freeswitch log (pastebin'd here: http://pastebin.com/f2a6f7945)
> , it seems to indicate that there is either a problem with an
> unhandled event 17 and/or a delay with the call being put in the
> GET_CALLERID state.
>
> In my case, I'm more concerned about answering the call rather than
> getting the caller id, so I tried setting <param name="enable-
> callerid" value="false"/> in openzap.conf.xml, as per the earlier
> post's author but this still doesn't get the dialplan to execute
> immediately.
>
> I'm running FS 1.0.4 built from the tarball, with zaptel 1.4.12.1 and
> wanpipe 3.5.4.18 in conjunction with a USBFXO device from Sangoma. I'm
> using the out -of-the-box configuration for FS, with the exception of
> one modified public dialplan. My openzap.conf.xml is:
>
> <configuration name="openzap.conf" description="OpenZAP  
> Configuration">
>   <settings>
>     <param name="debug" value="0"/>
>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>     <!--<param name="enable-analog-option" value="3-way"/>-->
>   </settings>
>    <pri_spans/>
>   <!-- one entry here per openzap span -->
>   <analog_spans>
>     <span id="1">
>       <!--<param name="hold-music" value="$${moh_uri}"/>-->
>       <!--<param name="enable-analog-option" value="call-swap"/>-->
>       <!--<param name="enable-analog-option" value="3-way"/>-->
>       <param name="tonegroup" value="uk"/>
>       <param name="digit-timeout" value="2000"/>
>       <param name="max-digits" value="11"/>
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="public"/>
>       <param name="enable-callerid" value="true"/>
>       <!-- regex to stop dialing when it matches -->
>       <!--<param name="dial-regex" value="5555"/>-->
>       <!-- regex to stop dialing when it does not match -->
>       <!--<param name="fail-dial-regex" value="^5"/>-->
>     </span>
>   </analog_spans>
> </configuration>
>
> Couldn't find any jiras for this issue, so not sure whether it was
> resolved. Any pointers would be welcome.
>
> BR
>
> Merul
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From fraunhofer.lists.freeswitch-001 at traced.net  Sat Aug  8 04:53:05 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Sat, 8 Aug 2009 13:53:05 +0200
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
In-Reply-To: <367751820908080348p6165b113kc3be468cb32fabb3@mail.gmail.com>
References: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>
	<367751820908080348p6165b113kc3be468cb32fabb3@mail.gmail.com>
Message-ID: <a37510170908080453m1e98ae3l544e3d8b301605e1@mail.gmail.com>

Hi Phillip,

2009/8/8 Phillip Jones <pjintheusa at gmail.com>:

> Not sure whether this helps but test this without set bypass_media. In
> my setup I have noticed the leg A session ends when bypass_media is
> true. Call/bridge continue successfully.

thx for that hint. unfortunately we can't do that due to the high
volume we anticipate
and i think i already tried that.

Note that it works in exactly this setup  (with the uuid_media()-call)
if there are only very few calls handled. It just starts to refuse
work once enough jobs are scheduled and not executed.

Cheers
  Benedikt


From frank at impactfax.com  Sat Aug  8 06:10:18 2009
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 8 Aug 2009 09:10:18 -0400
Subject: [Freeswitch-users] DTMF disable a few secs after call starts
Message-ID: <0B930F23C4A24E3CB0015945EA06C8B7@ws4>

FS is in the media path of an IVR call.
At the moment, the call is ulaw with DTMF in the audio I think coming
into FS and leaving FS.
The call is coming from an Asterisk server and going to an Asterisk
server.
 
Is there a way to disable FS from passing DTMF at some point in the
call?  For example, after 15 seconds, is there a way to get FS to stop
passing DTMF events?
 
Would I have to try to force asterisk to use rfc2833 when sending the
call to FS and when accepting it back from FS?
 
 
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From william.suffill at gmail.com  Sat Aug  8 06:33:49 2009
From: william.suffill at gmail.com (William Suffill)
Date: Sat, 8 Aug 2009 09:33:49 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <782241E8-A1A6-4074-8D0A-4D362CF45FA5@jerris.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local> 
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com> 
	<E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com> 
	<6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com> 
	<782241E8-A1A6-4074-8D0A-4D362CF45FA5@jerris.com>
Message-ID: <6b65470d0908080633m1b3c15b9me110eb0245a93b0f@mail.gmail.com>

I'll see what I can do. Got a few sections of the PHP ESL to finish
but should be able to
write up fs_ivrd as well.

-- W


From rupa at rupa.com  Sat Aug  8 07:16:37 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 8 Aug 2009 09:16:37 -0500
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
Message-ID: <b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>

While any use of loopback can be considered abuse, that is how I solve this
issue.  I use loopback for each dialed # and then in the dialplan use
mod_lcr to build a dialstring that does failover in order of cost.

So:

 <action application="bridge"
data="sofia/SIP_PROVIDER_1/6095551234,sofia/SIP_PROVIDER_1/7325551234"/>
becomes

 <action application="bridge"
data="loopback/6095551234,loopback/7325551234"/>

In the dialplan you then match on those (just regular 10digit dialing) and
do your normal failover dialstring.  Again, I do this with mod_lcr but that
isn't necessary.

<action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>


On Sat, Aug 8, 2009 at 5:59 AM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi there,
>
> I am trying to implement a scenario where I can terminate calls to
> multiple destinations AND have termination carrier fail over.
> Currently I can see how to do one or the other. But not both.
>
> Multiple destinations is easy:
>
>  <action application="bridge"
> data="sofia/SIP_PROVIDER_1/6095551234,sofia/SIP_PROVIDER_1/7325551234"/>
>
> Failover appears to use the same mechanism:
>
> <action application="bridge"
> data="sofia/gateway/primary/blah|sofia/gateway/secondary/blah"/>
>
> I can not get my head around how use these together, such that:
>
>
> - 6095551234 is dialed through SIP_PROVIDER_1
> - if NO_ROUTE_DESTINATION then dial 6095551234  through SIP_PROVIDER_2
> - Called party does not answer
> 7325551234 is dialed through SIP_PROVIDER_1
> - if NO_ROUTE_DESTINATION then dial 7325551234 through SIP_PROVIDER_2
> - Called party answers
>
>
> This must be a fairly common requirement so any ideas on what I might
> be missing would be very welcome.
>
> Thanks
>
>
> Phillip Jones
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From rupa at rupa.com  Sat Aug  8 07:18:59 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 8 Aug 2009 09:18:59 -0500
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
	<b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
Message-ID: <b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>

On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>
>

That of course, should be:

 <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$1..."/>
-- 
-Rupa
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From brian at freeswitch.org  Sat Aug  8 07:26:32 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 8 Aug 2009 09:26:32 -0500
Subject: [Freeswitch-users] ESL for Perl
In-Reply-To: <1452e2980908072228u5e5fc191ufa2cdf002d903396@mail.gmail.com>
References: <1452e2980908072228u5e5fc191ufa2cdf002d903396@mail.gmail.com>
Message-ID: <7A151806-653B-4DDA-84A4-F5EBC25E0D81@freeswitch.org>

You don't do make in the perl directory... You cd .. && 'make permod'  
from the top level dir.

/b

On Aug 8, 2009, at 12:28 AM, velusamy velu wrote:

> Dear all,
>    When I do make for Perl ESL libraries in "esl/perl" directory I  
> have got the following error.
>
> /usr/lib/gcc/i486-linux-gnu/4.1.2/../../../../lib/crt1.o: In  
> function `_start':
> ../sysdeps/i386/elf/start.S:115: undefined reference to `main'
> esl_wrap.o: In function `_wrap_eslSetLogLevel':
> esl_wrap.cpp:(.text+0x1c10): undefined reference to `eslSetLogLevel'
> esl_wrap.o: In function `_wrap_ESLconnection_setEventLock':
> esl_wrap.cpp:(.text+0x3395): undefined reference to  
> `ESLconnection::setEventLock(char const*)'
> esl_wrap.o: In function `_wrap_ESLconnection_setBlockingExecute':
> esl_wrap.cpp:(.text+0x3c65): undefined reference to  
> `ESLconnection::setBlockingExecute(char const*)'
> esl_wrap.o: In function `_wrap_ESLconnection_execute':
> esl_wrap.cpp:(.text+0x41ab): undefined reference to  
> `ESLconnection::execute(char const*, char const*, char const*)'
> ........
>
> I have understood the when creating soft link ESL.o that error has  
> occurred.
>
> What is the problem?
> Is there any dependency to create that link?
>
> please help me.......
>
> Thanks
>
> Regards,
> Velusamy
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From alan at chandlerfamily.org.uk  Sat Aug  8 07:31:02 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Sat, 08 Aug 2009 15:31:02 +0100
Subject: [Freeswitch-users] New to Freeswitch - some help needed
In-Reply-To: <33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
Message-ID: <4A7D8C26.4020408@chandlerfamily.org.uk>

Mark Campbell-Smith wrote:
> Hi Alan,
> 
> I hope you find your answers here as these are the sort of things that
> are hard to find on the wiki, which is somewhat outdated in areas.  If
> you do find your answers, please post them back here for everyone
> else.
> 
> I am new to FS also, so my comments below may not be 100% correct!
> 
> 1. Very similar to what I want to have setup as well.  Do you have a
> static IP address at home.  If not, get a dyndns account and setup an
> entry there so that your friends/family can register using your dns
> name instead of ip address

Thanks for the idea - but I am way ahead on this.

My IP address is allocated by dhcp, but since my router stays switched 
on 24/7, and even powering it off for short periods doesn't change 
things. It has not changed for maybe a year.

So I own my own domain name (chandlerfamily.org.uk) and can point 
home.chandlerfamily.org.uk at my ip address. If it ever changes, I can 
go to my domain name provider and change my dns entry very easily.

You can in fact tell freeswitch to ignore the domain names used by the 
clients with the two parameters in the internal sip profile

     <!--all inbound reg will look in this domain for the users -->
     <param name="force-register-domain" value="$${domain}"/>
     <!--all inbound reg will stored in the db using this domain -->
     <param name="force-register-db-domain" value="$${domain}"/>

(This is from the new 1.0.4 version of freeswitch)

After a discussion on IRC I decided to set $${domain} to 
chandlerfamily.org.uk so theoretically all my phones have addresses 
extn-no at chandlerfamily.org.uk

And in the pap2t box I am using tell it to use chandlerfamily.org.uk as 
its "proxy" and home.chandlerfamily.org.uk as its "outgoing proxy" - for 
phones outside the NAT box.  Inside my home I use the internal name of 
the freeswitch box as the "outgoing proxy".


> 
> 2. No idea.  Maybe try another stun server?

I still don't have any answers to this, but it doesn't appear to be 
important

> 
> 3. Not sure if double-NAT is needed now with the newer builds of
> FreeSwitch.  Download the latest 1.0.4 to be on the safeside and
> compile it again!  (I have FS 1.0.4 pre9 and it works I think).  As
> long as your clients can register remotely you should be okay. I think
> FS can work around most home NATs.  Make sure you have auto-nat set in
> your internal.xml file (I think its this one)
>

I have 1.0.4 installed as of yesterday, and it appears to be working. 
My daughter still has to change the user id's inside her PAP2T box to 
numbers as I could not make the "number-alias" function work - so 
although the sip part appears to work, I have still to find out if the 
voice part does.


> 4. SIP is the signaling.  RTP is the payload, or voice in your case.
> Any transition is done via the SIP signaling.  This is how FS can
> transfer calls etc or use the media bypass mode by specifying the IP
> address where the RTP should be sent, which does not have to be the
> same as the signaling.  Make sure you enable tracing in the
> internal.xml file so you can debug the signaling.
> 
> You don't need to take a laptop to your daughters to test this.  Use
> an internet sip phone like flaphone.com, which works through your web
> browser.  This will register with an external IP address exactly like
> your daughters and save you time traveling.  Note that sound isn't so
> clear for me using this service, but it helps with debugging.
> 
> I also would recommend a sip client on windows like Zoiper, or
> CounterPath's X-Lite.. both are free.  X-Lite is well known, Zoiper
> allows for multiple SIP registrations and comes in a portable version.
> 

I used zoiper before as an iax client - so I'll look again at this


-- 
Alan Chandler
http://www.chandlerfamily.org.uk



From chad at apartmentlines.com  Sat Aug  8 08:15:18 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Sat, 8 Aug 2009 11:15:18 -0400
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase macro is
	playing
In-Reply-To: <4A7D8C26.4020408@chandlerfamily.org.uk>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
Message-ID: <BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>

in a lua script, i've tried using session:setInputCallback() to catch  
DTMF tones while a phrase macro is playing, but it doesn't seem to  
work.  the same callback _does_ catch DTMF when i use  
session:streamFile() to play something.  is there some other way to do  
it?

below is an example of how i'm doing it now.  when voicemail/vm- 
mailbox_full.wav plays, i see key presses being printed to the  
console, but when the phrase plays, no key presses are displayed.

function key_press(session, input_type, data)
	if input_type == "dtmf" then
		freeswitch.consoleLog("info", "Key pressed: " .. data["digit"] ..  
"\n")
	end
end

if session:ready() then
	session:answer()
	session:execute("sleep", "1000")
	session:setInputCallback("key_press", "")
	session:streamFile("voicemail/vm-mailbox_full.wav")
	session:execute("phrase", "voicemail_menu,1:2:3:#")
end


From brian at freeswitch.org  Sat Aug  8 08:59:42 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 8 Aug 2009 10:59:42 -0500
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
Message-ID: <96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>

Examples exist in the scripts/lua folder called callback.lua.  Also  
please DO NOT hijack threads.  You can't click reply, change the  
subject as it messes up the mailing list threading and most mail  
readers will thread it incorrectly.

Thanks,
Brian

On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines wrote:

> in a lua script, i've tried using session:setInputCallback() to catch
> DTMF tones while a phrase macro is playing, but it doesn't seem to
> work.  the same callback _does_ catch DTMF when i use
> session:streamFile() to play something.  is there some other way to do
> it?
>
> below is an example of how i'm doing it now.  when voicemail/vm-
> mailbox_full.wav plays, i see key presses being printed to the
> console, but when the phrase plays, no key presses are displayed.
>
> function key_press(session, input_type, data)
> 	if input_type == "dtmf" then
> 		freeswitch.consoleLog("info", "Key pressed: " .. data["digit"] ..
> "\n")
> 	end
> end
>
> if session:ready() then
> 	session:answer()
> 	session:execute("sleep", "1000")
> 	session:setInputCallback("key_press", "")
> 	session:streamFile("voicemail/vm-mailbox_full.wav")
> 	session:execute("phrase", "voicemail_menu,1:2:3:#")
> end
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From vladrodionov at gmail.com  Sat Aug  8 09:26:05 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Sat, 8 Aug 2009 09:26:05 -0700
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
Message-ID: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>

Good morning,
This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
AS does some logic and transfers call (or forward) out of Voip provider
network to another PSTN number User2.


This is call bridge


    UA1     (PSTN) -                                 ->  UA2 (PSTN)
              -                                            -
                -  (1)                                   -  (4)
                  ->         PSTN Gateway->
                            -                        -
                       (2) -                        - (3)
                          -> FreeSWITCH ->


This is what I want to acomplish
                                    (4)
    UA1     (PSTN) ------------------------------- ->  UA2 (PSTN)
                    -
                       -   (1)
                        ->  PSTN Gateway->
                             -                        -
                       (2)  -                        - (3)
                            -> FreeSWITCH ->


1. Can it be implemented in FreeSWITCH?
2. Does anybody know Voip providers which support out of network call
transfer/forwarding to PSTN?

TIA

-Vladimir Rodionov
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From raffaele.p.guidi at gmail.com  Sat Aug  8 09:50:57 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sat, 8 Aug 2009 18:50:57 +0200
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
Message-ID: <b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>

Brian, I've read 8-9 of your emails asking to not hijack threads in threads
where this it didn't happen (like this). I see this as a new thread with a
single message (plus yours, of course) inside with "[Freeswitch-users] how
to catch DTMF in Lua while a phrase macro is playing" as the subject - and I
think is quite relevant to the message. I suspect you have something wrong
in your email client (mine is gmail) or server and suggest you to give it a
check.

Regards,
   Raffaele

On Sat, Aug 8, 2009 at 17:59, Brian West <brian at freeswitch.org> wrote:

> Examples exist in the scripts/lua folder called callback.lua.  Also
> please DO NOT hijack threads.  You can't click reply, change the
> subject as it messes up the mailing list threading and most mail
> readers will thread it incorrectly.
>
> Thanks,
> Brian
>
> On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines wrote:
>
> > in a lua script, i've tried using session:setInputCallback() to catch
> > DTMF tones while a phrase macro is playing, but it doesn't seem to
> > work.  the same callback _does_ catch DTMF when i use
> > session:streamFile() to play something.  is there some other way to do
> > it?
> >
> > below is an example of how i'm doing it now.  when voicemail/vm-
> > mailbox_full.wav plays, i see key presses being printed to the
> > console, but when the phrase plays, no key presses are displayed.
> >
> > function key_press(session, input_type, data)
> >       if input_type == "dtmf" then
> >               freeswitch.consoleLog("info", "Key pressed: " ..
> data["digit"] ..
> > "\n")
> >       end
> > end
> >
> > if session:ready() then
> >       session:answer()
> >       session:execute("sleep", "1000")
> >       session:setInputCallback("key_press", "")
> >       session:streamFile("voicemail/vm-mailbox_full.wav")
> >       session:execute("phrase", "voicemail_menu,1:2:3:#")
> > end
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From raffaele.p.guidi at gmail.com  Sat Aug  8 09:55:27 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sat, 8 Aug 2009 18:55:27 +0200
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
	<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
Message-ID: <b0a8792f0908080955k150badbcmadf154283c3d2238@mail.gmail.com>

A cut&paste of a gmail search for "label:voip hijack". The subjects seem to
be ok

Chad, Brian, Raffaele (3)
liste.voip

[Freeswitch-users] how to catch DTMF in Lua while a phrase macro is
playing? - ?
please DO NOT *hijack* threads. You can't click reply, change the subject as
it messes up the ?
 18:50Greg .. Brian, Michael (6)
liste.voip

[Freeswitch-users] phpmod compile error? - ? Please don't *hijack* threads...
please click new message and start a new thread. Thanks ?
 Aug 5mashudi, Brian (2)
liste.voip

[Freeswitch-users] video playback on FS? - ? btw don't *hijack* threads
please. thank you in advanced, best regard, mashudi Brian West
brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ?
 Jun 23Tim .. Michael, Chris (4)
liste.voip

[Freeswitch-users] Routing 911 calls? - ? p.s. Sorry to *hijack* the thread.
--- On Mon, 6/8/09, Michael Collins <msc at freeswitch.org> wrote: From:
Michael Collins <msc at freeswitch.org> Subject: Re ?
 Jun 8Lars .. Brian, Michael (13)
liste.voip

[Freeswitch-users] Total noob question? - ? please do not *hijack* threads..
Click new message and input the address
freeswitch-users at lists.freeswitch.org Thanks, Brian On May 7, 2009, at 8:13
PM, Lars Zeb wrote: I ?
 May 8wchao
liste.voip

[Freeswitch-users] Registration problem with multiple IP phones behind Linux
NAT firewa...? - ? is trying to *hijack* phone A's registration. The packet
capture was taken when running Freeswitch 1.0.4pre4, but I subsequently
upgraded to 1.0.4pre6 and it didn't ?
 May 4Chris, Brian, Anthony (3)
liste.voip

[Freeswitch-users] Audio delay when conferencing? - ? Also don't
*hijack* threads
please! :) /b On Apr 30, 2009, at 3:54 PM, Chris Fowler wrote: I'm using
FreeSWITCH (Build 13168M) and we're having intermittent ?
 Apr 30Chris, Brian (2)
liste.voip

[Freeswitch-users] mod_fifo uuid_transfer into mod_conference audio issue? - ?
PLEASE do not *hijack* threads. You clicked reply.. cleared the body and the
subject and sent the message. DO NOT DO THAT. Please click new message and
input freeswitch ?
 Apr 15Valentin, Brian, Anthony (3)
liste.voip

[Freeswitch-users] Nokia N800? - ? please do not *hijack* threads... you
clicked reply, changed the subject and body which causes it to thread your
message with the original posters thread. So please in the ?
 Apr 1
ConversationsSelect: All, None, Read, Unread, Starred, Unstarred

Regards,
   Raffaele


On Sat, Aug 8, 2009 at 18:50, Raffaele P. Guidi
<raffaele.p.guidi at gmail.com>wrote:

> Brian, I've read 8-9 of your emails asking to not hijack threads in threads
> where this it didn't happen (like this). I see this as a new thread with a
> single message (plus yours, of course) inside with "[Freeswitch-users] how
> to catch DTMF in Lua while a phrase macro is playing" as the subject - and I
> think is quite relevant to the message. I suspect you have something wrong
> in your email client (mine is gmail) or server and suggest you to give it a
> check.
>
> Regards,
>    Raffaele
>
> On Sat, Aug 8, 2009 at 17:59, Brian West <brian at freeswitch.org> wrote:
>
>> Examples exist in the scripts/lua folder called callback.lua.  Also
>> please DO NOT hijack threads.  You can't click reply, change the
>> subject as it messes up the mailing list threading and most mail
>> readers will thread it incorrectly.
>>
>> Thanks,
>> Brian
>>
>> On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines wrote:
>>
>> > in a lua script, i've tried using session:setInputCallback() to catch
>> > DTMF tones while a phrase macro is playing, but it doesn't seem to
>> > work.  the same callback _does_ catch DTMF when i use
>> > session:streamFile() to play something.  is there some other way to do
>> > it?
>> >
>> > below is an example of how i'm doing it now.  when voicemail/vm-
>> > mailbox_full.wav plays, i see key presses being printed to the
>> > console, but when the phrase plays, no key presses are displayed.
>> >
>> > function key_press(session, input_type, data)
>> >       if input_type == "dtmf" then
>> >               freeswitch.consoleLog("info", "Key pressed: " ..
>> data["digit"] ..
>> > "\n")
>> >       end
>> > end
>> >
>> > if session:ready() then
>> >       session:answer()
>> >       session:execute("sleep", "1000")
>> >       session:setInputCallback("key_press", "")
>> >       session:streamFile("voicemail/vm-mailbox_full.wav")
>> >       session:execute("phrase", "voicemail_menu,1:2:3:#")
>> > end
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From chad at apartmentlines.com  Sat Aug  8 10:16:19 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Sat, 8 Aug 2009 13:16:19 -0400
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
Message-ID: <566B33BA-4566-4544-9529-B577D26F6C87@apartmentlines.com>


On Aug 8, 2009, at 11:59 AM, Brian West wrote:

> Examples exist in the scripts/lua folder called callback.lua.

i read this example code, but it doesn't seem to really address my  
question.  if you re-read the code and explanation that i posted  
earlier, the input callback is working when i do something like  
session:streamFile("blah"), but *not* when i do something like  
session:execute("phrase", "blah,args")).  to be clear, the phrase  
plays just fine, but when i press keys during the phrase playback, the  
input callback doesn't get fired.  am i missing something obvious  
here?  you should be able to drop that code in that i provided earlier  
to verify what i'm talking about.

one thing i will add is that i'm using portaudio for making the call  
and sending the DTMF.  i'll try testing this out with a regular sip  
client and see if the same thing occurs...

> Also
> please DO NOT hijack threads.  You can't click reply, change the
> subject as it messes up the mailing list threading and most mail
> readers will thread it incorrectly.

sorry, i've never had that problem before -- i can certainly adjust  :)


From rlucente at teleuco.com  Sat Aug  8 10:25:51 2009
From: rlucente at teleuco.com (Rocco Lucente)
Date: Sat, 8 Aug 2009 19:25:51 +0200
Subject: [Freeswitch-users] question on att_xfer
Message-ID: <2969faac0908081025l41d33ed5r295b86635ca46853@mail.gmail.com>

hi, during a call made with att_xfer, you can resume the user A on ring? Let
me explain better, with bind_meta_app and att_xfer can do this: during a
bridge if B dials *1 (for example), A is parked and B can call C. After C to
answer, B can to dial # and return in conversation with A. B could resume A
while is calling C (for example because I noticed that I've dialed an
incorrect number)?
I tried with execute_on_ring, but without result. Any ideas?
Thank you and good job to all.
-- 
Rocco Lucente
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From pjintheusa at gmail.com  Sat Aug  8 10:40:33 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 10:40:33 -0700
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
In-Reply-To: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
References: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
Message-ID: <367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>

Hi there,

Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
INVITE) and only pass back the media to the network, or pass back
signaling also (SIP REFER)?

I know several suppliers who support SIP re INVITE but none that
support SIP REFER.

Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
and http://wiki.freeswitch.org/wiki/Bypass_Media



On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov at gmail.com> wrote:
> Good morning,
> This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
> through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
> AS does some logic and transfers call (or forward) out of Voip provider
> network to another PSTN number User2.
>
>
> This is call bridge
>
>
> ??? UA1???? (PSTN) -???????????????????????????????? ->? UA2 (PSTN)
> ????????????? -??????????????????????????????????????????? -
> ??????????????? -? (1)?????????????????????????????????? -? (4)
> ????????????????? ->???????? PSTN Gateway->
> ??????????????????????????? -??????????????????????? -
> ?????????????????????? (2) -??????????????????????? - (3)
> ????????????????????????? -> FreeSWITCH ->
>
>
> This is what I want to acomplish
> ??????????????????????????????????? (4)
> ??? UA1???? (PSTN) ------------------------------- ->? UA2 (PSTN)
> ??????????????????? -
> ???????????????????? ? - ? (1)
> ??????????????????????? ->? PSTN Gateway->
> ???????????????????????????? -??????????????????????? -
> ?????????????????????? (2)? -??????????????????????? - (3)
> ??????????????????????????? -> FreeSWITCH ->
>
>
> 1. Can it be implemented in FreeSWITCH?
> 2. Does anybody know Voip providers which support out of network call
> transfer/forwarding to PSTN?
>
> TIA
>
> -Vladimir Rodionov
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

From pjintheusa at gmail.com  Sat Aug  8 10:44:46 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 10:44:46 -0700
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
	<b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
	<b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>
Message-ID: <367751820908081044g67a85684hc34a0b242c67b6a4@mail.gmail.com>

Thanks very much for that - very help.

Why would loopback be considered "abuse"? What would be the downsides
of doing this?

On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker<rupa at rupa.com> wrote:
>
>
> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>> <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>
>
> That of course, should be:
>
> ?<action application="bridge" data="sofia/gw1/$1|sofia/gw2/$1..."/>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From stkn at freeswitch.org  Sat Aug  8 11:01:50 2009
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Sat, 08 Aug 2009 20:01:50 +0200
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
 macro is playing
In-Reply-To: <b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>	<4A7D8C26.4020408@chandlerfamily.org.uk>	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
	<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
Message-ID: <4A7DBD8E.4090608@freeswitch.org>

His message was a direct reply to Alan Chandler's mail and Brian's mail client is working perfectly fine
(as are both of my mail clients).

Look at the Message-Id and In-Reply-To headers of the mails if you still think our clients are broken :)


stkn


Raffaele P. Guidi wrote:
> Brian, I've read 8-9 of your emails asking to not hijack threads in
> threads where this it didn't happen (like this). I see this as a new
> thread with a single message (plus yours, of course) inside with
> "[Freeswitch-users] how to catch DTMF in Lua while a phrase macro is
> playing" as the subject - and I think is quite relevant to the message.
> I suspect you have something wrong in your email client (mine is gmail)
> or server and suggest you to give it a check.
> 
> Regards,
>    Raffaele
> 
> On Sat, Aug 8, 2009 at 17:59, Brian West <brian at freeswitch.org
> <mailto:brian at freeswitch.org>> wrote:
> 
>     Examples exist in the scripts/lua folder called callback.lua.  Also
>     please DO NOT hijack threads.  You can't click reply, change the
>     subject as it messes up the mailing list threading and most mail
>     readers will thread it incorrectly.
> 
>     Thanks,
>     Brian
> 
>     On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines wrote:
> 
>     > in a lua script, i've tried using session:setInputCallback() to catch
>     > DTMF tones while a phrase macro is playing, but it doesn't seem to
>     > work.  the same callback _does_ catch DTMF when i use
>     > session:streamFile() to play something.  is there some other way to do
>     > it?
>     >
>     > below is an example of how i'm doing it now.  when voicemail/vm-
>     > mailbox_full.wav plays, i see key presses being printed to the
>     > console, but when the phrase plays, no key presses are displayed.
>     >
>     > function key_press(session, input_type, data)
>     >       if input_type == "dtmf" then
>     >               freeswitch.consoleLog("info", "Key pressed: " ..
>     data["digit"] ..
>     > "\n")
>     >       end
>     > end
>     >
>     > if session:ready() then
>     >       session:answer()
>     >       session:execute("sleep", "1000")
>     >       session:setInputCallback("key_press", "")
>     >       session:streamFile("voicemail/vm-mailbox_full.wav")
>     >       session:execute("phrase", "voicemail_menu,1:2:3:#")
>     > end
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From william.suffill at gmail.com  Sat Aug  8 11:34:23 2009
From: william.suffill at gmail.com (William Suffill)
Date: Sat, 8 Aug 2009 14:34:23 -0400
Subject: [Freeswitch-users] mod_php needed
In-Reply-To: <191c3a030906150827t2e2e919bw396097bd637a0b91@mail.gmail.com>
References: <4A33E8A0.1070708@xpirio.com>
	<87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> 
	<A2B4F48489739E439D8845296A2FD9C4147238@sun.Noble.local>
	<4A361708.8020808@xpirio.com> 
	<191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> 
	<4A3660EB.4070203@xpirio.com>
	<191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com> 
	<6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com> 
	<191c3a030906150827t2e2e919bw396097bd637a0b91@mail.gmail.com>
Message-ID: <6b65470d0908081134x719bbef6r707644d877eee328@mail.gmail.com>

http://wiki.freeswitch.org/wiki/PHP_ESL

Formatting could use a bit of help just exported my working document
to MediaWiki. Hope to add some more information to ESL as I continue
to do more with it.

I'm going to start a page on fs_ivrd as well. Feedback welcome.

-- W

IRC: wsuff on freenode


From vladrodionov at gmail.com  Sat Aug  8 11:39:13 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Sat, 8 Aug 2009 11:39:13 -0700
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
In-Reply-To: <367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>
References: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
	<367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>
Message-ID: <3c233920908081139o769c1310yaee2857d4543da7d@mail.gmail.com>

Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty
sure it is doable because voxeo offers this
option for their Voice XML customers but I am not interested in a hosted
solution at the time - it is quite expensive. As far as I understood, Voip
provider MUST have pstn call transfer feature enabled by telecom provider
(AT&T for example) and this should work fine with SIP.

-Vladimir Rodionov


On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi there,
>
> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
> INVITE) and only pass back the media to the network, or pass back
> signaling also (SIP REFER)?
>
> I know several suppliers who support SIP re INVITE but none that
> support SIP REFER.
>
> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
> and http://wiki.freeswitch.org/wiki/Bypass_Media
>
>
>
> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov at gmail.com>
> wrote:
> > Good morning,
> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
> > through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
> > AS does some logic and transfers call (or forward) out of Voip provider
> > network to another PSTN number User2.
> >
> >
> > This is call bridge
> >
> >
> >     UA1     (PSTN) -                                 ->  UA2 (PSTN)
> >               -                                            -
> >                 -  (1)                                   -  (4)
> >                   ->         PSTN Gateway->
> >                             -                        -
> >                        (2) -                        - (3)
> >                           -> FreeSWITCH ->
> >
> >
> > This is what I want to acomplish
> >                                     (4)
> >     UA1     (PSTN) ------------------------------- ->  UA2 (PSTN)
> >                     -
> >                        -   (1)
> >                         ->  PSTN Gateway->
> >                              -                        -
> >                        (2)  -                        - (3)
> >                             -> FreeSWITCH ->
> >
> >
> > 1. Can it be implemented in FreeSWITCH?
> > 2. Does anybody know Voip providers which support out of network call
> > transfer/forwarding to PSTN?
> >
> > TIA
> >
> > -Vladimir Rodionov
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From raffaele.p.guidi at gmail.com  Sat Aug  8 11:48:28 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sat, 8 Aug 2009 20:48:28 +0200
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <4A7DBD8E.4090608@freeswitch.org>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
	<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
	<4A7DBD8E.4090608@freeswitch.org>
Message-ID: <b0a8792f0908081148l2174443fhcd210e66aec15bf@mail.gmail.com>

Uh, oh, ok. Message-Id and In-Reply-To headers don't mean too much in a
gmail world (I think gmail does a better job than any other mail client).
Well, at least I understood what Brian was talking about, thanks.
Regards,
   Raffaele

On Sat, Aug 8, 2009 at 20:01, Stefan Knoblich <stkn at freeswitch.org> wrote:

> His message was a direct reply to Alan Chandler's mail and Brian's mail
> client is working perfectly fine
> (as are both of my mail clients).
>
> Look at the Message-Id and In-Reply-To headers of the mails if you still
> think our clients are broken :)
>
>
> stkn
>
>
> Raffaele P. Guidi wrote:
> > Brian, I've read 8-9 of your emails asking to not hijack threads in
> > threads where this it didn't happen (like this). I see this as a new
> > thread with a single message (plus yours, of course) inside with
> > "[Freeswitch-users] how to catch DTMF in Lua while a phrase macro is
> > playing" as the subject - and I think is quite relevant to the message.
> > I suspect you have something wrong in your email client (mine is gmail)
> > or server and suggest you to give it a check.
> >
> > Regards,
> >    Raffaele
> >
> > On Sat, Aug 8, 2009 at 17:59, Brian West <brian at freeswitch.org
> > <mailto:brian at freeswitch.org>> wrote:
> >
> >     Examples exist in the scripts/lua folder called callback.lua.  Also
> >     please DO NOT hijack threads.  You can't click reply, change the
> >     subject as it messes up the mailing list threading and most mail
> >     readers will thread it incorrectly.
> >
> >     Thanks,
> >     Brian
> >
> >     On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines wrote:
> >
> >     > in a lua script, i've tried using session:setInputCallback() to
> catch
> >     > DTMF tones while a phrase macro is playing, but it doesn't seem to
> >     > work.  the same callback _does_ catch DTMF when i use
> >     > session:streamFile() to play something.  is there some other way to
> do
> >     > it?
> >     >
> >     > below is an example of how i'm doing it now.  when voicemail/vm-
> >     > mailbox_full.wav plays, i see key presses being printed to the
> >     > console, but when the phrase plays, no key presses are displayed.
> >     >
> >     > function key_press(session, input_type, data)
> >     >       if input_type == "dtmf" then
> >     >               freeswitch.consoleLog("info", "Key pressed: " ..
> >     data["digit"] ..
> >     > "\n")
> >     >       end
> >     > end
> >     >
> >     > if session:ready() then
> >     >       session:answer()
> >     >       session:execute("sleep", "1000")
> >     >       session:setInputCallback("key_press", "")
> >     >       session:streamFile("voicemail/vm-mailbox_full.wav")
> >     >       session:execute("phrase", "voicemail_menu,1:2:3:#")
> >     > end
> >     >
> >     > _______________________________________________
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vladrodionov at gmail.com  Sat Aug  8 11:52:09 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Sat, 8 Aug 2009 11:52:09 -0700
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
In-Reply-To: <3c233920908081139o769c1310yaee2857d4543da7d@mail.gmail.com>
References: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
	<367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>
	<3c233920908081139o769c1310yaee2857d4543da7d@mail.gmail.com>
Message-ID: <3c233920908081152r70acceeoe63cf15f8e1a40dd@mail.gmail.com>

Actually, this is what I need

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect

Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN
number?

-Vladimir Rodionov


On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov
<vladrodionov at gmail.com>wrote:

> Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty
> sure it is doable because voxeo offers this
> option for their Voice XML customers but I am not interested in a hosted
> solution at the time - it is quite expensive. As far as I understood, Voip
> provider MUST have pstn call transfer feature enabled by telecom provider
> (AT&T for example) and this should work fine with SIP.
>
> -Vladimir Rodionov
>
>
> On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> Hi there,
>>
>> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
>> INVITE) and only pass back the media to the network, or pass back
>> signaling also (SIP REFER)?
>>
>> I know several suppliers who support SIP re INVITE but none that
>> support SIP REFER.
>>
>> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
>> and http://wiki.freeswitch.org/wiki/Bypass_Media
>>
>>
>>
>> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov at gmail.com>
>> wrote:
>> > Good morning,
>> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
>> > through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
>> > AS does some logic and transfers call (or forward) out of Voip provider
>> > network to another PSTN number User2.
>> >
>> >
>> > This is call bridge
>> >
>> >
>> >     UA1     (PSTN) -                                 ->  UA2 (PSTN)
>> >               -                                            -
>> >                 -  (1)                                   -  (4)
>> >                   ->         PSTN Gateway->
>> >                             -                        -
>> >                        (2) -                        - (3)
>> >                           -> FreeSWITCH ->
>> >
>> >
>> > This is what I want to acomplish
>> >                                     (4)
>> >     UA1     (PSTN) ------------------------------- ->  UA2 (PSTN)
>> >                     -
>> >                        -   (1)
>> >                         ->  PSTN Gateway->
>> >                              -                        -
>> >                        (2)  -                        - (3)
>> >                             -> FreeSWITCH ->
>> >
>> >
>> > 1. Can it be implemented in FreeSWITCH?
>> > 2. Does anybody know Voip providers which support out of network call
>> > transfer/forwarding to PSTN?
>> >
>> > TIA
>> >
>> > -Vladimir Rodionov
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From pjintheusa at gmail.com  Sat Aug  8 12:00:39 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 12:00:39 -0700
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
In-Reply-To: <3c233920908081152r70acceeoe63cf15f8e1a40dd@mail.gmail.com>
References: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
	<367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>
	<3c233920908081139o769c1310yaee2857d4543da7d@mail.gmail.com>
	<3c233920908081152r70acceeoe63cf15f8e1a40dd@mail.gmail.com>
Message-ID: <367751820908081200y1ea230c7o12377ffe45bc75eb@mail.gmail.com>

Are your calls coming in on TDM or SIP trunks? Are your calls answered
by FreeSWITCH before you need to redirect them?

On Sat, Aug 8, 2009 at 11:52 AM, Vladimir
Rodionov<vladrodionov at gmail.com> wrote:
> Actually, this is what I need
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
>
> Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN
> number?
>
> -Vladimir Rodionov
>
>
> On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <vladrodionov at gmail.com>
> wrote:
>>
>> Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty
>> sure it is doable because voxeo offers this
>> option for their Voice XML customers but I am not interested in a hosted
>> solution at the time - it is quite expensive. As far as I understood, Voip
>> provider MUST have pstn call transfer feature enabled by telecom provider
>> (AT&T for example) and this should work fine with SIP.
>>
>> -Vladimir Rodionov
>>
>>
>> On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa at gmail.com>
>> wrote:
>>>
>>> Hi there,
>>>
>>> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
>>> INVITE) and only pass back the media to the network, or pass back
>>> signaling also (SIP REFER)?
>>>
>>> I know several suppliers who support SIP re INVITE but none that
>>> support SIP REFER.
>>>
>>> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
>>> and http://wiki.freeswitch.org/wiki/Bypass_Media
>>>
>>>
>>>
>>> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov at gmail.com>
>>> wrote:
>>> > Good morning,
>>> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
>>> > through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
>>> > AS does some logic and transfers call (or forward) out of Voip provider
>>> > network to another PSTN number User2.
>>> >
>>> >
>>> > This is call bridge
>>> >
>>> >
>>> > ??? UA1???? (PSTN) -???????????????????????????????? ->? UA2 (PSTN)
>>> > ????????????? -??????????????????????????????????????????? -
>>> > ??????????????? -? (1)?????????????????????????????????? -? (4)
>>> > ????????????????? ->???????? PSTN Gateway->
>>> > ??????????????????????????? -??????????????????????? -
>>> > ?????????????????????? (2) -??????????????????????? - (3)
>>> > ????????????????????????? -> FreeSWITCH ->
>>> >
>>> >
>>> > This is what I want to acomplish
>>> > ??????????????????????????????????? (4)
>>> > ??? UA1???? (PSTN) ------------------------------- ->? UA2 (PSTN)
>>> > ??????????????????? -
>>> > ???????????????????? ? - ? (1)
>>> > ??????????????????????? ->? PSTN Gateway->
>>> > ???????????????????????????? -??????????????????????? -
>>> > ?????????????????????? (2)? -??????????????????????? - (3)
>>> > ??????????????????????????? -> FreeSWITCH ->
>>> >
>>> >
>>> > 1. Can it be implemented in FreeSWITCH?
>>> > 2. Does anybody know Voip providers which support out of network call
>>> > transfer/forwarding to PSTN?
>>> >
>>> > TIA
>>> >
>>> > -Vladimir Rodionov
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From diego.viola at gmail.com  Sat Aug  8 12:06:57 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 8 Aug 2009 15:06:57 -0400
Subject: [Freeswitch-users] mod_php needed
In-Reply-To: <6b65470d0908081134x719bbef6r707644d877eee328@mail.gmail.com>
References: <4A33E8A0.1070708@xpirio.com>
	<87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147238@sun.Noble.local>
	<4A361708.8020808@xpirio.com>
	<191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com>
	<4A3660EB.4070203@xpirio.com>
	<191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com>
	<6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com>
	<191c3a030906150827t2e2e919bw396097bd637a0b91@mail.gmail.com>
	<6b65470d0908081134x719bbef6r707644d877eee328@mail.gmail.com>
Message-ID: <86a32abc0908081206l6932b87bo5b05098c4d2ffe5a@mail.gmail.com>

Damn, looks like we need to learn some formatting =D.

I made some changes on that article to fix the formatting a bit, lets take
care of the wiki as we care about the software :).

On Sat, Aug 8, 2009 at 2:34 PM, William Suffill
<william.suffill at gmail.com>wrote:

> http://wiki.freeswitch.org/wiki/PHP_ESL
>
> Formatting could use a bit of help just exported my working document
> to MediaWiki. Hope to add some more information to ESL as I continue
> to do more with it.
>
> I'm going to start a page on fs_ivrd as well. Feedback welcome.
>
> -- W
>
> IRC: wsuff on freenode
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From chad at apartmentlines.com  Sat Aug  8 12:09:48 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Sat, 8 Aug 2009 15:09:48 -0400
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <b0a8792f0908081148l2174443fhcd210e66aec15bf@mail.gmail.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
	<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
	<4A7DBD8E.4090608@freeswitch.org>
	<b0a8792f0908081148l2174443fhcd210e66aec15bf@mail.gmail.com>
Message-ID: <98727038-1A37-43A4-BFC4-F624451772AF@apartmentlines.com>

does anybody else find it amusing that my thread about Lua scripting  
was hijacked by people talking about hijacking threads??  ;)

/me waits patiently for discussion on the topic


On Aug 8, 2009, at 2:48 PM, Raffaele P. Guidi wrote:

> Uh, oh, ok. Message-Id and In-Reply-To headers don't mean too much  
> in a gmail world (I think gmail does a better job than any other  
> mail client). Well, at least I understood what Brian was talking  
> about, thanks.
>
> Regards,
>    Raffaele
>
> On Sat, Aug 8, 2009 at 20:01, Stefan Knoblich <stkn at freeswitch.org>  
> wrote:
> His message was a direct reply to Alan Chandler's mail and Brian's  
> mail client is working perfectly fine
> (as are both of my mail clients).
>
> Look at the Message-Id and In-Reply-To headers of the mails if you  
> still think our clients are broken :)
>
>
> stkn
>
>
> Raffaele P. Guidi wrote:
> > Brian, I've read 8-9 of your emails asking to not hijack threads in
> > threads where this it didn't happen (like this). I see this as a new
> > thread with a single message (plus yours, of course) inside with
> > "[Freeswitch-users] how to catch DTMF in Lua while a phrase macro is
> > playing" as the subject - and I think is quite relevant to the  
> message.
> > I suspect you have something wrong in your email client (mine is  
> gmail)
> > or server and suggest you to give it a check.
> >
> > Regards,
> >    Raffaele
> >
> > On Sat, Aug 8, 2009 at 17:59, Brian West <brian at freeswitch.org
> > <mailto:brian at freeswitch.org>> wrote:
> >
> >     Examples exist in the scripts/lua folder called callback.lua.   
> Also
> >     please DO NOT hijack threads.  You can't click reply, change the
> >     subject as it messes up the mailing list threading and most mail
> >     readers will thread it incorrectly.
> >
> >     Thanks,
> >     Brian
> >
> >     On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines  
> wrote:
> >
> >     > in a lua script, i've tried using session:setInputCallback()  
> to catch
> >     > DTMF tones while a phrase macro is playing, but it doesn't  
> seem to
> >     > work.  the same callback _does_ catch DTMF when i use
> >     > session:streamFile() to play something.  is there some other  
> way to do
> >     > it?
> >     >
> >     > below is an example of how i'm doing it now.  when voicemail/ 
> vm-
> >     > mailbox_full.wav plays, i see key presses being printed to the
> >     > console, but when the phrase plays, no key presses are  
> displayed.
> >     >
> >     > function key_press(session, input_type, data)
> >     >       if input_type == "dtmf" then
> >     >               freeswitch.consoleLog("info", "Key pressed: " ..
> >     data["digit"] ..
> >     > "\n")
> >     >       end
> >     > end
> >     >
> >     > if session:ready() then
> >     >       session:answer()
> >     >       session:execute("sleep", "1000")
> >     >       session:setInputCallback("key_press", "")
> >     >       session:streamFile("voicemail/vm-mailbox_full.wav")
> >     >       session:execute("phrase", "voicemail_menu,1:2:3:#")
> >     > end
> >     >
> >     > _______________________________________________
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >  
> ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vladrodionov at gmail.com  Sat Aug  8 12:12:13 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Sat, 8 Aug 2009 12:12:13 -0700
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
In-Reply-To: <367751820908081200y1ea230c7o12377ffe45bc75eb@mail.gmail.com>
References: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
	<367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>
	<3c233920908081139o769c1310yaee2857d4543da7d@mail.gmail.com>
	<3c233920908081152r70acceeoe63cf15f8e1a40dd@mail.gmail.com>
	<367751820908081200y1ea230c7o12377ffe45bc75eb@mail.gmail.com>
Message-ID: <3c233920908081212s29f6d42ft210755fe50282cba@mail.gmail.com>

A call is coming on SIP trunk. From PSTN.  I does not need to be answered,
actually - I need to do some logic before redirecting call but I can answer
call as well It won't break the app logic.

-Vladimir Rodionov

On Sat, Aug 8, 2009 at 12:00 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Are your calls coming in on TDM or SIP trunks? Are your calls answered
> by FreeSWITCH before you need to redirect them?
>
> On Sat, Aug 8, 2009 at 11:52 AM, Vladimir
> Rodionov<vladrodionov at gmail.com> wrote:
> > Actually, this is what I need
> >
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
> >
> > Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN
> > number?
> >
> > -Vladimir Rodionov
> >
> >
> > On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <
> vladrodionov at gmail.com>
> > wrote:
> >>
> >> Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am
> pretty
> >> sure it is doable because voxeo offers this
> >> option for their Voice XML customers but I am not interested in a hosted
> >> solution at the time - it is quite expensive. As far as I understood,
> Voip
> >> provider MUST have pstn call transfer feature enabled by telecom
> provider
> >> (AT&T for example) and this should work fine with SIP.
> >>
> >> -Vladimir Rodionov
> >>
> >>
> >> On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa at gmail.com>
> >> wrote:
> >>>
> >>> Hi there,
> >>>
> >>> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
> >>> INVITE) and only pass back the media to the network, or pass back
> >>> signaling also (SIP REFER)?
> >>>
> >>> I know several suppliers who support SIP re INVITE but none that
> >>> support SIP REFER.
> >>>
> >>> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
> >>> and http://wiki.freeswitch.org/wiki/Bypass_Media
> >>>
> >>>
> >>>
> >>> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<
> vladrodionov at gmail.com>
> >>> wrote:
> >>> > Good morning,
> >>> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls
> goes
> >>> > through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
> >>> > AS does some logic and transfers call (or forward) out of Voip
> provider
> >>> > network to another PSTN number User2.
> >>> >
> >>> >
> >>> > This is call bridge
> >>> >
> >>> >
> >>> >     UA1     (PSTN) -                                 ->  UA2 (PSTN)
> >>> >               -                                            -
> >>> >                 -  (1)                                   -  (4)
> >>> >                   ->         PSTN Gateway->
> >>> >                             -                        -
> >>> >                        (2) -                        - (3)
> >>> >                           -> FreeSWITCH ->
> >>> >
> >>> >
> >>> > This is what I want to acomplish
> >>> >                                     (4)
> >>> >     UA1     (PSTN) ------------------------------- ->  UA2 (PSTN)
> >>> >                     -
> >>> >                        -   (1)
> >>> >                         ->  PSTN Gateway->
> >>> >                              -                        -
> >>> >                        (2)  -                        - (3)
> >>> >                             -> FreeSWITCH ->
> >>> >
> >>> >
> >>> > 1. Can it be implemented in FreeSWITCH?
> >>> > 2. Does anybody know Voip providers which support out of network call
> >>> > transfer/forwarding to PSTN?
> >>> >
> >>> > TIA
> >>> >
> >>> > -Vladimir Rodionov
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vladrodionov at gmail.com  Sat Aug  8 12:21:38 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Sat, 8 Aug 2009 12:21:38 -0700
Subject: [Freeswitch-users] Lua on Windows and additional modules
In-Reply-To: <0EE782AE-FC28-4029-AA01-76D87BC08069@jerris.com>
References: <3c233920908061755h5d17aa0as3fb24743215a8298@mail.gmail.com>
	<0EE782AE-FC28-4029-AA01-76D87BC08069@jerris.com>
Message-ID: <3c233920908081221m58acbcdake04a2353b7fbb401@mail.gmail.com>

Thank you all, but I decided to go with ESL and Java.

On Fri, Aug 7, 2009 at 11:11 PM, Michael Jerris <mike at jerris.com> wrote:

> check the sample config files for options to specify these:
>
>
> http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/autoload_configs/lua.conf.xml?r=10747
>
>
> On Aug 6, 2009, at 8:55 PM, Vladimir Rodionov wrote:
>
> > Good evening,
> > This is newbie question.
> >
> > The FreeSWITCH lua module does not support sockets and sql out of
> > box that is why
> > I just installed LuaBinaries (including socket, sql modules). My dev
> > environment is Win XP not Linux/Unix.
> >
> > I am trying to understand what will happen when lua_module get this:
> >
> > require "socket" or
> > require "luasql.mysql"
> > ?
> >
> > How does lua_module look up additional lua modules on Windows
> > platform?
> >
> > Do I have to set some env variables?
> >
> > TIA
> > -Vladimir Rodionov
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From raffaele.p.guidi at gmail.com  Sat Aug  8 12:42:06 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sat, 8 Aug 2009 21:42:06 +0200
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <98727038-1A37-43A4-BFC4-F624451772AF@apartmentlines.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
	<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
	<4A7DBD8E.4090608@freeswitch.org>
	<b0a8792f0908081148l2174443fhcd210e66aec15bf@mail.gmail.com>
	<98727038-1A37-43A4-BFC4-F624451772AF@apartmentlines.com>
Message-ID: <b0a8792f0908081242yfb19d46ofefd68e52145554a@mail.gmail.com>

I did that. And I do find it quite amusing :D

On Sat, Aug 8, 2009 at 21:09, Chad Phillips -- Apartment Lines <
chad at apartmentlines.com> wrote:

> does anybody else find it amusing that my thread about Lua scripting was
> hijacked by people talking about hijacking threads??  ;)
> /me waits patiently for discussion on the topic
>
>
>
> On Aug 8, 2009, at 2:48 PM, Raffaele P. Guidi wrote:
>
> Uh, oh, ok. Message-Id and In-Reply-To headers don't mean too much in a
> gmail world (I think gmail does a better job than any other mail client).
> Well, at least I understood what Brian was talking about, thanks.
> Regards,
>    Raffaele
>
> On Sat, Aug 8, 2009 at 20:01, Stefan Knoblich <stkn at freeswitch.org> wrote:
>
>> His message was a direct reply to Alan Chandler's mail and Brian's mail
>> client is working perfectly fine
>> (as are both of my mail clients).
>>
>> Look at the Message-Id and In-Reply-To headers of the mails if you still
>> think our clients are broken :)
>>
>>
>> stkn
>>
>>
>> Raffaele P. Guidi wrote:
>> > Brian, I've read 8-9 of your emails asking to not hijack threads in
>> > threads where this it didn't happen (like this). I see this as a new
>> > thread with a single message (plus yours, of course) inside with
>> > "[Freeswitch-users] how to catch DTMF in Lua while a phrase macro is
>> > playing" as the subject - and I think is quite relevant to the message.
>> > I suspect you have something wrong in your email client (mine is gmail)
>> > or server and suggest you to give it a check.
>> >
>> > Regards,
>> >    Raffaele
>> >
>> > On Sat, Aug 8, 2009 at 17:59, Brian West <brian at freeswitch.org
>> > <mailto:brian at freeswitch.org>> wrote:
>> >
>> >     Examples exist in the scripts/lua folder called callback.lua.  Also
>> >     please DO NOT hijack threads.  You can't click reply, change the
>> >     subject as it messes up the mailing list threading and most mail
>> >     readers will thread it incorrectly.
>> >
>> >     Thanks,
>> >     Brian
>> >
>> >     On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines wrote:
>> >
>> >     > in a lua script, i've tried using session:setInputCallback() to
>> catch
>> >     > DTMF tones while a phrase macro is playing, but it doesn't seem to
>> >     > work.  the same callback _does_ catch DTMF when i use
>> >     > session:streamFile() to play something.  is there some other way
>> to do
>> >     > it?
>> >     >
>> >     > below is an example of how i'm doing it now.  when voicemail/vm-
>> >     > mailbox_full.wav plays, i see key presses being printed to the
>> >     > console, but when the phrase plays, no key presses are displayed.
>> >     >
>> >     > function key_press(session, input_type, data)
>> >     >       if input_type == "dtmf" then
>> >     >               freeswitch.consoleLog("info", "Key pressed: " ..
>> >     data["digit"] ..
>> >     > "\n")
>> >     >       end
>> >     > end
>> >     >
>> >     > if session:ready() then
>> >     >       session:answer()
>> >     >       session:execute("sleep", "1000")
>> >     >       session:setInputCallback("key_press", "")
>> >     >       session:streamFile("voicemail/vm-mailbox_full.wav")
>> >     >       session:execute("phrase", "voicemail_menu,1:2:3:#")
>> >     > end
>> >     >
>> >     > _______________________________________________
>> >     > FreeSWITCH-users mailing list
>> >     > FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     >
>> >     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >     > http://www.freeswitch.org
>> >
>> >
>> >     _______________________________________________
>> >     FreeSWITCH-users mailing list
>> >     FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >     http://www.freeswitch.org
>> >
>> >
>> >
>> > ------------------------------------------------------------------------
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sat Aug  8 12:59:10 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 15:59:10 -0400
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
In-Reply-To: <a37510170908080453m1e98ae3l544e3d8b301605e1@mail.gmail.com>
References: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>
	<367751820908080348p6165b113kc3be468cb32fabb3@mail.gmail.com>
	<a37510170908080453m1e98ae3l544e3d8b301605e1@mail.gmail.com>
Message-ID: <1F19675C-E556-4AF0-88C1-B5652D8C635F@jerris.com>

how many does it stop at?  is it the same number each time?

Mike

On Aug 8, 2009, at 7:53 AM, Benedikt Fraunhofer wrote:

> Hi Phillip,
>
> 2009/8/8 Phillip Jones <pjintheusa at gmail.com>:
>
>> Not sure whether this helps but test this without set bypass_media.  
>> In
>> my setup I have noticed the leg A session ends when bypass_media is
>> true. Call/bridge continue successfully.
>
> thx for that hint. unfortunately we can't do that due to the high
> volume we anticipate
> and i think i already tried that.
>
> Note that it works in exactly this setup  (with the uuid_media()-call)
> if there are only very few calls handled. It just starts to refuse
> work once enough jobs are scheduled and not executed.
>
> Cheers
>  Benedikt



From mike at jerris.com  Sat Aug  8 13:00:45 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:00:45 -0400
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <367751820908081044g67a85684hc34a0b242c67b6a4@mail.gmail.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
	<b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
	<b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>
	<367751820908081044g67a85684hc34a0b242c67b6a4@mail.gmail.com>
Message-ID: <F4D7EE24-6B62-4A46-83CB-3E412D3120D5@jerris.com>

loopback ends up using extra threads which we are only able to drop  
later in certain situations so it will decrease your total amount of  
calls you can do if your not careful with them.

Mike

On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:

> Thanks very much for that - very help.
>
> Why would loopback be considered "abuse"? What would be the downsides
> of doing this?
>
> On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker<rupa at rupa.com> wrote:
>>
>>
>> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>> <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>
>>
>> That of course, should be:
>>
>>  <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$1..."/>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Sat Aug  8 13:04:07 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:04:07 -0400
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
In-Reply-To: <3c233920908081212s29f6d42ft210755fe50282cba@mail.gmail.com>
References: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
	<367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>
	<3c233920908081139o769c1310yaee2857d4543da7d@mail.gmail.com>
	<3c233920908081152r70acceeoe63cf15f8e1a40dd@mail.gmail.com>
	<367751820908081200y1ea230c7o12377ffe45bc75eb@mail.gmail.com>
	<3c233920908081212s29f6d42ft210755fe50282cba@mail.gmail.com>
Message-ID: <6C03C620-C2BB-4B60-9EE2-4D014736F9CF@jerris.com>

I don't know of any sip carriers who will let you do refer.  you will  
need to find a carrier who supports it.  FreeSWITCH will have no  
problem sending it but I doubt you will find a carrier who will let  
you do it easily.

Mike

On Aug 8, 2009, at 3:12 PM, Vladimir Rodionov wrote:

> A call is coming on SIP trunk. From PSTN.  I does not need to be  
> answered, actually - I need to do some logic before redirecting call  
> but I can answer call as well It won't break the app logic.
>
> -Vladimir Rodionov
>
> On Sat, Aug 8, 2009 at 12:00 PM, Phillip Jones  
> <pjintheusa at gmail.com> wrote:
> Are your calls coming in on TDM or SIP trunks? Are your calls answered
> by FreeSWITCH before you need to redirect them?
>
> On Sat, Aug 8, 2009 at 11:52 AM, Vladimir
> Rodionov<vladrodionov at gmail.com> wrote:
> > Actually, this is what I need
> >
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
> >
> > Will it work with PSTN? Can I redirect incoming PSTN call to  
> another PSTN
> > number?
> >
> > -Vladimir Rodionov
> >
> >
> > On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <vladrodionov at gmail.com 
> >
> > wrote:
> >>
> >> Yes, I need SIP-REFER (pass back media and signaling to PSTN). I  
> am pretty
> >> sure it is doable because voxeo offers this
> >> option for their Voice XML customers but I am not interested in a  
> hosted
> >> solution at the time - it is quite expensive. As far as I  
> understood, Voip
> >> provider MUST have pstn call transfer feature enabled by telecom  
> provider
> >> (AT&T for example) and this should work fine with SIP.
> >>
> >> -Vladimir Rodionov
> >>
> >>
> >> On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa at gmail.com 
> >
> >> wrote:
> >>>
> >>> Hi there,
> >>>
> >>> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
> >>> INVITE) and only pass back the media to the network, or pass back
> >>> signaling also (SIP REFER)?
> >>>
> >>> I know several suppliers who support SIP re INVITE but none that
> >>> support SIP REFER.
> >>>
> >>> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
> >>> and http://wiki.freeswitch.org/wiki/Bypass_Media
> >>>
> >>>
> >>>
> >>> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov at gmail.com 
> >
> >>> wrote:
> >>> > Good morning,
> >>> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID).  
> Calls goes
> >>> > through PSTN Gateway (1) to freeSWITCH application server (AS)  
> (2).
> >>> > AS does some logic and transfers call (or forward) out of Voip  
> provider
> >>> > network to another PSTN number User2.
> >>> >
> >>> >
> >>> > This is call bridge
> >>> >
> >>> >
> >>> >     UA1     (PSTN) -                                 ->  UA2  
> (PSTN)
> >>> >               -                                            -
> >>> >                 -  (1)                                   -  (4)
> >>> >                   ->         PSTN Gateway->
> >>> >                             -                        -
> >>> >                        (2) -                        - (3)
> >>> >                           -> FreeSWITCH ->
> >>> >
> >>> >
> >>> > This is what I want to acomplish
> >>> >                                     (4)
> >>> >     UA1     (PSTN) ------------------------------- ->  UA2  
> (PSTN)
> >>> >                     -
> >>> >                        -   (1)
> >>> >                         ->  PSTN Gateway->
> >>> >                              -                        -
> >>> >                        (2)  -                        - (3)
> >>> >                             -> FreeSWITCH ->
> >>> >
> >>> >
> >>> > 1. Can it be implemented in FreeSWITCH?
> >>> > 2. Does anybody know Voip providers which support out of  
> network call
> >>> > transfer/forwarding to PSTN?
> >>> >
> >>> > TIA
> >>> >
> >>> > -Vladimir Rodionov
> >>> >
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From mike at jerris.com  Sat Aug  8 13:16:39 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:16:39 -0400
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
Message-ID: <32E54334-B499-4FA7-974B-705D8713A663@jerris.com>

use the SayPhrase method of sessin instead of executing the phrase  
application then your input callback should work as expected.

Mike

On Aug 8, 2009, at 11:15 AM, Chad Phillips -- Apartment Lines wrote:

> in a lua script, i've tried using session:setInputCallback() to catch
> DTMF tones while a phrase macro is playing, but it doesn't seem to
> work.  the same callback _does_ catch DTMF when i use
> session:streamFile() to play something.  is there some other way to do
> it?
>
> below is an example of how i'm doing it now.  when voicemail/vm-
> mailbox_full.wav plays, i see key presses being printed to the
> console, but when the phrase plays, no key presses are displayed.
>
> function key_press(session, input_type, data)
> 	if input_type == "dtmf" then
> 		freeswitch.consoleLog("info", "Key pressed: " .. data["digit"] ..
> "\n")
> 	end
> end
>
> if session:ready() then
> 	session:answer()
> 	session:execute("sleep", "1000")
> 	session:setInputCallback("key_press", "")
> 	session:streamFile("voicemail/vm-mailbox_full.wav")
> 	session:execute("phrase", "voicemail_menu,1:2:3:#")
> end



From mike at jerris.com  Sat Aug  8 13:19:48 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:19:48 -0400
Subject: [Freeswitch-users] FreeSwitch doesn't play music on hold
	forbriged channel
In-Reply-To: <EC1055E8A337451F9F5ADC7414BDFDD9@abisoft.biz>
References: <EC1055E8A337451F9F5ADC7414BDFDD9@abisoft.biz>
Message-ID: <E0DF0981-0AA7-4321-87E5-9AA9E7FFC580@jerris.com>

What do the debug logs on fs say when you try to put the call on hold?

Mike

On Aug 6, 2009, at 1:01 PM, Kozak Vladimir wrote:

>
> The scenario is the following:
> FS User A dial an extension
> Extention opens outbound socket channel to my application
> My application bridges the call to FS User B
> The application check for CHANNEL_BRIDGED event and stores Other-leg- 
> unique-id
> The application sends hold to the bridged channel using SendMsg with  
> Other-leg-unique-id
> User B is placed on hold but no music on hold is played to the  
> caller (User A)
>
>
> I have outbound socket channel and the following sequence of  
> commands/event:
> listening on [any] 8084 ...
> connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru  
> [172.26.200.250] 34000
> connect
>

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From mike at jerris.com  Sat Aug  8 13:21:11 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:21:11 -0400
Subject: [Freeswitch-users] State of originated call
In-Reply-To: <c55d46950908071644m7604eb57jf5018c2b378d3255@mail.gmail.com>
References: <c55d46950908071644m7604eb57jf5018c2b378d3255@mail.gmail.com>
Message-ID: <1E164379-D432-4309-8381-A2932868079C@jerris.com>

typically when these questions are asked we find people really want to  
be doing it the way these signals are automatically passed across.   
Can you describe what your trying to do a bit more?

Mike

On Aug 7, 2009, at 7:44 PM, Max Bridgewater wrote:

> Hi,
>
> using javascript, i do originate the call this way:
>
> Session s= new Session(originateStr);
>
> From this point, is it possible to know what states the call is  
> going through? In a previous message it was suggested that  
> variable_originate_disposition would give me the response code. Now,  
> how to i use this in practice in a script?
> How do i for instance retrieve a 180 response code when rining is  
> hapening on the remote end?
>
> Thanks,
> Max.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From pjintheusa at gmail.com  Sat Aug  8 13:24:45 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 13:24:45 -0700
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <F4D7EE24-6B62-4A46-83CB-3E412D3120D5@jerris.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
	<b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
	<b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>
	<367751820908081044g67a85684hc34a0b242c67b6a4@mail.gmail.com>
	<F4D7EE24-6B62-4A46-83CB-3E412D3120D5@jerris.com>
Message-ID: <367751820908081324r44c7ccf5lfc8ad78c96facb69@mail.gmail.com>

Mike/Rupa ,

Thanks for your help on this. So I am correct that summarizing that
FreeSWITCH does not really support fail over and multiple call
destinations because the same mechanism is used to achieve both? And
that loopback as a solution is possible but not recommended?

Is there any other solution to this? Perhaps a second FS box in the mix?

Phil


On Sat, Aug 8, 2009 at 1:00 PM, Michael Jerris<mike at jerris.com> wrote:
> loopback ends up using extra threads which we are only able to drop
> later in certain situations so it will decrease your total amount of
> calls you can do if your not careful with them.
>
> Mike
>
> On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:
>
>> Thanks very much for that - very help.
>>
>> Why would loopback be considered "abuse"? What would be the downsides
>> of doing this?
>>
>> On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker<rupa at rupa.com> wrote:
>>>
>>>
>>> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>
>>>> <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>
>>>
>>> That of course, should be:
>>>
>>> ?<action application="bridge" data="sofia/gw1/$1|sofia/gw2/$1..."/>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mike at jerris.com  Sat Aug  8 13:24:55 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:24:55 -0400
Subject: [Freeswitch-users] DTMF disable a few secs after call starts
In-Reply-To: <0B930F23C4A24E3CB0015945EA06C8B7@ws4>
References: <0B930F23C4A24E3CB0015945EA06C8B7@ws4>
Message-ID: <B1AF3C1E-6235-457A-AFF3-425F24BEB68E@jerris.com>

you would have to write a module in c that hooks the dtmf and throws  
it away.  This would work similar to how bind_meta works.  That being  
said, this is all much easier if your using 2833 as you don't need to  
go to the extra cpu of detecting tones.

Mike

On Aug 8, 2009, at 9:10 AM, Frank @ Impact wrote:

> FS is in the media path of an IVR call.
> At the moment, the call is ulaw with DTMF in the audio I think  
> coming into FS and leaving FS.
> The call is coming from an Asterisk server and going to an Asterisk  
> server.
>
> Is there a way to disable FS from passing DTMF at some point in the  
> call?  For example, after 15 seconds, is there a way to get FS to  
> stop passing DTMF events?
>
> Would I have to try to force asterisk to use rfc2833 when sending  
> the call to FS and when accepting it back from FS?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Sat Aug  8 13:32:12 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:32:12 -0400
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <367751820908081324r44c7ccf5lfc8ad78c96facb69@mail.gmail.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
	<b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
	<b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>
	<367751820908081044g67a85684hc34a0b242c67b6a4@mail.gmail.com>
	<F4D7EE24-6B62-4A46-83CB-3E412D3120D5@jerris.com>
	<367751820908081324r44c7ccf5lfc8ad78c96facb69@mail.gmail.com>
Message-ID: <287F078C-1132-413C-B456-A2630082571C@jerris.com>

  I think that summary is totally wrong.  Loopback should be used  
here, and this should work to do what you want, just be aware of what  
that means.

Mike


On Aug 8, 2009, at 4:24 PM, Phillip Jones wrote:

> Mike/Rupa ,
>
> Thanks for your help on this. So I am correct that summarizing that
> FreeSWITCH does not really support fail over and multiple call
> destinations because the same mechanism is used to achieve both? And
> that loopback as a solution is possible but not recommended?
>
> Is there any other solution to this? Perhaps a second FS box in the  
> mix?
>
> Phil
>
>
> On Sat, Aug 8, 2009 at 1:00 PM, Michael Jerris<mike at jerris.com> wrote:
>> loopback ends up using extra threads which we are only able to drop
>> later in certain situations so it will decrease your total amount of
>> calls you can do if your not careful with them.
>>
>> Mike
>>
>> On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:
>>
>>> Thanks very much for that - very help.
>>>
>>> Why would loopback be considered "abuse"? What would be the  
>>> downsides
>>> of doing this?
>>>
>>> On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker<rupa at rupa.com> wrote:
>>>>
>>>>
>>>> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker <rupa at rupa.com>  
>>>> wrote:
>>>>>
>>>>> <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>
>>>>
>>>> That of course, should be:
>>>>
>>>>  <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$1..."/>
>>>> --
>>>> -Rupa



From pjintheusa at gmail.com  Sat Aug  8 13:51:00 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 13:51:00 -0700
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <287F078C-1132-413C-B456-A2630082571C@jerris.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
	<b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
	<b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>
	<367751820908081044g67a85684hc34a0b242c67b6a4@mail.gmail.com>
	<F4D7EE24-6B62-4A46-83CB-3E412D3120D5@jerris.com>
	<367751820908081324r44c7ccf5lfc8ad78c96facb69@mail.gmail.com>
	<287F078C-1132-413C-B456-A2630082571C@jerris.com>
Message-ID: <367751820908081351pdc18bb9kf97ea308a669f14a@mail.gmail.com>

Ok - that's great. I will build this out - thanks both for your help
on this. Much appreciated.

On Sat, Aug 8, 2009 at 1:32 PM, Michael Jerris<mike at jerris.com> wrote:
> ?I think that summary is totally wrong. ?Loopback should be used
> here, and this should work to do what you want, just be aware of what
> that means.
>
> Mike
>
>
> On Aug 8, 2009, at 4:24 PM, Phillip Jones wrote:
>
>> Mike/Rupa ,
>>
>> Thanks for your help on this. So I am correct that summarizing that
>> FreeSWITCH does not really support fail over and multiple call
>> destinations because the same mechanism is used to achieve both? And
>> that loopback as a solution is possible but not recommended?
>>
>> Is there any other solution to this? Perhaps a second FS box in the
>> mix?
>>
>> Phil
>>
>>
>> On Sat, Aug 8, 2009 at 1:00 PM, Michael Jerris<mike at jerris.com> wrote:
>>> loopback ends up using extra threads which we are only able to drop
>>> later in certain situations so it will decrease your total amount of
>>> calls you can do if your not careful with them.
>>>
>>> Mike
>>>
>>> On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:
>>>
>>>> Thanks very much for that - very help.
>>>>
>>>> Why would loopback be considered "abuse"? What would be the
>>>> downsides
>>>> of doing this?
>>>>
>>>> On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker<rupa at rupa.com> wrote:
>>>>>
>>>>>
>>>>> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker <rupa at rupa.com>
>>>>> wrote:
>>>>>>
>>>>>> <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>
>>>>>
>>>>> That of course, should be:
>>>>>
>>>>> ?<action application="bridge" data="sofia/gw1/$1|sofia/gw2/$1..."/>
>>>>> --
>>>>> -Rupa
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From fraunhofer.lists.freeswitch-001 at traced.net  Sat Aug  8 14:21:25 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Sat, 8 Aug 2009 23:21:25 +0200
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
In-Reply-To: <1F19675C-E556-4AF0-88C1-B5652D8C635F@jerris.com>
References: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>
	<367751820908080348p6165b113kc3be468cb32fabb3@mail.gmail.com>
	<a37510170908080453m1e98ae3l544e3d8b301605e1@mail.gmail.com>
	<1F19675C-E556-4AF0-88C1-B5652D8C635F@jerris.com>
Message-ID: <a37510170908081421i6e6c9031vaa806e8adf3b5f91@mail.gmail.com>

Hello Mike,

2009/8/8 Michael Jerris <mike at jerris.com>:

> how many does it stop at? ?is it the same number each time?

i tried to express that non-wisdom using the words
"This only works fine if we've few concurrent calls. There is no magic
borderline where it starts to refuse work."

this is surely not the best english, i admit. I just have no exact
"lower-bound" when it suddenly happens. My manual test-calls work only
in the beginning, once the loadgen is on for ~3min even the manual
test-calls do not work. Once it's locked up, it initiates calls
successfully (it rings) but (in my example) M doesn't ACK the answer
event it get's from A.

In my current setup with 100 call-legs i can repeatably make it
happen. Even better (for debugging), it does happen each time.

If someone is interested in the exact setup i can place a tarball
somewhere or - if someone could advise me with some gdb commands he
wants to have -  i'll recompile with debugging symbols and get you
what you want. I'm definitely stuck here.

Thx.

  Beni.


From tayeb.meftah at gmail.com  Sat Aug  8 14:32:29 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 08 Aug 2009 21:32:29 +0000
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk, revamped
In-Reply-To: <7b197bef0907270708h1dd55de2m92b5de869da6404d@mail.gmail.com>
References: <7b197bef0907270708h1dd55de2m92b5de869da6404d@mail.gmail.com>
Message-ID: <4A7DEEED.1020006@gmail.com>

hello,
i'm using mod_skypiax
is working very perfectly in my ubuntu machine, no problem or bug
only remember the featur that i requested,
hide the skype client to the system tray and prev it from poping up
thanks!
Giovanni Maruzzelli wrote:
> Ciao FreeSWITCHers,
>
> please have a look at the much changed wiki page:
> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk and
> checkout and test the code in svn.
>
> Much has happened, various bug fixes and features added.
>
> Most notable:
> - multiple instances of the same Skype username on Linux (eg: running
> 20 concurrent channels as "Bob" Skype user)
> - adding and removing interfaces on the fly (patch sent by Muhammad Shahzad)
> - easier creation of Skype clients configuration directory
> - reduced latency
> - better robustness
> - running as Windows Service
> - customized ALSA driver for more devices with less IRQs and context switches
> - custom kernel tickless and 100HZ (eg. solves high load problems in
> CentOS and in virtualization)
> - interactive command line client for prototyping
>
> Also, please note that ALSA drivers version 1.0.20 seems to be much
> more stable in our kind of usage (snd-dummy).
>
> Various other enhancements will come, but in the mean time please give
> feedback on the current svn code (we want to be robust for the 1.0.4
> Release :-) )
>
> See you all at www.cluecon.com, talk on Skypiax August 4th at 4.30 pm !
>
> -giovanni
>
>
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4317 (20090808) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>   



__________ Information from ESET NOD32 Antivirus, version of virus signature database 4317 (20090808) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




From mike at jerris.com  Sat Aug  8 15:06:09 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 18:06:09 -0400
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
In-Reply-To: <a37510170908081421i6e6c9031vaa806e8adf3b5f91@mail.gmail.com>
References: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>
	<367751820908080348p6165b113kc3be468cb32fabb3@mail.gmail.com>
	<a37510170908080453m1e98ae3l544e3d8b301605e1@mail.gmail.com>
	<1F19675C-E556-4AF0-88C1-B5652D8C635F@jerris.com>
	<a37510170908081421i6e6c9031vaa806e8adf3b5f91@mail.gmail.com>
Message-ID: <26EDA0B5-7DD8-4CE4-829C-FEDCD3E84775@jerris.com>

please open a bug on jira.freeswitch.org with the details of exactly  
how to re-create this issue

Mike

On Aug 8, 2009, at 5:21 PM, Benedikt Fraunhofer wrote:

> Hello Mike,
>
> 2009/8/8 Michael Jerris <mike at jerris.com>:
>
>> how many does it stop at?  is it the same number each time?
>
> i tried to express that non-wisdom using the words
> "This only works fine if we've few concurrent calls. There is no magic
> borderline where it starts to refuse work."
>
> this is surely not the best english, i admit. I just have no exact
> "lower-bound" when it suddenly happens. My manual test-calls work only
> in the beginning, once the loadgen is on for ~3min even the manual
> test-calls do not work. Once it's locked up, it initiates calls
> successfully (it rings) but (in my example) M doesn't ACK the answer
> event it get's from A.
>
> In my current setup with 100 call-legs i can repeatably make it
> happen. Even better (for debugging), it does happen each time.
>
> If someone is interested in the exact setup i can place a tarball
> somewhere or - if someone could advise me with some gdb commands he
> wants to have -  i'll recompile with debugging symbols and get you
> what you want. I'm definitely stuck here.
>
> Thx.
>
>  Beni.



From mgg at giagnocavo.net  Sat Aug  8 17:17:35 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Sat, 8 Aug 2009 20:17:35 -0400
Subject: [Freeswitch-users] mod_php needed
In-Reply-To: <6b65470d0908081134x719bbef6r707644d877eee328@mail.gmail.com>
References: <4A33E8A0.1070708@xpirio.com>
	<87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147238@sun.Noble.local>
	<4A361708.8020808@xpirio.com>
	<191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com>
	<4A3660EB.4070203@xpirio.com>
	<191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com>
	<6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com>
	<191c3a030906150827t2e2e919bw396097bd637a0b91@mail.gmail.com>
	<6b65470d0908081134x719bbef6r707644d877eee328@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702CC213681@mse17be1.mse17.exchange.ms>

You should be able to use PHP with mod_managed just fine. You'll just need Phalanger:
http://www.codeplex.com/Phalanger
http://www.php-compiler.net/doku.php

I don't have any use for PHP, but if anyone is interested in this, I'd be happy to help out if it doesn't "just work" out of the box. It should work on MS CLR and Mono. It also has CodeDom support, so enabling it for dynamic-compilation/scripting should also be easy.

As an added bonus (apart from the claims of 2x the performance), it even has Visual Studio support, complete with some designers: 
http://www.php-compiler.net/doku.php?id=core%3aphp-in-vs2008

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William Suffill
Sent: Saturday, August 08, 2009 12:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_php needed

http://wiki.freeswitch.org/wiki/PHP_ESL

Formatting could use a bit of help just exported my working document
to MediaWiki. Hope to add some more information to ESL as I continue
to do more with it.

I'm going to start a page on fs_ivrd as well. Feedback welcome.

-- W

IRC: wsuff on freenode

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From brian at freeswitch.org  Sat Aug  8 18:18:01 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 8 Aug 2009 20:18:01 -0500
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
	<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
Message-ID: <012255F2-64AF-4B14-A063-590162E6DC9A@freeswitch.org>

Yes it happens 100% of the time I say not to do it..

Example this thread started with:

In-Reply-To: <4A7BE35B.8010709 at chandlerfamily.org.uk>
References: <4A7BE35B.8010709 at chandlerfamily.org.uk>


The hijacked thread has:

References: <4A7BE35B.8010709 at chandlerfamily.org.uk>
<33c87fa30908072139m9444779w80f5908fa624e9de at mail.gmail.com>
<4A7D8C26.4020408 at chandlerfamily.org.uk>
<BC708A39-BA09-4524-A072-92A5E5A870F3 at apartmentlines.com>
<96ADFBB4-A015-417A-B0AD-2C427376E010 at freeswitch.org>
<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de at mail.gmail.com>

Which causes it to thread it via the References header.

Hence it was hijacked.

/b


On Aug 8, 2009, at 11:50 AM, Raffaele P. Guidi wrote:

> Brian, I've read 8-9 of your emails asking to not hijack threads in  
> threads where this it didn't happen (like this). I see this as a new  
> thread with a single message (plus yours, of course) inside with  
> "[Freeswitch-users] how to catch DTMF in Lua while a phrase macro is  
> playing" as the subject - and I think is quite relevant to the  
> message. I suspect you have something wrong in your email client  
> (mine is gmail) or server and suggest you to give it a check.
>
> Regards,
>    Raffaele



From ivan at myrvold.org  Sun Aug  9 07:52:14 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Sun, 9 Aug 2009 16:52:14 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
Message-ID: <C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>

I tried to compile mod_skypiax, and am getting problem with X11. On OS  
X Leopard, X11 is installed in /usr/X11/lib/
See below.

What can I do to get past this error?

I can also let you ssh into my machine. Contact me off list in case.

Ivan

making all mod_skypiax
Compiling skypiax_protocol.c...
Compiling mod_skypiax.c...
mkdir .libs
Compiling mod_skypiax.c ...
Creating mod_skypiax.so...
ld: library not found for -lX11
collect2: ld returned 1 exit status
gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/ 
Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/Documents/ 
Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror - 
fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g - 
ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -shared - 
o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/ 
mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/Freeswitch/ 
freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/ 
imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/ 
expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/ 
libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/libiconv.dylib / 
Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr/.libs/ 
libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz - 
lncurses -lX11
make[5]: *** [mod_skypiax.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_skypiax-all] Error 1
make[2]: *** [all-recursive] Error 1


Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:

> No, it needs implementation of the message pump between the module and
> the Skype API.
>
> It's probably kind of trivial, if no other problems I'm not aware of.
>
> I do not have a Mac to implement it, tough :-(.
>
> -giovanni
>
>
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
>
>
>
> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>  
> wrote:
>> I'm not sure about that one.... I haven't tried lately because the  
>> API
>> differs on the Mac last I looked at it.
>>
>> /b
>>
>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>
>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>
>>> Ivan
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From gmaruzz at celliax.org  Sun Aug  9 08:10:57 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 9 Aug 2009 17:10:57 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org> 
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com> 
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
Message-ID: <7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>

Ciao Ivan,

it seems that you do not have the libX11 **development** package installed.

Unfortunately I don't know about OSX, so I cannot help you, but many
on the list know.

BTW: it will probably be of no use to you to compile mod_skypiax on
OSX, because Skype for MACOSX works in another way than Skype for
Linux.
If you know about MacOSX programming, please have a look at
https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
probably be simple enough to add a message pump for MacOSX.

-giovanni




Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org> wrote:
> I tried to compile mod_skypiax, and am getting problem with X11. On OS
> X Leopard, X11 is installed in /usr/X11/lib/
> See below.
>
> What can I do to get past this error?
>
> I can also let you ssh into my machine. Contact me off list in case.
>
> Ivan
>
> making all mod_skypiax
> Compiling skypiax_protocol.c...
> Compiling mod_skypiax.c...
> mkdir .libs
> Compiling mod_skypiax.c ...
> Creating mod_skypiax.so...
> ld: library not found for -lX11
> collect2: ld returned 1 exit status
> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/Documents/
> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -
> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -shared -
> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/
> mod_skypiax.o skypiax_protocol.o ?/Users/imyrvold/Documents/Freeswitch/
> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/
> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/libiconv.dylib /
> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr/.libs/
> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
> lncurses -lX11
> make[5]: *** [mod_skypiax.so] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_skypiax-all] Error 1
> make[2]: *** [all-recursive] Error 1
>
>
> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>
>> No, it needs implementation of the message pump between the module and
>> the Skype API.
>>
>> It's probably kind of trivial, if no other problems I'm not aware of.
>>
>> I do not have a Mac to implement it, tough :-(.
>>
>> -giovanni
>>
>>
>>
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> =========================================
>> www.celliax.org
>> via Pierlombardo 9, 20135 Milano
>> Italy
>> gmaruzz at celliax dot org
>> Cell : +39-347-2665618
>> Fax : +39-02-87390039
>>
>>
>>
>>
>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>> wrote:
>>> I'm not sure about that one.... I haven't tried lately because the
>>> API
>>> differs on the Mac last I looked at it.
>>>
>>> /b
>>>
>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>
>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>
>>>> Ivan
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dujinfang at gmail.com  Sun Aug  9 08:24:53 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 9 Aug 2009 23:24:53 +0800
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
Message-ID: <6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>


On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
> Ciao Ivan,
>
> it seems that you do not have the libX11 **development** package  
> installed.
>
> Unfortunately I don't know about OSX, so I cannot help you, but many
> on the list know.
> BTW: it will probably be of no use to you to compile mod_skypiax on
> OSX, because Skype for MACOSX works in another way than Skype for
> Linux.

That's right.

> If you know about MacOSX programming, please have a look at
> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
> probably be simple enough to add a message pump for MacOSX.
>
> -giovanni
>
>
Giovanni, I have a Mac and tried to get this work yesterday, but  
haven't got it work. Will try further if I have time. However, I don't  
think it's so useful because I don't know how to run and hence control  
multi-skype instances on Mac.

If someone interested to try this I can check the code into my branch.

>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
>
>
>
> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>  
> wrote:
>> I tried to compile mod_skypiax, and am getting problem with X11. On  
>> OS
>> X Leopard, X11 is installed in /usr/X11/lib/
>> See below.
>>
>> What can I do to get past this error?
>>
>> I can also let you ssh into my machine. Contact me off list in case.
>>
>> Ivan
>>
>> making all mod_skypiax
>> Compiling skypiax_protocol.c...
>> Compiling mod_skypiax.c...
>> mkdir .libs
>> Compiling mod_skypiax.c ...
>> Creating mod_skypiax.so...
>> ld: library not found for -lX11
>> collect2: ld returned 1 exit status
>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/ 
>> Documents/
>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -
>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -shared -
>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/
>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/ 
>> Freeswitch/
>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/
>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/ 
>> libiconv.dylib /
>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/ 
>> apr/.libs/
>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
>> lncurses -lX11
>> make[5]: *** [mod_skypiax.so] Error 1
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_skypiax-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>>
>>
>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>
>>> No, it needs implementation of the message pump between the module  
>>> and
>>> the Skype API.
>>>
>>> It's probably kind of trivial, if no other problems I'm not aware  
>>> of.
>>>
>>> I do not have a Mac to implement it, tough :-(.
>>>
>>> -giovanni
>>>
>>>
>>>
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> =========================================
>>> www.celliax.org
>>> via Pierlombardo 9, 20135 Milano
>>> Italy
>>> gmaruzz at celliax dot org
>>> Cell : +39-347-2665618
>>> Fax : +39-02-87390039
>>>
>>>
>>>
>>>
>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>> wrote:
>>>> I'm not sure about that one.... I haven't tried lately because the
>>>> API
>>>> differs on the Mac last I looked at it.
>>>>
>>>> /b
>>>>
>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>
>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>
>>>>> Ivan
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From ivan at myrvold.org  Sun Aug  9 08:33:09 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Sun, 9 Aug 2009 17:33:09 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
Message-ID: <4F6990B2-E9D0-4EC2-9457-973A643AFDD6@myrvold.org>

Thanks, Giovanni for the pointer!

Yes, I am a Cocoa developer, and have no problem compiling with the  
Skype.framework. But I still have no clue how to include this into the  
skypiax code.
Probably need just a little more hint of how to do it.

Ivan

Den 9. aug.. 2009 kl. 17:10 skrev Giovanni Maruzzelli:

> Ciao Ivan,
>
> it seems that you do not have the libX11 **development** package  
> installed.
>
> Unfortunately I don't know about OSX, so I cannot help you, but many
> on the list know.
>
> BTW: it will probably be of no use to you to compile mod_skypiax on
> OSX, because Skype for MACOSX works in another way than Skype for
> Linux.
> If you know about MacOSX programming, please have a look at
> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
> probably be simple enough to add a message pump for MacOSX.
>
> -giovanni
>
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
>
>
>
> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>  
> wrote:
>> I tried to compile mod_skypiax, and am getting problem with X11. On  
>> OS
>> X Leopard, X11 is installed in /usr/X11/lib/
>> See below.
>>
>> What can I do to get past this error?
>>
>> I can also let you ssh into my machine. Contact me off list in case.
>>
>> Ivan
>>
>> making all mod_skypiax
>> Compiling skypiax_protocol.c...
>> Compiling mod_skypiax.c...
>> mkdir .libs
>> Compiling mod_skypiax.c ...
>> Creating mod_skypiax.so...
>> ld: library not found for -lX11
>> collect2: ld returned 1 exit status
>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/ 
>> Documents/
>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -
>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -shared -
>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/
>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/ 
>> Freeswitch/
>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/
>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/ 
>> libiconv.dylib /
>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/ 
>> apr/.libs/
>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
>> lncurses -lX11
>> make[5]: *** [mod_skypiax.so] Error 1
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_skypiax-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>>
>>
>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>
>>> No, it needs implementation of the message pump between the module  
>>> and
>>> the Skype API.
>>>
>>> It's probably kind of trivial, if no other problems I'm not aware  
>>> of.
>>>
>>> I do not have a Mac to implement it, tough :-(.
>>>
>>> -giovanni
>>>
>>>
>>>
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> =========================================
>>> www.celliax.org
>>> via Pierlombardo 9, 20135 Milano
>>> Italy
>>> gmaruzz at celliax dot org
>>> Cell : +39-347-2665618
>>> Fax : +39-02-87390039
>>>
>>>
>>>
>>>
>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>> wrote:
>>>> I'm not sure about that one.... I haven't tried lately because the
>>>> API
>>>> differs on the Mac last I looked at it.
>>>>
>>>> /b
>>>>
>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>
>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>
>>>>> Ivan
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From ivan at myrvold.org  Sun Aug  9 08:34:37 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Sun, 9 Aug 2009 17:34:37 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
	<6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
Message-ID: <4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>

Yes, I am interested in this, and if you have any source I could have  
a look at it.

Ivan

Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:

>
> On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
>> Ciao Ivan,
>>
>> it seems that you do not have the libX11 **development** package
>> installed.
>>
>> Unfortunately I don't know about OSX, so I cannot help you, but many
>> on the list know.
>> BTW: it will probably be of no use to you to compile mod_skypiax on
>> OSX, because Skype for MACOSX works in another way than Skype for
>> Linux.
>
> That's right.
>
>> If you know about MacOSX programming, please have a look at
>> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
>> probably be simple enough to add a message pump for MacOSX.
>>
>> -giovanni
>>
>>
> Giovanni, I have a Mac and tried to get this work yesterday, but
> haven't got it work. Will try further if I have time. However, I don't
> think it's so useful because I don't know how to run and hence control
> multi-skype instances on Mac.
>
> If someone interested to try this I can check the code into my branch.
>
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> =========================================
>> www.celliax.org
>> via Pierlombardo 9, 20135 Milano
>> Italy
>> gmaruzz at celliax dot org
>> Cell : +39-347-2665618
>> Fax : +39-02-87390039
>>
>>
>>
>>
>> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>
>> wrote:
>>> I tried to compile mod_skypiax, and am getting problem with X11. On
>>> OS
>>> X Leopard, X11 is installed in /usr/X11/lib/
>>> See below.
>>>
>>> What can I do to get past this error?
>>>
>>> I can also let you ssh into my machine. Contact me off list in case.
>>>
>>> Ivan
>>>
>>> making all mod_skypiax
>>> Compiling skypiax_protocol.c...
>>> Compiling mod_skypiax.c...
>>> mkdir .libs
>>> Compiling mod_skypiax.c ...
>>> Creating mod_skypiax.so...
>>> ld: library not found for -lX11
>>> collect2: ld returned 1 exit status
>>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
>>> Documents/
>>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 - 
>>> g -
>>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE - 
>>> shared -
>>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/
>>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/
>>> Freeswitch/
>>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/
>>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
>>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
>>> libiconv.dylib /
>>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
>>> apr/.libs/
>>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
>>> lncurses -lX11
>>> make[5]: *** [mod_skypiax.so] Error 1
>>> make[4]: *** [all] Error 1
>>> make[3]: *** [mod_skypiax-all] Error 1
>>> make[2]: *** [all-recursive] Error 1
>>>
>>>
>>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>>
>>>> No, it needs implementation of the message pump between the module
>>>> and
>>>> the Skype API.
>>>>
>>>> It's probably kind of trivial, if no other problems I'm not aware
>>>> of.
>>>>
>>>> I do not have a Mac to implement it, tough :-(.
>>>>
>>>> -giovanni
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> =========================================
>>>> www.celliax.org
>>>> via Pierlombardo 9, 20135 Milano
>>>> Italy
>>>> gmaruzz at celliax dot org
>>>> Cell : +39-347-2665618
>>>> Fax : +39-02-87390039
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>>> wrote:
>>>>> I'm not sure about that one.... I haven't tried lately because the
>>>>> API
>>>>> differs on the Mac last I looked at it.
>>>>>
>>>>> /b
>>>>>
>>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>>
>>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>>
>>>>>> Ivan
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From gmaruzz at celliax.org  Sun Aug  9 08:52:07 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 9 Aug 2009 17:52:07 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org> 
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com> 
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com> 
	<6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
	<4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
Message-ID: <7b197bef0908090852t7243db6r761ddfce31093ed0@mail.gmail.com>

Seven,
thanks a lot for your effort, please let your stuff be available,
maybe Ivan can make use of it!

Ivan,
in the file src/mod/endpoints/mod_skypiax/skypiax_protocol.c add you
will find #ifdef WIN32 .

it conditional compiles code between WIN32 and linux.

You need to add another #ifdef, so it will compile for OSX.

You will probably be able to use the same pipe mechanism as in Linux
(normal POSIX pipes).
You will for sure need to implement the part that deals with the Skype
API. Maybe it will be not much more than reusing the example code to
interact with the API.

Please, let us know how it goes, and feel *very* free to ask for further info.

-giovanni




Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Sun, Aug 9, 2009 at 5:34 PM, Ivan C Myrvold<ivan at myrvold.org> wrote:
> Yes, I am interested in this, and if you have any source I could have
> a look at it.
>
> Ivan
>
> Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
>
>>
>> On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
>>> Ciao Ivan,
>>>
>>> it seems that you do not have the libX11 **development** package
>>> installed.
>>>
>>> Unfortunately I don't know about OSX, so I cannot help you, but many
>>> on the list know.
>>> BTW: it will probably be of no use to you to compile mod_skypiax on
>>> OSX, because Skype for MACOSX works in another way than Skype for
>>> Linux.
>>
>> That's right.
>>
>>> If you know about MacOSX programming, please have a look at
>>> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
>>> probably be simple enough to add a message pump for MacOSX.
>>>
>>> -giovanni
>>>
>>>
>> Giovanni, I have a Mac and tried to get this work yesterday, but
>> haven't got it work. Will try further if I have time. However, I don't
>> think it's so useful because I don't know how to run and hence control
>> multi-skype instances on Mac.
>>
>> If someone interested to try this I can check the code into my branch.
>>
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> =========================================
>>> www.celliax.org
>>> via Pierlombardo 9, 20135 Milano
>>> Italy
>>> gmaruzz at celliax dot org
>>> Cell : +39-347-2665618
>>> Fax : +39-02-87390039
>>>
>>>
>>>
>>>
>>> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>
>>> wrote:
>>>> I tried to compile mod_skypiax, and am getting problem with X11. On
>>>> OS
>>>> X Leopard, X11 is installed in /usr/X11/lib/
>>>> See below.
>>>>
>>>> What can I do to get past this error?
>>>>
>>>> I can also let you ssh into my machine. Contact me off list in case.
>>>>
>>>> Ivan
>>>>
>>>> making all mod_skypiax
>>>> Compiling skypiax_protocol.c...
>>>> Compiling mod_skypiax.c...
>>>> mkdir .libs
>>>> Compiling mod_skypiax.c ...
>>>> Creating mod_skypiax.so...
>>>> ld: library not found for -lX11
>>>> collect2: ld returned 1 exit status
>>>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>>>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
>>>> Documents/
>>>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>>>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -
>>>> g -
>>>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -
>>>> shared -
>>>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/
>>>> mod_skypiax.o skypiax_protocol.o ?/Users/imyrvold/Documents/
>>>> Freeswitch/
>>>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>>>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/
>>>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
>>>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
>>>> libiconv.dylib /
>>>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
>>>> apr/.libs/
>>>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
>>>> lncurses -lX11
>>>> make[5]: *** [mod_skypiax.so] Error 1
>>>> make[4]: *** [all] Error 1
>>>> make[3]: *** [mod_skypiax-all] Error 1
>>>> make[2]: *** [all-recursive] Error 1
>>>>
>>>>
>>>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>>>
>>>>> No, it needs implementation of the message pump between the module
>>>>> and
>>>>> the Skype API.
>>>>>
>>>>> It's probably kind of trivial, if no other problems I'm not aware
>>>>> of.
>>>>>
>>>>> I do not have a Mac to implement it, tough :-(.
>>>>>
>>>>> -giovanni
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> =========================================
>>>>> www.celliax.org
>>>>> via Pierlombardo 9, 20135 Milano
>>>>> Italy
>>>>> gmaruzz at celliax dot org
>>>>> Cell : +39-347-2665618
>>>>> Fax : +39-02-87390039
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>>>> wrote:
>>>>>> I'm not sure about that one.... I haven't tried lately because the
>>>>>> API
>>>>>> differs on the Mac last I looked at it.
>>>>>>
>>>>>> /b
>>>>>>
>>>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>>>
>>>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>>>
>>>>>>> Ivan
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From markmorreny at gmail.com  Sun Aug  9 08:57:06 2009
From: markmorreny at gmail.com (mark morreny)
Date: Sun, 9 Aug 2009 23:57:06 +0800
Subject: [Freeswitch-users] Scheduler in module
Message-ID: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>

Hi,

I would like to collect some data in my module which would listen to the
event socket and then a scheduler would kick off every so often to clean up
the data.

Does anyone know what is the best way to implement a scheduler in a mod?

Best Regards,
Mark
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From moises.silva at gmail.com  Sun Aug  9 09:05:47 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 9 Aug 2009 12:05:47 -0400
Subject: [Freeswitch-users] Scheduler in module
In-Reply-To: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
Message-ID: <c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>

On Sun, Aug 9, 2009 at 11:57 AM, mark morreny <markmorreny at gmail.com> wrote:

> Hi,
>
> I would like to collect some data in my module which would listen to the
> event socket and then a scheduler would kick off every so often to clean up
> the data.
>
> Does anyone know what is the best way to implement a scheduler in a mod?
>
> Best Regards,
> Mark
>

Just use the API that the FreeSWITCH core already has for you. Check
src/include/switch_scheduler.h for details, the comments in there should be
enough for you to learn to use it. If not, you can ask here more specific
questions.
-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
-------------- next part --------------
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From dujinfang at gmail.com  Sun Aug  9 11:02:34 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 10 Aug 2009 02:02:34 +0800
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
	<6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
	<4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
Message-ID: <092B4355-F911-4E37-A5D1-AC923AF9B8FB@gmail.com>

Ivan,

Good to know you are a cocoa dev. Unable to check in code right now,  
will send the diff to you offlist for now.

0) I'm not familiar with Mac dev, just tried my best
1) It doesn't work yet, but should be able to compile, sure you  
already have the Skype framework in place :)
2) if run the skype delegate from a threat, then cannot get event  
callback. e.g. mac_client.c works but mac_client2.c doesn't. Since  
skypiax is running in a thread, we need to figure out this first.
3) it uses Carbon, since I think we only need to low level api, no  
need to bother the complicate of Cocoa.
4) strsep shows some warning on compile, haven't figured out why
5) perhaps you should only add one interface in skypiax.conf.xml
6) do you want to run multi-instances like on Linux?
7) I really not sure if it will work or not :)

Let me know if it helps. I bet you can make it work. Also code will be  
in my branch soon.

7.


On Aug 9, 2009, at 11:34 PM, Ivan C Myrvold wrote:
> Yes, I am interested in this, and if you have any source I could have
> a look at it.
>
> Ivan
>
> Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
>
>>
>> On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
>>> Ciao Ivan,
>>>
>>> it seems that you do not have the libX11 **development** package
>>> installed.
>>>
>>> Unfortunately I don't know about OSX, so I cannot help you, but many
>>> on the list know.
>>> BTW: it will probably be of no use to you to compile mod_skypiax on
>>> OSX, because Skype for MACOSX works in another way than Skype for
>>> Linux.
>>
>> That's right.
>>
>>> If you know about MacOSX programming, please have a look at
>>> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
>>> probably be simple enough to add a message pump for MacOSX.
>>>
>>> -giovanni
>>>
>>>
>> Giovanni, I have a Mac and tried to get this work yesterday, but
>> haven't got it work. Will try further if I have time. However, I  
>> don't
>> think it's so useful because I don't know how to run and hence  
>> control
>> multi-skype instances on Mac.
>>
>> If someone interested to try this I can check the code into my  
>> branch.
>>
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> =========================================
>>> www.celliax.org
>>> via Pierlombardo 9, 20135 Milano
>>> Italy
>>> gmaruzz at celliax dot org
>>> Cell : +39-347-2665618
>>> Fax : +39-02-87390039
>>>
>>>
>>>
>>>
>>> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>
>>> wrote:
>>>> I tried to compile mod_skypiax, and am getting problem with X11. On
>>>> OS
>>>> X Leopard, X11 is installed in /usr/X11/lib/
>>>> See below.
>>>>
>>>> What can I do to get past this error?
>>>>
>>>> I can also let you ssh into my machine. Contact me off list in  
>>>> case.
>>>>
>>>> Ivan
>>>>
>>>> making all mod_skypiax
>>>> Compiling skypiax_protocol.c...
>>>> Compiling mod_skypiax.c...
>>>> mkdir .libs
>>>> Compiling mod_skypiax.c ...
>>>> Creating mod_skypiax.so...
>>>> ld: library not found for -lX11
>>>> collect2: ld returned 1 exit status
>>>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>>>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
>>>> Documents/
>>>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>>>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -
>>>> g -
>>>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -
>>>> shared -
>>>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat- 
>>>> namespace .libs/
>>>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/
>>>> Freeswitch/
>>>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>>>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/ 
>>>> xml/
>>>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
>>>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
>>>> libiconv.dylib /
>>>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
>>>> apr/.libs/
>>>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
>>>> lncurses -lX11
>>>> make[5]: *** [mod_skypiax.so] Error 1
>>>> make[4]: *** [all] Error 1
>>>> make[3]: *** [mod_skypiax-all] Error 1
>>>> make[2]: *** [all-recursive] Error 1
>>>>
>>>>
>>>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>>>
>>>>> No, it needs implementation of the message pump between the module
>>>>> and
>>>>> the Skype API.
>>>>>
>>>>> It's probably kind of trivial, if no other problems I'm not aware
>>>>> of.
>>>>>
>>>>> I do not have a Mac to implement it, tough :-(.
>>>>>
>>>>> -giovanni
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> =========================================
>>>>> www.celliax.org
>>>>> via Pierlombardo 9, 20135 Milano
>>>>> Italy
>>>>> gmaruzz at celliax dot org
>>>>> Cell : +39-347-2665618
>>>>> Fax : +39-02-87390039
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>>>> wrote:
>>>>>> I'm not sure about that one.... I haven't tried lately because  
>>>>>> the
>>>>>> API
>>>>>> differs on the Mac last I looked at it.
>>>>>>
>>>>>> /b
>>>>>>
>>>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>>>
>>>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>>>
>>>>>>> Ivan
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dujinfang at gmail.com  Sun Aug  9 17:44:32 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 10 Aug 2009 08:44:32 +0800
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <7b197bef0908090852t7243db6r761ddfce31093ed0@mail.gmail.com>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
	<6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
	<4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
	<7b197bef0908090852t7243db6r761ddfce31093ed0@mail.gmail.com>
Message-ID: <09A9F94D-D271-4B70-9BF0-4C9DBB14E5F5@gmail.com>


On Aug 9, 2009, at 11:52 PM, Giovanni Maruzzelli wrote:
> Seven,
> thanks a lot for your effort, please let your stuff be available,
> maybe Ivan can make use of it!
>
svn diff http://svn.freeswitch.org/svn/freeswitch/branches/seven -r  
14473
:14475

When this done I think it's better to split codes into

skypiax_protocol.c
skypiax_protocol_mac.c
skypiax_protocol_linux.c
skypiax_protocol_windows.c

:)

7.
> Ivan,
> in the file src/mod/endpoints/mod_skypiax/skypiax_protocol.c add you
> will find #ifdef WIN32 .
>
> it conditional compiles code between WIN32 and linux.
>
> You need to add another #ifdef, so it will compile for OSX.
>
> You will probably be able to use the same pipe mechanism as in Linux
> (normal POSIX pipes).
> You will for sure need to implement the part that deals with the Skype
> API. Maybe it will be not much more than reusing the example code to
> interact with the API.
>
> Please, let us know how it goes, and feel *very* free to ask for  
> further info.
>
> -giovanni
>
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
>
>
>
> On Sun, Aug 9, 2009 at 5:34 PM, Ivan C Myrvold<ivan at myrvold.org>  
> wrote:
>> Yes, I am interested in this, and if you have any source I could have
>> a look at it.
>>
>> Ivan
>>
>> Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
>>
>>>
>>> On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
>>>> Ciao Ivan,
>>>>
>>>> it seems that you do not have the libX11 **development** package
>>>> installed.
>>>>
>>>> Unfortunately I don't know about OSX, so I cannot help you, but  
>>>> many
>>>> on the list know.
>>>> BTW: it will probably be of no use to you to compile mod_skypiax on
>>>> OSX, because Skype for MACOSX works in another way than Skype for
>>>> Linux.
>>>
>>> That's right.
>>>
>>>> If you know about MacOSX programming, please have a look at
>>>> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
>>>> probably be simple enough to add a message pump for MacOSX.
>>>>
>>>> -giovanni
>>>>
>>>>
>>> Giovanni, I have a Mac and tried to get this work yesterday, but
>>> haven't got it work. Will try further if I have time. However, I  
>>> don't
>>> think it's so useful because I don't know how to run and hence  
>>> control
>>> multi-skype instances on Mac.
>>>
>>> If someone interested to try this I can check the code into my  
>>> branch.
>>>
>>>>
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> =========================================
>>>> www.celliax.org
>>>> via Pierlombardo 9, 20135 Milano
>>>> Italy
>>>> gmaruzz at celliax dot org
>>>> Cell : +39-347-2665618
>>>> Fax : +39-02-87390039
>>>>
>>>>
>>>>
>>>>
>>>> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>
>>>> wrote:
>>>>> I tried to compile mod_skypiax, and am getting problem with X11.  
>>>>> On
>>>>> OS
>>>>> X Leopard, X11 is installed in /usr/X11/lib/
>>>>> See below.
>>>>>
>>>>> What can I do to get past this error?
>>>>>
>>>>> I can also let you ssh into my machine. Contact me off list in  
>>>>> case.
>>>>>
>>>>> Ivan
>>>>>
>>>>> making all mod_skypiax
>>>>> Compiling skypiax_protocol.c...
>>>>> Compiling mod_skypiax.c...
>>>>> mkdir .libs
>>>>> Compiling mod_skypiax.c ...
>>>>> Creating mod_skypiax.so...
>>>>> ld: library not found for -lX11
>>>>> collect2: ld returned 1 exit status
>>>>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>>>>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
>>>>> Documents/
>>>>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>>>>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -
>>>>> g -
>>>>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -
>>>>> shared -
>>>>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat- 
>>>>> namespace .libs/
>>>>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/
>>>>> Freeswitch/
>>>>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>>>>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/ 
>>>>> xml/
>>>>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch. 
>>>>> 09-08-09/
>>>>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
>>>>> libiconv.dylib /
>>>>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
>>>>> apr/.libs/
>>>>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto - 
>>>>> lz -
>>>>> lncurses -lX11
>>>>> make[5]: *** [mod_skypiax.so] Error 1
>>>>> make[4]: *** [all] Error 1
>>>>> make[3]: *** [mod_skypiax-all] Error 1
>>>>> make[2]: *** [all-recursive] Error 1
>>>>>
>>>>>
>>>>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>>>>
>>>>>> No, it needs implementation of the message pump between the  
>>>>>> module
>>>>>> and
>>>>>> the Skype API.
>>>>>>
>>>>>> It's probably kind of trivial, if no other problems I'm not aware
>>>>>> of.
>>>>>>
>>>>>> I do not have a Mac to implement it, tough :-(.
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Sincerely,
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> =========================================
>>>>>> www.celliax.org
>>>>>> via Pierlombardo 9, 20135 Milano
>>>>>> Italy
>>>>>> gmaruzz at celliax dot org
>>>>>> Cell : +39-347-2665618
>>>>>> Fax : +39-02-87390039
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>>>>> wrote:
>>>>>>> I'm not sure about that one.... I haven't tried lately because  
>>>>>>> the
>>>>>>> API
>>>>>>> differs on the Mac last I looked at it.
>>>>>>>
>>>>>>> /b
>>>>>>>
>>>>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>>>>
>>>>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>>>>
>>>>>>>> Ivan
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From godson.g at gmail.com  Sun Aug  9 22:26:23 2009
From: godson.g at gmail.com (Godson Gera)
Date: Mon, 10 Aug 2009 10:56:23 +0530
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
Message-ID: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>

Hi FS Team,


        The files.freeswitch.org is resetting connection since 3 days. As a
result I was not able to download latest release of FS. Got the trunk
version from svn. But still it suffers from the lack of sound files.  When
ever I do 'make uhd-sounds-install' http://files.freeswitch.org resets
connection immediately wget tries 20 times and gives up. Other users on IRC
also reported this issue.

-- 
Thanks & Regards,
Godson Gera
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From ryder86 at googlemail.com  Sun Aug  9 22:44:46 2009
From: ryder86 at googlemail.com (Artem Vasiliev)
Date: Mon, 10 Aug 2009 09:44:46 +0400
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <5a8712120908071350v2eac4613j7fd53e5158680742@mail.gmail.com>
References: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
	<ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>
	<b0a8792f0908071150x68f3e3b5v596bad90bc56d1ad@mail.gmail.com>
	<5a8712120908071350v2eac4613j7fd53e5158680742@mail.gmail.com>
Message-ID: <cb6be45b0908092244w3d75dd7i7f0bab93682ffdc2@mail.gmail.com>

>Jo?o Mesquita
Nice thing. I made similar one for Event_Socket (C#).

>Raffaele P. Guidi
Yes, I'm working on CTI. We use a CTI application called WebAgent, but
it's TAPI-based, so we have to create a special DLL to make it work
with FreeSWITCH.

>Kevin Green
Thanks for explanation. It works as you said.
Answer - auto (let it be for now)
Hold/Unhold - via uuid_hold (hold button doesn't work)
Hangup - the only that works properly
Make call - made simple javascript that dials both ends and then bridges them

2009/8/8 Jo?o Mesquita <jmesquita at gmail.com>:
> Stay tuned on fsgui. It will get there really soon.
>
> jmesquita
>
> On Fri, Aug 7, 2009 at 3:50 PM, Raffaele P. Guidi
> <raffaele.p.guidi at gmail.com> wrote:
>>
>> Maybe Artem is interested in CTI (computer telephony integration) -
>> click2dial, opening a url (or statrting a program) on incoming call...?
>>
>> On Fri, Aug 7, 2009 at 17:00, Kevin Green <kevin at johnnyvoip.com> wrote:
>>>
>>> From what I am aware you can't use FreeSWITCH to control a softphone
>>> directly though you can make it do things that will have a similar end
>>> result. You could set eyeBeam to auto-answer calls if you want them to
>>> answer right away or orginiate a call that is auto-answered but not bridge
>>> the call?until a user on the eyeBeam presses a digit or a socket control
>>> tells it to connect the two ends. You can also use FreeSWITCH to place the
>>> line on hold using event sockets, this will place it on hold in the server
>>> and not directly like placing it on hold in eyeBeam (i.e. the hold button in
>>> eyeBeam likely wont show it as being on hold).
>>> Beyond that if you want to directly control the clients you would need to
>>> look at getting an API access into the eyeBeam client.
>>> I hope this will help.
>>> Regards,
>>> ? ?Kevin Green
>>>
>>>
>>> On Fri, Aug 7, 2009 at 7:02 AM, Artem Vasiliev <ryder86 at googlemail.com>
>>> wrote:
>>>>
>>>> No, I don't want to make softphone from FreeSwitch
>>>>
>>>> I have FS and several users with eyeBeam softphones. I need to control
>>>> those eyeBeams
>>>>
>>>> >You can run FreeSWITCH as a softphone and control it.
>>>> >http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>>>>
>>>> >2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>
>>>>
>>>> >> Hi
>>>> >>
>>>> >> I have FreeSwitch and external application, which communicates to it
>>>> >> via
>>>> >> event socket - listens for events for certain number and gives some
>>>> >> commands.
>>>> >> Is it possible for this application to control client softphones, for
>>>> >> example, make them answer or hold, using the event socket or other
>>>> >> FreeSwitch capabilities?


From diego.viola at gmail.com  Sun Aug  9 22:57:00 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 10 Aug 2009 01:57:00 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
Message-ID: <86a32abc0908092257k3cc7695ay3a8017a874bf186a@mail.gmail.com>

I been having this problem also.

On Mon, Aug 10, 2009 at 1:26 AM, Godson Gera <godson.g at gmail.com> wrote:

> Hi FS Team,
>
>
>         The files.freeswitch.org is resetting connection since 3 days. As
> a result I was not able to download latest release of FS. Got the trunk
> version from svn. But still it suffers from the lack of sound files.  When
> ever I do 'make uhd-sounds-install' http://files.freeswitch.org resets
> connection immediately wget tries 20 times and gives up. Other users on IRC
> also reported this issue.
>
> --
> Thanks & Regards,
> Godson Gera
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Sun Aug  9 23:06:05 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 10 Aug 2009 02:06:05 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908092257k3cc7695ay3a8017a874bf186a@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<86a32abc0908092257k3cc7695ay3a8017a874bf186a@mail.gmail.com>
Message-ID: <86a32abc0908092306k446f5118oa6526769a26fb76c@mail.gmail.com>

http://files-sync.freeswitch.org/

On Mon, Aug 10, 2009 at 1:57 AM, Diego Viola <diego.viola at gmail.com> wrote:

> I been having this problem also.
>
> On Mon, Aug 10, 2009 at 1:26 AM, Godson Gera <godson.g at gmail.com> wrote:
>
>> Hi FS Team,
>>
>>
>>         The files.freeswitch.org is resetting connection since 3 days. As
>> a result I was not able to download latest release of FS. Got the trunk
>> version from svn. But still it suffers from the lack of sound files.  When
>> ever I do 'make uhd-sounds-install' http://files.freeswitch.org resets
>> connection immediately wget tries 20 times and gives up. Other users on IRC
>> also reported this issue.
>>
>> --
>> Thanks & Regards,
>> Godson Gera
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From velu.technical at gmail.com  Sun Aug  9 23:38:38 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Mon, 10 Aug 2009 12:08:38 +0530
Subject: [Freeswitch-users] ALARM signal in esl libraries
Message-ID: <1452e2980908092338j15c4e33cn2cff799bb64464d0@mail.gmail.com>

Dear All,
       I have registered  ALARM signal  in my perl program to handle the
DTMF digit timeout.
When ALARM signal generated  the  connection with  ESL  is  automatically
closed.

I have checked the connection with "connected: function, it returns 0.

Why the connection was closed?
Is there any idea to alive the connection after ALARM signal generation??

Please help me.........

Thanks,
Velusamy
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From lakindia89 at gmail.com  Mon Aug 10 01:00:49 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Mon, 10 Aug 2009 01:00:49 -0700 (PDT)
Subject: [Freeswitch-users] Error while creating object
In-Reply-To: <7d79b3930908060347xe5be545yfeeafad761aba274@mail.gmail.com>
References: <7d79b3930908060347xe5be545yfeeafad761aba274@mail.gmail.com>
Message-ID: <24895716.post@talk.nabble.com>


Can any one please say what I did wrong here?

regards,
Lakshmanan G.


lakshmanan wrote:
> 
> Hi all,
> Greets.
> 
> I am in the process of controlling the freeswitch with perl.
> I have read about mod_perl and I wrote some scripts to test which works
> fine.
> Yesterday I tried to access the digit_set function.
> So I create an object for the freeswitch::DTMF.
> But it reported the following error.
> 
> 2009-08-06 15:53:46 [ERR] mod_perl.c:69 Perl_safe_eval() [require
> '/usr/local/freeswitch/conf/test.pl';]
> No matching function for overloaded 'new_DTMF' at
> /usr/local/freeswitch/perl/freeswitch.pm line 197.
> Compilation failed in require at (eval 2) line 1.
> 
> Here is my code.
> 
> #!/usr/bin/perl
> use strict;
> use freeswitch;
> our $session;
> $session->execute("bridge","user/1010");
> my $sess=&freeswitch::DTMF::new;
> return 1;
> 
> The bridge is working fine. But while creating the object it said error.
> 
> Can any one explain why this happens and how can I correct it?
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Error-while-creating-object-tp24849065p24895716.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From wangdq.no1 at gmail.com  Sun Aug  9 23:16:39 2009
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Mon, 10 Aug 2009 14:16:39 +0800
Subject: [Freeswitch-users] can freeswitch group_call to gateway extension ?
Message-ID: <ecd05ce30908092316s66051433p506df518942ec93f@mail.gmail.com>

Hello every one :
  I tested FS. and when I use group_call.  But I can't call the extension in
group and not registered in FS.
  Why ? and What  ?
  thanks .
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From michal.bielicki at halo2.pl  Mon Aug 10 02:48:33 2009
From: michal.bielicki at halo2.pl (Michal Bielicki)
Date: Mon, 10 Aug 2009 11:48:33 +0200
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>
Message-ID: <17FB8F3C-0661-4594-912E-9541476DFDAC@halo2.pl>

It builds on both.
Can you tell me how you are trying to build ?

Am 08.08.2009 um 06:36 schrieb Michael Jerris:

> This is not currently a supported platform, it only builds on 64 bit
> right now I think on solaris.
>
> Mike
>
> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>
>> Hello,
>>
>> Does anyone have, or know where to get, a pre-compiled copy of
>> FreeSwitch for Solaris 10/x86?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. info at halokwadrat.de
e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
Hauptgesch?ftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 0000153539

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From vkozak at abisoft.spb.ru  Mon Aug 10 03:00:11 2009
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Mon, 10 Aug 2009 14:00:11 +0400
Subject: [Freeswitch-users] Fw: FreeSwitch doesn't play music on hold
	forbriged channel
Message-ID: <C5CEF36D42B347B49C82CDE05C3B0E5D@abisoft.biz>

2009-08-12 03:02:03 [DEBUG] switch_core_session.c:706 switch_core_session_queue_private_event() Send signal sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:228 audio_bridge_thread() sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes receive message [UNBRIDGE]
2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
2009-08-12 03:02:03 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes Command Execute hold()
2009-08-12 03:02:03 [DEBUG] switch_ivr.c:1054 switch_ivr_hold() sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes receive message [HOLD]
2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
2009-08-12 03:02:03 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes entering state [calling]
2009-08-12 03:02:03 [WARNING] switch_ivr_async.c:2239 switch_ivr_broadcast() Channel [sofia/internal/1001 at 172.26.200.250][local_stream://moh] already broadcasting...broadcast aborted
2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:231 audio_bridge_thread() sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes receive message [BRIDGE]
2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:233 audio_bridge_thread() Send signal sofia/internal/1001 at 172.26.200.250 [BREAK]
2009-08-12 03:02:03 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes entering state [ready]
2009-08-12 03:02:03 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP:
v=0
o=- 6 3 IN IP4 172.26.10.39
s=CounterPath eyeBeam 1.5
c=IN IP4 172.26.10.39
t=0 0
m=audio 15712 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=recvonly

2009-08-12 03:02:03 [DEBUG] sofia_glue.c:2483 sofia_glue_negotiate_sdp() Our existing sdp is still good [PCMU 172.26.10.39:15712], let's keep it.
2009-08-12 03:02:03 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101

  ----- Original Message ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, August 09, 2009 12:19 AM
  Subject: Re: [Freeswitch-users] FreeSwitch doesn't play music on holdforbriged channel


  What do the debug logs on fs say when you try to put the call on hold? 


  Mike


  On Aug 6, 2009, at 1:01 PM, Kozak Vladimir wrote:



    The scenario is the following:
    FS User A dial an extension
    Extention opens outbound socket channel to my application
    My application bridges the call to FS User B
    The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id
    The application sends hold to the bridged channel using SendMsg with Other-leg-unique-id
    User B is placed on hold but no music on hold is played to the caller (User A)


    I have outbound socket channel and the following sequence of commands/event:
    listening on [any] 8084 ...
    connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250] 34000
    connect





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org



  The scenario is the following:
  FS User A dial an extension
  Extention opens outbound socket channel to my application
  My application bridges the call to FS User B
  The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id
  The application sends hold to the bridged channel using SendMsg with Other-leg-unique-id
  User B is placed on hold but no music on hold is played to the caller (User A)


  I have outbound socket channel and the following sequence of commands/event:
  listening on [any] 8084 ...
  connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250] 34000
  connect

  myevents

  SendMsg
  call-command: execute
  execute-app-name: bridge
  execute-app-arg:user/1000 at uat.agent.starpoundtech.net

  Channel-Username: 1001
  Channel-Dialplan: XML
  Channel-Caller-ID-Name: 1001
  Channel-Caller-ID-Number: 1001
  Channel-Network-Addr: 172.26.10.39
  Channel-Destination-Number: 6666
  Channel-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Channel-Source: mod_sofia
  Channel-Context: default
  Channel-Channel-Name: sofia/internal/1001%40172.26.200.250
  Channel-Profile-Index: 1
  Channel-Profile-Created-Time: 1249142681680114
  Channel-Channel-Created-Time: 1249142681680114
  Channel-Channel-Answered-Time: 0
  Channel-Channel-Progress-Time: 0
  Channel-Channel-Progress-Media-Time: 1249142681809352
  Channel-Channel-Hangup-Time: 0
  Channel-Channel-Transfer-Time: 0
  Channel-Screen-Bit: true
  Channel-Privacy-Hide-Name: false
  Channel-Privacy-Hide-Number: false
  Channel-State: CS_EXECUTE
  Channel-State-Number: 4
  Channel-Name: sofia/internal/1001%40172.26.200.250
  Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Call-Direction: inbound
  Answer-State: early
  Channel-Read-Codec-Name: PCMU
  Channel-Read-Codec-Rate: 8000
  Channel-Write-Codec-Name: PCMU
  Channel-Write-Codec-Rate: 8000
  Caller-Username: 1001
  Caller-Dialplan: XML
  Caller-Caller-ID-Name: 1001
  Caller-Caller-ID-Number: 1001
  Caller-Network-Addr: 172.26.10.39
  Caller-Destination-Number: 6666
  Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Caller-Source: mod_sofia
  Caller-Context: default
  Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
  Caller-Profile-Index: 1
  Caller-Profile-Created-Time: 1249142681680114
  Caller-Channel-Created-Time: 1249142681680114
  Caller-Channel-Answered-Time: 0
  Caller-Channel-Progress-Time: 0
  Caller-Channel-Progress-Media-Time: 1249142681809352
  Caller-Channel-Hangup-Time: 0
  Caller-Channel-Transfer-Time: 0
  Caller-Screen-Bit: true
  Caller-Privacy-Hide-Name: false
  Caller-Privacy-Hide-Number: false
  variable_sip_received_ip: 172.26.10.39
  variable_sip_received_port: 13488
  variable_sip_via_protocol: udp
  variable_sip_authorized: true
  variable_sip_mailbox: 1001
  variable_sip_auth_username: 1001
  variable_sip_auth_realm: 172.26.200.250
  variable_mailbox: 1001
  variable_toll_allow: domestic,international,local
  variable_accountcode: 1001
  variable_user_context: default
  variable_effective_caller_id_name: Extension%201001
  variable_effective_caller_id_number: 1001
  variable_outbound_caller_id_name: StarPound%20FreeSWITCH
  variable_outbound_caller_id_number: 0000000000
  variable_callgroup: techsupport
  variable_sip_from_user: 1001
  variable_sip_from_uri: 1001%40172.26.200.250
  variable_sip_from_host: 172.26.200.250
  variable_sip_from_user_stripped: 1001
  variable_sip_from_tag: bd11f93c
  variable_sofia_profile_name: internal
  variable_sip_req_user: 6666
  variable_sip_req_uri: 6666%40172.26.200.250
  variable_sip_req_host: 172.26.200.250
  variable_sip_to_user: 6666
  variable_sip_to_uri: 6666%40172.26.200.250
  variable_sip_to_host: 172.26.200.250
  variable_sip_contact_user: 1001
  variable_sip_contact_port: 13488
  variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
  variable_sip_contact_host: 172.26.10.39
  variable_channel_name: sofia/internal/1001%40172.26.200.250
  variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
  variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
  variable_sip_via_host: 172.26.10.39
  variable_sip_via_port: 13488
  variable_sip_via_rport: 13488
  variable_max_forwards: 70
  variable_presence_id: 1001%40172.26.200.250
  variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
  variable_remote_media_ip: 172.26.10.39
  variable_remote_media_port: 29826
  variable_read_codec: PCMU
  variable_read_rate: 8000
  variable_write_codec: PCMU
  variable_write_rate: 8000
  variable_use_profile: nat
  variable_record_stereo: true
  variable_transfer_fallback_extension: operator
  variable_numbering_plan: US
  variable_default_areacode: 918
  variable_default_gateway: example.com
  variable_user_name: default
  variable_domain_name: 172.26.200.250
  variable_current_application_data: 172.26.200.251%3A8084%20async%20full
  variable_current_application: socket
  variable_socket_host: 172.26.200.251
  variable_local_media_ip: 172.26.200.250
  variable_local_media_port: 29370
  variable_endpoint_disposition: EARLY%20MEDIA
  variable_sip_nat_detected: true
  Content-Type: command/reply
  Reply-Text: %2BOK%0A
  Socket-Mode: async
  Control: full

  Content-Type: command/reply
  Reply-Text: +OK Events Enabled

  Content-Type: command/reply
  Reply-Text: +OK

  Content-Length: 1541
  Content-Type: text/event-plain

  Channel-State: CS_EXECUTE
  Channel-State-Number: 4
  Channel-Name: sofia/internal/1001%40172.26.200.250
  Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Call-Direction: inbound
  Answer-State: early
  Channel-Read-Codec-Name: PCMU
  Channel-Read-Codec-Rate: 8000
  Channel-Write-Codec-Name: PCMU
  Channel-Write-Codec-Rate: 8000
  Caller-Username: 1001
  Caller-Dialplan: XML
  Caller-Caller-ID-Name: 1001
  Caller-Caller-ID-Number: 1001
  Caller-Network-Addr: 172.26.10.39
  Caller-Destination-Number: 6666
  Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Caller-Source: mod_sofia
  Caller-Context: default
  Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
  Caller-Profile-Index: 1
  Caller-Profile-Created-Time: 1249142681680114
  Caller-Channel-Created-Time: 1249142681680114
  Caller-Channel-Answered-Time: 0
  Caller-Channel-Progress-Time: 0
  Caller-Channel-Progress-Media-Time: 1249142681809352
  Caller-Channel-Hangup-Time: 0
  Caller-Channel-Transfer-Time: 0
  Caller-Screen-Bit: true
  Caller-Privacy-Hide-Name: false
  Caller-Privacy-Hide-Number: false
  Application: bridge
  Application-Data: user/1000%40uat.agent.starpoundtech.net
  Event-Name: CHANNEL_EXECUTE
  Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
  FreeSWITCH-Hostname: centos4-4-vm
  FreeSWITCH-IPv4: 172.26.200.250
  FreeSWITCH-IPv6: %3A%3A1
  Event-Date-Local: 2009-08-01%2020%3A04%3A51
  Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A51%20GMT
  Event-Date-Timestamp: 1249142691754598
  Event-Calling-File: switch_core_session.c
  Event-Calling-Function: switch_core_session_exec
  Event-Calling-Line-Number: 1333

  Content-Length: 5242
  Content-Type: text/event-plain

  Channel-State: CS_EXECUTE
  Channel-State-Number: 4
  Channel-Name: sofia/internal/1001%40172.26.200.250
  Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Call-Direction: inbound
  Answer-State: answered
  Channel-Read-Codec-Name: PCMU
  Channel-Read-Codec-Rate: 8000
  Channel-Write-Codec-Name: PCMU
  Channel-Write-Codec-Rate: 8000
  Caller-Username: 1001
  Caller-Dialplan: XML
  Caller-Caller-ID-Name: 1001
  Caller-Caller-ID-Number: 1001
  Caller-Network-Addr: 172.26.10.39
  Caller-Destination-Number: 6666
  Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Caller-Source: mod_sofia
  Caller-Context: default
  Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
  Caller-Profile-Index: 1
  Caller-Profile-Created-Time: 1249142681680114
  Caller-Channel-Created-Time: 1249142681680114
  Caller-Channel-Answered-Time: 1249142692414509
  Caller-Channel-Progress-Time: 1249142691898434
  Caller-Channel-Progress-Media-Time: 1249142681809352
  Caller-Channel-Hangup-Time: 0
  Caller-Channel-Transfer-Time: 0
  Caller-Screen-Bit: true
  Caller-Privacy-Hide-Name: false
  Caller-Privacy-Hide-Number: false
  Other-Leg-Username: 1001
  Other-Leg-Dialplan: XML
  Other-Leg-Caller-ID-Name: Extension%201001
  Other-Leg-Caller-ID-Number: 1001
  Other-Leg-Network-Addr: 172.26.10.39
  Other-Leg-Destination-Number: sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
  Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
  Other-Leg-Source: mod_sofia
  Other-Leg-Context: default
  Other-Leg-Channel-Name: sofia/internal/sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
  Other-Leg-Screen-Bit: true
  Other-Leg-Privacy-Hide-Name: false
  Other-Leg-Privacy-Hide-Number: false
  variable_sip_received_ip: 172.26.10.39
  variable_sip_received_port: 13488
  variable_sip_via_protocol: udp
  variable_sip_authorized: true
  variable_sip_mailbox: 1001
  variable_sip_auth_username: 1001
  variable_sip_auth_realm: 172.26.200.250
  variable_mailbox: 1001
  variable_toll_allow: domestic,international,local
  variable_accountcode: 1001
  variable_user_context: default
  variable_effective_caller_id_name: Extension%201001
  variable_effective_caller_id_number: 1001
  variable_outbound_caller_id_name: StarPound%20FreeSWITCH
  variable_outbound_caller_id_number: 0000000000
  variable_callgroup: techsupport
  variable_sip_from_user: 1001
  variable_sip_from_uri: 1001%40172.26.200.250
  variable_sip_from_host: 172.26.200.250
  variable_sip_from_user_stripped: 1001
  variable_sip_from_tag: bd11f93c
  variable_sofia_profile_name: internal
  variable_sip_req_user: 6666
  variable_sip_req_uri: 6666%40172.26.200.250
  variable_sip_req_host: 172.26.200.250
  variable_sip_to_user: 6666
  variable_sip_to_uri: 6666%40172.26.200.250
  variable_sip_to_host: 172.26.200.250
  variable_sip_contact_user: 1001
  variable_sip_contact_port: 13488
  variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
  variable_sip_contact_host: 172.26.10.39
  variable_channel_name: sofia/internal/1001%40172.26.200.250
  variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
  variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
  variable_sip_via_host: 172.26.10.39
  variable_sip_via_port: 13488
  variable_sip_via_rport: 13488
  variable_max_forwards: 70
  variable_presence_id: 1001%40172.26.200.250
  variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
  variable_remote_media_ip: 172.26.10.39
  variable_remote_media_port: 29826
  variable_read_codec: PCMU
  variable_read_rate: 8000
  variable_write_codec: PCMU
  variable_write_rate: 8000
  variable_use_profile: nat
  variable_record_stereo: true
  variable_transfer_fallback_extension: operator
  variable_numbering_plan: US
  variable_default_areacode: 918
  variable_default_gateway: example.com
  variable_user_name: default
  variable_domain_name: 172.26.200.250
  variable_socket_host: 172.26.200.251
  variable_local_media_ip: 172.26.200.250
  variable_local_media_port: 29370
  variable_endpoint_disposition: EARLY%20MEDIA
  variable_current_application_data: user/1000%40uat.agent.starpoundtech.net
  variable_current_application: bridge
  variable_dialed_user: 1000
  variable_dialed_domain: uat.agent.starpoundtech.net
  variable_originate_disposition: failure
  variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
  variable_sip_redirect_contact_user_0: 1000
  variable_sip_redirect_contact_host_0: 172.26.10.39
  variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
  variable_sip_nat_detected: true
  Event-Name: CHANNEL_ANSWER
  Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
  FreeSWITCH-Hostname: centos4-4-vm
  FreeSWITCH-IPv4: 172.26.200.250
  FreeSWITCH-IPv6: %3A%3A1
  Event-Date-Local: 2009-08-01%2020%3A04%3A52
  Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
  Event-Date-Timestamp: 1249142692414509
  Event-Calling-File: switch_channel.c
  Event-Calling-Function: switch_channel_perform_mark_answered
  Event-Calling-Line-Number: 1776

  Content-Length: 5233
  Content-Type: text/event-plain

  Channel-State: CS_EXECUTE
  Channel-State-Number: 4
  Channel-Name: sofia/internal/1001%40172.26.200.250
  Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Call-Direction: inbound
  Answer-State: answered
  Channel-Read-Codec-Name: PCMU
  Channel-Read-Codec-Rate: 8000
  Channel-Write-Codec-Name: PCMU
  Channel-Write-Codec-Rate: 8000
  Caller-Username: 1001
  Caller-Dialplan: XML
  Caller-Caller-ID-Name: 1001
  Caller-Caller-ID-Number: 1001
  Caller-Network-Addr: 172.26.10.39
  Caller-Destination-Number: 6666
  Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Caller-Source: mod_sofia
  Caller-Context: default
  Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
  Caller-Profile-Index: 1
  Caller-Profile-Created-Time: 1249142681680114
  Caller-Channel-Created-Time: 1249142681680114
  Caller-Channel-Answered-Time: 1249142692414509
  Caller-Channel-Progress-Time: 1249142691898434
  Caller-Channel-Progress-Media-Time: 1249142681809352
  Caller-Channel-Hangup-Time: 0
  Caller-Channel-Transfer-Time: 0
  Caller-Screen-Bit: true
  Caller-Privacy-Hide-Name: false
  Caller-Privacy-Hide-Number: false
  Other-Leg-Username: 1001
  Other-Leg-Dialplan: XML
  Other-Leg-Caller-ID-Name: Extension%201001
  Other-Leg-Caller-ID-Number: 1001
  Other-Leg-Network-Addr: 172.26.10.39
  Other-Leg-Destination-Number: sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
  Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
  Other-Leg-Source: mod_sofia
  Other-Leg-Context: default
  Other-Leg-Channel-Name: sofia/internal/sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
  Other-Leg-Screen-Bit: true
  Other-Leg-Privacy-Hide-Name: false
  Other-Leg-Privacy-Hide-Number: false
  variable_sip_received_ip: 172.26.10.39
  variable_sip_received_port: 13488
  variable_sip_via_protocol: udp
  variable_sip_authorized: true
  variable_sip_mailbox: 1001
  variable_sip_auth_username: 1001
  variable_sip_auth_realm: 172.26.200.250
  variable_mailbox: 1001
  variable_toll_allow: domestic,international,local
  variable_accountcode: 1001
  variable_user_context: default
  variable_effective_caller_id_name: Extension%201001
  variable_effective_caller_id_number: 1001
  variable_outbound_caller_id_name: StarPound%20FreeSWITCH
  variable_outbound_caller_id_number: 0000000000
  variable_callgroup: techsupport
  variable_sip_from_user: 1001
  variable_sip_from_uri: 1001%40172.26.200.250
  variable_sip_from_host: 172.26.200.250
  variable_sip_from_user_stripped: 1001
  variable_sip_from_tag: bd11f93c
  variable_sofia_profile_name: internal
  variable_sip_req_user: 6666
  variable_sip_req_uri: 6666%40172.26.200.250
  variable_sip_req_host: 172.26.200.250
  variable_sip_to_user: 6666
  variable_sip_to_uri: 6666%40172.26.200.250
  variable_sip_to_host: 172.26.200.250
  variable_sip_contact_user: 1001
  variable_sip_contact_port: 13488
  variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
  variable_sip_contact_host: 172.26.10.39
  variable_channel_name: sofia/internal/1001%40172.26.200.250
  variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
  variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
  variable_sip_via_host: 172.26.10.39
  variable_sip_via_port: 13488
  variable_sip_via_rport: 13488
  variable_max_forwards: 70
  variable_presence_id: 1001%40172.26.200.250
  variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
  variable_remote_media_ip: 172.26.10.39
  variable_remote_media_port: 29826
  variable_read_codec: PCMU
  variable_read_rate: 8000
  variable_write_codec: PCMU
  variable_write_rate: 8000
  variable_use_profile: nat
  variable_record_stereo: true
  variable_transfer_fallback_extension: operator
  variable_numbering_plan: US
  variable_default_areacode: 918
  variable_default_gateway: example.com
  variable_user_name: default
  variable_domain_name: 172.26.200.250
  variable_socket_host: 172.26.200.251
  variable_local_media_ip: 172.26.200.250
  variable_local_media_port: 29370
  variable_current_application_data: user/1000%40uat.agent.starpoundtech.net
  variable_current_application: bridge
  variable_dialed_user: 1000
  variable_dialed_domain: uat.agent.starpoundtech.net
  variable_sip_redirect_contact_user_0: 1000
  variable_sip_redirect_contact_host_0: 172.26.10.39
  variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
  variable_sip_nat_detected: true
  variable_endpoint_disposition: ANSWER
  variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
  variable_originate_disposition: SUCCESS
  Event-Name: CHANNEL_BRIDGE
  Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
  FreeSWITCH-Hostname: centos4-4-vm
  FreeSWITCH-IPv4: 172.26.200.250
  FreeSWITCH-IPv6: %3A%3A1
  Event-Date-Local: 2009-08-01%2020%3A04%3A52
  Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
  Event-Date-Timestamp: 1249142692414509
  Event-Calling-File: switch_ivr_bridge.c
  Event-Calling-Function: switch_ivr_multi_threaded_bridge
  Event-Calling-Line-Number: 828

  SendMsg 94b59a38-57c4-4703-9c6e-9985d832d119
  call-command: execute
  execute-app-name: hold

  Content-Type: command/reply
  Reply-Text: +OK




  - I don't see the variable hold_music ... did you remove it? 

  - I didn't. Moreover, I tried to set it explicitly using api uuid_setvar.


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From markmorreny at gmail.com  Mon Aug 10 03:58:04 2009
From: markmorreny at gmail.com (mark morreny)
Date: Mon, 10 Aug 2009 18:58:04 +0800
Subject: [Freeswitch-users] Fwd:  Scheduler in module
In-Reply-To: <c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
Message-ID: <20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>

Hi,

Thank you for pointing out sched_api.

What about if I want to do a recurring schedule. is it possible or I should
just call sched_api another time in the execution of my clean up code.

Thanks,
Mark

---------- Forwarded message ----------
From: Moises Silva <moises.silva at gmail.com>
Date: Mon, Aug 10, 2009 at 12:05 AM
Subject: Re: [Freeswitch-users] Scheduler in module
To: freeswitch-users at lists.freeswitch.org


On Sun, Aug 9, 2009 at 11:57 AM, mark morreny <markmorreny at gmail.com> wrote:

> Hi,
>
> I would like to collect some data in my module which would listen to the
> event socket and then a scheduler would kick off every so often to clean up
> the data.
>
> Does anyone know what is the best way to implement a scheduler in a mod?
>
> Best Regards,
> Mark
>

Just use the API that the FreeSWITCH core already has for you. Check
src/include/switch_scheduler.h for details, the comments in there should be
enough for you to learn to use it. If not, you can ask here more specific
questions.
-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From mike at jerris.com  Mon Aug 10 05:01:20 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 Aug 2009 08:01:20 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
Message-ID: <266B3B3C-73E6-4264-AFFA-73551794365F@jerris.com>

Anyone experiencing this issue please email me the source ip and the  
address that files.freeswitch.org is resolving as.

Mike

On Aug 10, 2009, at 1:26 AM, Godson Gera <godson.g at gmail.com> wrote:

> Hi FS Team,
>
>
>         The files.freeswitch.org is resetting connection since 3  
> days. As a result I was not able to download latest release of FS.  
> Got the trunk version from svn. But still it suffers from the lack  
> of sound files.  When ever I do 'make uhd-sounds-install' http://files.freeswitch.org 
>  resets connection immediately wget tries 20 times and gives up.  
> Other users on IRC also reported this issue.
>
> -- 
> Thanks & Regards,
> Godson Gera
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Mon Aug 10 05:04:39 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 Aug 2009 08:04:39 -0400
Subject: [Freeswitch-users] can freeswitch group_call to gateway
	extension ?
In-Reply-To: <ecd05ce30908092316s66051433p506df518942ec93f@mail.gmail.com>
References: <ecd05ce30908092316s66051433p506df518942ec93f@mail.gmail.com>
Message-ID: <261A7939-0C0E-48DF-B4D4-9A02D5616898@jerris.com>

Can you rephrase your question?

On Aug 10, 2009, at 2:16 AM, daqiang wang <wangdq.no1 at gmail.com> wrote:

> Hello every one :
>   I tested FS. and when I use group_call.  But I can't call the  
> extension in group and not registered in FS.
>   Why ? and What  ?
>   thanks .
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From t.mahe at telemaque.fr  Mon Aug 10 02:00:26 2009
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Mon, 10 Aug 2009 11:00:26 +0200
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <20090808025613.GA19871@hijacked.us>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
	<20090808025613.GA19871@hijacked.us>
Message-ID: <4A7FE1AA.6030309@telemaque.fr>

Morning guys,

Yes the latest make swigall broke php code generation.

Simple workaround:

edit libs/esl/ESL.i to this content:

----------------------Cut----------------------
%{
#include "esl.h"
#include "esl_oop.h"
%}

%include "esl_oop.h"
---------------------Cut-----------------------

and make reswig

It should be a swig bug, but I'm not sure.

Regards,

Gled

Andrew Thompson a ?crit :
> On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
>   
>> Hi,
>>
>> I'm trying to get started with the ESL using PHP. I compiled the ESL, then
>> phpmod according to the wiki instructions, but then when I try the examples
>> in the libs/esl/php dir, they fail saying:
>>
>> PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
>> /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132
>>
>> Checking ESL.php on line 132, I see there are several different declarations
>> for the function __construct() with different parameters each, which makes
>> sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
>> higher than that or something? What could be the problem?
>>
>>     
>
> Someone in the IRC channel mentioned this too. I looked at it briefly
> and it looks like the latest 'swigall' screwed it up. The original
> reporter said he'd file a jira, but you may want to check yourself and
> if not make one yourself. In the meantime, the previous version of the
> file was reported to work if you really need it.
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
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From mike at jerris.com  Mon Aug 10 05:10:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 Aug 2009 08:10:32 -0400
Subject: [Freeswitch-users] Fw: FreeSwitch doesn't play music on hold
	forbriged channel
In-Reply-To: <C5CEF36D42B347B49C82CDE05C3B0E5D@abisoft.biz>
References: <C5CEF36D42B347B49C82CDE05C3B0E5D@abisoft.biz>
Message-ID: <E1B1030B-D1EA-4EB1-80A7-650EBD3B44D7@jerris.com>

The logs indicate that you are already doing a broadcast to the  
channel.  Can you please come up with the simplest test script to  
reproduce this issue and post to Jira.freeswitch.org

Mike

On Aug 10, 2009, at 6:00 AM, "Kozak Vladimir" <vkozak at abisoft.spb.ru>  
wrote:

> 2009-08-12 03:02:03 [DEBUG] switch_core_session.c:706  
> switch_core_session_queue_private_event() Send signal sofia/internal/ 
> sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> [BREAK]
> 2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:228  
> audio_bridge_thread() sofia/internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> receive message [UNBRIDGE]
> 2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523  
> switch_core_session_perform_receive_message() Send signal sofia/ 
> internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
> 2009-08-12 03:02:03 [DEBUG] switch_ivr.c:540  
> switch_ivr_parse_event()  sofia/internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> Command Execute hold()
> 2009-08-12 03:02:03 [DEBUG] switch_ivr.c:1054 switch_ivr_hold()  
> sofia/internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> receive message [HOLD]
> 2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523  
> switch_core_session_perform_receive_message() Send signal sofia/ 
> internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
> 2009-08-12 03:02:03 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state()  
> Channel sofia/internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> entering state [calling]
> 2009-08-12 03:02:03 [WARNING] switch_ivr_async.c:2239  
> switch_ivr_broadcast() Channel [sofia/internal/1001 at 172.26.200.250] 
> [local_stream://moh] already broadcasting...broadcast aborted
> 2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:231  
> audio_bridge_thread() sofia/internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> receive message [BRIDGE]
> 2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523  
> switch_core_session_perform_receive_message() Send signal sofia/ 
> internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
> 2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:233  
> audio_bridge_thread() Send signal sofia/internal/1001 at 172.26.200.250  
> [BREAK]
> 2009-08-12 03:02:03 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state()  
> Channel sofia/internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> entering state [ready]
> 2009-08-12 03:02:03 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state()  
> Remote SDP:
> v=0
> o=- 6 3 IN IP4 172.26.10.39
> s=CounterPath eyeBeam 1.5
> c=IN IP4 172.26.10.39
> t=0 0
> m=audio 15712 RTP/AVP 0 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=recvonly
>
> 2009-08-12 03:02:03 [DEBUG] sofia_glue.c:2483  
> sofia_glue_negotiate_sdp() Our existing sdp is still good [PCMU 172.26.10.39 
> :15712], let's keep it.
> 2009-08-12 03:02:03 [DEBUG] sofia_glue.c:2509  
> sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101
>
> ----- Original Message -----
> From: Michael Jerris
> To: freeswitch-users at lists.freeswitch.org
> Sent: Sunday, August 09, 2009 12:19 AM
> Subject: Re: [Freeswitch-users] FreeSwitch doesn't play music on  
> holdforbriged channel
>
> What do the debug logs on fs say when you try to put the call on hold?
>
> Mike
>
> On Aug 6, 2009, at 1:01 PM, Kozak Vladimir wrote:
>
>>
>> The scenario is the following:
>> FS User A dial an extension
>> Extention opens outbound socket channel to my application
>> My application bridges the call to FS User B
>> The application check for CHANNEL_BRIDGED event and stores Other- 
>> leg-unique-id
>> The application sends hold to the bridged channel using SendMsg  
>> with Other-leg-unique-id
>> User B is placed on hold but no music on hold is played to the  
>> caller (User A)
>>
>>
>> I have outbound socket channel and the following sequence of  
>> commands/event:
>> listening on [any] 8084 ...
>> connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250 
>> ] 34000
>> connect
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> The scenario is the following:
> FS User A dial an extension
> Extention opens outbound socket channel to my application
> My application bridges the call to FS User B
> The application check for CHANNEL_BRIDGED event and stores Other-leg- 
> unique-id
> The application sends hold to the bridged channel using SendMsg with  
> Other-leg-unique-id
> User B is placed on hold but no music on hold is played to the  
> caller (User A)
>
>
> I have outbound socket channel and the following sequence of  
> commands/event:
> listening on [any] 8084 ...
> connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250 
> ] 34000
> connect
>
> myevents
>
> SendMsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg:user/1000 at uat.agent.starpoundtech.net
>
> Channel-Username: 1001
> Channel-Dialplan: XML
> Channel-Caller-ID-Name: 1001
> Channel-Caller-ID-Number: 1001
> Channel-Network-Addr: 172.26.10.39
> Channel-Destination-Number: 6666
> Channel-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Channel-Source: mod_sofia
> Channel-Context: default
> Channel-Channel-Name: sofia/internal/1001%40172.26.200.250
> Channel-Profile-Index: 1
> Channel-Profile-Created-Time: 1249142681680114
> Channel-Channel-Created-Time: 1249142681680114
> Channel-Channel-Answered-Time: 0
> Channel-Channel-Progress-Time: 0
> Channel-Channel-Progress-Media-Time:    1249142681809352
> Channel-Channel-Hangup-Time: 0
> Channel-Channel-Transfer-Time: 0
> Channel-Screen-Bit: true
> Channel-Privacy-Hide-Name: false
> Channel-Privacy-Hide-Number: false
> Channel-State: CS_EXECUTE
> Channel-State-Number: 4
> Channel-Name: sofia/internal/1001%40172.26.200.250
> Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Call-Direction: inbound
> Answer-State: early
> Channel-Read-Codec-Name: PCMU
> Channel-Read-Codec-Rate: 8000
> Channel-Write-Codec-Name: PCMU
> Channel-Write-Codec-Rate: 8000
> Caller-Username: 1001
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: 1001
> Caller-Caller-ID-Number: 1001
> Caller-Network-Addr: 172.26.10.39
> Caller-Destination-Number: 6666
> Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Caller-Source: mod_sofia
> Caller-Context: default
> Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1249142681680114
> Caller-Channel-Created-Time: 1249142681680114
> Caller-Channel-Answered-Time: 0
> Caller-Channel-Progress-Time: 0
> Caller-Channel-Progress-Media-Time: 1249142681809352
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
> variable_sip_received_ip: 172.26.10.39
> variable_sip_received_port: 13488
> variable_sip_via_protocol: udp
> variable_sip_authorized: true
> variable_sip_mailbox: 1001
> variable_sip_auth_username: 1001
> variable_sip_auth_realm: 172.26.200.250
> variable_mailbox: 1001
> variable_toll_allow: domestic,international,local
> variable_accountcode: 1001
> variable_user_context: default
> variable_effective_caller_id_name: Extension%201001
> variable_effective_caller_id_number: 1001
> variable_outbound_caller_id_name: StarPound%20FreeSWITCH
> variable_outbound_caller_id_number: 0000000000
> variable_callgroup: techsupport
> variable_sip_from_user: 1001
> variable_sip_from_uri: 1001%40172.26.200.250
> variable_sip_from_host: 172.26.200.250
> variable_sip_from_user_stripped: 1001
> variable_sip_from_tag: bd11f93c
> variable_sofia_profile_name: internal
> variable_sip_req_user: 6666
> variable_sip_req_uri: 6666%40172.26.200.250
> variable_sip_req_host: 172.26.200.250
> variable_sip_to_user: 6666
> variable_sip_to_uri: 6666%40172.26.200.250
> variable_sip_to_host: 172.26.200.250
> variable_sip_contact_user: 1001
> variable_sip_contact_port: 13488
> variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
> variable_sip_contact_host: 172.26.10.39
> variable_channel_name: sofia/internal/1001%40172.26.200.250
> variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
> variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
> variable_sip_via_host: 172.26.10.39
> variable_sip_via_port: 13488
> variable_sip_via_rport: 13488
> variable_max_forwards: 70
> variable_presence_id: 1001%40172.26.200.250
> variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN 
> %20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN 
> %20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP 
> %200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D 
> %0Aa%3Dfmtp%3A101%200-15%0D%0A
> variable_remote_media_ip: 172.26.10.39
> variable_remote_media_port: 29826
> variable_read_codec: PCMU
> variable_read_rate: 8000
> variable_write_codec: PCMU
> variable_write_rate: 8000
> variable_use_profile: nat
> variable_record_stereo: true
> variable_transfer_fallback_extension: operator
> variable_numbering_plan: US
> variable_default_areacode: 918
> variable_default_gateway: example.com
> variable_user_name: default
> variable_domain_name: 172.26.200.250
> variable_current_application_data: 172.26.200.251%3A8084%20async 
> %20full
> variable_current_application: socket
> variable_socket_host: 172.26.200.251
> variable_local_media_ip: 172.26.200.250
> variable_local_media_port: 29370
> variable_endpoint_disposition: EARLY%20MEDIA
> variable_sip_nat_detected: true
> Content-Type: command/reply
> Reply-Text: %2BOK%0A
> Socket-Mode: async
> Control: full
>
> Content-Type: command/reply
> Reply-Text: +OK Events Enabled
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> Content-Length: 1541
> Content-Type: text/event-plain
>
> Channel-State: CS_EXECUTE
> Channel-State-Number: 4
> Channel-Name: sofia/internal/1001%40172.26.200.250
> Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Call-Direction: inbound
> Answer-State: early
> Channel-Read-Codec-Name: PCMU
> Channel-Read-Codec-Rate: 8000
> Channel-Write-Codec-Name: PCMU
> Channel-Write-Codec-Rate: 8000
> Caller-Username: 1001
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: 1001
> Caller-Caller-ID-Number: 1001
> Caller-Network-Addr: 172.26.10.39
> Caller-Destination-Number: 6666
> Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Caller-Source: mod_sofia
> Caller-Context: default
> Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1249142681680114
> Caller-Channel-Created-Time: 1249142681680114
> Caller-Channel-Answered-Time: 0
> Caller-Channel-Progress-Time: 0
> Caller-Channel-Progress-Media-Time: 1249142681809352
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
> Application: bridge
> Application-Data: user/1000%40uat.agent.starpoundtech.net
> Event-Name: CHANNEL_EXECUTE
> Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
> FreeSWITCH-Hostname: centos4-4-vm
> FreeSWITCH-IPv4: 172.26.200.250
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-08-01%2020%3A04%3A51
> Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A51%20GMT
> Event-Date-Timestamp: 1249142691754598
> Event-Calling-File: switch_core_session.c
> Event-Calling-Function: switch_core_session_exec
> Event-Calling-Line-Number: 1333
>
> Content-Length: 5242
> Content-Type: text/event-plain
>
> Channel-State: CS_EXECUTE
> Channel-State-Number: 4
> Channel-Name: sofia/internal/1001%40172.26.200.250
> Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Call-Direction: inbound
> Answer-State: answered
> Channel-Read-Codec-Name: PCMU
> Channel-Read-Codec-Rate: 8000
> Channel-Write-Codec-Name: PCMU
> Channel-Write-Codec-Rate: 8000
> Caller-Username: 1001
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: 1001
> Caller-Caller-ID-Number: 1001
> Caller-Network-Addr: 172.26.10.39
> Caller-Destination-Number: 6666
> Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Caller-Source: mod_sofia
> Caller-Context: default
> Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1249142681680114
> Caller-Channel-Created-Time: 1249142681680114
> Caller-Channel-Answered-Time: 1249142692414509
> Caller-Channel-Progress-Time: 1249142691898434
> Caller-Channel-Progress-Media-Time: 1249142681809352
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
> Other-Leg-Username: 1001
> Other-Leg-Dialplan: XML
> Other-Leg-Caller-ID-Name: Extension%201001
> Other-Leg-Caller-ID-Number: 1001
> Other-Leg-Network-Addr: 172.26.10.39
> Other-Leg-Destination-Number: sip 
> %3A1000%40172.26.10.39%3A60152%3Brinstance 
> %3D393ff32df3ec5e33%3Bfs_nat%3Dyes
> Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
> Other-Leg-Source: mod_sofia
> Other-Leg-Context: default
> Other-Leg-Channel-Name: sofia/internal/sip 
> %3A1000%40172.26.10.39%3A60152%3Brinstance 
> %3D393ff32df3ec5e33%3Bfs_nat%3Dyes
> Other-Leg-Screen-Bit: true
> Other-Leg-Privacy-Hide-Name: false
> Other-Leg-Privacy-Hide-Number: false
> variable_sip_received_ip: 172.26.10.39
> variable_sip_received_port: 13488
> variable_sip_via_protocol: udp
> variable_sip_authorized: true
> variable_sip_mailbox: 1001
> variable_sip_auth_username: 1001
> variable_sip_auth_realm: 172.26.200.250
> variable_mailbox: 1001
> variable_toll_allow: domestic,international,local
> variable_accountcode: 1001
> variable_user_context: default
> variable_effective_caller_id_name: Extension%201001
> variable_effective_caller_id_number: 1001
> variable_outbound_caller_id_name: StarPound%20FreeSWITCH
> variable_outbound_caller_id_number: 0000000000
> variable_callgroup: techsupport
> variable_sip_from_user: 1001
> variable_sip_from_uri: 1001%40172.26.200.250
> variable_sip_from_host: 172.26.200.250
> variable_sip_from_user_stripped: 1001
> variable_sip_from_tag: bd11f93c
> variable_sofia_profile_name: internal
> variable_sip_req_user: 6666
> variable_sip_req_uri: 6666%40172.26.200.250
> variable_sip_req_host: 172.26.200.250
> variable_sip_to_user: 6666
> variable_sip_to_uri: 6666%40172.26.200.250
> variable_sip_to_host: 172.26.200.250
> variable_sip_contact_user: 1001
> variable_sip_contact_port: 13488
> variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
> variable_sip_contact_host: 172.26.10.39
> variable_channel_name: sofia/internal/1001%40172.26.200.250
> variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
> variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
> variable_sip_via_host: 172.26.10.39
> variable_sip_via_port: 13488
> variable_sip_via_rport: 13488
> variable_max_forwards: 70
> variable_presence_id: 1001%40172.26.200.250
> variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN 
> %20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN 
> %20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP 
> %200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D 
> %0Aa%3Dfmtp%3A101%200-15%0D%0A
> variable_remote_media_ip: 172.26.10.39
> variable_remote_media_port: 29826
> variable_read_codec: PCMU
> variable_read_rate: 8000
> variable_write_codec: PCMU
> variable_write_rate: 8000
> variable_use_profile: nat
> variable_record_stereo: true
> variable_transfer_fallback_extension: operator
> variable_numbering_plan: US
> variable_default_areacode: 918
> variable_default_gateway: example.com
> variable_user_name: default
> variable_domain_name: 172.26.200.250
> variable_socket_host: 172.26.200.251
> variable_local_media_ip: 172.26.200.250
> variable_local_media_port: 29370
> variable_endpoint_disposition: EARLY%20MEDIA
> variable_current_application_data: user/ 
> 1000%40uat.agent.starpoundtech.net
> variable_current_application: bridge
> variable_dialed_user: 1000
> variable_dialed_domain: uat.agent.starpoundtech.net
> variable_originate_disposition: failure
> variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
> variable_sip_redirect_contact_user_0: 1000
> variable_sip_redirect_contact_host_0: 172.26.10.39
> variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN 
> %20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac 
> %3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio 
> %2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap 
> %3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
> variable_sip_nat_detected: true
> Event-Name: CHANNEL_ANSWER
> Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
> FreeSWITCH-Hostname: centos4-4-vm
> FreeSWITCH-IPv4: 172.26.200.250
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-08-01%2020%3A04%3A52
> Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
> Event-Date-Timestamp: 1249142692414509
> Event-Calling-File: switch_channel.c
> Event-Calling-Function: switch_channel_perform_mark_answered
> Event-Calling-Line-Number: 1776
>
> Content-Length: 5233
> Content-Type: text/event-plain
>
> Channel-State: CS_EXECUTE
> Channel-State-Number: 4
> Channel-Name: sofia/internal/1001%40172.26.200.250
> Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Call-Direction: inbound
> Answer-State: answered
> Channel-Read-Codec-Name: PCMU
> Channel-Read-Codec-Rate: 8000
> Channel-Write-Codec-Name: PCMU
> Channel-Write-Codec-Rate: 8000
> Caller-Username: 1001
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: 1001
> Caller-Caller-ID-Number: 1001
> Caller-Network-Addr: 172.26.10.39
> Caller-Destination-Number: 6666
> Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Caller-Source: mod_sofia
> Caller-Context: default
> Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1249142681680114
> Caller-Channel-Created-Time: 1249142681680114
> Caller-Channel-Answered-Time: 1249142692414509
> Caller-Channel-Progress-Time: 1249142691898434
> Caller-Channel-Progress-Media-Time: 1249142681809352
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
> Other-Leg-Username: 1001
> Other-Leg-Dialplan: XML
> Other-Leg-Caller-ID-Name: Extension%201001
> Other-Leg-Caller-ID-Number: 1001
> Other-Leg-Network-Addr: 172.26.10.39
> Other-Leg-Destination-Number: sip 
> %3A1000%40172.26.10.39%3A60152%3Brinstance 
> %3D393ff32df3ec5e33%3Bfs_nat%3Dyes
> Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
> Other-Leg-Source: mod_sofia
> Other-Leg-Context: default
> Other-Leg-Channel-Name: sofia/internal/sip 
> %3A1000%40172.26.10.39%3A60152%3Brinstance 
> %3D393ff32df3ec5e33%3Bfs_nat%3Dyes
> Other-Leg-Screen-Bit: true
> Other-Leg-Privacy-Hide-Name: false
> Other-Leg-Privacy-Hide-Number: false
> variable_sip_received_ip: 172.26.10.39
> variable_sip_received_port: 13488
> variable_sip_via_protocol: udp
> variable_sip_authorized: true
> variable_sip_mailbox: 1001
> variable_sip_auth_username: 1001
> variable_sip_auth_realm: 172.26.200.250
> variable_mailbox: 1001
> variable_toll_allow: domestic,international,local
> variable_accountcode: 1001
> variable_user_context: default
> variable_effective_caller_id_name: Extension%201001
> variable_effective_caller_id_number: 1001
> variable_outbound_caller_id_name: StarPound%20FreeSWITCH
> variable_outbound_caller_id_number: 0000000000
> variable_callgroup: techsupport
> variable_sip_from_user: 1001
> variable_sip_from_uri: 1001%40172.26.200.250
> variable_sip_from_host: 172.26.200.250
> variable_sip_from_user_stripped: 1001
> variable_sip_from_tag: bd11f93c
> variable_sofia_profile_name: internal
> variable_sip_req_user: 6666
> variable_sip_req_uri: 6666%40172.26.200.250
> variable_sip_req_host: 172.26.200.250
> variable_sip_to_user: 6666
> variable_sip_to_uri: 6666%40172.26.200.250
> variable_sip_to_host: 172.26.200.250
> variable_sip_contact_user: 1001
> variable_sip_contact_port: 13488
> variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
> variable_sip_contact_host: 172.26.10.39
> variable_channel_name: sofia/internal/1001%40172.26.200.250
> variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
> variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
> variable_sip_via_host: 172.26.10.39
> variable_sip_via_port: 13488
> variable_sip_via_rport: 13488
> variable_max_forwards: 70
> variable_presence_id: 1001%40172.26.200.250
> variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN 
> %20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN 
> %20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP 
> %200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D 
> %0Aa%3Dfmtp%3A101%200-15%0D%0A
> variable_remote_media_ip: 172.26.10.39
> variable_remote_media_port: 29826
> variable_read_codec: PCMU
> variable_read_rate: 8000
> variable_write_codec: PCMU
> variable_write_rate: 8000
> variable_use_profile: nat
> variable_record_stereo: true
> variable_transfer_fallback_extension: operator
> variable_numbering_plan: US
> variable_default_areacode: 918
> variable_default_gateway: example.com
> variable_user_name: default
> variable_domain_name: 172.26.200.250
> variable_socket_host: 172.26.200.251
> variable_local_media_ip: 172.26.200.250
> variable_local_media_port: 29370
> variable_current_application_data: user/ 
> 1000%40uat.agent.starpoundtech.net
> variable_current_application: bridge
> variable_dialed_user: 1000
> variable_dialed_domain: uat.agent.starpoundtech.net
> variable_sip_redirect_contact_user_0: 1000
> variable_sip_redirect_contact_host_0: 172.26.10.39
> variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN 
> %20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac 
> %3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio 
> %2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap 
> %3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
> variable_sip_nat_detected: true
> variable_endpoint_disposition: ANSWER
> variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
> variable_originate_disposition: SUCCESS
> Event-Name: CHANNEL_BRIDGE
> Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
> FreeSWITCH-Hostname: centos4-4-vm
> FreeSWITCH-IPv4: 172.26.200.250
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-08-01%2020%3A04%3A52
> Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
> Event-Date-Timestamp: 1249142692414509
> Event-Calling-File: switch_ivr_bridge.c
> Event-Calling-Function: switch_ivr_multi_threaded_bridge
> Event-Calling-Line-Number: 828
>
> SendMsg 94b59a38-57c4-4703-9c6e-9985d832d119
> call-command: execute
> execute-app-name: hold
>
> Content-Type: command/reply
> Reply-Text: +OK
>
>
>
>
> - I don't see the variable hold_music ... did you remove it?
>
> - I didn't. Moreover, I tried to set it explicitly using api  
> uuid_setvar.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> DIV>
> - I don't see the variable hold_music ... did you remove it?
>
> - I didn't. Moreover, I tried to set it explicitly using api  
> uuid_setvar.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> TE>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Mon Aug 10 05:13:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 Aug 2009 08:13:27 -0400
Subject: [Freeswitch-users] Fwd:  Scheduler in module
In-Reply-To: <20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
Message-ID: <985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>

Re schedule is done in your callback, take a look at places that use  
these apis in the code for details.

On Aug 10, 2009, at 6:58 AM, mark morreny <markmorreny at gmail.com> wrote:

> Hi,
>
> Thank you for pointing out sched_api.
>
> What about if I want to do a recurring schedule. is it possible or I  
> should just call sched_api another time in the execution of my clean  
> up code.
>
> Thanks,
> Mark
>
> ---------- Forwarded message ----------
> From: Moises Silva <moises.silva at gmail.com>
> Date: Mon, Aug 10, 2009 at 12:05 AM
> Subject: Re: [Freeswitch-users] Scheduler in module
> To: freeswitch-users at lists.freeswitch.org
>
>
> On Sun, Aug 9, 2009 at 11:57 AM, mark morreny  
> <markmorreny at gmail.com> wrote:
> Hi,
>
> I would like to collect some data in my module which would listen to  
> the event socket and then a scheduler would kick off every so often  
> to clean up the data.
>
> Does anyone know what is the best way to implement a scheduler in a  
> mod?
>
> Best Regards,
> Mark
>
> Just use the API that the FreeSWITCH core already has for you. Check  
> src/include/switch_scheduler.h for details, the comments in there  
> should be enough for you to learn to use it. If not, you can ask  
> here more specific questions.
>
> -- 
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON  
> L3R 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Mon Aug 10 05:22:29 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 Aug 2009 08:22:29 -0400
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <4A7FE1AA.6030309@telemaque.fr>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
	<20090808025613.GA19871@hijacked.us>
	<4A7FE1AA.6030309@telemaque.fr>
Message-ID: <F123F0B3-9CD9-429B-97EE-83BBA6D74E46@jerris.com>

Can you please post a patch to Jira.freswitch.org and assign it to me.

Mike

On Aug 10, 2009, at 5:00 AM, Tristan Mah? <t.mahe at telemaque.fr> wrote:

> Morning guys,
>
> Yes the latest make swigall broke php code generation.
>
> Simple workaround:
>
> edit libs/esl/ESL.i to this content:
>
> ----------------------Cut----------------------
> %{
> #include "esl.h"
> #include "esl_oop.h"
> %}
>
> %include "esl_oop.h"
> ---------------------Cut-----------------------
>
> and make reswig
>
> It should be a swig bug, but I'm not sure.
>
> Regards,
>
> Gled
>
> Andrew Thompson a ?crit :
>>
>> On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
>>
>>> Hi,
>>>
>>> I'm trying to get started with the ESL using PHP. I compiled the  
>>> ESL, then
>>> phpmod according to the wiki instructions, but then when I try the  
>>> examples
>>> in the libs/esl/php dir, they fail saying:
>>>
>>> PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
>>> /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132
>>>
>>> Checking ESL.php on line 132, I see there are several different  
>>> declarations
>>> for the function __construct() with different parameters each,  
>>> which makes
>>> sense, but doens't work. I am using PHP 5.1.6, is there a required  
>>> minimum
>>> higher than that or something? What could be the problem?
>>>
>>>
>> Someone in the IRC channel mentioned this too. I looked at it briefly
>> and it looks like the latest 'swigall' screwed it up. The original
>> reporter said he'd file a jira, but you may want to check yourself  
>> and
>> if not make one yourself. In the meantime, the previous version of  
>> the
>> file was reported to work if you really need it.
>>
>> Andrew
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From wangdq.no1 at gmail.com  Mon Aug 10 05:26:18 2009
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Mon, 10 Aug 2009 20:26:18 +0800
Subject: [Freeswitch-users] can freeswitch group_call to gateway
	extension ?
In-Reply-To: <261A7939-0C0E-48DF-B4D4-9A02D5616898@jerris.com>
References: <ecd05ce30908092316s66051433p506df518942ec93f@mail.gmail.com>
	<261A7939-0C0E-48DF-B4D4-9A02D5616898@jerris.com>
Message-ID: <ecd05ce30908100526j31f5d5ddl2df548d3da6f7a36@mail.gmail.com>

I mean:
   if the telephone number . is in ghe group . (defined in
/directory/default.xml ) .
  but the telephone number is not registered on the FS . (for example: the
telephone number is a mobile number).
 when I dialed the group_call  extension.
 the mobile can't ring.
 why ?   thanks.


2009/8/10 Michael Jerris <mike at jerris.com>

> Can you rephrase your question?
>
> On Aug 10, 2009, at 2:16 AM, daqiang wang <wangdq.no1 at gmail.com> wrote:
>
> > Hello every one :
> >   I tested FS. and when I use group_call.  But I can't call the
> > extension in group and not registered in FS.
> >   Why ? and What  ?
> >   thanks .
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From t.mahe at telemaque.fr  Mon Aug 10 05:34:33 2009
From: t.mahe at telemaque.fr (=?UTF-8?B?VHJpc3RhbiBNYWjDqQ==?=)
Date: Mon, 10 Aug 2009 14:34:33 +0200
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <F123F0B3-9CD9-429B-97EE-83BBA6D74E46@jerris.com>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>	<20090808025613.GA19871@hijacked.us>	<4A7FE1AA.6030309@telemaque.fr>
	<F123F0B3-9CD9-429B-97EE-83BBA6D74E46@jerris.com>
Message-ID: <4A8013D9.30806@telemaque.fr>

Hello Mike,

No problem to post a patch, but it would break perl/python/etc... as
actually there are some functions defined in ESL.i, wouldn't it ?
I don't know about swig, never used that.

Maybe we could have two swig files instead ? one for generating php and
one for the other languages ?
I'm on IRC if you want to discuss it.

Gled.

Michael Jerris a ?crit :
> Can you please post a patch to Jira.freswitch.org and assign it to me.
>
> Mike
>
> On Aug 10, 2009, at 5:00 AM, Tristan Mah? <t.mahe at telemaque.fr
> <mailto:t.mahe at telemaque.fr>> wrote:
>
>> Morning guys,
>>
>> Yes the latest make swigall broke php code generation.
>>
>> Simple workaround:
>>
>> edit libs/esl/ESL.i to this content:
>>
>> ----------------------Cut----------------------
>> %{
>> #include "esl.h"
>> #include "esl_oop.h"
>> %}
>>
>> %include "esl_oop.h"
>> ---------------------Cut-----------------------
>>
>> and make reswig
>>
>> It should be a swig bug, but I'm not sure.
>>
>> Regards,
>>
>> Gled
>>
>> Andrew Thompson a ?crit :
>>> On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
>>>   
>>>> Hi,
>>>>
>>>> I'm trying to get started with the ESL using PHP. I compiled the ESL, then
>>>> phpmod according to the wiki instructions, but then when I try the examples
>>>> in the libs/esl/php dir, they fail saying:
>>>>
>>>> PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
>>>> /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132
>>>>
>>>> Checking ESL.php on line 132, I see there are several different declarations
>>>> for the function __construct() with different parameters each, which makes
>>>> sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
>>>> higher than that or something? What could be the problem?
>>>>
>>>>     
>>> Someone in the IRC channel mentioned this too. I looked at it briefly
>>> and it looks like the latest 'swigall' screwed it up. The original
>>> reporter said he'd file a jira, but you may want to check yourself and
>>> if not make one yourself. In the meantime, the previous version of the
>>> file was reported to work if you really need it.
>>>
>>> Andrew
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>   
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
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From brian at freeswitch.org  Mon Aug 10 05:53:31 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 07:53:31 -0500
Subject: [Freeswitch-users] Fwd:  Scheduler in module
In-Reply-To: <985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
Message-ID: <5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>

switch_rtp.c has a simple one for the zrtp cache storing.

/b

On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:

> Re schedule is done in your callback, take a look at places that use  
> these apis in the code for details.



From brian at freeswitch.org  Mon Aug 10 06:31:10 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 08:31:10 -0500
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
Message-ID: <80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>

As MikeJ pointed out please report the IP address files.freeswitch.org  
resolves to.  We have that on a content delivery network so that the  
files are closer to you geographically and you can download them  
faster but if you're having an issue I'll need the IP so I can report  
it correctly.

Thanks,
Brian

On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:

> Hi FS Team,
>
>
>         The files.freeswitch.org is resetting connection since 3  
> days. As a result I was not able to download latest release of FS.  
> Got the trunk version from svn. But still it suffers from the lack  
> of sound files.  When ever I do 'make uhd-sounds-install'http://files.freeswitch.org 
>  resets connection immediately wget tries 20 times and gives up.  
> Other users on IRC also reported this issue.
>
> -- 
> Thanks & Regards,
> Godson Gera

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From nicolas at medularis.com  Mon Aug 10 06:39:19 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 10 Aug 2009 09:39:19 -0400
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <4A8013D9.30806@telemaque.fr>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
	<20090808025613.GA19871@hijacked.us> <4A7FE1AA.6030309@telemaque.fr>
	<F123F0B3-9CD9-429B-97EE-83BBA6D74E46@jerris.com>
	<4A8013D9.30806@telemaque.fr>
Message-ID: <1b46b4e80908100639mb2b6087jbc006f9e420da45e@mail.gmail.com>

Hi, I tried the ESL.i modification and when I do make reswig I get:

make -C php reswig
make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php'
rm -f esl_wrap.* ESL.so php_ESL.* ESL.php
swig -module ESL -php5 -c++ -DMULTIPLICITY -I../src/include -o esl_wrap.cpp
../ESL.i
swig error : Unrecognized option -php5
Use 'swig -help' for available options.
make[1]: *** [esl_wrap.cpp] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php'
make: *** [reswig] Error 2


On Mon, Aug 10, 2009 at 8:34 AM, Tristan Mah? <t.mahe at telemaque.fr> wrote:

>  Hello Mike,
>
> No problem to post a patch, but it would break perl/python/etc... as
> actually there are some functions defined in ESL.i, wouldn't it ?
> I don't know about swig, never used that.
>
> Maybe we could have two swig files instead ? one for generating php and one
> for the other languages ?
> I'm on IRC if you want to discuss it.
>
> Gled.
>
> Michael Jerris a ?crit :
>
> Can you please post a patch to Jira.freswitch.org and assign it to me.
>
>  Mike
>
> On Aug 10, 2009, at 5:00 AM, Tristan Mah? <t.mahe at telemaque.fr> wrote:
>
>  Morning guys,
>
> Yes the latest make swigall broke php code generation.
>
> Simple workaround:
>
> edit libs/esl/ESL.i to this content:
>
> ----------------------Cut----------------------
> %{
> #include "esl.h"
> #include "esl_oop.h"
> %}
>
> %include "esl_oop.h"
> ---------------------Cut-----------------------
>
> and make reswig
>
> It should be a swig bug, but I'm not sure.
>
> Regards,
>
> Gled
>
> Andrew Thompson a ?crit :
>
> On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
>
>
>  Hi,
>
> I'm trying to get started with the ESL using PHP. I compiled the ESL, then
> phpmod according to the wiki instructions, but then when I try the examples
> in the libs/esl/php dir, they fail saying:
>
> PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
> /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132
>
> Checking ESL.php on line 132, I see there are several different declarations
> for the function __construct() with different parameters each, which makes
> sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
> higher than that or something? What could be the problem?
>
>
>
>  Someone in the IRC channel mentioned this too. I looked at it briefly
> and it looks like the latest 'swigall' screwed it up. The original
> reporter said he'd file a jira, but you may want to check yourself and
> if not make one yourself. In the meantime, the previous version of the
> file was reported to work if you really need it.
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>     _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Aug 10 06:41:50 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 08:41:50 -0500
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <1b46b4e80908100639mb2b6087jbc006f9e420da45e@mail.gmail.com>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
	<20090808025613.GA19871@hijacked.us>
	<4A7FE1AA.6030309@telemaque.fr>
	<F123F0B3-9CD9-429B-97EE-83BBA6D74E46@jerris.com>
	<4A8013D9.30806@telemaque.fr>
	<1b46b4e80908100639mb2b6087jbc006f9e420da45e@mail.gmail.com>
Message-ID: <0DA5E116-7E6E-4A3E-B86D-88329FB2C2D8@freeswitch.org>

If you're removing the newobject lines you're introducing a huge  
memory leak :P  also I'm guessing your swig is older and doesn't have  
php5 support.

/b

On Aug 10, 2009, at 8:39 AM, Nicolas Brenner wrote:

> Hi, I tried the ESL.i modification and when I do make reswig I get:
>
> make -C php reswig
> make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php'
> rm -f esl_wrap.* ESL.so php_ESL.* ESL.php
> swig -module ESL -php5 -c++ -DMULTIPLICITY -I../src/include -o  
> esl_wrap.cpp ../ESL.i
> swig error : Unrecognized option -php5
> Use 'swig -help' for available options.
> make[1]: *** [esl_wrap.cpp] Error 1
> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php'
> make: *** [reswig] Error 2



From ivan at myrvold.org  Mon Aug 10 08:03:36 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Mon, 10 Aug 2009 17:03:36 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <092B4355-F911-4E37-A5D1-AC923AF9B8FB@gmail.com>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
	<6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
	<4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
	<092B4355-F911-4E37-A5D1-AC923AF9B8FB@gmail.com>
Message-ID: <E6E26B16-4905-49DE-A167-BEFC7A0D1638@myrvold.org>

Seven,
I am afraid I will not be able to help you much with the Carbon code,  
as I am only good at Cocoa programming. You said you chose Carbon  
because you only needed low level API, and that is fair enough, but I  
will also add that you can do the same with only linking to the  
Foundation framework in Cocoa.

I looked a little at the diff file yesterday, and will investigate  
more today, to try to understand how you have done the Skype  
integration to the Freeswitch in the Carbon code.
And I am glad that someone have contributed to get skypiax working in  
OS X. Great work so far!

Ivan

Den 9. aug.. 2009 kl. 20:02 skrev Seven Du:

> Ivan,
>
> Good to know you are a cocoa dev. Unable to check in code right now,
> will send the diff to you offlist for now.
>
> 0) I'm not familiar with Mac dev, just tried my best
> 1) It doesn't work yet, but should be able to compile, sure you
> already have the Skype framework in place :)
> 2) if run the skype delegate from a threat, then cannot get event
> callback. e.g. mac_client.c works but mac_client2.c doesn't. Since
> skypiax is running in a thread, we need to figure out this first.
> 3) it uses Carbon, since I think we only need to low level api, no
> need to bother the complicate of Cocoa.
> 4) strsep shows some warning on compile, haven't figured out why
> 5) perhaps you should only add one interface in skypiax.conf.xml
> 6) do you want to run multi-instances like on Linux?
> 7) I really not sure if it will work or not :)
>
> Let me know if it helps. I bet you can make it work. Also code will be
> in my branch soon.
>
> 7.
>
>
> On Aug 9, 2009, at 11:34 PM, Ivan C Myrvold wrote:
>> Yes, I am interested in this, and if you have any source I could have
>> a look at it.
>>
>> Ivan
>>
>> Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
>>
>>>
>>> On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
>>>> Ciao Ivan,
>>>>
>>>> it seems that you do not have the libX11 **development** package
>>>> installed.
>>>>
>>>> Unfortunately I don't know about OSX, so I cannot help you, but  
>>>> many
>>>> on the list know.
>>>> BTW: it will probably be of no use to you to compile mod_skypiax on
>>>> OSX, because Skype for MACOSX works in another way than Skype for
>>>> Linux.
>>>
>>> That's right.
>>>
>>>> If you know about MacOSX programming, please have a look at
>>>> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
>>>> probably be simple enough to add a message pump for MacOSX.
>>>>
>>>> -giovanni
>>>>
>>>>
>>> Giovanni, I have a Mac and tried to get this work yesterday, but
>>> haven't got it work. Will try further if I have time. However, I
>>> don't
>>> think it's so useful because I don't know how to run and hence
>>> control
>>> multi-skype instances on Mac.
>>>
>>> If someone interested to try this I can check the code into my
>>> branch.
>>>
>>>>
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> =========================================
>>>> www.celliax.org
>>>> via Pierlombardo 9, 20135 Milano
>>>> Italy
>>>> gmaruzz at celliax dot org
>>>> Cell : +39-347-2665618
>>>> Fax : +39-02-87390039
>>>>
>>>>
>>>>
>>>>
>>>> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>
>>>> wrote:
>>>>> I tried to compile mod_skypiax, and am getting problem with X11.  
>>>>> On
>>>>> OS
>>>>> X Leopard, X11 is installed in /usr/X11/lib/
>>>>> See below.
>>>>>
>>>>> What can I do to get past this error?
>>>>>
>>>>> I can also let you ssh into my machine. Contact me off list in
>>>>> case.
>>>>>
>>>>> Ivan
>>>>>
>>>>> making all mod_skypiax
>>>>> Compiling skypiax_protocol.c...
>>>>> Compiling mod_skypiax.c...
>>>>> mkdir .libs
>>>>> Compiling mod_skypiax.c ...
>>>>> Creating mod_skypiax.so...
>>>>> ld: library not found for -lX11
>>>>> collect2: ld returned 1 exit status
>>>>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>>>>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
>>>>> Documents/
>>>>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>>>>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -
>>>>> g -
>>>>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -
>>>>> shared -
>>>>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-
>>>>> namespace .libs/
>>>>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/
>>>>> Freeswitch/
>>>>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>>>>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/
>>>>> xml/
>>>>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch. 
>>>>> 09-08-09/
>>>>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
>>>>> libiconv.dylib /
>>>>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
>>>>> apr/.libs/
>>>>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto - 
>>>>> lz -
>>>>> lncurses -lX11
>>>>> make[5]: *** [mod_skypiax.so] Error 1
>>>>> make[4]: *** [all] Error 1
>>>>> make[3]: *** [mod_skypiax-all] Error 1
>>>>> make[2]: *** [all-recursive] Error 1
>>>>>
>>>>>
>>>>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>>>>
>>>>>> No, it needs implementation of the message pump between the  
>>>>>> module
>>>>>> and
>>>>>> the Skype API.
>>>>>>
>>>>>> It's probably kind of trivial, if no other problems I'm not aware
>>>>>> of.
>>>>>>
>>>>>> I do not have a Mac to implement it, tough :-(.
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Sincerely,
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> =========================================
>>>>>> www.celliax.org
>>>>>> via Pierlombardo 9, 20135 Milano
>>>>>> Italy
>>>>>> gmaruzz at celliax dot org
>>>>>> Cell : +39-347-2665618
>>>>>> Fax : +39-02-87390039
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>>>>> wrote:
>>>>>>> I'm not sure about that one.... I haven't tried lately because
>>>>>>> the
>>>>>>> API
>>>>>>> differs on the Mac last I looked at it.
>>>>>>>
>>>>>>> /b
>>>>>>>
>>>>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>>>>
>>>>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>>>>
>>>>>>>> Ivan
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From email.list.subscriber at gmail.com  Mon Aug 10 08:49:14 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Mon, 10 Aug 2009 11:49:14 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>
Message-ID: <4a804139.1408c00a.6102.38b8@mx.google.com>

Thanks for the response(s):

I ran the "./compile" script with a set PREFIX.  This took a few
attempts with errors before it was able to complete error-free, as I
had to install libtool.

Since then, I have tried running 'make', 'gmake', and
'/opt/gnu/bin/make', but each results with an error.  This is the
error when running 'make' or 'gmake':

<snip>
make: Fatal error: Command failed for target `all-recursive'
Current working directory /home/vmorales/freeswitch-1.0.4
*** Error code 1
make: Fatal error: Command failed for target `all'
</snip>


This is the error when running '/opt/gnu/bin/make':

<snip>
make[5]: *** [mod_amr.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_amr-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               /opt/gnu/bin/make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
</snip>


I re-untar'd before each compile/make attempt.  Let me know if this is
something that I can resolve.

Vladimir


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: Saturday, August 08, 2009 12:37 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

This is not currently a supported platform, it only builds on 64 bit  
right now I think on solaris.

Mike

On Aug 6, 2009, at 6:03 PM, vmorales wrote:

> Hello,
>
> Does anyone have, or know where to get, a pre-compiled copy of
> FreeSwitch for Solaris 10/x86?


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
http://www.freeswitch.org



From david.nembrot at sogeti.com  Mon Aug 10 09:56:47 2009
From: david.nembrot at sogeti.com (David Nembrot)
Date: Mon, 10 Aug 2009 18:56:47 +0200
Subject: [Freeswitch-users] Freeswitch :: VoIP issues using ILBC audio codec
Message-ID: <20090810185647.a3ymeb6d9w804s84@mail.sogeti.com>



   Hello world,

   I've got two FS servers configured as gateways for each other and  
I'm currently testing the telephony. Usinge the ILBC audio codec, I  
figured out that one of the FS servers doesn't forward RTP streams  
correctly to the other server. Here is its status-quo:
INPUT = proper ILBC payload type (97 or 108)
OUTPUT = unknown payload type (97 or 102)

   I've already changed the parameters in internal.xml & external.xml:
<param name="inbound-codec-negotiation" value="greedy"/>
<param name="disable-transcoding" value="true"/>

   When dialing out, I also use the following  
syntax:{absolute_codec_string='GSM,PCMU'}sofia/gateway/mygateway/mynumber

   Is there another thing to do to have proper ILBC streams passing  
through the gateways ?
Thanking y'all in advance ;)

   BR,
David N.

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   Hello world,

   I've got two FS servers configured as gateways for each other and  
I'm currently testing the telephony. Usinge the ILBC audio codec, I  
figured out that one of the FS servers doesn't forward RTP streams  
correctly to the other server. Here is its status-quo:

   INPUT = proper ILBC payload type (97 or 108)
OUTPUT = unknown payload type (97 or 102)

   I've already changed the parameters in internal.xml and also in  
external.xml:
<param name="inbound-codec-negotiation" value="greedy"/>
<param name="disable-transcoding" value="true"/>

   Is there another thing to do to have proper ILBC streams passing  
through the gateways ?
Thanking y'all in advance ;)

   BR,
David N.

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From brian at freeswitch.org  Mon Aug 10 10:03:37 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 12:03:37 -0500
Subject: [Freeswitch-users] Freeswitch :: VoIP issues using ILBC audio
	codec
In-Reply-To: <20090810185647.a3ymeb6d9w804s84@mail.sogeti.com>
References: <20090810185647.a3ymeb6d9w804s84@mail.sogeti.com>
Message-ID: <A0D83BF0-CF9F-41C8-AE0E-1863D7922D36@freeswitch.org>

Well since the codec is in the dynamic range the payload number  
doesn't matter.  I would have to see an RTP trace to see this... you  
must also include all the SIP traffic so wireshark can properly figure  
things out.

/b

On Aug 10, 2009, at 11:56 AM, David Nembrot wrote:

> Hello world,
>
> I've got two FS servers configured as gateways for each other and  
> I'm currently testing the telephony. Usinge the ILBC audio codec, I  
> figured out that one of the FS servers doesn't forward RTP streams  
> correctly to the other server. Here is its status-quo:
> INPUT = proper ILBC payload type (97 or 108)
> OUTPUT = unknown payload type (97 or 102)
>
> I've already changed the parameters in internal.xml & external.xml:
> <param name="inbound-codec-negotiation" value="greedy"/>
> <param name="disable-transcoding" value="true"/>
>
> When dialing out, I also use the following syntax: 
> {absolute_codec_string='GSM,PCMU'}sofia/gateway/mygateway/mynumber
>
> Is there another thing to do to have proper ILBC streams passing  
> through the gateways ?
> Thanking y'all in advance ;)
>
> BR,
> David N.
>
> Hello world,
>
> I've got two FS servers configured as gateways for each other and  
> I'm currently testing the telephony. Usinge the ILBC audio codec, I  
> figured out that one of the FS servers doesn't forward RTP streams  
> correctly to the other server. Here is its status-quo:
>
> INPUT = proper ILBC payload type (97 or 108)
> OUTPUT = unknown payload type (97 or 102)
>
> I've already changed the parameters in internal.xml and also in  
> external.xml:
> <param name="inbound-codec-negotiation" value="greedy"/>
> <param name="disable-transcoding" value="true"/>
>
> Is there another thing to do to have proper ILBC streams passing  
> through the gateways ?
> Thanking y'all in advance ;)
>
> BR,
> David N.
>



From email.list.subscriber at gmail.com  Mon Aug 10 11:14:09 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Mon, 10 Aug 2009 14:14:09 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>
	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>
Message-ID: <4a80632f.1508c00a.4d3c.090d@mx.google.com>

By "./compile" I was referring to "./configure"

Vladimir

-----Original Message-----
From: vmorales [mailto:email.list.subscriber at gmail.com] 
Sent: Monday, August 10, 2009 11:49 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

Thanks for the response(s):

I ran the "./compile" script with a set PREFIX.  This took a few
attempts with errors before it was able to complete error-free, as I
had to install libtool.

Since then, I have tried running 'make', 'gmake', and
'/opt/gnu/bin/make', but each results with an error.  This is the
error when running 'make' or 'gmake':

<snip>
make: Fatal error: Command failed for target `all-recursive'
Current working directory /home/vmorales/freeswitch-1.0.4
*** Error code 1
make: Fatal error: Command failed for target `all'
</snip>


This is the error when running '/opt/gnu/bin/make':

<snip>
make[5]: *** [mod_amr.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_amr-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               /opt/gnu/bin/make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
</snip>


I re-untar'd before each compile/make attempt.  Let me know if this is
something that I can resolve.

Vladimir


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: Saturday, August 08, 2009 12:37 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

This is not currently a supported platform, it only builds on 64 bit  
right now I think on solaris.

Mike

On Aug 6, 2009, at 6:03 PM, vmorales wrote:

> Hello,
>
> Does anyone have, or know where to get, a pre-compiled copy of
> FreeSwitch for Solaris 10/x86?


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
http://www.freeswitch.org



From pete at privateconnect.com  Mon Aug 10 11:28:06 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Mon, 10 Aug 2009 11:28:06 -0700
Subject: [Freeswitch-users] VoiceMail transcription
Message-ID: <20090810112806.2ad02225396a31c9de30536f2e338977.8f8b5c9b40.wbe@email04.secureserver.net>

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From asannucci at gmail.com  Mon Aug 10 11:43:43 2009
From: asannucci at gmail.com (bakko)
Date: Mon, 10 Aug 2009 20:43:43 +0200
Subject: [Freeswitch-users] Ivr and variables
Message-ID: <8D85DC2E55A74F41AE2190E05A44F028@voztovoice>

I'm tryng to put a variable in a IVR digits line like this:

<entry action="menu-exec-app" digits="${some}" param="transfer 1001 XML 
default"/>

but don't work

Is It possible tu use this solution?

thank you

BR




From dave at 3c.co.uk  Mon Aug 10 11:51:24 2009
From: dave at 3c.co.uk (David Knell)
Date: Mon, 10 Aug 2009 21:51:24 +0300
Subject: [Freeswitch-users] VoiceMail transcription
In-Reply-To: <20090810112806.2ad02225396a31c9de30536f2e338977.8f8b5c9b40.wbe@email04.secureserver.net>
References: <20090810112806.2ad02225396a31c9de30536f2e338977.8f8b5c9b40.wbe@email04.secureserver.net>
Message-ID: <1249930284.20224.34.camel@dk-d820>

Good evening Pete,

The only way to do this is, I'm afraid, to use a human.  We use Amazon's
Mechanical Turk to good effect.

Cheers --

Dave

> Good morning all,
> 
> I realize this is slightly off the FS topic, but I am wondering if
> anyone out there has experience with software packages designed for
> the transcription of voicemails to text.  I've used pocketsphinx  with
> FS to handle IVR menus, but now have the task of figuring out how to
> convert recorded phone conversations (voicemails mostly) to text.
> 
> This does not have to be a real-time process, I can store the audio
> files and process them over time.  This would need to be a software
> (preferable open source) solution.  ASPs like VoiceCloud would not
> work for this application.
> 
> Thanks for any help
> -pete
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From max.bridgewater at gmail.com  Mon Aug 10 11:51:36 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 14:51:36 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
Message-ID: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>

Hi get the following messages while running "make" on CentOS 5.3 (Of course
after bootstrap and configure).

os_Linux_x86_64.s: Assembler messages:
os_Linux_x86_64.s:46: Error: bad register name `%rdi)'
os_Linux_x86_64.s:61: Error: bad register name `%rdi)'
os_Linux_x86_64.s:75: Error: bad register name `%rdi)'
os_Linux_x86_64.s:89: Error: bad register name `%rdi)'
make[10]: *** [os_Linux_x86_64.o] Error 1
make[9]: *** [export] Error 2
make[8]: *** [export] Error 2
make[7]: *** [export] Error 2
make[6]: *** [export] Error 2
make[5]: *** [/home/installs/fstrunk/libs/js/libjs.la] Error 2
make[4]: *** [all] Error 1
make[3]: *** [mod_spidermonkey-all] Error 1
make[2]: *** [all-recursive] Error 1

Any idea?

Max.
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From brian at freeswitch.org  Mon Aug 10 11:53:44 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 13:53:44 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
Message-ID: <F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>

Sounds like you have a hardware problem... I just built fine on 5.3 ..  
just make sure you don't do make -j on the first build.

/b

On Aug 10, 2009, at 1:51 PM, Max Bridgewater wrote:

> Hi get the following messages while running "make" on CentOS 5.3 (Of  
> course after bootstrap and configure).
>
> os_Linux_x86_64.s: Assembler messages:
> os_Linux_x86_64.s:46: Error: bad register name `%rdi)'
> os_Linux_x86_64.s:61: Error: bad register name `%rdi)'
> os_Linux_x86_64.s:75: Error: bad register name `%rdi)'
> os_Linux_x86_64.s:89: Error: bad register name `%rdi)'
> make[10]: *** [os_Linux_x86_64.o] Error 1
> make[9]: *** [export] Error 2
> make[8]: *** [export] Error 2
> make[7]: *** [export] Error 2
> make[6]: *** [export] Error 2
> make[5]: *** [/home/installs/fstrunk/libs/js/libjs.la] Error 2
> make[4]: *** [all] Error 1
> make[3]: *** [mod_spidermonkey-all] Error 1
> make[2]: *** [all-recursive] Error 1
>
> Any idea?
>
> Max.

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From pete at privateconnect.com  Mon Aug 10 11:59:49 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Mon, 10 Aug 2009 11:59:49 -0700
Subject: [Freeswitch-users] VoiceMail transcription
Message-ID: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>

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From max.bridgewater at gmail.com  Mon Aug 10 12:02:58 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 15:02:58 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
Message-ID: <c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>

Hmm too bad. It's a new VPS that i purchased. I'm not using make -j and l'm
really sure what to do next.


On Mon, Aug 10, 2009 at 2:53 PM, Brian West <brian at freeswitch.org> wrote:

> Sounds like you have a hardware problem... I just
> built fine on 5.3 .. just make sure you don't do make -j on the first build.
> /b
>
> On Aug 10, 2009, at 1:51 PM, Max Bridgewater wrote:
>
> Hi get the following messages while running "make" on CentOS 5.3 (Of course
> after bootstrap and configure).
>
> os_Linux_x86_64.s: Assembler messages:
> os_Linux_x86_64.s:46: Error: bad register name `%rdi)'
> os_Linux_x86_64.s:61: Error: bad register name `%rdi)'
> os_Linux_x86_64.s:75: Error: bad register name `%rdi)'
> os_Linux_x86_64.s:89: Error: bad register name `%rdi)'
> make[10]: *** [os_Linux_x86_64.o] Error 1
> make[9]: *** [export] Error 2
> make[8]: *** [export] Error 2
> make[7]: *** [export] Error 2
> make[6]: *** [export] Error 2
> make[5]: *** [/home/installs/fstrunk/libs/js/libjs.la] Error 2
> make[4]: *** [all] Error 1
> make[3]: *** [mod_spidermonkey-all] Error 1
> make[2]: *** [all-recursive] Error 1
>
> Any idea?
>
> Max.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Mon Aug 10 12:04:31 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 10 Aug 2009 15:04:31 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
Message-ID: <40A2F6B2-BEA0-423A-A1E3-86EDE0EEC3AC@avgs.ca>

What arch? do a cat /proc/cpuinfo

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 10-Aug-09, at 3:02 PM, Max Bridgewater wrote:

>
> Hmm too bad. It's a new VPS that i purchased. I'm not using make -j  
> and l'm really sure what to do next.
>
>
> On Mon, Aug 10, 2009 at 2:53 PM, Brian West <brian at freeswitch.org>  
> wrote:
> Sounds like you have a hardware problem... I just built fine on  
> 5.3 .. just make sure you don't do make -j on the first build.
>
> /b
>
> On Aug 10, 2009, at 1:51 PM, Max Bridgewater wrote:
>
>> Hi get the following messages while running "make" on CentOS 5.3  
>> (Of course after bootstrap and configure).
>>
>> os_Linux_x86_64.s: Assembler messages:
>> os_Linux_x86_64.s:46: Error: bad register name `%rdi)'
>> os_Linux_x86_64.s:61: Error: bad register name `%rdi)'
>> os_Linux_x86_64.s:75: Error: bad register name `%rdi)'
>> os_Linux_x86_64.s:89: Error: bad register name `%rdi)'
>> make[10]: *** [os_Linux_x86_64.o] Error 1
>> make[9]: *** [export] Error 2
>> make[8]: *** [export] Error 2
>> make[7]: *** [export] Error 2
>> make[6]: *** [export] Error 2
>> make[5]: *** [/home/installs/fstrunk/libs/js/libjs.la] Error 2
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_spidermonkey-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>>
>> Any idea?
>>
>> Max.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Mon Aug 10 12:09:28 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 14:09:28 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
Message-ID: <30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>

Sounds like a 64bit kernel and 32bit userspace maybe?  Just  
guessing... why oh why oh VPS providers have recicarnail interface  
problems.

/b

On Aug 10, 2009, at 2:02 PM, Max Bridgewater wrote:

>
> Hmm too bad. It's a new VPS that i purchased. I'm not using make -j  
> and l'm really sure what to do next.
>



From max.bridgewater at gmail.com  Mon Aug 10 12:20:28 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 15:20:28 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
Message-ID: <c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>

OK, here is the result of cat /proc/cpuinfo. What conclusion can i draw from
this?


processor       : 0
vendor_id       : GenuineIntel
cpu family      : 6
model           : 15
model name      : Intel(R) Core(TM)2 Quad CPU    Q6600  @ 2.40GHz
stepping        : 11
cpu MHz         : 2400.084
cache size      : 4096 KB
physical id     : 0
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mca cmov pat pse36 clflush
dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc pni monitor
ds_cpl vmx est tm2 cx16 xtpr lahf_lm
bogomips        : 6005.15
clflush size    : 64
cache_alignment : 64
address sizes   : 36 bits physical, 48 bits virtual
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From brian at freeswitch.org  Mon Aug 10 12:25:41 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 14:25:41 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
Message-ID: <D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>

and uname -a says?

/b

On Aug 10, 2009, at 2:20 PM, Max Bridgewater wrote:

> OK, here is the result of cat /proc/cpuinfo. What conclusion can i  
> draw from this?
>
>
> processor       : 0
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 15
> model name      : Intel(R) Core(TM)2 Quad CPU    Q6600  @ 2.40GHz
> stepping        : 11
> cpu MHz         : 2400.084
> cache size      : 4096 KB
> physical id     : 0
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mca cmov pat pse36  
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm  
> constant_tsc pni monitor ds_cpl vmx est tm2 cx16 xtpr lahf_lm
> bogomips        : 6005.15
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 36 bits physical, 48 bits virtual
>



From max.bridgewater at gmail.com  Mon Aug 10 12:31:04 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 15:31:04 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
Message-ID: <c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>

and uname -a says?


Linux miriam 2.6.18-128.1.10.el5xen #1 SMP Thu May 7 11:07:18 EDT 2009
x86_64 x86_64 x86_64 GNU/Linux
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From brian at freeswitch.org  Mon Aug 10 12:32:50 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 14:32:50 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
Message-ID: <06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>

I would call them and make sure they know they have something broken.

/b

On Aug 10, 2009, at 2:31 PM, Max Bridgewater wrote:

>
>
> and uname -a says?
>
> Linux miriam 2.6.18-128.1.10.el5xen #1 SMP Thu May 7 11:07:18 EDT  
> 2009 x86_64 x86_64 x86_64 GNU/Linux
>



From max.bridgewater at gmail.com  Mon Aug 10 12:49:07 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 15:49:07 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
Message-ID: <c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>

I have no idea what is wrong and therefore what to tell them. Without
telling precisely what is wrong it would be easy for them to just categorize
this as an application issue. Any recommendation?

On Mon, Aug 10, 2009 at 3:32 PM, Brian West <brian at freeswitch.org> wrote:

> I would call them and make sure they know they have something broken.
>
> /b
>
> On Aug 10, 2009, at 2:31 PM, Max Bridgewater wrote:
>
> >
> >
> > and uname -a says?
> >
> > Linux miriam 2.6.18-128.1.10.el5xen #1 SMP Thu May 7 11:07:18 EDT
> > 2009 x86_64 x86_64 x86_64 GNU/Linux
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Aug 10 12:51:35 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 14:51:35 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
	<c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
Message-ID: <5E790EFC-8A43-428B-B1BB-C84014D3422F@freeswitch.org>

Just be like everyone else... "Its broken, FIX IT!"  :P  ... Honestly  
I am not sure what to tell them... Could be a Xen bug... never know.

/b

On Aug 10, 2009, at 2:49 PM, Max Bridgewater wrote:

> I have no idea what is wrong and therefore what to tell them.  
> Without telling precisely what is wrong it would be easy for them to  
> just categorize this as an application issue. Any recommendation?
>
> On Mon, Aug 10, 2009 at 3:32 PM, Brian West <brian at freeswitch.org>  
> wrote:
> I would call them and make sure they know they have something broken.
>
> /b

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From mike at jerris.com  Mon Aug 10 13:01:04 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 Aug 2009 13:01:04 -0700
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
	<c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
Message-ID: <C41A1A58-090E-4A71-9AA7-FEEFFBDD99D3@jerris.com>

Is this a 32 bit default compiler?  Try CFLAGS=-m64 on the configure  
line.

On Aug 10, 2009, at 12:49 PM, Max Bridgewater  
<max.bridgewater at gmail.com> wrote:

> I have no idea what is wrong and therefore what to tell them.  
> Without telling precisely what is wrong it would be easy for them to  
> just categorize this as an application issue. Any recommendation?
>
> On Mon, Aug 10, 2009 at 3:32 PM, Brian West <brian at freeswitch.org>  
> wrote:
> I would call them and make sure they know they have something broken.
>
> /b
>
> On Aug 10, 2009, at 2:31 PM, Max Bridgewater wrote:
>
> >
> >
> > and uname -a says?
> >
> > Linux miriam 2.6.18-128.1.10.el5xen #1 SMP Thu May 7 11:07:18 EDT
> > 2009 x86_64 x86_64 x86_64 GNU/Linux
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From pjintheusa at gmail.com  Mon Aug 10 13:35:50 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 10 Aug 2009 16:35:50 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
Message-ID: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>

Hi there,

I am attempting to do a simple bridge. Leg A is an inbound DID. Leg B
is terminated through a SIP carrier.

I am unable to use loopback AND bypass_media_after_bridge=true. The
bridge fails.

Here is my three line application:

Session.Answer();

Session.Execute("set", "bypass_media_after_bridge=true");

Session.Execute("bridge", "loopback/6095553828/default");

FreeSWITCH log: http://pastebin.freeswitch.org/9949

I believe the bridge fails as the SIP-REINVITE is issued and fails.

Using loopback and bypass_media_after_bridge=false works fine. As does
using bypass_media_after_bridge=true and sending call directly through
the gateway to which the loopback points.

e.g.  http://pastebin.freeswitch.org/9948

Has anyone else encountered this? Am I missing something?

Any help would be much appreciated!

Thanks


Phillip Jones


From msc at freeswitch.org  Mon Aug 10 15:56:14 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Aug 2009 17:56:14 -0500
Subject: [Freeswitch-users] question about latest version of mod_limit
In-Reply-To: <20ad6b920908060226t2dcf532aobab23dc0299d0f05@mail.gmail.com>
References: <20ad6b920908060226t2dcf532aobab23dc0299d0f05@mail.gmail.com>
Message-ID: <87f2f3b90908101556s6836143j7103124c94bdd9ad@mail.gmail.com>

Possibly the calls are being disconnected too quickly? Try adding a sleep
for 1000ms on each one and see what happens.
-MC

On Thu, Aug 6, 2009 at 4:26 AM, mark morreny <markmorreny at gmail.com> wrote:

> Hello,
>
> I have the following setup in the dialplan.  Then, I fire up sipp to send
> 5calls/s and I expect to get limit-pass=false in most of the INFO output.
> However, I am getting all "limit-pass=pass".
>
> Does anyone know what is wrong with my dialplan?
>
>
>  <context name="internal">
>
>   <extension name="test-limit">
>
>     <condition field="destination_number" expression="^9(.*)$">
>
>         <action application="limit_hash_execute" data="192.168.1.102 6000
> 1/1 transfer to-next" />
>         <action application="set" data="limit-pass=false"/>
>         <action application="info" />
>         <action application="hangup" />
>
>     </condition>
>
>   </extension>
>     <extension name="to-next">
>         <condition field="destination_number" expression="^to-next$">
>         <action application="set" data="limit-pass=true"/>
>         <action application="info" />
>         <action application="hangup" />
>
>
>
>     </condition>
>     </extension>
> </context>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Mon Aug 10 16:00:18 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 10 Aug 2009 19:00:18 -0400
Subject: [Freeswitch-users] question about latest version of mod_limit
In-Reply-To: <20ad6b920908060226t2dcf532aobab23dc0299d0f05@mail.gmail.com>
References: <20ad6b920908060226t2dcf532aobab23dc0299d0f05@mail.gmail.com>
Message-ID: <3DF62608-6E4E-4322-85EC-5384FEC58A71@avgs.ca>

You are using it wrong.
<action application="limit_hash_execute" data="192.168.1.102 6000 1/1  
bridge sofia/gateway/etc" />

The purpose of limit_hash_execute is to have a resource that is  
released if the command doesnt complete properly.
If you want to transfer when the res. is over-limit, use limit_hash,  
not limit_hash_execute.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 6-Aug-09, at 5:26 AM, mark morreny wrote:

> Hello,
>
> I have the following setup in the dialplan.  Then, I fire up sipp to  
> send 5calls/s and I expect to get limit-pass=false in most of the  
> INFO output.  However, I am getting all "limit-pass=pass".
>
> Does anyone know what is wrong with my dialplan?
>
>
>  <context name="internal">
>
>   <extension name="test-limit">
>
>     <condition field="destination_number" expression="^9(.*)$">
>
>         <action application="limit_hash_execute" data="192.168.1.102  
> 6000 1/1 transfer to-next" />
>         <action application="set" data="limit-pass=false"/>
>         <action application="info" />
>         <action application="hangup" />
>
>     </condition>
>
>   </extension>
>     <extension name="to-next">
>         <condition field="destination_number" expression="^to-next$">
>         <action application="set" data="limit-pass=true"/>
>         <action application="info" />
>         <action application="hangup" />
>
>
>
>     </condition>
>     </extension>
> </context>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Mon Aug 10 16:04:24 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Aug 2009 18:04:24 -0500
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
Message-ID: <87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>

Be sure to hop on IRC and speak with jmesquita because he's been working on
this also. It would be very good to have Spanish-speaking users review the
Spanish phrase_es.xml file for correctness. Also, we need Spanish-speaking
programmers to assist with the mod_say application for Spanish - there are
things that you have to do in Spanish that you don't have to do in English.

Thanks,
MC

On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

> On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
> edition .
>
> Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for
> FS playback.
>
> Here's a guide that has been put together for reference on what to record.
>
> http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml
>
> Regards,
>
>
>
>
> On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:
>
>> I'd like to begin record spanish prompts for FS.
>>
>> Do you know any software/hardware to make it?
>>
>> Thank you
>>
>> BR
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Aug 10 16:29:50 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Aug 2009 18:29:50 -0500
Subject: [Freeswitch-users] Error while creating object
In-Reply-To: <24895716.post@talk.nabble.com>
References: <7d79b3930908060347xe5be545yfeeafad761aba274@mail.gmail.com>
	<24895716.post@talk.nabble.com>
Message-ID: <87f2f3b90908101629p71048eb1vbccd19a19d328fff@mail.gmail.com>

On Mon, Aug 10, 2009 at 3:00 AM, lakshmanan <lakindia89 at gmail.com> wrote:

>
> Can any one please say what I did wrong here?
>

Maybe this instead?
my $sess=&freeswitch::DTMF->new();
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From kevin at kgolding.co.uk  Mon Aug 10 16:36:02 2009
From: kevin at kgolding.co.uk (Kevin Golding)
Date: Tue, 11 Aug 2009 00:36:02 +0100
Subject: [Freeswitch-users] ACL issue
Message-ID: <4A80AEE2.7050306@kgolding.co.uk>

Hello,

Am in the progress of setting up a test freeswitch box. Have got
internal extensions working, along with an outgoing gateway.

My issue is getting the incoming calls.

The error I'm getting is: "2009-08-10 21:47:42 [DEBUG] sofia.c:3785
sofia_handle_sip_i_invite() IP 213.166.5.129 Rejected by acl "domains".
Falling back to Digest auth."

After reading the docs, I believed I needed to add the 213.166.5.129 IP
to the "domains" list, but alas I could not find such a list by default.
It appears I was missing an acl.conf.xml file. I have created a new
acl.conf.xml file, and have put the following in it:

<configuration name="acl.conf" description="Network Lists">
   <network-lists>
     <list name="domains" value="allow">
         <node type="allow" domain="$${domain}"/>
         <node type="allow" domain="213.166.5.128/28"/>
         <node type="allow" domain="87.238.72.128/26"/>
     </list>
   </network-lists>
</configuration>

But I am still getting the same error for an incoming call.

Best regards,
--
Kevin Golding


From brian at freeswitch.org  Mon Aug 10 16:46:20 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 18:46:20 -0500
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <4A80AEE2.7050306@kgolding.co.uk>
References: <4A80AEE2.7050306@kgolding.co.uk>
Message-ID: <7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>

First off edit the profile to use your own custom ACL... not the  
domains one... secondly you'll have to use cidr= instead of domain=.

Domain= will search the user directory building an ACL list from all  
users with the cidr= attribute.

/b


On Aug 10, 2009, at 6:36 PM, Kevin Golding wrote:

> acl.conf.xml file, and have put the following in it:
>
> <configuration name="acl.conf" description="Network Lists">
>   <network-lists>
>     <list name="domains" value="allow">
>         <node type="allow" domain="$${domain}"/>
>         <node type="allow" domain="213.166.5.128/28"/>
>         <node type="allow" domain="87.238.72.128/26"/>
>     </list>
>   </network-lists>
> </configuration>



From max.bridgewater at gmail.com  Mon Aug 10 17:01:27 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 20:01:27 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <C41A1A58-090E-4A71-9AA7-FEEFFBDD99D3@jerris.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
	<c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
	<C41A1A58-090E-4A71-9AA7-FEEFFBDD99D3@jerris.com>
Message-ID: <c55d46950908101701o50d94acle2f183f744d97591@mail.gmail.com>

I suppose it is a 32 bit default compiler. When i compile with the suggested
flag it fails right away. I've attached the log file for your curiosity.
Meanwhil i've submited the issue to the hosting company. But i doubt
anything meaningful or useful will come from there.

Cheers,
Max.


On Mon, Aug 10, 2009 at 4:01 PM, Michael Jerris <mike at jerris.com> wrote:

> Is this a 32 bit default compiler?  Try CFLAGS=-m64 on the configure line.
>
>
> On Aug 10, 2009, at 12:49 PM, Max Bridgewater <max.bridgewater at gmail.com>
> wrote:
>
> I have no idea what is wrong and therefore what to tell them. Without
> telling precisely what is wrong it would be easy for them to just categorize
> this as an application issue. Any recommendation?
>
> On Mon, Aug 10, 2009 at 3:32 PM, Brian West < <brian at freeswitch.org>
> brian at freeswitch.org> wrote:
>
>> I would call them and make sure they know they have something broken.
>>
>> /b
>>
>> On Aug 10, 2009, at 2:31 PM, Max Bridgewater wrote:
>>
>> >
>> >
>> > and uname -a says?
>> >
>> > Linux miriam 2.6.18-128.1.10.el5xen #1 SMP Thu May 7 11:07:18 EDT
>> > 2009 x86_64 x86_64 x86_64 GNU/Linux
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Aug 10 17:06:08 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 19:06:08 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101701o50d94acle2f183f744d97591@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
	<c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
	<C41A1A58-090E-4A71-9AA7-FEEFFBDD99D3@jerris.com>
	<c55d46950908101701o50d94acle2f183f744d97591@mail.gmail.com>
Message-ID: <228DA52C-066F-4165-A568-DF9688EF4C61@freeswitch.org>

What flags are you using?
/b

On Aug 10, 2009, at 7:01 PM, Max Bridgewater wrote:

> I suppose it is a 32 bit default compiler. When i compile with the  
> suggested flag it fails right away. I've attached the log file for  
> your curiosity. Meanwhil i've submited the issue to the hosting  
> company. But i doubt anything meaningful or useful will come from  
> there.
>
> Cheers,
> Max.
>



From max.bridgewater at gmail.com  Mon Aug 10 17:14:41 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 20:14:41 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <228DA52C-066F-4165-A568-DF9688EF4C61@freeswitch.org>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
	<c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
	<C41A1A58-090E-4A71-9AA7-FEEFFBDD99D3@jerris.com>
	<c55d46950908101701o50d94acle2f183f744d97591@mail.gmail.com>
	<228DA52C-066F-4165-A568-DF9688EF4C61@freeswitch.org>
Message-ID: <c55d46950908101714x6690cf17o5b81bbc6ea04f7@mail.gmail.com>

CFLAGS=-m64 on the configure line.



On Mon, Aug 10, 2009 at 8:06 PM, Brian West <brian at freeswitch.org> wrote:

> What flags are you using?
> /b
>
> On Aug 10, 2009, at 7:01 PM, Max Bridgewater wrote:
>
> > I suppose it is a 32 bit default compiler. When i compile with the
> > suggested flag it fails right away. I've attached the log file for
> > your curiosity. Meanwhil i've submited the issue to the hosting
> > company. But i doubt anything meaningful or useful will come from
> > there.
> >
> > Cheers,
> > Max.
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Aug 10 17:53:18 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 19:53:18 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101714x6690cf17o5b81bbc6ea04f7@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
	<c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
	<C41A1A58-090E-4A71-9AA7-FEEFFBDD99D3@jerris.com>
	<c55d46950908101701o50d94acle2f183f744d97591@mail.gmail.com>
	<228DA52C-066F-4165-A568-DF9688EF4C61@freeswitch.org>
	<c55d46950908101714x6690cf17o5b81bbc6ea04f7@mail.gmail.com>
Message-ID: <580A1245-3E46-49AB-A95F-1F276F59B738@freeswitch.org>

You shouldn't be doing that.

/b

On Aug 10, 2009, at 7:14 PM, Max Bridgewater wrote:

> CFLAGS=-m64 on the configure line.



From jmesquita at gmail.com  Mon Aug 10 20:00:43 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 11 Aug 2009 00:00:43 -0300
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
Message-ID: <5a8712120908102000m421ea27fv371620bf66d69f6b@mail.gmail.com>

Hey! I have recorded a couple of samples and I will patch whatever is needed
to support portuguese(Brazil) and spanish on say. Don't worry, I am on top
of it.

jmesquita

On Mon, Aug 10, 2009 at 8:04 PM, Michael Collins <msc at freeswitch.org> wrote:

> Be sure to hop on IRC and speak with jmesquita because he's been working on
> this also. It would be very good to have Spanish-speaking users review the
> Spanish phrase_es.xml file for correctness. Also, we need Spanish-speaking
> programmers to assist with the mod_say application for Spanish - there are
> things that you have to do in Spanish that you don't have to do in English.
>
> Thanks,
> MC
>
>
> On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>
>> On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
>> edition .
>>
>> Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for
>> FS playback.
>>
>> Here's a guide that has been put together for reference on what to record.
>>
>> http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml
>>
>> Regards,
>>
>>
>>
>>
>> On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:
>>
>>> I'd like to begin record spanish prompts for FS.
>>>
>>> Do you know any software/hardware to make it?
>>>
>>> Thank you
>>>
>>> BR
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Mon Aug 10 21:03:37 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 11 Aug 2009 12:03:37 +0800
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <E6E26B16-4905-49DE-A167-BEFC7A0D1638@myrvold.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
	<6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
	<4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
	<092B4355-F911-4E37-A5D1-AC923AF9B8FB@gmail.com>
	<E6E26B16-4905-49DE-A167-BEFC7A0D1638@myrvold.org>
Message-ID: <23f91030908102103g50f49af4o97dd97ef0df666a4@mail.gmail.com>

2009/8/10 Ivan C Myrvold <ivan at myrvold.org>

> Seven,
> I am afraid I will not be able to help you much with the Carbon code,
> as I am only good at Cocoa programming. You said you chose Carbon
> because you only needed low level API, and that is fair enough, but I
> will also add that you can do the same with only linking to the
> Foundation framework in Cocoa.
>

Link with Cocoa is OK but just need extra hack to the standard Makefile and
not so necessary. And I think Carbon code is more C friendly.
I made a jira to Skype-Dev, hope someone can help us.


>
> I looked a little at the diff file yesterday, and will investigate
> more today, to try to understand how you have done the Skype
> integration to the Freeswitch in the Carbon code.
> And I am glad that someone have contributed to get skypiax working in
> OS X. Great work so far!
>
> Ivan
>
> Den 9. aug.. 2009 kl. 20:02 skrev Seven Du:
>
> > Ivan,
> >
> > Good to know you are a cocoa dev. Unable to check in code right now,
> > will send the diff to you offlist for now.
> >
> > 0) I'm not familiar with Mac dev, just tried my best
> > 1) It doesn't work yet, but should be able to compile, sure you
> > already have the Skype framework in place :)
> > 2) if run the skype delegate from a threat, then cannot get event
> > callback. e.g. mac_client.c works but mac_client2.c doesn't. Since
> > skypiax is running in a thread, we need to figure out this first.
> > 3) it uses Carbon, since I think we only need to low level api, no
> > need to bother the complicate of Cocoa.
> > 4) strsep shows some warning on compile, haven't figured out why
> > 5) perhaps you should only add one interface in skypiax.conf.xml
> > 6) do you want to run multi-instances like on Linux?
> > 7) I really not sure if it will work or not :)
> >
> > Let me know if it helps. I bet you can make it work. Also code will be
> > in my branch soon.
> >
> > 7.
> >
> >
> > On Aug 9, 2009, at 11:34 PM, Ivan C Myrvold wrote:
> >> Yes, I am interested in this, and if you have any source I could have
> >> a look at it.
> >>
> >> Ivan
> >>
> >> Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
> >>
> >>>
> >>> On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
> >>>> Ciao Ivan,
> >>>>
> >>>> it seems that you do not have the libX11 **development** package
> >>>> installed.
> >>>>
> >>>> Unfortunately I don't know about OSX, so I cannot help you, but
> >>>> many
> >>>> on the list know.
> >>>> BTW: it will probably be of no use to you to compile mod_skypiax on
> >>>> OSX, because Skype for MACOSX works in another way than Skype for
> >>>> Linux.
> >>>
> >>> That's right.
> >>>
> >>>> If you know about MacOSX programming, please have a look at
> >>>> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
> >>>> probably be simple enough to add a message pump for MacOSX.
> >>>>
> >>>> -giovanni
> >>>>
> >>>>
> >>> Giovanni, I have a Mac and tried to get this work yesterday, but
> >>> haven't got it work. Will try further if I have time. However, I
> >>> don't
> >>> think it's so useful because I don't know how to run and hence
> >>> control
> >>> multi-skype instances on Mac.
> >>>
> >>> If someone interested to try this I can check the code into my
> >>> branch.
> >>>
> >>>>
> >>>>
> >>>> Sincerely,
> >>>>
> >>>> Giovanni Maruzzelli
> >>>> =========================================
> >>>> www.celliax.org
> >>>> via Pierlombardo 9, 20135 Milano
> >>>> Italy
> >>>> gmaruzz at celliax dot org
> >>>> Cell : +39-347-2665618
> >>>> Fax : +39-02-87390039
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>
> >>>> wrote:
> >>>>> I tried to compile mod_skypiax, and am getting problem with X11.
> >>>>> On
> >>>>> OS
> >>>>> X Leopard, X11 is installed in /usr/X11/lib/
> >>>>> See below.
> >>>>>
> >>>>> What can I do to get past this error?
> >>>>>
> >>>>> I can also let you ssh into my machine. Contact me off list in
> >>>>> case.
> >>>>>
> >>>>> Ivan
> >>>>>
> >>>>> making all mod_skypiax
> >>>>> Compiling skypiax_protocol.c...
> >>>>> Compiling mod_skypiax.c...
> >>>>> mkdir .libs
> >>>>> Compiling mod_skypiax.c ...
> >>>>> Creating mod_skypiax.so...
> >>>>> ld: library not found for -lX11
> >>>>> collect2: ld returned 1 exit status
> >>>>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
> >>>>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
> >>>>> Documents/
> >>>>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
> >>>>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -
> >>>>> g -
> >>>>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -
> >>>>> shared -
> >>>>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-
> >>>>> namespace .libs/
> >>>>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/
> >>>>> Freeswitch/
> >>>>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
> >>>>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/
> >>>>> xml/
> >>>>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.
> >>>>> 09-08-09/
> >>>>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
> >>>>> libiconv.dylib /
> >>>>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
> >>>>> apr/.libs/
> >>>>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -
> >>>>> lz -
> >>>>> lncurses -lX11
> >>>>> make[5]: *** [mod_skypiax.so] Error 1
> >>>>> make[4]: *** [all] Error 1
> >>>>> make[3]: *** [mod_skypiax-all] Error 1
> >>>>> make[2]: *** [all-recursive] Error 1
> >>>>>
> >>>>>
> >>>>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
> >>>>>
> >>>>>> No, it needs implementation of the message pump between the
> >>>>>> module
> >>>>>> and
> >>>>>> the Skype API.
> >>>>>>
> >>>>>> It's probably kind of trivial, if no other problems I'm not aware
> >>>>>> of.
> >>>>>>
> >>>>>> I do not have a Mac to implement it, tough :-(.
> >>>>>>
> >>>>>> -giovanni
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> Sincerely,
> >>>>>>
> >>>>>> Giovanni Maruzzelli
> >>>>>> =========================================
> >>>>>> www.celliax.org
> >>>>>> via Pierlombardo 9, 20135 Milano
> >>>>>> Italy
> >>>>>> gmaruzz at celliax dot org
> >>>>>> Cell : +39-347-2665618
> >>>>>> Fax : +39-02-87390039
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
> >>>>>> wrote:
> >>>>>>> I'm not sure about that one.... I haven't tried lately because
> >>>>>>> the
> >>>>>>> API
> >>>>>>> differs on the Mac last I looked at it.
> >>>>>>>
> >>>>>>> /b
> >>>>>>>
> >>>>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
> >>>>>>>
> >>>>>>>> Is skypiax now working on Mac OS X in Freeswitch?
> >>>>>>>>
> >>>>>>>> Ivan
> >>>>>>>
> >>>>>>>
> >>>>>>> _______________________________________________
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> http://www.freeswitch.org
> >>>>>>>
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From velu.technical at gmail.com  Mon Aug 10 21:29:44 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Tue, 11 Aug 2009 09:59:44 +0530
Subject: [Freeswitch-users] Fwd: ALARM signal in esl libraries
In-Reply-To: <1452e2980908092338j15c4e33cn2cff799bb64464d0@mail.gmail.com>
References: <1452e2980908092338j15c4e33cn2cff799bb64464d0@mail.gmail.com>
Message-ID: <1452e2980908102129g51255d5bt430a3e73bd6f2e99@mail.gmail.com>

Dear Greats,
    Could you please help me to solve this problem?

---------- Forwarded message ----------
From: velusamy velu <velu.technical at gmail.com>
Date: Mon, Aug 10, 2009 at 12:08 PM
Subject: ALARM signal in esl libraries
To: freeswitch-users at lists.freeswitch.org


Dear All,
       I have registered  ALARM signal  in my perl program to handle the
DTMF digit timeout.
When ALARM signal generated  the  connection with  ESL  is  automatically
closed.

I have checked the connection with "connected: function, it returns 0.

Why the connection was closed?
Is there any idea to alive the connection after ALARM signal generation??

Please help me.........

Thanks,
Velusamy
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From godson.g at gmail.com  Mon Aug 10 22:20:33 2009
From: godson.g at gmail.com (Godson Gera)
Date: Tue, 11 Aug 2009 10:50:33 +0530
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
Message-ID: <f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>

Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
name resolution. But the problem is still there even if I use my ISP's  DNS
(mostly because its resolving to the same to IP address) .

On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org> wrote:

> As MikeJ pointed out please report the IP address files.freeswitch.orgresolves to.  We have that on a content delivery network so that the files
> are closer to you geographically and you can download them faster but if
> you're having an issue I'll need the IP so I can report it correctly.
> Thanks,
> Brian
>
> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>
> Hi FS Team,
>
>
>         The files.freeswitch.org is resetting connection since 3 days. As
> a result I was not able to download latest release of FS. Got the trunk
> version from svn. But still it suffers from the lack of sound files.  When
> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
> connection immediately wget tries 20 times and gives up. Other users on IRC
> also reported this issue.
>
> --
> Thanks & Regards,
> Godson Gera
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Thanks & Regards,
Godson Gera
http://godson.in
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From velu.technical at gmail.com  Mon Aug 10 22:27:15 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Tue, 11 Aug 2009 10:57:15 +0530
Subject: [Freeswitch-users] Timeouts in Dial plan
Message-ID: <1452e2980908102227o90aaef2i2f8fd36a2eaeefb6@mail.gmail.com>

Dear All,
      How to handle timeouts in Dialplan?

Thanks,
Velusamy.
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From peter.olsson at visionutveckling.se  Mon Aug 10 23:16:44 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 11 Aug 2009 08:16:44 +0200
Subject: [Freeswitch-users] OpenZAP/Sangoma in Windows
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED9435@cooper>

Hi, I'm trying to evaluate the OpenZAP/Sangoma-support in Windows, using PRI E1 connections.

I'm aware it's not yet to be considered as stable, but I'd still want to try it out some, and also help detecting bugs while I'm at it :)

Anyway, I have a couple of questions:

1. Has anyone tested it in Windows at all? I know the build-files for Visual Studio has only been checked in for a couple of months, so that's why I'm asking.

2. Does anyone have any directions how to configure the driver within Windows? Should I use BitStream or HDLC, and how should the channel groups be configured?

I have built everything correctly, and I've succeeded to open (some of) the channels when I configure each channel in a separate group, but I don't think that's the way it should be done...

Any suggestions welcome!

/Peter


From peter.olsson at visionutveckling.se  Mon Aug 10 23:20:50 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 11 Aug 2009 08:20:50 +0200
Subject: [Freeswitch-users] UniMRCP support for Windows...
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED9436@cooper>

Yet another Windows question from me.. :)

I've seen that vcproj-files for mod_unimrcp has been added to SVN, but they are not yet included in the main FreeSWITCH solution-file.

What's the curent status of this, should it work if I compile it separately, or is it not yet complete for Windows?

/Peter


From diego.viola at gmail.com  Mon Aug 10 23:54:39 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 11 Aug 2009 02:54:39 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
Message-ID: <86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>

The address for me is: 69.174.57.101

On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera <godson.g at gmail.com> wrote:

> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
> name resolution. But the problem is still there even if I use my ISP's  DNS
> (mostly because its resolving to the same to IP address) .
>
> On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org> wrote:
>
>> As MikeJ pointed out please report the IP address files.freeswitch.orgresolves to.  We have that on a content delivery network so that the files
>> are closer to you geographically and you can download them faster but if
>> you're having an issue I'll need the IP so I can report it correctly.
>> Thanks,
>> Brian
>>
>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>
>> Hi FS Team,
>>
>>
>>         The files.freeswitch.org is resetting connection since 3 days. As
>> a result I was not able to download latest release of FS. Got the trunk
>> version from svn. But still it suffers from the lack of sound files.  When
>> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>> connection immediately wget tries 20 times and gives up. Other users on IRC
>> also reported this issue.
>>
>> --
>> Thanks & Regards,
>> Godson Gera
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Thanks & Regards,
> Godson Gera
> http://godson.in
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From maxim.tsvetov at gmail.com  Tue Aug 11 00:10:36 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Tue, 11 Aug 2009 11:10:36 +0400
Subject: [Freeswitch-users] answer command
Message-ID: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>

Hello,

I found in WIKI list of supported commands
http://wiki.freeswitch.org/wiki/Mod_commands.

I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)

That's what Freeswitch replies for this command:

"answer: Command not found!"

?ould you please help?


Regards,
Maxim Tsvetov
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From kevin at kgolding.co.uk  Tue Aug 11 00:38:20 2009
From: kevin at kgolding.co.uk (Kevin Golding)
Date: Tue, 11 Aug 2009 08:38:20 +0100
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>
References: <4A80AEE2.7050306@kgolding.co.uk>
	<7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>
Message-ID: <4A811FEC.7040808@kgolding.co.uk>

Thanks Brian,

Well spotted with the domain/cidr :)

I changed the line <param name="apply-inbound-acl" value="domains"/> in 
the internal.xml with a new value, and changed the list to match but I 
still get the same 'Rejected by acl "domains"' error.

And yes I reloaded the xml. :)

Kevin

Brian West wrote:
> First off edit the profile to use your own custom ACL... not the  
> domains one... secondly you'll have to use cidr= instead of domain=.
> 
> Domain= will search the user directory building an ACL list from all  
> users with the cidr= attribute.
> 
> /b
> 
> 
> On Aug 10, 2009, at 6:36 PM, Kevin Golding wrote:
> 
>> acl.conf.xml file, and have put the following in it:
>>
>> <configuration name="acl.conf" description="Network Lists">
>>   <network-lists>
>>     <list name="domains" value="allow">
>>         <node type="allow" domain="$${domain}"/>
>>         <node type="allow" domain="213.166.5.128/28"/>
>>         <node type="allow" domain="87.238.72.128/26"/>
>>     </list>
>>   </network-lists>
>> </configuration>



From solko at gcdf.pl  Tue Aug 11 00:49:13 2009
From: solko at gcdf.pl (Szymon Olko)
Date: Tue, 11 Aug 2009 09:49:13 +0200
Subject: [Freeswitch-users] answer command
In-Reply-To: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
Message-ID: <4A812279.6080404@gcdf.pl>

Maxim Tsvetov pisze:
> Hello,
> 
> I found in WIKI list of supported commands
> http://wiki.freeswitch.org/wiki/Mod_commands.
> 
> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
> 
> That's what Freeswitch replies for this command:
> 
> "answer: Command not found!"
> 
> ?ould you please help?

Answer is in dialplan commands.

http://wiki.freeswitch.org/wiki/Mod_dptools

Szymon


From lakindia89 at gmail.com  Tue Aug 11 01:59:59 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 11 Aug 2009 14:29:59 +0530
Subject: [Freeswitch-users] Error while creating object
In-Reply-To: <87f2f3b90908101629p71048eb1vbccd19a19d328fff@mail.gmail.com>
References: <7d79b3930908060347xe5be545yfeeafad761aba274@mail.gmail.com>
	<24895716.post@talk.nabble.com>
	<87f2f3b90908101629p71048eb1vbccd19a19d328fff@mail.gmail.com>
Message-ID: <7d79b3930908110159m1a1602a4tcccfb46e33873fbb@mail.gmail.com>

Thanks for your replay.
I've tried that. But it says following error message.

2009-08-11 14:23:09 [ERR] mod_perl.c:69 Perl_safe_eval() [require
'/usr/local/freeswitch/conf/test.pl';]
Undefined subroutine &freeswitch::DTMF called at
/usr/local/freeswitch/conf/test.pl line 6.Compilation failed in require at
(eval 2) line 1.

Please help me to solve this issue!!!

On Tue, Aug 11, 2009 at 4:59 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Aug 10, 2009 at 3:00 AM, lakshmanan <lakindia89 at gmail.com> wrote:
>
>>
>> Can any one please say what I did wrong here?
>>
>
> Maybe this instead?
> my $sess=&freeswitch::DTMF->new();
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From michal.bielicki at halo2.pl  Tue Aug 11 02:33:06 2009
From: michal.bielicki at halo2.pl (Michal Bielicki)
Date: Tue, 11 Aug 2009 11:33:06 +0200
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4a80632f.1508c00a.4d3c.090d@mx.google.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>
	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>
	<4a80632f.1508c00a.4d3c.090d@mx.google.com>
Message-ID: <A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>

I'll retst it later today and give you a link with instructions

Am 10.08.2009 um 20:14 schrieb vmorales:

> By "./compile" I was referring to "./configure"
>
> Vladimir
>
> -----Original Message-----
> From: vmorales [mailto:email.list.subscriber at gmail.com]
> Sent: Monday, August 10, 2009 11:49 AM
> To: 'freeswitch-users at lists.freeswitch.org'
> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> Thanks for the response(s):
>
> I ran the "./compile" script with a set PREFIX.  This took a few
> attempts with errors before it was able to complete error-free, as I
> had to install libtool.
>
> Since then, I have tried running 'make', 'gmake', and
> '/opt/gnu/bin/make', but each results with an error.  This is the
> error when running 'make' or 'gmake':
>
> <snip>
> make: Fatal error: Command failed for target `all-recursive'
> Current working directory /home/vmorales/freeswitch-1.0.4
> *** Error code 1
> make: Fatal error: Command failed for target `all'
> </snip>
>
>
> This is the error when running '/opt/gnu/bin/make':
>
> <snip>
> make[5]: *** [mod_amr.so] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_amr-all] Error 1
> make[2]: *** [all-recursive] Error 1
> Making all in build
> +-------- FreeSWITCH Build Complete -----------+
> + FreeSWITCH has been successfully built.      +
> + Install by running:                          +
> +                                              +
> +               /opt/gnu/bin/make install                   +
> +----------------------------------------------+
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> </snip>
>
>
> I re-untar'd before each compile/make attempt.  Let me know if this is
> something that I can resolve.
>
> Vladimir
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Jerris
> Sent: Saturday, August 08, 2009 12:37 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> This is not currently a supported platform, it only builds on 64 bit
> right now I think on solaris.
>
> Mike
>
> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>
>> Hello,
>>
>> Does anyone have, or know where to get, a pre-compiled copy of
>> FreeSwitch for Solaris 10/x86?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. info at halokwadrat.de
e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
Hauptgesch?ftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 0000153539

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From maxim.tsvetov at gmail.com  Tue Aug 11 05:34:19 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Tue, 11 Aug 2009 05:34:19 -0700 (PDT)
Subject: [Freeswitch-users] answer command
In-Reply-To: <4A812279.6080404@gcdf.pl>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl>
Message-ID: <24916765.post@talk.nabble.com>


Thank you.

Is it possible to use this command from command line or via event_socket
interface?



Szymon Olko wrote:
> 
> Maxim Tsvetov pisze:
>> Hello,
>> 
>> I found in WIKI list of supported commands
>> http://wiki.freeswitch.org/wiki/Mod_commands.
>> 
>> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
>> 
>> That's what Freeswitch replies for this command:
>> 
>> "answer: Command not found!"
>> 
>> ?ould you please help?
> 
> Answer is in dialplan commands.
> 
> http://wiki.freeswitch.org/wiki/Mod_dptools
> 
> Szymon
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/answer-command-tp24912812p24916765.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From samu60 at gmail.com  Tue Aug 11 00:49:52 2009
From: samu60 at gmail.com (samuel)
Date: Tue, 11 Aug 2009 09:49:52 +0200
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
Message-ID: <d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>

I've taken a look at phrase_es.xml and would like other spanish native
speakers to check the following (i preferred to write down here my
impressions so others can contribute instead of sending a patch):

1) Centavos

<prompt phrase="Centavos por minuto" filename="cents-per-minute.wav"/>
<prompt phrase="Centavos" filename="cents.wav"/>

probably in european spanish the most apropiate word for cents is "c?ntimo"
but, if i'm not mistaken, in south american spanish is more common to use
"centavos" so this is more a localization issue...as you wish.
Since I live in Spain I would use the following:
<prompt phrase="C?ntimos por minuto" filename="cents-per-minute.wav"/>
<prompt phrase="C?ntimos" filename="cents.wav"/>

2) Minnor stress corrections:
instead of Dolar, you should use the stressed word D?lar:
-<prompt phrase="Dolar" filename="dollar.wav"/>
+<prompt phrase="D?lar" filename="dollar.wav"/>

instead of numero, you should use the stressed word n?mero:
-<prompt phrase="por favor ingrese su numero de usuario, seguido por"
filename="vm-enter_id.wav"/>
+<prompt phrase="por favor ingrese su n?mero de usuario, seguido por"
filename="vm-enter_id.wav"/>

3) en instead of a
-<prompt phrase="La persona a la extensi?n" filename="vm-person.wav"/>
+<prompt phrase="La persona en la extensi?n" filename="vm-person.wav"/>

4) femenin genre of contrase?a changes the word seguido for seguida
-<prompt phrase="por favor ingrese su contrase?a, seguido por"
filename="vm-enter_pass.wav"/>
+<prompt phrase="por favor ingrese su contrase?a, seguida por"
filename="vm-enter_pass.wav"/>

5) I would propose the word "almohadilla" as the translation of pound so I
would change following sentences:
-<prompt phrase="Signo de n?mero" filename="35.wav"/>
+<prompt phrase="Almohadilla" filename="35.wav"/>

-<prompt phrase="Signo de n?mero" filename="pound.wav"/>
+<prompt phrase="Almohadilla" filename="pound.wav"/>

-<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del
signo de n?mero" filename="ivr-enter_ext.wav"/>
+<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido de
almohadilla" filename="ivr-enter_ext.wav"/>

-<prompt phrase="Seguido del signo de n?mero" filename="followed.wav"/>
+<prompt phrase="Seguido de almohadilla" filename="followed.wav"/>

6)changing prepositions:
-<prompt phrase="Un error ha ocurrido, por favor contacte al administrador"
filename="error.wav"/>
+<prompt phrase="Un error ha ocurrido, por favor contacte con el
administrador" filename="error.wav"/>

7)instead of using ingrese, it sounds more natural to use introduzca when
interacting with an IVR asking for inserting data. So:
-<prompt phrase="por favor ingrese su numero de usuario, seguido por"
filename="vm-enter_id.wav"/>
+<prompt phrase="por favor introduzca su n?mero de usuario, seguido por"
filename="vm-enter_id.wav"/>

-<prompt phrase="por favor ingrese su contrase?a, seguido por"
filename="vm-enter_pass.wav"/>
+<prompt phrase="por favor introduzca su contrase?a, seguida por"
filename="vm-enter_pass.wav"/>

-<prompt phrase="Por favor ingrese el pin de la conferencia."
filename="conf-pin.wav"/>
+<prompt phrase="Por favor introduzca el pin de la conferencia."
filename="conf-pin.wav"/>

-<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del
signo de n?mero" filename="ivr-enter_ext.wav"/>
+<prompt phrase="por favor introduzca el n?mero de una extensi?n seguido de
almohadilla" filename="ivr-enter_ext.wav"/>



Comments?

Samuel


2009/8/11 Michael Collins <msc at freeswitch.org>

> Be sure to hop on IRC and speak with jmesquita because he's been working on
> this also. It would be very good to have Spanish-speaking users review the
> Spanish phrase_es.xml file for correctness. Also, we need Spanish-speaking
> programmers to assist with the mod_say application for Spanish - there are
> things that you have to do in Spanish that you don't have to do in English.
>
> Thanks,
> MC
>
>
> On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>
>> On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
>> edition .
>>
>> Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for
>> FS playback.
>>
>> Here's a guide that has been put together for reference on what to record.
>>
>> http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml
>>
>> Regards,
>>
>>
>>
>>
>> On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:
>>
>>> I'd like to begin record spanish prompts for FS.
>>>
>>> Do you know any software/hardware to make it?
>>>
>>> Thank you
>>>
>>> BR
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From asannucci at gmail.com  Tue Aug 11 06:01:24 2009
From: asannucci at gmail.com (bakko)
Date: Tue, 11 Aug 2009 15:01:24 +0200
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice><f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com><87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
Message-ID: <E97B464EBED44E2AB7D71362840E1836@voztovoice>

Maybe the best solution is make two spanish translation. 

One for american y one for Espa?a.

I'm italian but i know this little differences beacouse i live in Colombia and i have studied in Spain.

Maybe i can help with italian prompts but i don't understand how.

BR

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From testeador01 at gmail.com  Tue Aug 11 06:09:53 2009
From: testeador01 at gmail.com (Milena)
Date: Tue, 11 Aug 2009 08:09:53 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <24916765.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
Message-ID: <c29b930b0908110609y3c224641x6c2cfa5a263b53e5@mail.gmail.com>

sendmsg
call-command: execute
execute-app-name: answer\n\n

the example is in this wiki page:
http://wiki.freeswitch.org/wiki/Event_socket_outbound
<http://wiki.freeswitch.org/wiki/Event_socket_outbound>
2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>

>
> Thank you.
>
> Is it possible to use this command from command line or via event_socket
> interface?
>
>
>
> Szymon Olko wrote:
> >
> > Maxim Tsvetov pisze:
> >> Hello,
> >>
> >> I found in WIKI list of supported commands
> >> http://wiki.freeswitch.org/wiki/Mod_commands.
> >>
> >> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
> >>
> >> That's what Freeswitch replies for this command:
> >>
> >> "answer: Command not found!"
> >>
> >> ?ould you please help?
> >
> > Answer is in dialplan commands.
> >
> > http://wiki.freeswitch.org/wiki/Mod_dptools
> >
> > Szymon
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/answer-command-tp24912812p24916765.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From darren at aleph-com.net  Tue Aug 11 06:18:25 2009
From: darren at aleph-com.net (Darren Wiebe)
Date: Tue, 11 Aug 2009 07:18:25 -0600
Subject: [Freeswitch-users] VoiceMail transcription
In-Reply-To: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
References: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
Message-ID: <4A816FA1.3050701@aleph-com.net>

We've been using mycaption for a number of months and have been very 
happy with the results.  I'd suggest giving them a try.

Darren Wiebe
darren at aleph-com.net

Pete Mueller wrote:
> I apologize, I should have been more clear.  We will be using humans 
> to scan the translated results.  But we are looking for a system to 
> perform the "first pass" on the audio to hopefully help the human type 
> less.
>
> Although the question has been raised if it's faster to have a human 
> just transcribe the whole thing, or fix up what the computer spit 
> out.  If you have any insights on this, that would be great.
>
> -pete
>
>     -------- Original Message --------
>     Subject: Re: [Freeswitch-users] VoiceMail transcription
>     From: David Knell <dave at 3c.co.uk>
>     Date: Mon, August 10, 2009 11:51 am
>     To: freeswitch-users at lists.freeswitch.org
>
>     Good evening Pete,
>
>     The only way to do this is, I'm afraid, to use a human. We use
>     Amazon's
>     Mechanical Turk to good effect.
>
>     Cheers --
>
>     Dave
>
>     > Good morning all,
>     >
>     > I realize this is slightly off the FS topic, but I am wondering if
>     > anyone out there has experience with software packages designed for
>     > the transcription of voicemails to text. I've used pocketsphinx with
>     > FS to handle IVR menus, but now have the task of figuring out how to
>     > convert recorded phone conversations (voicemails mostly) to text.
>     >
>     > This does not have to be a real-time process, I can store the audio
>     > files and process them over time. This would need to be a software
>     > (preferable open source) solution. ASPs like VoiceCloud would not
>     > work for this application.
>     >
>     > Thanks for any help
>     > -pete
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     -- 
>     David Knell, Director, 3C Limited
>     T: +44 20 3298 2000
>     E: dave at 3c.co.uk
>     W: http://www.3c.co.uk
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From wangdq.no1 at gmail.com  Tue Aug 11 06:19:13 2009
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Tue, 11 Aug 2009 21:19:13 +0800
Subject: [Freeswitch-users] answer command
In-Reply-To: <24916765.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
Message-ID: <ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>

why not use:
    session:answer()


2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>

>
> Thank you.
>
> Is it possible to use this command from command line or via event_socket
> interface?
>
>
>
> Szymon Olko wrote:
> >
> > Maxim Tsvetov pisze:
> >> Hello,
> >>
> >> I found in WIKI list of supported commands
> >> http://wiki.freeswitch.org/wiki/Mod_commands.
> >>
> >> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
> >>
> >> That's what Freeswitch replies for this command:
> >>
> >> "answer: Command not found!"
> >>
> >> ?ould you please help?
> >
> > Answer is in dialplan commands.
> >
> > http://wiki.freeswitch.org/wiki/Mod_dptools
> >
> > Szymon
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/answer-command-tp24912812p24916765.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From maxim.tsvetov at gmail.com  Tue Aug 11 06:30:40 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Tue, 11 Aug 2009 06:30:40 -0700 (PDT)
Subject: [Freeswitch-users] answer command
In-Reply-To: <c29b930b0908110609y3c224641x6c2cfa5a263b53e5@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<c29b930b0908110609y3c224641x6c2cfa5a263b53e5@mail.gmail.com>
Message-ID: <24917680.post@talk.nabble.com>


I've tried all this command from FS console 
and all of them return "Unknown command".



Milena-6 wrote:
> 
> sendmsg
> call-command: execute
> execute-app-name: answer\n\n
> 
> the example is in this wiki page:
> http://wiki.freeswitch.org/wiki/Event_socket_outbound
> <http://wiki.freeswitch.org/wiki/Event_socket_outbound>
> 2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>
> 
>>
>> Thank you.
>>
>> Is it possible to use this command from command line or via event_socket
>> interface?
>>
>>
>>
>> Szymon Olko wrote:
>> >
>> > Maxim Tsvetov pisze:
>> >> Hello,
>> >>
>> >> I found in WIKI list of supported commands
>> >> http://wiki.freeswitch.org/wiki/Mod_commands.
>> >>
>> >> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
>> >>
>> >> That's what Freeswitch replies for this command:
>> >>
>> >> "answer: Command not found!"
>> >>
>> >> ?ould you please help?
>> >
>> > Answer is in dialplan commands.
>> >
>> > http://wiki.freeswitch.org/wiki/Mod_dptools
>> >
>> > Szymon
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> --
>> View this message in context:
>> http://www.nabble.com/answer-command-tp24912812p24916765.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/answer-command-tp24912812p24917680.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From maxim.tsvetov at gmail.com  Tue Aug 11 06:31:32 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Tue, 11 Aug 2009 06:31:32 -0700 (PDT)
Subject: [Freeswitch-users] answer command
In-Reply-To: <ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
Message-ID: <24917706.post@talk.nabble.com>


I've tried all this command from FS console 
and all of them return "Unknown command"


daqiang wang wrote:
> 
> why not use:
>     session:answer()
> 
> 
> 2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>
> 
>>
>> Thank you.
>>
>> Is it possible to use this command from command line or via event_socket
>> interface?
>>
>>
>>
>> Szymon Olko wrote:
>> >
>> > Maxim Tsvetov pisze:
>> >> Hello,
>> >>
>> >> I found in WIKI list of supported commands
>> >> http://wiki.freeswitch.org/wiki/Mod_commands.
>> >>
>> >> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
>> >>
>> >> That's what Freeswitch replies for this command:
>> >>
>> >> "answer: Command not found!"
>> >>
>> >> ?ould you please help?
>> >
>> > Answer is in dialplan commands.
>> >
>> > http://wiki.freeswitch.org/wiki/Mod_dptools
>> >
>> > Szymon
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> --
>> View this message in context:
>> http://www.nabble.com/answer-command-tp24912812p24916765.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/answer-command-tp24912812p24917706.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From dave at 3c.co.uk  Tue Aug 11 06:40:14 2009
From: dave at 3c.co.uk (David Knell)
Date: Tue, 11 Aug 2009 16:40:14 +0300
Subject: [Freeswitch-users] VoiceMail transcription
In-Reply-To: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
References: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
Message-ID: <1249998014.20224.82.camel@dk-d820>

Hi Pete,

I'm afraid that the answer's still the same: use a human.  Here's an
article describing the state of the art:
http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/
- the links to previous stories at the bottom provide good background.

--Dave

> I apologize, I should have been more clear.  We will be using humans
> to scan the translated results.  But we are looking for a system to
> perform the "first pass" on the audio to hopefully help the human type
> less.
> 
> Although the question has been raised if it's faster to have a human
> just transcribe the whole thing, or fix up what the computer spit out.
> If you have any insights on this, that would be great.
> 
> -pete
> 
>         -------- Original Message --------
>         Subject: Re: [Freeswitch-users] VoiceMail transcription
>         From: David Knell <dave at 3c.co.uk>
>         Date: Mon, August 10, 2009 11:51 am
>         To: freeswitch-users at lists.freeswitch.org
>         
>         Good evening Pete,
>         
>         The only way to do this is, I'm afraid, to use a human. We use
>         Amazon's
>         Mechanical Turk to good effect.
>         
>         Cheers --
>         
>         Dave
>         
>         > Good morning all,
>         > 
>         > I realize this is slightly off the FS topic, but I am
>         wondering if
>         > anyone out there has experience with software packages
>         designed for
>         > the transcription of voicemails to text. I've used
>         pocketsphinx with
>         > FS to handle IVR menus, but now have the task of figuring
>         out how to
>         > convert recorded phone conversations (voicemails mostly) to
>         text.
>         > 
>         > This does not have to be a real-time process, I can store
>         the audio
>         > files and process them over time. This would need to be a
>         software
>         > (preferable open source) solution. ASPs like VoiceCloud
>         would not
>         > work for this application.
>         > 
>         > Thanks for any help
>         > -pete
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         -- 
>         David Knell, Director, 3C Limited
>         T: +44 20 3298 2000
>         E: dave at 3c.co.uk
>         W: http://www.3c.co.uk
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From dujinfang at gmail.com  Tue Aug 11 06:40:41 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 11 Aug 2009 21:40:41 +0800
Subject: [Freeswitch-users] answer command
In-Reply-To: <24917706.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
Message-ID: <2AFE75D2-99F3-4A12-BADC-F6753C76166D@gmail.com>

answer only works on outbound event socket. why you don't answer in a  
dialplan? what's scenario you use this?

On Aug 11, 2009, at 9:31 PM, Maxim Tsvetov wrote:
>
> I've tried all this command from FS console
> and all of them return "Unknown command"
>
>
> daqiang wang wrote:
>>
>> why not use:
>>    session:answer()
>>
>>
>> 2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>
>>
>>>
>>> Thank you.
>>>
>>> Is it possible to use this command from command line or via  
>>> event_socket
>>> interface?
>>>
>>>
>>>
>>> Szymon Olko wrote:
>>>>
>>>> Maxim Tsvetov pisze:
>>>>> Hello,
>>>>>
>>>>> I found in WIKI list of supported commands
>>>>> http://wiki.freeswitch.org/wiki/Mod_commands.
>>>>>
>>>>> I need command "answer" but it doesn't exist (Freeswitch 1.04,  
>>>>> Win32)
>>>>>
>>>>> That's what Freeswitch replies for this command:
>>>>>
>>>>> "answer: Command not found!"
>>>>>
>>>>> ?ould you please help?
>>>>
>>>> Answer is in dialplan commands.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Mod_dptools
>>>>
>>>> Szymon
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> View this message in context:
>>> http://www.nabble.com/answer-command-tp24912812p24916765.html
>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://www.nabble.com/answer-command-tp24912812p24917706.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From testeador01 at gmail.com  Tue Aug 11 06:46:01 2009
From: testeador01 at gmail.com (Milena)
Date: Tue, 11 Aug 2009 08:46:01 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <24917706.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
Message-ID: <c29b930b0908110646w470a0ec0mc65cd34e5f04825b@mail.gmail.com>

you can send the "answer" on the console too;this is *how to* use
*sendmsg*from the console:
http://wiki.freeswitch.org/wiki/Event_Socket#sendmsg

<http://wiki.freeswitch.org/wiki/Event_Socket#sendmsg>btw, the previous
command i gave you works just fine on event socket, it will work for you if
you replicate the Netcat example.

2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>

>
> I've tried all this command from FS console
> and all of them return "Unknown command"
>
>
> daqiang wang wrote:
> >
> > why not use:
> >     session:answer()
> >
> >
> > 2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>
> >
> >>
> >> Thank you.
> >>
> >> Is it possible to use this command from command line or via event_socket
> >> interface?
> >>
> >>
> >>
> >> Szymon Olko wrote:
> >> >
> >> > Maxim Tsvetov pisze:
> >> >> Hello,
> >> >>
> >> >> I found in WIKI list of supported commands
> >> >> http://wiki.freeswitch.org/wiki/Mod_commands.
> >> >>
> >> >> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
> >> >>
> >> >> That's what Freeswitch replies for this command:
> >> >>
> >> >> "answer: Command not found!"
> >> >>
> >> >> ?ould you please help?
> >> >
> >> > Answer is in dialplan commands.
> >> >
> >> > http://wiki.freeswitch.org/wiki/Mod_dptools
> >> >
> >> > Szymon
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> --
> >> View this message in context:
> >> http://www.nabble.com/answer-command-tp24912812p24916765.html
> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/answer-command-tp24912812p24917706.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Aug 11 06:49:53 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 08:49:53 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <24917706.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
Message-ID: <59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>

Its not an api command the docs are wrong and should be fixed.

/b

On Aug 11, 2009, at 8:31 AM, Maxim Tsvetov wrote:

>
> I've tried all this command from FS console
> and all of them return "Unknown command"
>



From pjintheusa at gmail.com  Tue Aug 11 06:59:54 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 11 Aug 2009 09:59:54 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
Message-ID: <367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>

It occurred to me that I better check that this still occurs using lua
and that this is not a mod_manged issue.

The lua script:

session:answer();
session:execute("set", "bypass_media_after_bridge=true");
session:execute("bridge", "loopback/6095553828/default");

does produce the same issue. The bridge fails.

I am wondering whether this is something I should just put straight into jira.

On Mon, Aug 10, 2009 at 4:35 PM, Phillip Jones<pjintheusa at gmail.com> wrote:
> Hi there,
>
> I am attempting to do a simple bridge. Leg A is an inbound DID. Leg B
> is terminated through a SIP carrier.
>
> I am unable to use loopback AND bypass_media_after_bridge=true. The
> bridge fails.
>
> Here is my three line application:
>
> Session.Answer();
>
> Session.Execute("set", "bypass_media_after_bridge=true");
>
> Session.Execute("bridge", "loopback/6095553828/default");
>
> FreeSWITCH log: http://pastebin.freeswitch.org/9949
>
> I believe the bridge fails as the SIP-REINVITE is issued and fails.
>
> Using loopback and bypass_media_after_bridge=false works fine. As does
> using bypass_media_after_bridge=true and sending call directly through
> the gateway to which the loopback points.
>
> e.g. ?http://pastebin.freeswitch.org/9948
>
> Has anyone else encountered this? Am I missing something?
>
> Any help would be much appreciated!
>
> Thanks
>
>
> Phillip Jones
>


From testeador01 at gmail.com  Tue Aug 11 07:05:56 2009
From: testeador01 at gmail.com (Milena)
Date: Tue, 11 Aug 2009 09:05:56 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
Message-ID: <c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>

Hello Brian,

I wanna fix the wiki, but to make sure i got it right, does it only work on
outbound event socket? or is there any other scenario where it would work.

Thank you.


2009/8/11 Brian West <brian at freeswitch.org>

> Its not an api command the docs are wrong and should be fixed.
>
> /b
>
> On Aug 11, 2009, at 8:31 AM, Maxim Tsvetov wrote:
>
> >
> > I've tried all this command from FS console
> > and all of them return "Unknown command"
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Aug 11 07:07:27 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 09:07:27 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
Message-ID: <6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>

loopback requires media you can't be using it in this manner.  There  
is NO reason to use loopback in the first place.  Can you show me what  
you're doing in the dialplan that requires you to use loopback?

We gave you the rope... now you just have to stop from hanging  
yourself!  :P

/b

On Aug 11, 2009, at 8:59 AM, Phillip Jones wrote:

> It occurred to me that I better check that this still occurs using lua
> and that this is not a mod_manged issue.
>
> The lua script:
>
> session:answer();
> session:execute("set", "bypass_media_after_bridge=true");
> session:execute("bridge", "loopback/6095553828/default");
>
> does produce the same issue. The bridge fails.
>
> I am wondering whether this is something I should just put straight  
> into jira.



From email.list.subscriber at gmail.com  Tue Aug 11 07:12:57 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Tue, 11 Aug 2009 10:12:57 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>
	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>
Message-ID: <4a817c27.14b48c0a.5c23.6dfe@mx.google.com>

Would truly appreciate that.

Thanks Michal!


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michal Bielicki
Sent: Tuesday, August 11, 2009 5:33 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

I'll retst it later today and give you a link with instructions

Am 10.08.2009 um 20:14 schrieb vmorales:

> By "./compile" I was referring to "./configure"
>
> Vladimir
>
> -----Original Message-----
> From: vmorales [mailto:email.list.subscriber at gmail.com]
> Sent: Monday, August 10, 2009 11:49 AM
> To: 'freeswitch-users at lists.freeswitch.org'
> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> Thanks for the response(s):
>
> I ran the "./compile" script with a set PREFIX.  This took a few
> attempts with errors before it was able to complete error-free, as I
> had to install libtool.
>
> Since then, I have tried running 'make', 'gmake', and
> '/opt/gnu/bin/make', but each results with an error.  This is the
> error when running 'make' or 'gmake':
>
> <snip>
> make: Fatal error: Command failed for target `all-recursive'
> Current working directory /home/vmorales/freeswitch-1.0.4
> *** Error code 1
> make: Fatal error: Command failed for target `all'
> </snip>
>
>
> This is the error when running '/opt/gnu/bin/make':
>
> <snip>
> make[5]: *** [mod_amr.so] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_amr-all] Error 1
> make[2]: *** [all-recursive] Error 1
> Making all in build
> +-------- FreeSWITCH Build Complete -----------+
> + FreeSWITCH has been successfully built.      +
> + Install by running:                          +
> +                                              +
> +               /opt/gnu/bin/make install                   +
> +----------------------------------------------+
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> </snip>
>
>
> I re-untar'd before each compile/make attempt.  Let me know if this
is
> something that I can resolve.
>
> Vladimir
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Jerris
> Sent: Saturday, August 08, 2009 12:37 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> This is not currently a supported platform, it only builds on 64 bit
> right now I think on solaris.
>
> Mike
>
> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>
>> Hello,
>>
>> Does anyone have, or know where to get, a pre-compiled copy of
>> FreeSwitch for Solaris 10/x86?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
> http://www.freeswitch.org

Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. info at halokwadrat.de
e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
Hauptgesch?ftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 0000153539




From kirk.bateman at gmail.com  Tue Aug 11 07:17:47 2009
From: kirk.bateman at gmail.com (Kirk Bateman)
Date: Tue, 11 Aug 2009 15:17:47 +0100
Subject: [Freeswitch-users] VoiceMail transcription
In-Reply-To: <1249998014.20224.82.camel@dk-d820>
References: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
	<1249998014.20224.82.camel@dk-d820>
Message-ID: <2bee4fc40908110717p43649678ue2b94ecb571bd09@mail.gmail.com>

I'm still interested in getting pocketsphinx to attempt speech recognition
on an audio file.

To be honest, most of the problem is that at 8Khz (mobile phone call rate),
speech detection is NOT very accurate, at 16Khz it IS significantly better.

I'm planning to have a play with the speechtools module and mod_pocketsphinx
etc to try and get an audio file parsed, spare time permitting.

Will let the list know if I get anywhere.

Regards

Kirk Bateman


2009/8/11 David Knell <dave at 3c.co.uk>

> Hi Pete,
>
> I'm afraid that the answer's still the same: use a human.  Here's an
> article describing the state of the art:
> http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/
> - the links to previous stories at the bottom provide good background.
>
> --Dave
>
> > I apologize, I should have been more clear.  We will be using humans
> > to scan the translated results.  But we are looking for a system to
> > perform the "first pass" on the audio to hopefully help the human type
> > less.
> >
> > Although the question has been raised if it's faster to have a human
> > just transcribe the whole thing, or fix up what the computer spit out.
> > If you have any insights on this, that would be great.
> >
> > -pete
> >
> >         -------- Original Message --------
> >         Subject: Re: [Freeswitch-users] VoiceMail transcription
> >         From: David Knell <dave at 3c.co.uk>
> >         Date: Mon, August 10, 2009 11:51 am
> >         To: freeswitch-users at lists.freeswitch.org
> >
> >         Good evening Pete,
> >
> >         The only way to do this is, I'm afraid, to use a human. We use
> >         Amazon's
> >         Mechanical Turk to good effect.
> >
> >         Cheers --
> >
> >         Dave
> >
> >         > Good morning all,
> >         >
> >         > I realize this is slightly off the FS topic, but I am
> >         wondering if
> >         > anyone out there has experience with software packages
> >         designed for
> >         > the transcription of voicemails to text. I've used
> >         pocketsphinx with
> >         > FS to handle IVR menus, but now have the task of figuring
> >         out how to
> >         > convert recorded phone conversations (voicemails mostly) to
> >         text.
> >         >
> >         > This does not have to be a real-time process, I can store
> >         the audio
> >         > files and process them over time. This would need to be a
> >         software
> >         > (preferable open source) solution. ASPs like VoiceCloud
> >         would not
> >         > work for this application.
> >         >
> >         > Thanks for any help
> >         > -pete
> >         > _______________________________________________
> >         > FreeSWITCH-users mailing list
> >         > FreeSWITCH-users at lists.freeswitch.org
> >         >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > http://www.freeswitch.org
> >         --
> >         David Knell, Director, 3C Limited
> >         T: +44 20 3298 2000
> >         E: dave at 3c.co.uk
> >         W: http://www.3c.co.uk
> >
> >
> >         _______________________________________________
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nicolas at medularis.com  Tue Aug 11 07:27:27 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Tue, 11 Aug 2009 10:27:27 -0400
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
Message-ID: <1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>

I think bakko is right, I have a few observations about the changes proposed
by samuel, and I think it makes a lot of sense to have different
localizations.


On Tue, Aug 11, 2009 at 3:49 AM, samuel <samu60 at gmail.com> wrote:

> I've taken a look at phrase_es.xml and would like other spanish native
> speakers to check the following (i preferred to write down here my
> impressions so others can contribute instead of sending a patch):
>
> 1) Centavos
>
> <prompt phrase="Centavos por minuto" filename="cents-per-minute.wav"/>
> <prompt phrase="Centavos" filename="cents.wav"/>


> probably in european spanish the most apropiate word for cents is "c?ntimo"
> but, if i'm not mistaken, in south american spanish is more common to use
> "centavos" so this is more a localization issue...as you wish.
> Since I live in Spain I would use the following:
> <prompt phrase="C?ntimos por minuto" filename="cents-per-minute.wav"/>
> <prompt phrase="C?ntimos" filename="cents.wav"/>
>

You are right, in South America we use Centavos.


>
> 2) Minnor stress corrections:
> instead of Dolar, you should use the stressed word D?lar:
> -<prompt phrase="Dolar" filename="dollar.wav"/>
> +<prompt phrase="D?lar" filename="dollar.wav"/>
>
> instead of numero, you should use the stressed word n?mero:
> -<prompt phrase="por favor ingrese su numero de usuario, seguido por"
> filename="vm-enter_id.wav"/>
> +<prompt phrase="por favor ingrese su n?mero de usuario, seguido por"
> filename="vm-enter_id.wav"/>
>
> 3) en instead of a
> -<prompt phrase="La persona a la extensi?n" filename="vm-person.wav"/>
> +<prompt phrase="La persona en la extensi?n" filename="vm-person.wav"/>
>
> 4) femenin genre of contrase?a changes the word seguido for seguida
> -<prompt phrase="por favor ingrese su contrase?a, seguido por"
> filename="vm-enter_pass.wav"/>
> +<prompt phrase="por favor ingrese su contrase?a, seguida por"
> filename="vm-enter_pass.wav"/>
>
> 5) I would propose the word "almohadilla" as the translation of pound so I
> would change following sentences:
> -<prompt phrase="Signo de n?mero" filename="35.wav"/>
> +<prompt phrase="Almohadilla" filename="35.wav"/>
>
> -<prompt phrase="Signo de n?mero" filename="pound.wav"/>
> +<prompt phrase="Almohadilla" filename="pound.wav"/>
>

I don't know about the rest of the countries in South America, but at least
in Chile it would me much clearer "Signo de n?mero" instead of Almohadilla
(first time I've heard that name for the pound sign).


>
> -<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del
> signo de n?mero" filename="ivr-enter_ext.wav"/>
> +<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido de
> almohadilla" filename="ivr-enter_ext.wav"/>
>
> -<prompt phrase="Seguido del signo de n?mero" filename="followed.wav"/>
> +<prompt phrase="Seguido de almohadilla" filename="followed.wav"/>
>
> 6)changing prepositions:
> -<prompt phrase="Un error ha ocurrido, por favor contacte al administrador"
> filename="error.wav"/>
> +<prompt phrase="Un error ha ocurrido, por favor contacte con el
> administrador" filename="error.wav"/>
>

Here it would be:
<prompt phrase="Ha ocurrido un error, por favor contacte al administrador"
filename="error.wav"/>


> 7)instead of using ingrese, it sounds more natural to use introduzca when
> interacting with an IVR asking for inserting data. So:
> -<prompt phrase="por favor ingrese su numero de usuario, seguido por"
> filename="vm-enter_id.wav"/>
> +<prompt phrase="por favor introduzca su n?mero de usuario, seguido por"
> filename="vm-enter_id.wav"/>
>
> -<prompt phrase="por favor ingrese su contrase?a, seguido por"
> filename="vm-enter_pass.wav"/>
> +<prompt phrase="por favor introduzca su contrase?a, seguida por"
> filename="vm-enter_pass.wav"/>
>
> -<prompt phrase="Por favor ingrese el pin de la conferencia."
> filename="conf-pin.wav"/>
> +<prompt phrase="Por favor introduzca el pin de la conferencia."
> filename="conf-pin.wav"/>
>
> -<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del
> signo de n?mero" filename="ivr-enter_ext.wav"/>
> +<prompt phrase="por favor introduzca el n?mero de una extensi?n seguido de
> almohadilla" filename="ivr-enter_ext.wav"/>
>
>
>
> Comments?
>

I am not very convinced by the use of stresses in the words, I know it's
"more correct", but sometimes it's a pain, especially if you are working on
a remote server and don't have the right keyboard setting or something like
that. I've also had a few headaches with file encoding, so depending on that
and changes to the encoding you can screw your config files really easily.
For simplification I'd leave all stresses out.


>
> Samuel
>
>
> 2009/8/11 Michael Collins <msc at freeswitch.org>
>
> Be sure to hop on IRC and speak with jmesquita because he's been working on
>> this also. It would be very good to have Spanish-speaking users review the
>> Spanish phrase_es.xml file for correctness. Also, we need Spanish-speaking
>> programmers to assist with the mod_say application for Spanish - there are
>> things that you have to do in Spanish that you don't have to do in English.
>>
>> Thanks,
>> MC
>>
>>
>> On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>>
>>> On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
>>> edition .
>>>
>>> Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for
>>> FS playback.
>>>
>>> Here's a guide that has been put together for reference on what to
>>> record.
>>>
>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml
>>>
>>> Regards,
>>>
>>>
>>>
>>>
>>> On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:
>>>
>>>> I'd like to begin record spanish prompts for FS.
>>>>
>>>> Do you know any software/hardware to make it?
>>>>
>>>> Thank you
>>>>
>>>> BR
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Aug 11 07:30:59 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 09:30:59 -0500
Subject: [Freeswitch-users] Fwd: ALARM signal in esl libraries
In-Reply-To: <1452e2980908102129g51255d5bt430a3e73bd6f2e99@mail.gmail.com>
References: <1452e2980908092338j15c4e33cn2cff799bb64464d0@mail.gmail.com>
	<1452e2980908102129g51255d5bt430a3e73bd6f2e99@mail.gmail.com>
Message-ID: <ADEDE29B-B1C2-4563-A20B-C16225ED7881@freeswitch.org>

Replying again asking for help on the same thing over and over won't  
get you far on this list.  It'll usually get you ignored.  Can you  
explain what you're trying to do better?  Why are you needing an  
alarm? what purpose?

/b

On Aug 10, 2009, at 11:29 PM, velusamy velu wrote:

> Dear Greats,
>     Could you please help me to solve this problem?
>
> ---------- Forwarded message ----------
> From: velusamy velu <velu.technical at gmail.com>
> Date: Mon, Aug 10, 2009 at 12:08 PM
> Subject: ALARM signal in esl libraries
> To: freeswitch-users at lists.freeswitch.org
>
>
> Dear All,
>        I have registered  ALARM signal  in my perl program to handle  
> the DTMF digit timeout.
> When ALARM signal generated  the  connection with  ESL  is   
> automatically  closed.
>
> I have checked the connection with "connected: function, it returns 0.
>
> Why the connection was closed?
> Is there any idea to alive the connection after ALARM signal  
> generation??
>
> Please help me.........
>
> Thanks,
> Velusamy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From pjintheusa at gmail.com  Tue Aug 11 07:46:55 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 11 Aug 2009 10:46:55 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
Message-ID: <367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>

Thanks for the reply Brian.

I went through this fairly carefully with Mike Jerris - please see
http://www.nabble.com/Calling-Multiple-Destinations-with-Failover-td24877157.html

I am very open to any other mechanism that allows the calling multiple
destinations with carrier failover support. I would have thought this
is a fairly common requirement and is a fundamental requirement in my
application.

As always - your help is appreciated.

On Tue, Aug 11, 2009 at 10:07 AM, Brian West<brian at freeswitch.org> wrote:
> loopback requires media you can't be using it in this manner. ?There
> is NO reason to use loopback in the first place. ?Can you show me what
> you're doing in the dialplan that requires you to use loopback?
>
> We gave you the rope... now you just have to stop from hanging
> yourself! ?:P
>
> /b
>
> On Aug 11, 2009, at 8:59 AM, Phillip Jones wrote:
>
>> It occurred to me that I better check that this still occurs using lua
>> and that this is not a mod_manged issue.
>>
>> The lua script:
>>
>> session:answer();
>> session:execute("set", "bypass_media_after_bridge=true");
>> session:execute("bridge", "loopback/6095553828/default");
>>
>> does produce the same issue. The bridge fails.
>>
>> I am wondering whether this is something I should just put straight
>> into jira.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From samu60 at gmail.com  Tue Aug 11 07:48:36 2009
From: samu60 at gmail.com (samuel)
Date: Tue, 11 Aug 2009 16:48:36 +0200
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
Message-ID: <d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>

I'm also for different spanish localization if it's not too complicated. It
was also for me the first time I see "signo de n?mero" for pound ;)


Samuel.

2009/8/11 Nicolas Brenner <nicolas at medularis.com>

> I think bakko is right, I have a few observations about the changes
> proposed by samuel, and I think it makes a lot of sense to have different
> localizations.
>
>
> On Tue, Aug 11, 2009 at 3:49 AM, samuel <samu60 at gmail.com> wrote:
>
>> I've taken a look at phrase_es.xml and would like other spanish native
>> speakers to check the following (i preferred to write down here my
>> impressions so others can contribute instead of sending a patch):
>>
>> 1) Centavos
>>
>> <prompt phrase="Centavos por minuto" filename="cents-per-minute.wav"/>
>> <prompt phrase="Centavos" filename="cents.wav"/>
>
>
>> probably in european spanish the most apropiate word for cents is
>> "c?ntimo" but, if i'm not mistaken, in south american spanish is more common
>> to use "centavos" so this is more a localization issue...as you wish.
>> Since I live in Spain I would use the following:
>> <prompt phrase="C?ntimos por minuto" filename="cents-per-minute.wav"/>
>> <prompt phrase="C?ntimos" filename="cents.wav"/>
>>
>
> You are right, in South America we use Centavos.
>
>
>>
>> 2) Minnor stress corrections:
>> instead of Dolar, you should use the stressed word D?lar:
>> -<prompt phrase="Dolar" filename="dollar.wav"/>
>> +<prompt phrase="D?lar" filename="dollar.wav"/>
>>
>> instead of numero, you should use the stressed word n?mero:
>> -<prompt phrase="por favor ingrese su numero de usuario, seguido por"
>> filename="vm-enter_id.wav"/>
>> +<prompt phrase="por favor ingrese su n?mero de usuario, seguido por"
>> filename="vm-enter_id.wav"/>
>>
>> 3) en instead of a
>> -<prompt phrase="La persona a la extensi?n" filename="vm-person.wav"/>
>> +<prompt phrase="La persona en la extensi?n" filename="vm-person.wav"/>
>>
>> 4) femenin genre of contrase?a changes the word seguido for seguida
>> -<prompt phrase="por favor ingrese su contrase?a, seguido por"
>> filename="vm-enter_pass.wav"/>
>> +<prompt phrase="por favor ingrese su contrase?a, seguida por"
>> filename="vm-enter_pass.wav"/>
>>
>> 5) I would propose the word "almohadilla" as the translation of pound so I
>> would change following sentences:
>> -<prompt phrase="Signo de n?mero" filename="35.wav"/>
>> +<prompt phrase="Almohadilla" filename="35.wav"/>
>>
>> -<prompt phrase="Signo de n?mero" filename="pound.wav"/>
>> +<prompt phrase="Almohadilla" filename="pound.wav"/>
>>
>
> I don't know about the rest of the countries in South America, but at least
> in Chile it would me much clearer "Signo de n?mero" instead of Almohadilla
> (first time I've heard that name for the pound sign).
>
>
>>
>> -<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del
>> signo de n?mero" filename="ivr-enter_ext.wav"/>
>> +<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido de
>> almohadilla" filename="ivr-enter_ext.wav"/>
>>
>> -<prompt phrase="Seguido del signo de n?mero" filename="followed.wav"/>
>> +<prompt phrase="Seguido de almohadilla" filename="followed.wav"/>
>>
>> 6)changing prepositions:
>> -<prompt phrase="Un error ha ocurrido, por favor contacte al
>> administrador" filename="error.wav"/>
>> +<prompt phrase="Un error ha ocurrido, por favor contacte con el
>> administrador" filename="error.wav"/>
>>
>
> Here it would be:
> <prompt phrase="Ha ocurrido un error, por favor contacte al administrador"
> filename="error.wav"/>
>
>
>> 7)instead of using ingrese, it sounds more natural to use introduzca when
>> interacting with an IVR asking for inserting data. So:
>> -<prompt phrase="por favor ingrese su numero de usuario, seguido por"
>> filename="vm-enter_id.wav"/>
>> +<prompt phrase="por favor introduzca su n?mero de usuario, seguido por"
>> filename="vm-enter_id.wav"/>
>>
>> -<prompt phrase="por favor ingrese su contrase?a, seguido por"
>> filename="vm-enter_pass.wav"/>
>> +<prompt phrase="por favor introduzca su contrase?a, seguida por"
>> filename="vm-enter_pass.wav"/>
>>
>> -<prompt phrase="Por favor ingrese el pin de la conferencia."
>> filename="conf-pin.wav"/>
>> +<prompt phrase="Por favor introduzca el pin de la conferencia."
>> filename="conf-pin.wav"/>
>>
>> -<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del
>> signo de n?mero" filename="ivr-enter_ext.wav"/>
>> +<prompt phrase="por favor introduzca el n?mero de una extensi?n seguido
>> de almohadilla" filename="ivr-enter_ext.wav"/>
>>
>>
>>
>> Comments?
>>
>
> I am not very convinced by the use of stresses in the words, I know it's
> "more correct", but sometimes it's a pain, especially if you are working on
> a remote server and don't have the right keyboard setting or something like
> that. I've also had a few headaches with file encoding, so depending on that
> and changes to the encoding you can screw your config files really easily.
> For simplification I'd leave all stresses out.
>
>
>>
>> Samuel
>>
>>
>> 2009/8/11 Michael Collins <msc at freeswitch.org>
>>
>> Be sure to hop on IRC and speak with jmesquita because he's been working
>>> on this also. It would be very good to have Spanish-speaking users review
>>> the Spanish phrase_es.xml file for correctness. Also, we need
>>> Spanish-speaking programmers to assist with the mod_say application for
>>> Spanish - there are things that you have to do in Spanish that you don't
>>> have to do in English.
>>>
>>> Thanks,
>>> MC
>>>
>>>
>>> On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>>>
>>>> On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
>>>> edition .
>>>>
>>>> Record and edit at 48000Hz and then I use sox to downsample to 8000Hz
>>>> for FS playback.
>>>>
>>>> Here's a guide that has been put together for reference on what to
>>>> record.
>>>>
>>>>
>>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml
>>>>
>>>> Regards,
>>>>
>>>>
>>>>
>>>>
>>>> On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:
>>>>
>>>>> I'd like to begin record spanish prompts for FS.
>>>>>
>>>>> Do you know any software/hardware to make it?
>>>>>
>>>>> Thank you
>>>>>
>>>>> BR
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Aug 11 07:58:22 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 09:58:22 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
Message-ID: <D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>

This can all be accomplished with standard bridge lines stacked one on  
top of the other with continue_on_fail set to the list of codes you  
wish to fail over with.  You would also want to set  
hangup_after_bridge=true.  Loopback should NEVER be used in this case.

http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge

/b

On Aug 11, 2009, at 9:46 AM, Phillip Jones wrote:

> Thanks for the reply Brian.
>
> I went through this fairly carefully with Mike Jerris - please see
> http://www.nabble.com/Calling-Multiple-Destinations-with-Failover-td24877157.html
>
> I am very open to any other mechanism that allows the calling multiple
> destinations with carrier failover support. I would have thought this
> is a fairly common requirement and is a fundamental requirement in my
> application.
>
> As always - your help is appreciated.

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From moises.silva at gmail.com  Tue Aug 11 07:59:15 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 11 Aug 2009 10:59:15 -0400
Subject: [Freeswitch-users] OpenZAP/Sangoma in Windows
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED9435@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED9435@cooper>
Message-ID: <c4d05cbe0908110759j2c929796g1ce96f530b7ad681@mail.gmail.com>

On Tue, Aug 11, 2009 at 2:16 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Hi, I'm trying to evaluate the OpenZAP/Sangoma-support in Windows, using
> PRI E1 connections.
>

Thanks for testing this :-)

I have been meaning to install FreeSWITCH on Windows but just could not find
the time.


> 1. Has anyone tested it in Windows at all? I know the build-files for
> Visual Studio has only been checked in for a couple of months, so that's why
> I'm asking.
>

The drivers have been tested quite well but not using FreeSWITCH.


> 2. Does anyone have any directions how to configure the driver within
> Windows? Should I use BitStream or HDLC, and how should the channel groups
> be configured?
>

You should use HDLC for the D-channel and Bitstream for the B-Channels.
Typically you would create 2 groups, one with channels 1-23 and the other
with just channel 24. The first group would work in bitstream and
TDM_CHAN_VOICE_API operational mode and the second in HDLC/API mode.

You can find me in #openzap, #freeswitch or #freeswitch-dev as "moy" if you
have more questions.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From dftoro at yahoo.com  Tue Aug 11 08:07:27 2009
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 11 Aug 2009 08:07:27 -0700 (PDT)
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
Message-ID: <805599.45498.qm@web33502.mail.mud.yahoo.com>

Hi all,
?
I have some suggested code to?support currency and number pronounced in spanish,
?
http://jira.freeswitch.org/browse/MODAPP-317?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
?

Diego 
http://lacarretade.blogspot.com
?
?


--- On Tue, 8/11/09, Nicolas Brenner <nicolas at medularis.com> wrote:


From: Nicolas Brenner <nicolas at medularis.com>
Subject: Re: [Freeswitch-users] Spanish Prompts
To: freeswitch-users at lists.freeswitch.org
Date: Tuesday, August 11, 2009, 9:27 AM


I think bakko is right, I have a few observations about the changes proposed by samuel, and I think it makes a lot of sense to have different localizations.



On Tue, Aug 11, 2009 at 3:49 AM, samuel <samu60 at gmail.com> wrote:

I've taken a look at phrase_es.xml and would like other spanish native speakers to check the following (i preferred to write down here my impressions so others can contribute instead of sending a patch):

1) Centavos

<prompt phrase="Centavos por minuto" filename="cents-per-minute.wav"/>
<prompt phrase="Centavos" filename="cents.wav"/>?

probably in european spanish the most apropiate word for cents is "c?ntimo" but, if i'm not mistaken, in south american spanish is more common to use "centavos" so this is more a localization issue...as you wish.
Since I live in Spain I would use the following:
<prompt phrase="C?ntimos por minuto" filename="cents-per-minute.wav"/>
<prompt phrase="C?ntimos" filename="cents.wav"/>


You are right, in South America we use Centavos.
?

2) Minnor stress corrections:
instead of Dolar, you should use the stressed word D?lar:
-<prompt phrase="Dolar" filename="dollar.wav"/>
+<prompt phrase="D?lar" filename="dollar.wav"/>

instead of numero, you should use the stressed word n?mero:
-<prompt phrase="por favor ingrese su numero de usuario, seguido por" filename="vm-enter_id.wav"/>
+<prompt phrase="por favor ingrese su n?mero de usuario, seguido por" filename="vm-enter_id.wav"/>

3) en instead of a
-<prompt phrase="La persona a la extensi?n" filename="vm-person.wav"/>
+<prompt phrase="La persona en la extensi?n" filename="vm-person.wav"/>

4) femenin genre of contrase?a changes the word seguido for seguida
-<prompt phrase="por favor ingrese su contrase?a, seguido por" filename="vm-enter_pass.wav"/>
+<prompt phrase="por favor ingrese su contrase?a, seguida por" filename="vm-enter_pass.wav"/>

5) I would propose the word "almohadilla" as the translation of pound so I would change following sentences:
-<prompt phrase="Signo de n?mero" filename="35.wav"/>
+<prompt phrase="Almohadilla" filename="35.wav"/>

-<prompt phrase="Signo de n?mero" filename="pound.wav"/>
+<prompt phrase="Almohadilla" filename="pound.wav"/>


I don't know about the rest of the countries in South America, but at least in Chile it would me much clearer "Signo de n?mero" instead of Almohadilla (first time I've heard that name for the pound sign).
?

-<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del signo de n?mero" filename="ivr-enter_ext.wav"/>
+<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido de almohadilla" filename="ivr-enter_ext.wav"/>

-<prompt phrase="Seguido del signo de n?mero" filename="followed.wav"/>
+<prompt phrase="Seguido de almohadilla" filename="followed.wav"/>

6)changing prepositions:
-<prompt phrase="Un error ha ocurrido, por favor contacte al administrador" filename="error.wav"/>
+<prompt phrase="Un error ha ocurrido, por favor contacte con el administrador" filename="error.wav"/>


Here it would be:
<prompt phrase="Ha ocurrido un error, por favor contacte al administrador" filename="error.wav"/> 



7)instead of using ingrese, it sounds more natural to use introduzca when interacting with an IVR asking for inserting data. So:
-<prompt phrase="por favor ingrese su numero de usuario, seguido por" filename="vm-enter_id.wav"/>
+<prompt phrase="por favor introduzca su n?mero de usuario, seguido por" filename="vm-enter_id.wav"/>

-<prompt phrase="por favor ingrese su contrase?a, seguido por" filename="vm-enter_pass.wav"/>
+<prompt phrase="por favor introduzca su contrase?a, seguida por" filename="vm-enter_pass.wav"/>

-<prompt phrase="Por favor ingrese el pin de la conferencia." filename="conf-pin.wav"/>
+<prompt phrase="Por favor introduzca el pin de la conferencia." filename="conf-pin.wav"/>

-<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del signo de n?mero" filename="ivr-enter_ext.wav"/>
+<prompt phrase="por favor introduzca el n?mero de una extensi?n seguido de almohadilla" filename="ivr-enter_ext.wav"/>



Comments?


I am not very convinced by the use of stresses in the words, I know it's "more correct", but sometimes it's a pain, especially if you are working on a remote server and don't have the right keyboard setting or something like that. I've also had a few headaches with file encoding, so depending on that and changes to the encoding you can screw your config files really easily. For simplification I'd leave all stresses out.
?

Samuel



2009/8/11 Michael Collins <msc at freeswitch.org>




Be sure to hop on IRC and speak with jmesquita because he's been working on this also. It would be very good to have Spanish-speaking users review the Spanish phrase_es.xml file for correctness. Also, we need Spanish-speaking programmers to assist with the mod_say application for Spanish - there are things that you have to do in Spanish that you don't have to do in English.

Thanks,
MC





On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for edition .

Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for FS playback.

Here's a guide that has been put together for reference on what to record.

http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml

Regards,







On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:

I'd like to begin record spanish prompts for FS.

Do you know any software/hardware to make it?

Thank you

BR


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From mike at jerris.com  Tue Aug 11 08:38:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 Aug 2009 11:38:27 -0400
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
Message-ID: <1513B1DC-2CD5-484F-B2CD-74B25E5AB822@jerris.com>

Correct me if I am wrong, but can't all these differences be done in a  
different soundset without modifying any code or other config?

Mike

On Aug 11, 2009, at 10:27 AM, Nicolas Brenner wrote:

> I think bakko is right, I have a few observations about the changes  
> proposed by samuel, and I think it makes a lot of sense to have  
> different localizations.
>
>
> On Tue, Aug 11, 2009 at 3:49 AM, samuel <samu60 at gmail.com> wrote:
> I've taken a look at phrase_es.xml and would like other spanish  
> native speakers to check the following (i preferred to write down  
> here my impressions so others can contribute instead of sending a  
> patch):
>
> 1) Centavos
>
> <prompt phrase="Centavos por minuto" filename="cents-per-minute.wav"/>
> <prompt phrase="Centavos" filename="cents.wav"/>
>
> probably in european spanish the most apropiate word for cents is  
> "c?ntimo" but, if i'm not mistaken, in south american spanish is  
> more common to use "centavos" so this is more a localization  
> issue...as you wish.
> Since I live in Spain I would use the following:
> <prompt phrase="C?ntimos por minuto" filename="cents-per-minute.wav"/>
> <prompt phrase="C?ntimos" filename="cents.wav"/>
>
> You are right, in South America we use Centavos.
>
>
> 2) Minnor stress corrections:
> instead of Dolar, you should use the stressed word D?lar:
> -<prompt phrase="Dolar" filename="dollar.wav"/>
> +<prompt phrase="D?lar" filename="dollar.wav"/>
>
> instead of numero, you should use the stressed word n?mero:
> -<prompt phrase="por favor ingrese su numero de usuario, seguido  
> por" filename="vm-enter_id.wav"/>
> +<prompt phrase="por favor ingrese su n?mero de usuario, seguido  
> por" filename="vm-enter_id.wav"/>
>
> 3) en instead of a
> -<prompt phrase="La persona a la extensi?n" filename="vm-person.wav"/>
> +<prompt phrase="La persona en la extensi?n" filename="vm- 
> person.wav"/>
>
> 4) femenin genre of contrase?a changes the word seguido for seguida
> -<prompt phrase="por favor ingrese su contrase?a, seguido por"  
> filename="vm-enter_pass.wav"/>
> +<prompt phrase="por favor ingrese su contrase?a, seguida por"  
> filename="vm-enter_pass.wav"/>
>
> 5) I would propose the word "almohadilla" as the translation of  
> pound so I would change following sentences:
> -<prompt phrase="Signo de n?mero" filename="35.wav"/>
> +<prompt phrase="Almohadilla" filename="35.wav"/>
>
> -<prompt phrase="Signo de n?mero" filename="pound.wav"/>
> +<prompt phrase="Almohadilla" filename="pound.wav"/>
>
> I don't know about the rest of the countries in South America, but  
> at least in Chile it would me much clearer "Signo de n?mero" instead  
> of Almohadilla (first time I've heard that name for the pound sign).
>
>
> -<prompt phrase="por favor ingrese el n?mero de una extensi?n  
> seguido del signo de n?mero" filename="ivr-enter_ext.wav"/>
> +<prompt phrase="por favor ingrese el n?mero de una extensi?n  
> seguido de almohadilla" filename="ivr-enter_ext.wav"/>
>
> -<prompt phrase="Seguido del signo de n?mero"  
> filename="followed.wav"/>
> +<prompt phrase="Seguido de almohadilla" filename="followed.wav"/>
>
> 6)changing prepositions:
> -<prompt phrase="Un error ha ocurrido, por favor contacte al  
> administrador" filename="error.wav"/>
> +<prompt phrase="Un error ha ocurrido, por favor contacte con el  
> administrador" filename="error.wav"/>
>
> Here it would be:
> <prompt phrase="Ha ocurrido un error, por favor contacte al  
> administrador" filename="error.wav"/>
>
>
> 7)instead of using ingrese, it sounds more natural to use introduzca  
> when interacting with an IVR asking for inserting data. So:
> -<prompt phrase="por favor ingrese su numero de usuario, seguido  
> por" filename="vm-enter_id.wav"/>
> +<prompt phrase="por favor introduzca su n?mero de usuario, seguido  
> por" filename="vm-enter_id.wav"/>
>
> -<prompt phrase="por favor ingrese su contrase?a, seguido por"  
> filename="vm-enter_pass.wav"/>
> +<prompt phrase="por favor introduzca su contrase?a, seguida por"  
> filename="vm-enter_pass.wav"/>
>
> -<prompt phrase="Por favor ingrese el pin de la conferencia."  
> filename="conf-pin.wav"/>
> +<prompt phrase="Por favor introduzca el pin de la conferencia."  
> filename="conf-pin.wav"/>
>
> -<prompt phrase="por favor ingrese el n?mero de una extensi?n  
> seguido del signo de n?mero" filename="ivr-enter_ext.wav"/>
> +<prompt phrase="por favor introduzca el n?mero de una extensi?n  
> seguido de almohadilla" filename="ivr-enter_ext.wav"/>
>
>
>
> Comments?
>
> I am not very convinced by the use of stresses in the words, I know  
> it's "more correct", but sometimes it's a pain, especially if you  
> are working on a remote server and don't have the right keyboard  
> setting or something like that. I've also had a few headaches with  
> file encoding, so depending on that and changes to the encoding you  
> can screw your config files really easily. For simplification I'd  
> leave all stresses out.
>
>
> Samuel
>
>
> 2009/8/11 Michael Collins <msc at freeswitch.org>
>
> Be sure to hop on IRC and speak with jmesquita because he's been  
> working on this also. It would be very good to have Spanish-speaking  
> users review the Spanish phrase_es.xml file for correctness. Also,  
> we need Spanish-speaking programmers to assist with the mod_say  
> application for Spanish - there are things that you have to do in  
> Spanish that you don't have to do in English.
>
> Thanks,
> MC
>
>
> On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com>  
> wrote:
> On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound  
> for edition .
>
> Record and edit at 48000Hz and then I use sox to downsample to  
> 8000Hz for FS playback.
>
> Here's a guide that has been put together for reference on what to  
> record.
>
> http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml
>
> Regards,
>
>
>
>
> On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:
> I'd like to begin record spanish prompts for FS.
>
> Do you know any software/hardware to make it?
>
> Thank you
>
> BR
>
>

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From pjintheusa at gmail.com  Tue Aug 11 09:08:03 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 11 Aug 2009 12:08:03 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
Message-ID: <367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>

Can you use this method to make simultaneous calls though while
preserving carrier order? i.e. each call must try carrier 1 first then
carrier 2 (because each carrier terminates a different subset of
numbers)

So - if I did NOT want to support failover I would use:

session:execute("bridge",
"sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828");
 << call 6095553828 and 7325553828 at the same time

If I were to use failover and not need multiple destinations then I would use:

session:execute("bridge",
"sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828");
<< call using broadvox first and quest it that fails

I need to combine:

Call 6095553828  and 7325553828 simultaneously and each trying
broadvox then on fail quest.

session:execute("bridge",
"(sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828),(sofia/gateway/broadvox/7325553828,sofia/gateway/quest/7325553828)");

But of course that syntax does not work.

I can not see another syntax that would achieve this?


On Tue, Aug 11, 2009 at 10:58 AM, Brian West<brian at freeswitch.org> wrote:
> This can all be accomplished with standard bridge lines stacked one on top
> of the other with continue_on_fail set to the list of codes you wish to fail
> over with. ?You would also want to set hangup_after_bridge=true. ?Loopback
> should NEVER be used in this case.
> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
> http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge
> /b
> On Aug 11, 2009, at 9:46 AM, Phillip Jones wrote:
>
> Thanks for the reply Brian.
>
> I went through this fairly carefully with Mike Jerris - please see
> http://www.nabble.com/Calling-Multiple-Destinations-with-Failover-td24877157.html
>
> I am very open to any other mechanism that allows the calling multiple
> destinations with carrier failover support. I would have thought this
> is a fairly common requirement and is a fundamental requirement in my
> application.
>
> As always - your help is appreciated.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jmesquita at gmail.com  Tue Aug 11 09:30:59 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 11 Aug 2009 13:30:59 -0300
Subject: [Freeswitch-users] VoiceMail transcription
In-Reply-To: <2bee4fc40908110717p43649678ue2b94ecb571bd09@mail.gmail.com>
References: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
	<1249998014.20224.82.camel@dk-d820>
	<2bee4fc40908110717p43649678ue2b94ecb571bd09@mail.gmail.com>
Message-ID: <5a8712120908110930s74b84bf8x673ca1209fbae965@mail.gmail.com>

I am sorry for the ignorance on the matter, but how does google voice does?
Do they also have humans?

jmesquita

On Tue, Aug 11, 2009 at 11:17 AM, Kirk Bateman <kirk.bateman at gmail.com>wrote:

> I'm still interested in getting pocketsphinx to attempt speech recognition
> on an audio file.
>
> To be honest, most of the problem is that at 8Khz (mobile phone call rate),
> speech detection is NOT very accurate, at 16Khz it IS significantly better.
>
> I'm planning to have a play with the speechtools module and
> mod_pocketsphinx etc to try and get an audio file parsed, spare time
> permitting.
>
> Will let the list know if I get anywhere.
>
> Regards
>
> Kirk Bateman
>
>
> 2009/8/11 David Knell <dave at 3c.co.uk>
>
> Hi Pete,
>>
>> I'm afraid that the answer's still the same: use a human.  Here's an
>> article describing the state of the art:
>> http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/
>> - the links to previous stories at the bottom provide good background.
>>
>> --Dave
>>
>> > I apologize, I should have been more clear.  We will be using humans
>> > to scan the translated results.  But we are looking for a system to
>> > perform the "first pass" on the audio to hopefully help the human type
>> > less.
>> >
>> > Although the question has been raised if it's faster to have a human
>> > just transcribe the whole thing, or fix up what the computer spit out.
>> > If you have any insights on this, that would be great.
>> >
>> > -pete
>> >
>> >         -------- Original Message --------
>> >         Subject: Re: [Freeswitch-users] VoiceMail transcription
>> >         From: David Knell <dave at 3c.co.uk>
>> >         Date: Mon, August 10, 2009 11:51 am
>> >         To: freeswitch-users at lists.freeswitch.org
>> >
>> >         Good evening Pete,
>> >
>> >         The only way to do this is, I'm afraid, to use a human. We use
>> >         Amazon's
>> >         Mechanical Turk to good effect.
>> >
>> >         Cheers --
>> >
>> >         Dave
>> >
>> >         > Good morning all,
>> >         >
>> >         > I realize this is slightly off the FS topic, but I am
>> >         wondering if
>> >         > anyone out there has experience with software packages
>> >         designed for
>> >         > the transcription of voicemails to text. I've used
>> >         pocketsphinx with
>> >         > FS to handle IVR menus, but now have the task of figuring
>> >         out how to
>> >         > convert recorded phone conversations (voicemails mostly) to
>> >         text.
>> >         >
>> >         > This does not have to be a real-time process, I can store
>> >         the audio
>> >         > files and process them over time. This would need to be a
>> >         software
>> >         > (preferable open source) solution. ASPs like VoiceCloud
>> >         would not
>> >         > work for this application.
>> >         >
>> >         > Thanks for any help
>> >         > -pete
>> >         > _______________________________________________
>> >         > FreeSWITCH-users mailing list
>> >         > FreeSWITCH-users at lists.freeswitch.org
>> >         >
>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >         >
>> >         UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >         > http://www.freeswitch.org
>> >         --
>> >         David Knell, Director, 3C Limited
>> >         T: +44 20 3298 2000
>> >         E: dave at 3c.co.uk
>> >         W: http://www.3c.co.uk
>> >
>> >
>> >         _______________________________________________
>> >         FreeSWITCH-users mailing list
>> >         FreeSWITCH-users at lists.freeswitch.org
>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >         UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >         http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> --
>> David Knell, Director, 3C Limited
>> T: +44 20 3298 2000
>> E: dave at 3c.co.uk
>> W: http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Aug 11 09:55:26 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 11:55:26 -0500
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <4A811FEC.7040808@kgolding.co.uk>
References: <4A80AEE2.7050306@kgolding.co.uk>
	<7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>
	<4A811FEC.7040808@kgolding.co.uk>
Message-ID: <87f2f3b90908110955s3f67e710je913a34c7cc3055a@mail.gmail.com>

On Tue, Aug 11, 2009 at 2:38 AM, Kevin Golding <kevin at kgolding.co.uk> wrote:

> Thanks Brian,
>
> Well spotted with the domain/cidr :)
>
> I changed the line <param name="apply-inbound-acl" value="domains"/> in
> the internal.xml with a new value, and changed the list to match but I
> still get the same 'Rejected by acl "domains"' error.
>

You need to apply the name of your new ACL in the sip profile. In the file
internal.xml look for this line:
<param name="apply-inbound-acl" value="domains"/>

And change "domains" to whatever your new ACL is named:
<param name="apply-inbound-acl" value="YOUR_NEW_ACL_NAME_HERE"/>

Then restart FS or go to the command line and execute:
reloadacl reloadxml

Should be good after that!
-MC

>
> And yes I reloaded the xml. :)
>
> Kevin
>
> Brian West wrote:
> > First off edit the profile to use your own custom ACL... not the
> > domains one... secondly you'll have to use cidr= instead of domain=.
> >
> > Domain= will search the user directory building an ACL list from all
> > users with the cidr= attribute.
> >
> > /b
> >
> >
> > On Aug 10, 2009, at 6:36 PM, Kevin Golding wrote:
> >
> >> acl.conf.xml file, and have put the following in it:
> >>
> >> <configuration name="acl.conf" description="Network Lists">
> >>   <network-lists>
> >>     <list name="domains" value="allow">
> >>         <node type="allow" domain="$${domain}"/>
> >>         <node type="allow" domain="213.166.5.128/28"/>
> >>         <node type="allow" domain="87.238.72.128/26"/>
> >>     </list>
> >>   </network-lists>
> >> </configuration>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bsnipes at snipes.org  Tue Aug 11 09:56:04 2009
From: bsnipes at snipes.org (bsnipes)
Date: Tue, 11 Aug 2009 09:56:04 -0700 (PDT)
Subject: [Freeswitch-users]  Multiple phones and single extensions
Message-ID: <24921557.post@talk.nabble.com>


I've tried to do this a couple of ways in the past but can't quite seem to
get it right. The scenario is this:

attorney at extension 326 with 2 phones
secretary at extension 327 with 1 phone and a blf that monitors 326

Initially I was simply adding the attorney's extension to the secretary's
phone as a second registration but the issue comes up that the monitor light
for the 326 line lights up as if the attorney is on the phone and to other
people it shows the attorney's extension in use as if he were on the phone. 
Ok says I... I'll change the dialstring and have it ring both his and her
extension so that if she picks up it will be on her extension and it will
still show her with her blf if he is on his 326 extension. That works but
the blf light stays lit a lot after she picks up on a call destined for his
extension.  I use the following dialstring lines:

<action application="set" data="pcaller_id_name=ATTY:${caller_id_name}"/>
<action application="set"
data="dialstring=${sofia_contact(internal/${dialed_ext}@192.168.5.4)},[origination_caller_id_name=${pcaller_id_name}]sofia/default/327"/>

The phones are SNOM 370s and what  I am not sure of is if the light
constantly blinking is a firmware issue on the phone or a notification issue
from FS. I've noticed that the same constant blinking happens with groups
also. If I monitor a phone that is a member of a group and the group gets
called the light for that monitor key and the message light will blink even
if no one is on the phone. Doesn't happen every time but multiple times a
day it does.

Any suggestions?
Brian

-- 
View this message in context: http://www.nabble.com/Multiple-phones-and-single-extensions-tp24921557p24921557.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From msc at freeswitch.org  Tue Aug 11 10:09:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 12:09:51 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
Message-ID: <87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>

On Tue, Aug 11, 2009 at 9:05 AM, Milena <testeador01 at gmail.com> wrote:

>
> Hello Brian,
>
> I wanna fix the wiki, but to make sure i got it right, does it only work on
> outbound event socket? or is there any other scenario where it would work.
>

FYI, Diego Viola fixed the wiki. (Thanks Diego!)
-MC
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From lfurrea at gmail.com  Tue Aug 11 10:11:41 2009
From: lfurrea at gmail.com (Luis F Urrea)
Date: Tue, 11 Aug 2009 11:11:41 -0600
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <1513B1DC-2CD5-484F-B2CD-74B25E5AB822@jerris.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<1513B1DC-2CD5-484F-B2CD-74B25E5AB822@jerris.com>
Message-ID: <f120f4c80908111011h75ced645g74e363bbdabf1d19@mail.gmail.com>

On Tue, Aug 11, 2009 at 9:38 AM, Michael Jerris <mike at jerris.com> wrote:

> Correct me if I am wrong, but can't all these differences be done in a
> different soundset without modifying any code or other config?
>

Particularly the changes in discussion only refer to different ways to say
things.
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From brian at freeswitch.org  Tue Aug 11 10:37:55 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 12:37:55 -0500
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <4A811FEC.7040808@kgolding.co.uk>
References: <4A80AEE2.7050306@kgolding.co.uk>
	<7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>
	<4A811FEC.7040808@kgolding.co.uk>
Message-ID: <0C9B302E-C1A6-4D1F-831B-63FA2AEB8C37@freeswitch.org>

you have to now apply the ACL correctly to the sofia profile.

/b

On Aug 11, 2009, at 2:38 AM, Kevin Golding wrote:

> Thanks Brian,
>
> Well spotted with the domain/cidr :)
>
> I changed the line <param name="apply-inbound-acl" value="domains"/>  
> in
> the internal.xml with a new value, and changed the list to match but I
> still get the same 'Rejected by acl "domains"' error.
>
> And yes I reloaded the xml. :)
>
> Kevin



From msc at freeswitch.org  Tue Aug 11 11:07:23 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 13:07:23 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
Message-ID: <87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>

On Tue, Aug 11, 2009 at 11:08 AM, Phillip Jones <pjintheusa at gmail.com>wrote:

> Can you use this method to make simultaneous calls though while
> preserving carrier order? i.e. each call must try carrier 1 first then
> carrier 2 (because each carrier terminates a different subset of
> numbers)
>
> So - if I did NOT want to support failover I would use:
>
> session:execute("bridge",
> "sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828");
>  << call 6095553828 and 7325553828 at the same time
>
> If I were to use failover and not need multiple destinations then I would
> use:
>
> session:execute("bridge",
> "sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828");
> << call using broadvox first and quest it that fails
>
> I need to combine:
>
> Call 6095553828  and 7325553828 simultaneously and each trying
> broadvox then on fail quest.
>
> session:execute("bridge",
>
> "(sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828),(sofia/gateway/broadvox/7325553828,sofia/gateway/quest/7325553828)");
>
> But of course that syntax does not work.
>
> I can not see another syntax that would achieve this?


Just to make sure I'm reading you correctly I want to clarify...

You are trying to maintain carrier order and also trying to dial
simultaneously. If I understand what you mean, you have two separate
dialstrings that you want to dial simultaneously and if BOTH fail, THEN you
want to dial the third dialstring?

If the above is correct then this dialplan snippet SHOULD work:

<action application="set" data="continue_on_fail=true"/>
<action application="bridge"
data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
<!-- if either of the above dialstrings is successful then this next line
won't execute... -->
<action application="bridge" data="sofia/gateway/quest/7325553828"/>

I also noticed that you're using js in your example... I hope you will
consider using the dialplan instead. The DP is quick and clean. At the very
least, if you really need js for some logic then let the js do the logic and
then transfer back out to the dialplan. That way you will use fewer system
resources and your project can scale more easily.

-MC
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From diego.viola at gmail.com  Tue Aug 11 11:10:42 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 11 Aug 2009 14:10:42 -0400
Subject: [Freeswitch-users] answer command
In-Reply-To: <87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
Message-ID: <86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>

Michael, you're welcome :).

Milena, answer is a mod_dptools command, you can use it from the XML
dialplan or from the event socket outbound. mod_commands API are APIs that
you execute from the socket, event socket inbound, etc. But you can also
execute them from event socket outbound using the "api" command.

I hope that makes sense, correct me if I'm wrong =D.

On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Aug 11, 2009 at 9:05 AM, Milena <testeador01 at gmail.com> wrote:
>
>>
>> Hello Brian,
>>
>> I wanna fix the wiki, but to make sure i got it right, does it only work
>> on outbound event socket? or is there any other scenario where it would
>> work.
>>
>
> FYI, Diego Viola fixed the wiki. (Thanks Diego!)
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Aug 11 11:23:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 13:23:41 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
Message-ID: <87f2f3b90908111123x7316d515lf7afe91e56aad842@mail.gmail.com>

On Tue, Aug 11, 2009 at 1:10 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Michael, you're welcome :).
>
> Milena, answer is a mod_dptools command, you can use it from the XML
> dialplan or from the event socket outbound. mod_commands API are APIs that
> you execute from the socket, event socket inbound, etc. But you can also
> execute them from event socket outbound using the "api" command.
>
> I hope that makes sense, correct me if I'm wrong =D.


A simple way to remember it is like this:

mod_dptools are dialplan tools that go in the dialplan, aka dialplan apps
mod_commands are APIs that are typed at the FS command line

Of course, there's some crossover, but as long as you separate the idea of
an API and an app you'll be fine.
-MC
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From diego.viola at gmail.com  Tue Aug 11 11:24:08 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 11 Aug 2009 14:24:08 -0400
Subject: [Freeswitch-users] answer command
In-Reply-To: <86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
Message-ID: <86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>

I suggest that you learn the differences between mod_commands commands and
mod_dptools applications, and also the interfaces where you can access and
use them.

As said before, mod_dptools is accessible from dialplan, event socket
outbound, etc. and mod_commands is accessible from the CLI, event socket
(inbound/outbound), XML RPC, etc.

That's all described in the wiki I think.

Let us know if you have any questions =D.

On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Michael, you're welcome :).
>
> Milena, answer is a mod_dptools command, you can use it from the XML
> dialplan or from the event socket outbound. mod_commands API are APIs that
> you execute from the socket, event socket inbound, etc. But you can also
> execute them from event socket outbound using the "api" command.
>
> I hope that makes sense, correct me if I'm wrong =D.
>
> On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, Aug 11, 2009 at 9:05 AM, Milena <testeador01 at gmail.com> wrote:
>>
>>>
>>> Hello Brian,
>>>
>>> I wanna fix the wiki, but to make sure i got it right, does it only work
>>> on outbound event socket? or is there any other scenario where it would
>>> work.
>>>
>>
>> FYI, Diego Viola fixed the wiki. (Thanks Diego!)
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From boonedox at gmail.com  Tue Aug 11 11:29:44 2009
From: boonedox at gmail.com (Jeremiah Johnson)
Date: Tue, 11 Aug 2009 12:29:44 -0600
Subject: [Freeswitch-users] hangup_after_bridge=false only works if
	bypass_media=false
Message-ID: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>

This is an integral part of my application.  I need to have FreeSWITCH
outside of the media path as well as be able to do multiple bridges for the
same "A" leg.
/*WORKS*/
<action application="set" data="hangup_after_bridge=false"/>
<action application="bridge"
data="sofia/gateway/${mygateway}/1${client_one}"/>
<action application="bridge"
data="sofia/gateway/${mygateway}/1${client_two}"/>

/*DOES NOT WORK*/
<action application="set" data="hangup_after_bridge=false"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge"
data="sofia/gateway/${mygateway}/1${client_one}"/>
<action application="bridge"
data="sofia/gateway/${mygateway}/1${client_two}"/>

In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the leg for
client_one hangs up.
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From pjintheusa at gmail.com  Tue Aug 11 11:43:36 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 11 Aug 2009 14:43:36 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
Message-ID: <367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>

Hi Michael,

Thanks for trying to help with this.

> <action application="set" data="continue_on_fail=true"/>
> <action application="bridge"
> data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
> <!-- if either of the above dialstrings is successful then this next line
> won't execute... -->
> <action application="bridge" data="sofia/gateway/quest/7325553828"/>

In the dial plan above, consider that target is sitting at their
6095553828 phone. And that broadvox does not route to 6095553828  and
passes me a NO_ROUTE_DESTINATION.

The target will never get the call.

So if I extend this to:

<action application="bridge"  data="sofia/gateway/broadvox/6095553828,
sofia/gateway/broadvox/7325553828"/>
<action application="bridge" data="sofia/gateway/quest/6095553828.
sofia/gateway/quest/7325553828"/>

In this plan you are not calling the numbers simultaneously if the
broadvox is not able to route both calls. I can not see that you can
stack this way and get the ability to call multiple destinations, so I
must be missing something.

"Call blast" or "Followme" is basically the functionality I am trying
to implement. Separately we use multiple carriers. This is fairly
common requirement.

I am using .NET by the way - but point taken.

Any other ideas on how to achieve multiple destinations with carrier
failover would great!

Thanks again.


On Tue, Aug 11, 2009 at 2:07 PM, Michael Collins<msc at freeswitch.org> wrote:
>
>
> On Tue, Aug 11, 2009 at 11:08 AM, Phillip Jones <pjintheusa at gmail.com>
> wrote:
>>
>> Can you use this method to make simultaneous calls though while
>> preserving carrier order? i.e. each call must try carrier 1 first then
>> carrier 2 (because each carrier terminates a different subset of
>> numbers)
>>
>> So - if I did NOT want to support failover I would use:
>>
>> session:execute("bridge",
>> "sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828");
>> ?<< call 6095553828 and 7325553828 at the same time
>>
>> If I were to use failover and not need multiple destinations then I would
>> use:
>>
>> session:execute("bridge",
>> "sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828");
>> << call using broadvox first and quest it that fails
>>
>> I need to combine:
>>
>> Call 6095553828 ?and 7325553828 simultaneously and each trying
>> broadvox then on fail quest.
>>
>> session:execute("bridge",
>>
>> "(sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828),(sofia/gateway/broadvox/7325553828,sofia/gateway/quest/7325553828)");
>>
>> But of course that syntax does not work.
>>
>> I can not see another syntax that would achieve this?
>
> Just to make sure I'm reading you correctly I want to clarify...
>
> You are trying to maintain carrier order and also trying to dial
> simultaneously. If I understand what you mean, you have two separate
> dialstrings that you want to dial simultaneously and if BOTH fail, THEN you
> want to dial the third dialstring?
>
> If the above is correct then this dialplan snippet SHOULD work:
>
> <action application="set" data="continue_on_fail=true"/>
> <action application="bridge"
> data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
> <!-- if either of the above dialstrings is successful then this next line
> won't execute... -->
> <action application="bridge" data="sofia/gateway/quest/7325553828"/>
>
> I also noticed that you're using js in your example... I hope you will
> consider using the dialplan instead. The DP is quick and clean. At the very
> least, if you really need js for some logic then let the js do the logic and
> then transfer back out to the dialplan. That way you will use fewer system
> resources and your project can scale more easily.
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From pjintheusa at gmail.com  Tue Aug 11 11:50:42 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 11 Aug 2009 14:50:42 -0400
Subject: [Freeswitch-users] hangup_after_bridge=false only works if
	bypass_media=false
In-Reply-To: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
References: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
Message-ID: <367751820908111150s468a1717l20cf409ef8b7f66b@mail.gmail.com>

Hi there,

I the 'does not work' example your media stream is back up at the
carrier as the called party hangs up.

You would have to cause another SIP REINVITE and re capture the media.

See http://wiki.freeswitch.org/wiki/Bypass_Media and the "How to
disable/enable it on the fly?" section as a start.




On Tue, Aug 11, 2009 at 2:29 PM, Jeremiah Johnson<boonedox at gmail.com> wrote:
> This is an integral part of my application. ?I need to have FreeSWITCH
> outside of the media path as well as be able to do multiple bridges for the
> same "A" leg.
> /*WORKS*/
> <action application="set" data="hangup_after_bridge=false"/>
> <action application="bridge"
> data="sofia/gateway/${mygateway}/1${client_one}"/>
> <action application="bridge"
> data="sofia/gateway/${mygateway}/1${client_two}"/>
> /*DOES NOT WORK*/
> <action application="set" data="hangup_after_bridge=false"/>
> <action application="set" data="bypass_media=true"/>
> <action application="bridge"
> data="sofia/gateway/${mygateway}/1${client_one}"/>
> <action application="bridge"
> data="sofia/gateway/${mygateway}/1${client_two}"/>
> In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the leg for
> client_one hangs up.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From lfurrea at gmail.com  Tue Aug 11 11:55:45 2009
From: lfurrea at gmail.com (Luis F Urrea)
Date: Tue, 11 Aug 2009 12:55:45 -0600
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <f120f4c80908111011h75ced645g74e363bbdabf1d19@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<1513B1DC-2CD5-484F-B2CD-74B25E5AB822@jerris.com>
	<f120f4c80908111011h75ced645g74e363bbdabf1d19@mail.gmail.com>
Message-ID: <f120f4c80908111155y7704c08fi922706487ab5b3d9@mail.gmail.com>

The patch submitted by Diego Toro, however would fix the actual
pronunciation of currency and dates  from 20th - 29th which are not working
with the "english" implementation of mod_say_es.c

In regards the dates pronunciation for voicemail for example we used the
following changes to mod_say_es.c

372,375c412,427
-               say_file("time/day-%d.wav", tm.tm_wday);
-               say_file("time/mon-%d.wav", tm.tm_mon);
-               say_num(tm.tm_mday, SSM_COUNTED);
-               say_num(tm.tm_year + 1900, SSM_PRONOUNCED);
---
+               if (tm.tm_mday == 1) {
+                       say_file("time/day-%d.wav", tm.tm_wday);
+                       say_num(tm.tm_mday, SSM_COUNTED);
+                       say_file("time/de.wav");
+                       say_file("time/mon-%d.wav", tm.tm_mon);
+                       say_file("time/de.wav");
+                       say_num(tm.tm_year + 1900, SSM_PRONOUNCED);
+
+               } else {
+                       say_file("time/day-%d.wav", tm.tm_wday);
+                       say_num(tm.tm_mday, SSM_PRONOUNCED);
+                       say_file("time/de.wav");
+                       say_file("time/mon-%d.wav", tm.tm_mon);
+                       say_file("time/de.wav");
+                       say_num(tm.tm_year + 1900, SSM_PRONOUNCED);
+               }
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From gshfreesw at gmail.com  Tue Aug 11 12:10:33 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Tue, 11 Aug 2009 15:10:33 -0400
Subject: [Freeswitch-users] Call transfer issues
Message-ID: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>

Hello,

Good work guys. I am having good fun using freeswitch so far. Currently, I
am having a serious issue on making a call transfer happen. The scenario is
simple.

1. Caller A arrives on extension 1 and is waiting on a fifo queue.
2. Caller B arrives on  extension 2 and dial plan bridges the call to Caller
C.

Now after a certain period of time, I want to hang up Caller B and put
Caller C in the fifo queue where Caller A is waiting. I know the UUIDs of
the all 3 calls. Naively, I tried the following and failed.

1. I put the bridged call to Caller C on park .
2. I did a "hang up" to caller B.
3. I did "fifo in" for Caller C into fifo queue of caller A.

This did not work. What is the right way to do this? Do I have make sure
uuid_transfer? I am using Event socket to do all of this.

-Shameem.
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From brian at freeswitch.org  Tue Aug 11 12:19:15 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 14:19:15 -0500
Subject: [Freeswitch-users] Call transfer issues
In-Reply-To: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>
References: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>
Message-ID: <D554D137-805E-4924-A43B-3930275451BE@freeswitch.org>

OK I try so hard to follow what you're doing but it makes little sense  
to me.

I'm guessing you'll need to set/export hangup_after_bridge=false to  
prevent the hangup from taking place on B and C!

The other option is to uuid_transfer both B and C to park using the - 
both then hangup on C and then uuid_bridge A and C.  Because once you  
break the bridge on B and C by hanging up on B you left C hanging so  
its naturally going to hangup.

-USAGE: <uuid> [-bleg|-both] <dest-exten> [<dialplan>] [<context>]

/b

On Aug 11, 2009, at 2:10 PM, Shameem Shiek wrote:

> Hello,
>
> Good work guys. I am having good fun using freeswitch so far.  
> Currently, I am having a serious issue on making a call transfer  
> happen. The scenario is simple.
>
> 1. Caller A arrives on extension 1 and is waiting on a fifo queue.
> 2. Caller B arrives on  extension 2 and dial plan bridges the call  
> to Caller C.
>
> Now after a certain period of time, I want to hang up Caller B and  
> put Caller C in the fifo queue where Caller A is waiting. I know the  
> UUIDs of the all 3 calls. Naively, I tried the following and failed.
>
> 1. I put the bridged call to Caller C on park .
> 2. I did a "hang up" to caller B.
> 3. I did "fifo in" for Caller C into fifo queue of caller A.
>
> This did not work. What is the right way to do this? Do I have make  
> sure uuid_transfer? I am using Event socket to do all of this.
>
> -Shameem.
> _______________



From msc at freeswitch.org  Tue Aug 11 12:19:43 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 14:19:43 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
Message-ID: <87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>

On Tue, Aug 11, 2009 at 1:43 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi Michael,
>
> Thanks for trying to help with this.
>
> > <action application="set" data="continue_on_fail=true"/>
> > <action application="bridge"
> >
> data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
> > <!-- if either of the above dialstrings is successful then this next line
> > won't execute... -->
> > <action application="bridge" data="sofia/gateway/quest/7325553828"/>
>
> In the dial plan above, consider that target is sitting at their
> 6095553828 phone. And that broadvox does not route to 6095553828  and
> passes me a NO_ROUTE_DESTINATION.
>
> The target will never get the call.
>
> So if I extend this to:
>
> <action application="bridge"  data="sofia/gateway/broadvox/6095553828,
> sofia/gateway/broadvox/7325553828"/>
> <action application="bridge" data="sofia/gateway/quest/6095553828.
> sofia/gateway/quest/7325553828"/>
>
> In this plan you are not calling the numbers simultaneously if the
> broadvox is not able to route both calls. I can not see that you can
> stack this way and get the ability to call multiple destinations, so I
> must be missing something.


When you say "call multiple destinations" what exactly do you mean? If the
Broadvox is unable to connect either call then the dialplan moves on to the
Quest dial attempts. If that isn't what you need then can you clarify? Under
what condition(s) would the dialplan need to attempt to call via Quest?


>
> "Call blast" or "Followme" is basically the functionality I am trying
> to implement. Separately we use multiple carriers. This is fairly
> common requirement.
>

Yep, makes sense. Could you maybe write it out in pseudo code? Something
like this:
#1 Call phone numbers ABC and XYZ simultaneously via Broadvox
#2 If BOTH ABC AND XYZ fail, then call ABC and XYZ via Quest


>
> I am using .NET by the way - but point taken.
>

My mistake. :)


>
> Any other ideas on how to achieve multiple destinations with carrier
> failover would great!
>

I'm 100% certain that this is possible with FreeSWITCH. You might just need
a different approach, depending upon the circumstances with your carriers.
Try the above suggestions first and let's see what happens. We'll take it
from there.
-MC
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From alan at chandlerfamily.org.uk  Tue Aug 11 12:25:14 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Tue, 11 Aug 2009 20:25:14 +0100
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
Message-ID: <4A81C59A.2020705@chandlerfamily.org.uk>

samuel wrote:
> I'm also for different spanish localization if it's not too complicated. 
> It was also for me the first time I see "signo de n?mero" for pound ;)
> 


I just idly noticed this - so just a comment from a Brit who iteracts 
quite a bit with Americans.

The # symbol in the UK is not called "pound" because (I don't know if 
this will come out on your screens correctly) we use ? for our currency.

I would refer to the # symbol as "hash" or just possibly the "number 
symbol".


-- 
Alan Chandler
http://www.chandlerfamily.org.uk




From gshfreesw at gmail.com  Tue Aug 11 12:33:50 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Tue, 11 Aug 2009 15:33:50 -0400
Subject: [Freeswitch-users] Call transfer issues
In-Reply-To: <D554D137-805E-4924-A43B-3930275451BE@freeswitch.org>
References: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>
	<D554D137-805E-4924-A43B-3930275451BE@freeswitch.org>
Message-ID: <5070fcbd0908111233w6ecefeeex92b1304f04744f23@mail.gmail.com>

Thanks for the explanation. I was trying to find the correct way to do this
as I saw several ways of doing this. I will try to explain how I did on the
asterisk land.

In asterisk land, Caller A will be parked at extension 701 and Caller B
"transfers" the call to extension 701 and and Caller A and C are connected
and Caller B is hung up right after transfer.

In Freeswitch, Can Caller B do a "transfer" and connect caller A and C? How
can I do this? Since Freeswitch does not have a concept of parked
extensions.

Thank you for the help.


On Tue, Aug 11, 2009 at 3:19 PM, Brian West <brian at freeswitch.org> wrote:

> OK I try so hard to follow what you're doing but it makes little sense
> to me.
>
> I'm guessing you'll need to set/export hangup_after_bridge=false to
> prevent the hangup from taking place on B and C!
>
> The other option is to uuid_transfer both B and C to park using the -
> both then hangup on C and then uuid_bridge A and C.  Because once you
> break the bridge on B and C by hanging up on B you left C hanging so
> its naturally going to hangup.
>
> -USAGE: <uuid> [-bleg|-both] <dest-exten> [<dialplan>] [<context>]
>
> /b
>
> On Aug 11, 2009, at 2:10 PM, Shameem Shiek wrote:
>
> > Hello,
> >
> > Good work guys. I am having good fun using freeswitch so far.
> > Currently, I am having a serious issue on making a call transfer
> > happen. The scenario is simple.
> >
> > 1. Caller A arrives on extension 1 and is waiting on a fifo queue.
> > 2. Caller B arrives on  extension 2 and dial plan bridges the call
> > to Caller C.
> >
> > Now after a certain period of time, I want to hang up Caller B and
> > put Caller C in the fifo queue where Caller A is waiting. I know the
> > UUIDs of the all 3 calls. Naively, I tried the following and failed.
> >
> > 1. I put the bridged call to Caller C on park .
> > 2. I did a "hang up" to caller B.
> > 3. I did "fifo in" for Caller C into fifo queue of caller A.
> >
> > This did not work. What is the right way to do this? Do I have make
> > sure uuid_transfer? I am using Event socket to do all of this.
> >
> > -Shameem.
> > _______________
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Aug 11 12:40:14 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 14:40:14 -0500
Subject: [Freeswitch-users] Call transfer issues
In-Reply-To: <5070fcbd0908111233w6ecefeeex92b1304f04744f23@mail.gmail.com>
References: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>
	<D554D137-805E-4924-A43B-3930275451BE@freeswitch.org>
	<5070fcbd0908111233w6ecefeeex92b1304f04744f23@mail.gmail.com>
Message-ID: <55901B44-E104-4829-8B40-81B36A3522CA@freeswitch.org>


On Aug 11, 2009, at 2:33 PM, Shameem Shiek wrote:

> Thanks for the explanation. I was trying to find the correct way to  
> do this as I saw several ways of doing this. I will try to explain  
> how I did on the asterisk land.
>
> In asterisk land, Caller A will be parked at extension 701 and  
> Caller B "transfers" the call to extension 701 and and Caller A and  
> C are connected and Caller B is hung up right after transfer.
>
> In Freeswitch, Can Caller B do a "transfer" and connect caller A and  
> C? How can I do this? Since Freeswitch does not have a concept of  
> parked extensions.

This is where you're wrong... you can park people with the park  
application just transfer them to it... But you have to take off the  
asterisk hat.. I told you in the last email how to do it correctly  
using nothing but API commands.

/b


>
> Thank you for the help.



From mike at jerris.com  Tue Aug 11 12:45:38 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 Aug 2009 15:45:38 -0400
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <4A81C59A.2020705@chandlerfamily.org.uk>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
	<4A81C59A.2020705@chandlerfamily.org.uk>
Message-ID: <06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>

again, this issue should be addressed when you do a sound set for that  
dialect, we are attempting to keep the c code common for all dialects  
within a language, we will see if this works unless anyone can point  
to a place this will not work.

Mike

On Aug 11, 2009, at 3:25 PM, Alan Chandler wrote:

> samuel wrote:
>> I'm also for different spanish localization if it's not too  
>> complicated.
>> It was also for me the first time I see "signo de n?mero" for  
>> pound ;)
>>
>
>
> I just idly noticed this - so just a comment from a Brit who iteracts
> quite a bit with Americans.
>
> The # symbol in the UK is not called "pound" because (I don't know if
> this will come out on your screens correctly) we use ? for our  
> currency.
>
> I would refer to the # symbol as "hash" or just possibly the "number
> symbol".
>
>
> -- 
> Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jmesquita at gmail.com  Tue Aug 11 12:49:31 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 11 Aug 2009 16:49:31 -0300
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
	<4A81C59A.2020705@chandlerfamily.org.uk>
	<06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>
Message-ID: <5a8712120908111249t3e23ceeese1c18d6fef52a227@mail.gmail.com>

Mike, the gender thing will eventually have to change code, I guess. I have
not yet looked at the say code, so I am just imagining here.



On Tue, Aug 11, 2009 at 4:45 PM, Michael Jerris <mike at jerris.com> wrote:

> again, this issue should be addressed when you do a sound set for that
> dialect, we are attempting to keep the c code common for all dialects
> within a language, we will see if this works unless anyone can point
> to a place this will not work.
>
> Mike
>
> On Aug 11, 2009, at 3:25 PM, Alan Chandler wrote:
>
> > samuel wrote:
> >> I'm also for different spanish localization if it's not too
> >> complicated.
> >> It was also for me the first time I see "signo de n?mero" for
> >> pound ;)
> >>
> >
> >
> > I just idly noticed this - so just a comment from a Brit who iteracts
> > quite a bit with Americans.
> >
> > The # symbol in the UK is not called "pound" because (I don't know if
> > this will come out on your screens correctly) we use ? for our
> > currency.
> >
> > I would refer to the # symbol as "hash" or just possibly the "number
> > symbol".
> >
> >
> > --
> > Alan Chandler
> > http://www.chandlerfamily.org.uk
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Aug 11 12:54:12 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 14:54:12 -0500
Subject: [Freeswitch-users] Ivr and variables
In-Reply-To: <8D85DC2E55A74F41AE2190E05A44F028@voztovoice>
References: <8D85DC2E55A74F41AE2190E05A44F028@voztovoice>
Message-ID: <87f2f3b90908111254i4479d3f9m68ab8cce6a62554e@mail.gmail.com>

What value is in the variable ${some} ?
-MC

On Mon, Aug 10, 2009 at 1:43 PM, bakko <asannucci at gmail.com> wrote:

> I'm tryng to put a variable in a IVR digits line like this:
>
> <entry action="menu-exec-app" digits="${some}" param="transfer 1001 XML
> default"/>
>
> but don't work
>
> Is It possible tu use this solution?
>
> thank you
>
> BR
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Tue Aug 11 12:54:16 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 11 Aug 2009 15:54:16 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
Message-ID: <367751820908111254h4f39b8d5hf793e3e9dce78cbb@mail.gmail.com>

Thanks Michael,

pseudo code would look this.

1) Call 6095556666 and 7325556666 simultaneously.
2a) As you dial 6095556666 go to the next carrier if you receive
NO_ROUTE_DESTINATION.
2b) As you dial 7325556666 go to the next carrier if you receive
NO_ROUTE_DESTINATION
3) The first number to answer and accept the call - bridge to leg a.
Drop the other.


Brian maintains that this can be done without loopback and loopback
should never be used. Mike Jerris believes that loop back handles this
and should be used. I am happy for all the help, but I am a bit
confused at this stage.

If this can not be done in the dailplan then I will need to pursue
loopback and see how far I get with that. At the moment it works but
not with bypass_media.

As I said "call followme" and carrier failover are fairly common
requirements so I do feel I am missing something. This must be
implemented elsewhere.

Again - thank you all for your help. You do a great job of getting
getting your heads around all this stuff day in day out.

Phillip Jones


On Tue, Aug 11, 2009 at 3:19 PM, Michael Collins<msc at freeswitch.org> wrote:
>
>
> On Tue, Aug 11, 2009 at 1:43 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
>>
>> Hi Michael,
>>
>> Thanks for trying to help with this.
>>
>> > <action application="set" data="continue_on_fail=true"/>
>> > <action application="bridge"
>> >
>> > data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
>> > <!-- if either of the above dialstrings is successful then this next
>> > line
>> > won't execute... -->
>> > <action application="bridge" data="sofia/gateway/quest/7325553828"/>
>>
>> In the dial plan above, consider that target is sitting at their
>> 6095553828 phone. And that broadvox does not route to 6095553828 ?and
>> passes me a NO_ROUTE_DESTINATION.
>>
>> The target will never get the call.
>>
>> So if I extend this to:
>>
>> <action application="bridge" ?data="sofia/gateway/broadvox/6095553828,
>> sofia/gateway/broadvox/7325553828"/>
>> <action application="bridge" data="sofia/gateway/quest/6095553828.
>> sofia/gateway/quest/7325553828"/>
>>
>> In this plan you are not calling the numbers simultaneously if the
>> broadvox is not able to route both calls. I can not see that you can
>> stack this way and get the ability to call multiple destinations, so I
>> must be missing something.
>
> When you say "call multiple destinations" what exactly do you mean? If the
> Broadvox is unable to connect either call then the dialplan moves on to the
> Quest dial attempts. If that isn't what you need then can you clarify? Under
> what condition(s) would the dialplan need to attempt to call via Quest?
>
>>
>>
>> "Call blast" or "Followme" is basically the functionality I am trying
>> to implement. Separately we use multiple carriers. This is fairly
>> common requirement.
>
> Yep, makes sense. Could you maybe write it out in pseudo code? Something
> like this:
> #1 Call phone numbers ABC and XYZ simultaneously via Broadvox
> #2 If BOTH ABC AND XYZ fail, then call ABC and XYZ via Quest
>
>>
>> I am using .NET by the way - but point taken.
>
> My mistake. :)
>
>>
>> Any other ideas on how to achieve multiple destinations with carrier
>> failover would great!
>
> I'm 100% certain that this is possible with FreeSWITCH. You might just need
> a different approach, depending upon the circumstances with your carriers.
> Try the above suggestions first and let's see what happens. We'll take it
> from there.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Tue Aug 11 12:54:53 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 14:54:53 -0500
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <5a8712120908111249t3e23ceeese1c18d6fef52a227@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
	<4A81C59A.2020705@chandlerfamily.org.uk>
	<06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>
	<5a8712120908111249t3e23ceeese1c18d6fef52a227@mail.gmail.com>
Message-ID: <87f2f3b90908111254pf443a35qa306f7da93fb336a@mail.gmail.com>

2009/8/11 Jo?o Mesquita <jmesquita at gmail.com>

> Mike, the gender thing will eventually have to change code, I guess. I have
> not yet looked at the say code, so I am just imagining here.
>

Are there gender differences between dialects of the same language?
-MC
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From dave at 3c.co.uk  Tue Aug 11 12:55:12 2009
From: dave at 3c.co.uk (David Knell)
Date: Tue, 11 Aug 2009 22:55:12 +0300
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
Message-ID: <1250020512.4659.16.camel@dk-d820>

Just to add my $0.02-worth (if you're feeling generous..) - I don't
think that the dialplan is expressive enough to do what's needed here,
and that's where the trouble's coming from.  It's not enormously tricky
to build a generic "dial this set of numbers according to these rules"
service using something hanging off the event socket - there's a writeup
here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
be done on SoftIVR.

To roll something similar yourself using the event socket, you'd need to
map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
and have some way of passing messages around between the threads
handling the different call legs, assuming that you're using one thread
per leg.

--Dave



From brian at freeswitch.org  Tue Aug 11 12:55:51 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 14:55:51 -0500
Subject: [Freeswitch-users] Ivr and variables
In-Reply-To: <87f2f3b90908111254i4479d3f9m68ab8cce6a62554e@mail.gmail.com>
References: <8D85DC2E55A74F41AE2190E05A44F028@voztovoice>
	<87f2f3b90908111254i4479d3f9m68ab8cce6a62554e@mail.gmail.com>
Message-ID: <3332EB9D-AFA8-4D51-835D-9C6D812237C8@freeswitch.org>

What he is wanting is not possible.

/b

On Aug 11, 2009, at 2:54 PM, Michael Collins wrote:

> What value is in the variable ${some} ?
> -MC



From jmesquita at gmail.com  Tue Aug 11 12:56:28 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 11 Aug 2009 16:56:28 -0300
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <87f2f3b90908111254pf443a35qa306f7da93fb336a@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
	<4A81C59A.2020705@chandlerfamily.org.uk>
	<06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>
	<5a8712120908111249t3e23ceeese1c18d6fef52a227@mail.gmail.com>
	<87f2f3b90908111254pf443a35qa306f7da93fb336a@mail.gmail.com>
Message-ID: <5a8712120908111256v4b072587r4c5ead9fa4d4afb3@mail.gmail.com>

Oops, I thought you were saying different languages. Sorry about that.

jmesquita

On Tue, Aug 11, 2009 at 4:54 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> 2009/8/11 Jo?o Mesquita <jmesquita at gmail.com>
>
>> Mike, the gender thing will eventually have to change code, I guess. I
>> have not yet looked at the say code, so I am just imagining here.
>>
>
> Are there gender differences between dialects of the same language?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Tue Aug 11 12:59:47 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 11 Aug 2009 15:59:47 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
Message-ID: <86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>

It resolves fine for me already.

On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola <diego.viola at gmail.com> wrote:

> The address for me is: 69.174.57.101
>
>
> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera <godson.g at gmail.com> wrote:
>
>> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
>> name resolution. But the problem is still there even if I use my ISP's  DNS
>> (mostly because its resolving to the same to IP address) .
>>
>> On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> As MikeJ pointed out please report the IP address files.freeswitch.orgresolves to.  We have that on a content delivery network so that the files
>>> are closer to you geographically and you can download them faster but if
>>> you're having an issue I'll need the IP so I can report it correctly.
>>> Thanks,
>>> Brian
>>>
>>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>>
>>> Hi FS Team,
>>>
>>>
>>>         The files.freeswitch.org is resetting connection since 3 days.
>>> As a result I was not able to download latest release of FS. Got the trunk
>>> version from svn. But still it suffers from the lack of sound files.  When
>>> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>>> connection immediately wget tries 20 times and gives up. Other users on IRC
>>> also reported this issue.
>>>
>>> --
>>> Thanks & Regards,
>>> Godson Gera
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Thanks & Regards,
>> Godson Gera
>> http://godson.in
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From brian at freeswitch.org  Tue Aug 11 13:01:28 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 15:01:28 -0500
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
Message-ID: <8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>

Does it still give you connection refused?

/b

On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:

> It resolves fine for me already.
>
> On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> The address for me is: 69.174.57.101
>
>
> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera <godson.g at gmail.com>  
> wrote:
> Ok here is the address 69.174.57.101 . Most of the time I use  
> OpenDNS for name resolution. But the problem is still there even if  
> I use my ISP's  DNS (mostly because its resolving to the same to IP  
> address) .
>
> On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org>  
> wrote:
> As MikeJ pointed out please report the IP address  
> files.freeswitch.org resolves to.  We have that on a content  
> delivery network so that the files are closer to you geographically  
> and you can download them faster but if you're having an issue I'll  
> need the IP so I can report it correctly.
>
> Thanks,
> Brian
>
> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>
>> Hi FS Team,
>>
>>
>>         The files.freeswitch.org is resetting connection since 3  
>> days. As a result I was not able to download latest release of FS.  
>> Got the trunk version from svn. But still it suffers from the lack  
>> of sound files.  When ever I do 'make uhd-sounds-install'http://files.freeswitch.org 
>>  resets connection immediately wget tries 20 times and gives up.  
>> Other users on IRC also reported this issue.
>>
>> -- 
>> Thanks & Regards,
>> Godson Gera
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Thanks & Regards,
> Godson Gera
> http://godson.in
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From diego.viola at gmail.com  Tue Aug 11 13:03:28 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 11 Aug 2009 16:03:28 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
	<8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
Message-ID: <86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>

Nope.

On Tue, Aug 11, 2009 at 4:01 PM, Brian West <brian at freeswitch.org> wrote:

> Does it still give you connection refused?
> /b
>
> On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:
>
> It resolves fine for me already.
>
> On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> The address for me is: 69.174.57.101
>>
>>
>> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera <godson.g at gmail.com> wrote:
>>
>>> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
>>> name resolution. But the problem is still there even if I use my ISP's  DNS
>>> (mostly because its resolving to the same to IP address) .
>>>
>>> On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org>wrote:
>>>
>>>> As MikeJ pointed out please report the IP address files.freeswitch.orgresolves to.  We have that on a content delivery network so that the files
>>>> are closer to you geographically and you can download them faster but if
>>>> you're having an issue I'll need the IP so I can report it correctly.
>>>> Thanks,
>>>> Brian
>>>>
>>>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>>>
>>>> Hi FS Team,
>>>>
>>>>
>>>>         The files.freeswitch.org is resetting connection since 3 days.
>>>> As a result I was not able to download latest release of FS. Got the trunk
>>>> version from svn. But still it suffers from the lack of sound files.  When
>>>> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>>>> connection immediately wget tries 20 times and gives up. Other users on IRC
>>>> also reported this issue.
>>>>
>>>> --
>>>> Thanks & Regards,
>>>> Godson Gera
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Thanks & Regards,
>>> Godson Gera
>>> http://godson.in
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From boonedox at gmail.com  Tue Aug 11 13:13:40 2009
From: boonedox at gmail.com (Jeremiah Johnson)
Date: Tue, 11 Aug 2009 14:13:40 -0600
Subject: [Freeswitch-users] hangup_after_bridge=false only works if
	bypass_media=false
In-Reply-To: <367751820908111150s468a1717l20cf409ef8b7f66b@mail.gmail.com>
References: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
	<367751820908111150s468a1717l20cf409ef8b7f66b@mail.gmail.com>
Message-ID: <c626f0a50908111313q2844f4abn2b988c699325fc2e@mail.gmail.com>

I guess that makes sense, but that's not the answer I wanted to hear.  I
don't necessarily know when the "B" leg is going to hangup so I may not be
able to do the re-invite beforehand.  Is there some sort of workaround?

On Tue, Aug 11, 2009 at 12:50 PM, Phillip Jones <pjintheusa at gmail.com>wrote:

> Hi there,
>
> I the 'does not work' example your media stream is back up at the
> carrier as the called party hangs up.
>
> You would have to cause another SIP REINVITE and re capture the media.
>
> See http://wiki.freeswitch.org/wiki/Bypass_Media and the "How to
> disable/enable it on the fly?" section as a start.
>
>
>
>
> On Tue, Aug 11, 2009 at 2:29 PM, Jeremiah Johnson<boonedox at gmail.com>
> wrote:
> > This is an integral part of my application.  I need to have FreeSWITCH
> > outside of the media path as well as be able to do multiple bridges for
> the
> > same "A" leg.
> > /*WORKS*/
> > <action application="set" data="hangup_after_bridge=false"/>
> > <action application="bridge"
> > data="sofia/gateway/${mygateway}/1${client_one}"/>
> > <action application="bridge"
> > data="sofia/gateway/${mygateway}/1${client_two}"/>
> > /*DOES NOT WORK*/
> > <action application="set" data="hangup_after_bridge=false"/>
> > <action application="set" data="bypass_media=true"/>
> > <action application="bridge"
> > data="sofia/gateway/${mygateway}/1${client_one}"/>
> > <action application="bridge"
> > data="sofia/gateway/${mygateway}/1${client_two}"/>
> > In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the leg
> for
> > client_one hangs up.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From boonedox at gmail.com  Tue Aug 11 13:17:47 2009
From: boonedox at gmail.com (Jeremiah Johnson)
Date: Tue, 11 Aug 2009 14:17:47 -0600
Subject: [Freeswitch-users] hangup_after_bridge=false only works if
	bypass_media=false
In-Reply-To: <c626f0a50908111313q2844f4abn2b988c699325fc2e@mail.gmail.com>
References: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
	<367751820908111150s468a1717l20cf409ef8b7f66b@mail.gmail.com>
	<c626f0a50908111313q2844f4abn2b988c699325fc2e@mail.gmail.com>
Message-ID: <c626f0a50908111317v327d4db1sb609eef4ba568da6@mail.gmail.com>

Also, if I understand bypass_media, the SIP signaling is going through
FreeSWITCH and the audio is endpoint to endpoint, so I would think the
manner in which the "B" leg hangup is handled is controlled by FreeSWITCH.
 That is what appears to be happening in the debug.

On Tue, Aug 11, 2009 at 2:13 PM, Jeremiah Johnson <boonedox at gmail.com>wrote:

> I guess that makes sense, but that's not the answer I wanted to hear.  I
> don't necessarily know when the "B" leg is going to hangup so I may not be
> able to do the re-invite beforehand.  Is there some sort of workaround?
>
>
> On Tue, Aug 11, 2009 at 12:50 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> Hi there,
>>
>> I the 'does not work' example your media stream is back up at the
>> carrier as the called party hangs up.
>>
>> You would have to cause another SIP REINVITE and re capture the media.
>>
>> See http://wiki.freeswitch.org/wiki/Bypass_Media and the "How to
>> disable/enable it on the fly?" section as a start.
>>
>>
>>
>>
>> On Tue, Aug 11, 2009 at 2:29 PM, Jeremiah Johnson<boonedox at gmail.com>
>> wrote:
>> > This is an integral part of my application.  I need to have FreeSWITCH
>> > outside of the media path as well as be able to do multiple bridges for
>> the
>> > same "A" leg.
>> > /*WORKS*/
>> > <action application="set" data="hangup_after_bridge=false"/>
>> > <action application="bridge"
>> > data="sofia/gateway/${mygateway}/1${client_one}"/>
>> > <action application="bridge"
>> > data="sofia/gateway/${mygateway}/1${client_two}"/>
>> > /*DOES NOT WORK*/
>> > <action application="set" data="hangup_after_bridge=false"/>
>> > <action application="set" data="bypass_media=true"/>
>> > <action application="bridge"
>> > data="sofia/gateway/${mygateway}/1${client_one}"/>
>> > <action application="bridge"
>> > data="sofia/gateway/${mygateway}/1${client_two}"/>
>> > In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the leg
>> for
>> > client_one hangs up.
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From gshfreesw at gmail.com  Tue Aug 11 13:27:55 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Tue, 11 Aug 2009 16:27:55 -0400
Subject: [Freeswitch-users] Call transfer issues
In-Reply-To: <55901B44-E104-4829-8B40-81B36A3522CA@freeswitch.org>
References: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>
	<D554D137-805E-4924-A43B-3930275451BE@freeswitch.org>
	<5070fcbd0908111233w6ecefeeex92b1304f04744f23@mail.gmail.com>
	<55901B44-E104-4829-8B40-81B36A3522CA@freeswitch.org>
Message-ID: <5070fcbd0908111327i641ebdd6pe6dc389c7707624f@mail.gmail.com>

OK. I am little confused here. You say I can "transfer" to a parked call
using "uuid_transfer" . I do not see any options there where would or how
would I specify parked call in the options.

Can I do:

uuid_transfer <uuid> -both "park"

Where "park" is the destination?


On Tue, Aug 11, 2009 at 3:40 PM, Brian West <brian at freeswitch.org> wrote:

>
> On Aug 11, 2009, at 2:33 PM, Shameem Shiek wrote:
>
> > Thanks for the explanation. I was trying to find the correct way to
> > do this as I saw several ways of doing this. I will try to explain
> > how I did on the asterisk land.
> >
> > In asterisk land, Caller A will be parked at extension 701 and
> > Caller B "transfers" the call to extension 701 and and Caller A and
> > C are connected and Caller B is hung up right after transfer.
> >
> > In Freeswitch, Can Caller B do a "transfer" and connect caller A and
> > C? How can I do this? Since Freeswitch does not have a concept of
> > parked extensions.
>
> This is where you're wrong... you can park people with the park
> application just transfer them to it... But you have to take off the
> asterisk hat.. I told you in the last email how to do it correctly
> using nothing but API commands.
>
> /b
>
>
> >
> > Thank you for the help.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Aug 11 13:34:50 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 15:34:50 -0500
Subject: [Freeswitch-users] Call transfer issues
In-Reply-To: <5070fcbd0908111327i641ebdd6pe6dc389c7707624f@mail.gmail.com>
References: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>
	<D554D137-805E-4924-A43B-3930275451BE@freeswitch.org>
	<5070fcbd0908111233w6ecefeeex92b1304f04744f23@mail.gmail.com>
	<55901B44-E104-4829-8B40-81B36A3522CA@freeswitch.org>
	<5070fcbd0908111327i641ebdd6pe6dc389c7707624f@mail.gmail.com>
Message-ID: <5F3CDC9A-6E29-4EA3-8C8B-BB821A2D6E25@freeswitch.org>

You also have uuid_park... but uuid_transfer to the dialplan were the  
extension calls the park application.

/b

On Aug 11, 2009, at 3:27 PM, Shameem Shiek wrote:

> OK. I am little confused here. You say I can "transfer" to a parked  
> call using "uuid_transfer" . I do not see any options there where  
> would or how would I specify parked call in the options.
>
> Can I do:
>
> uuid_transfer <uuid> -both "park"
>
> Where "park" is the destination?



From mike at jerris.com  Tue Aug 11 13:45:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 Aug 2009 16:45:51 -0400
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <5a8712120908111256v4b072587r4c5ead9fa4d4afb3@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
	<4A81C59A.2020705@chandlerfamily.org.uk>
	<06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>
	<5a8712120908111249t3e23ceeese1c18d6fef52a227@mail.gmail.com>
	<87f2f3b90908111254pf443a35qa306f7da93fb336a@mail.gmail.com>
	<5a8712120908111256v4b072587r4c5ead9fa4d4afb3@mail.gmail.com>
Message-ID: <3D7EF5AC-2204-4B74-87D2-11554E8D27C3@jerris.com>

We have a plan to address this already, I can't recall if we added the  
gender types in code yet.

Mike

On Aug 11, 2009, at 3:56 PM, Jo?o Mesquita wrote:

> Oops, I thought you were saying different languages. Sorry about that.
>
> jmesquita
>
> On Tue, Aug 11, 2009 at 4:54 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> 2009/8/11 Jo?o Mesquita <jmesquita at gmail.com>
>
> Mike, the gender thing will eventually have to change code, I guess.  
> I have not yet looked at the say code, so I am just imagining here.
>
> Are there gender differences between dialects of the same language?
> -MC
>

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From asannucci at gmail.com  Tue Aug 11 14:00:14 2009
From: asannucci at gmail.com (bakko)
Date: Tue, 11 Aug 2009 23:00:14 +0200
Subject: [Freeswitch-users] Ivr and variables
In-Reply-To: <87f2f3b90908111254i4479d3f9m68ab8cce6a62554e@mail.gmail.com>
References: <8D85DC2E55A74F41AE2190E05A44F028@voztovoice>
	<87f2f3b90908111254i4479d3f9m68ab8cce6a62554e@mail.gmail.com>
Message-ID: <1C081CD9FB8642238779D6BCC8D0CCF8@voztovoice>

The value of  ${some} is 1234.

Thank you.

BR

> What value is in the variable ${some} ?
> -MC
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From kevin at kgolding.co.uk  Tue Aug 11 14:21:35 2009
From: kevin at kgolding.co.uk (Kevin Golding)
Date: Tue, 11 Aug 2009 22:21:35 +0100
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <0C9B302E-C1A6-4D1F-831B-63FA2AEB8C37@freeswitch.org>
References: <4A80AEE2.7050306@kgolding.co.uk>	<7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>	<4A811FEC.7040808@kgolding.co.uk>
	<0C9B302E-C1A6-4D1F-831B-63FA2AEB8C37@freeswitch.org>
Message-ID: <4A81E0DF.3060609@kgolding.co.uk>

Sorry, but could you give me a pointer on what this involves or where to 
read up on it please?

Brian West wrote:
> you have to now apply the ACL correctly to the sofia profile.
> 
> /b
> 
> On Aug 11, 2009, at 2:38 AM, Kevin Golding wrote:
> 
>> Thanks Brian,
>>
>> Well spotted with the domain/cidr :)
>>
>> I changed the line <param name="apply-inbound-acl" value="domains"/>  
>> in
>> the internal.xml with a new value, and changed the list to match but I
>> still get the same 'Rejected by acl "domains"' error.
>>
>> And yes I reloaded the xml. :)
>>
>> Kevin
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From brian at freeswitch.org  Tue Aug 11 14:25:36 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 16:25:36 -0500
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <4A81E0DF.3060609@kgolding.co.uk>
References: <4A80AEE2.7050306@kgolding.co.uk>	<7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>	<4A811FEC.7040808@kgolding.co.uk>
	<0C9B302E-C1A6-4D1F-831B-63FA2AEB8C37@freeswitch.org>
	<4A81E0DF.3060609@kgolding.co.uk>
Message-ID: <09C4F1BE-A4F3-466F-A102-5F1F490201CD@freeswitch.org>

I would guess in conf/sip_profiles/internal.xml where it applies the  
domains ACL to the profile... you can change that or add additional  
lines to apply more ACL's to the profile.

/b

On Aug 11, 2009, at 4:21 PM, Kevin Golding wrote:

> Sorry, but could you give me a pointer on what this involves or  
> where to
> read up on it please?
>
> Brian West wrote:
>> you have to now apply the ACL correctly to the sofia profile.
>>
>> /b
>



From msc at freeswitch.org  Tue Aug 11 14:30:48 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 16:30:48 -0500
Subject: [Freeswitch-users] Ivr and variables
In-Reply-To: <1C081CD9FB8642238779D6BCC8D0CCF8@voztovoice>
References: <8D85DC2E55A74F41AE2190E05A44F028@voztovoice>
	<87f2f3b90908111254i4479d3f9m68ab8cce6a62554e@mail.gmail.com>
	<1C081CD9FB8642238779D6BCC8D0CCF8@voztovoice>
Message-ID: <87f2f3b90908111430n2587cceeid8fa10e16a9bb63c@mail.gmail.com>

As Brian mentioned earlier, what you are trying to do is not possible in the
ivr.conf.xml file. Variables are not interpolated in the digits param. If
you need this dynamic functionality you will need to investigate
mod_xml_curl which allows you to create dynamic dialplans, etc.

-MC

On Tue, Aug 11, 2009 at 4:00 PM, bakko <asannucci at gmail.com> wrote:

>  The value of  ${some} is 1234.
>
> Thank you.
>
> BR
>
> > What value is in the variable ${some} ?
> > -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kevin at kgolding.co.uk  Tue Aug 11 14:58:16 2009
From: kevin at kgolding.co.uk (Kevin Golding)
Date: Tue, 11 Aug 2009 22:58:16 +0100
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <09C4F1BE-A4F3-466F-A102-5F1F490201CD@freeswitch.org>
References: <4A80AEE2.7050306@kgolding.co.uk>	<7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>	<4A811FEC.7040808@kgolding.co.uk>	<0C9B302E-C1A6-4D1F-831B-63FA2AEB8C37@freeswitch.org>	<4A81E0DF.3060609@kgolding.co.uk>
	<09C4F1BE-A4F3-466F-A102-5F1F490201CD@freeswitch.org>
Message-ID: <4A81E978.1000107@kgolding.co.uk>

Thank you Brian. Have got the ACL issue sorted with your help - thank 
you. See new thread for my next problem :)

kevin

Brian West wrote:
> I would guess in conf/sip_profiles/internal.xml where it applies the  
> domains ACL to the profile... you can change that or add additional  
> lines to apply more ACL's to the profile.
> 
> /b
> 
> On Aug 11, 2009, at 4:21 PM, Kevin Golding wrote:
> 
>> Sorry, but could you give me a pointer on what this involves or  
>> where to
>> read up on it please?
>>
>> Brian West wrote:
>>> you have to now apply the ACL correctly to the sofia profile.
>>>
>>> /b
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From mgg at giagnocavo.net  Tue Aug 11 15:13:05 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 11 Aug 2009 18:13:05 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <1250020512.4659.16.camel@dk-d820>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>

It's also simple enough to write a plugin in one of the scripting languages to add an app to do exactly what you want...

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Knell
Sent: Tuesday, August 11, 2009 1:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Loopback and bypass_media

Just to add my $0.02-worth (if you're feeling generous..) - I don't
think that the dialplan is expressive enough to do what's needed here,
and that's where the trouble's coming from.  It's not enormously tricky
to build a generic "dial this set of numbers according to these rules"
service using something hanging off the event socket - there's a writeup
here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
be done on SoftIVR.

To roll something similar yourself using the event socket, you'd need to
map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
and have some way of passing messages around between the threads
handling the different call legs, assuming that you're using one thread
per leg.

--Dave


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From diego.viola at gmail.com  Tue Aug 11 21:56:29 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 12 Aug 2009 00:56:29 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
	<8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
	<86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>
Message-ID: <86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>

Resolving files.freeswitch.org... failed: Temporary failure in name
resolution.

Again...

On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Nope.
>
>
> On Tue, Aug 11, 2009 at 4:01 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Does it still give you connection refused?
>> /b
>>
>> On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:
>>
>> It resolves fine for me already.
>>
>> On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> The address for me is: 69.174.57.101
>>>
>>>
>>> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera <godson.g at gmail.com> wrote:
>>>
>>>> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS
>>>> for name resolution. But the problem is still there even if I use my ISP's
>>>> DNS (mostly because its resolving to the same to IP address) .
>>>>
>>>> On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org>wrote:
>>>>
>>>>> As MikeJ pointed out please report the IP address files.freeswitch.orgresolves to.  We have that on a content delivery network so that the files
>>>>> are closer to you geographically and you can download them faster but if
>>>>> you're having an issue I'll need the IP so I can report it correctly.
>>>>> Thanks,
>>>>> Brian
>>>>>
>>>>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>>>>
>>>>> Hi FS Team,
>>>>>
>>>>>
>>>>>         The files.freeswitch.org is resetting connection since 3 days.
>>>>> As a result I was not able to download latest release of FS. Got the trunk
>>>>> version from svn. But still it suffers from the lack of sound files.  When
>>>>> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>>>>> connection immediately wget tries 20 times and gives up. Other users on IRC
>>>>> also reported this issue.
>>>>>
>>>>> --
>>>>> Thanks & Regards,
>>>>> Godson Gera
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Thanks & Regards,
>>>> Godson Gera
>>>> http://godson.in
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From diego.viola at gmail.com  Tue Aug 11 22:01:12 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 12 Aug 2009 01:01:12 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
	<8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
	<86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>
	<86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
Message-ID: <86a32abc0908112201n1e505d33w74d46a5570104500@mail.gmail.com>

http://files-sync.freeswitch.org/ works fine.

On Wed, Aug 12, 2009 at 12:56 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Resolving files.freeswitch.org... failed: Temporary failure in name
> resolution.
>
> Again...
>
>
> On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Nope.
>>
>>
>> On Tue, Aug 11, 2009 at 4:01 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Does it still give you connection refused?
>>> /b
>>>
>>> On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:
>>>
>>> It resolves fine for me already.
>>>
>>> On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> The address for me is: 69.174.57.101
>>>>
>>>>
>>>> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera <godson.g at gmail.com>wrote:
>>>>
>>>>> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS
>>>>> for name resolution. But the problem is still there even if I use my ISP's
>>>>> DNS (mostly because its resolving to the same to IP address) .
>>>>>
>>>>> On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org>wrote:
>>>>>
>>>>>> As MikeJ pointed out please report the IP address
>>>>>> files.freeswitch.org resolves to.  We have that on a content delivery
>>>>>> network so that the files are closer to you geographically and you can
>>>>>> download them faster but if you're having an issue I'll need the IP so I can
>>>>>> report it correctly.
>>>>>> Thanks,
>>>>>> Brian
>>>>>>
>>>>>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>>>>>
>>>>>> Hi FS Team,
>>>>>>
>>>>>>
>>>>>>         The files.freeswitch.org is resetting connection since 3
>>>>>> days. As a result I was not able to download latest release of FS. Got the
>>>>>> trunk version from svn. But still it suffers from the lack of sound files.
>>>>>> When ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>>>>>> connection immediately wget tries 20 times and gives up. Other users on IRC
>>>>>> also reported this issue.
>>>>>>
>>>>>> --
>>>>>> Thanks & Regards,
>>>>>> Godson Gera
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Thanks & Regards,
>>>>> Godson Gera
>>>>> http://godson.in
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From jason at jasonjgw.net  Tue Aug 11 22:30:18 2009
From: jason at jasonjgw.net (Jason White)
Date: Wed, 12 Aug 2009 15:30:18 +1000
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
	<8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
	<86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>
	<86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
Message-ID: <20090812053018.GA3821@jdc.jasonjgw.net>

Diego Viola <diego.viola at gmail.com> wrote:
> Resolving files.freeswitch.org... failed: Temporary failure in name
> resolution.

It must be a problem at your end.

jason at jdc:~$ host files.freeswitch.org
files.freeswitch.org is an alias for filessync.freeswitch.netdna-cdn.com.
filessync.freeswitch.netdna-cdn.com has address 69.174.57.101

Note that I am running my own Bind 9 daemon on this host. The record wasn't
cached, as it has been a few weeks since I last upgraded FreeSWITCH.



From brian at freeswitch.org  Tue Aug 11 22:57:38 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 00:57:38 -0500
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
	<8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
	<86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>
	<86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
Message-ID: <2A5563F4-3197-42B0-851B-ABAB804A08F9@freeswitch.org>

I would have to say its YOUR system and not ours.

/b

On Aug 11, 2009, at 11:56 PM, Diego Viola wrote:

> Resolving files.freeswitch.org... failed: Temporary failure in name  
> resolution.
>
> Again...
>
> On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> Nope.
>

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From plite2012 at gmail.com  Tue Aug 11 22:58:41 2009
From: plite2012 at gmail.com (Paul Li)
Date: Wed, 12 Aug 2009 00:58:41 -0500
Subject: [Freeswitch-users] How to delay IVR answer during an outbound call
Message-ID: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com>

I have a dummy question. Say, you have an outbound call to the demo
IVR as below:

originate sofia/gateway/myvoip/19876543210 5000

How do I delay the IVR response until the recipient at 19876543210
picks up the call? I tried "ignore_early_media=true", which had no
effect.

Many thanks in advance.


From brian at freeswitch.org  Tue Aug 11 23:01:58 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 01:01:58 -0500
Subject: [Freeswitch-users] How to delay IVR answer during an outbound
	call
In-Reply-To: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com>
References: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com>
Message-ID: <11E6F5A9-D997-473E-8880-0E5AC996D3C4@freeswitch.org>

Is your provider answering the call before its connected?  If so then  
they should be shot.  I can't imagine other way the call would be  
answered unless you're using ignore_early_media wrong.... can you show  
me who you're doing this?

/b

On Aug 12, 2009, at 12:58 AM, Paul Li wrote:

> I have a dummy question. Say, you have an outbound call to the demo
> IVR as below:
>
> originate sofia/gateway/myvoip/19876543210 5000
>
> How do I delay the IVR response until the recipient at 19876543210
> picks up the call? I tried "ignore_early_media=true", which had no
> effect.
>
> Many thanks in advance.



From plite2012 at gmail.com  Tue Aug 11 23:23:33 2009
From: plite2012 at gmail.com (Paul Li)
Date: Wed, 12 Aug 2009 01:23:33 -0500
Subject: [Freeswitch-users] How to delay IVR answer during an outbound call
In-Reply-To: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com>
References: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com>
Message-ID: <c43084ae0908112323m1c8d5856kc05acc6664b635cd@mail.gmail.com>

I am actually doing a lua script for IVR as follows

-- answer the call
session:answer();

while session:ready() == true do
	-- sleep a second
	session:sleep(1000);

	-- play a file
	session:streamFile("/path/to/blah.wav");
	
	-- hangup
	session:hangup();
end

The problem lies in: when I picked up my phone, blah.wav was already
played for a while, instead of from the beginning.

I shall greatly appreciate any input.

On Wed, Aug 12, 2009 at 12:58 AM, Paul Li<plite2012 at gmail.com> wrote:
> I have a dummy question. Say, you have an outbound call to the demo
> IVR as below:
>
> originate sofia/gateway/myvoip/19876543210 5000
>
> How do I delay the IVR response until the recipient at 19876543210
> picks up the call? I tried "ignore_early_media=true", which had no
> effect.
>
> Many thanks in advance.
>


From tzury.by at reguluslabs.com  Tue Aug 11 23:44:49 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Wed, 12 Aug 2009 09:44:49 +0300
Subject: [Freeswitch-users] problem when adding more extension
Message-ID: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>

Hi,

I wanted to add more extension to freeswitch.
to add extension 1050 with password 1234 I did the following:

$ cd /usr/local/freeswitch/conf/directory/default

created 1050.xml having all '1000' strings replaced by '1050' by typing
$ sed s/1000/1050/g < 1000.xml > 1050.xml

rescan and reload the xml by typing into the CLI

freeswitch at internal> sofia profile internal rescan reloadxml

However, when I tried to login with these credentials I got the
following in the fs_cli:

2009-08-12 06:35:01 [DEBUG] sofia_reg.c:1469 sofia_reg_parse_auth()
SIP username 1050 does not match auth username
2009-08-12 06:35:01 [DEBUG] sofia_reg.c:869
sofia_reg_handle_register() Send challenge for
[1050 at server_address.net]

below are the content of 1000 and 1050 xml files

please advise.


$ cat 1050.xml
<include>
 <user id="1050" mailbox="1050">
   <params>
     <param name="password" value="1234"/>
     <param name="vm-password" value="1050"/>
   </params>
   <variables>
     <variable name="toll_allow" value="domestic,international,local"/>
     <variable name="accountcode" value="1050"/>
     <variable name="user_context" value="default"/>
     <variable name="effective_caller_id_name" value="Extension 1050"/>
     <variable name="effective_caller_id_number" value="1050"/>
     <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
     <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
     <variable name="callgroup" value="techsupport"/>
   </variables>
 </user>
</include>

$ cat 1000.xml
<include>
 <user id="1000" mailbox="1000">
   <params>
     <param name="password" value="1234"/>
     <param name="vm-password" value="1000"/>
   </params>
   <variables>
     <variable name="toll_allow" value="domestic,international,local"/>
     <variable name="accountcode" value="1000"/>
     <variable name="user_context" value="default"/>
     <variable name="effective_caller_id_name" value="Extension 1000"/>
     <variable name="effective_caller_id_number" value="1000"/>
     <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
     <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
     <variable name="callgroup" value="techsupport"/>
   </variables>
 </user>
</include>


From diego.viola at gmail.com  Wed Aug 12 00:19:04 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 12 Aug 2009 03:19:04 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <2A5563F4-3197-42B0-851B-ABAB804A08F9@freeswitch.org>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
	<8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
	<86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>
	<86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
	<2A5563F4-3197-42B0-851B-ABAB804A08F9@freeswitch.org>
Message-ID: <86a32abc0908120019k5a78eb8fi603d1bf4bd082a9@mail.gmail.com>

Aww, ok.

Bad luck to me :).

On Wed, Aug 12, 2009 at 1:57 AM, Brian West <brian at freeswitch.org> wrote:

> I would have to say its YOUR system and not ours.
> /b
>
> On Aug 11, 2009, at 11:56 PM, Diego Viola wrote:
>
> Resolving files.freeswitch.org... failed: Temporary failure in name
> resolution.
>
> Again...
>
> On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola <diego.viola at gmail.com>
> wrote:
>
>> Nope.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From samu60 at gmail.com  Wed Aug 12 00:25:25 2009
From: samu60 at gmail.com (samuel)
Date: Wed, 12 Aug 2009 09:25:25 +0200
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <87f2f3b90908111254pf443a35qa306f7da93fb336a@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
	<4A81C59A.2020705@chandlerfamily.org.uk>
	<06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>
	<5a8712120908111249t3e23ceeese1c18d6fef52a227@mail.gmail.com>
	<87f2f3b90908111254pf443a35qa306f7da93fb336a@mail.gmail.com>
Message-ID: <d18bd3a10908120025y203a887axb0567d7e122d8207@mail.gmail.com>

I would say there are no changes in gender for dialects...but with so many
languages around I can't assure it 100% ;)

Samuel.

2009/8/11 Michael Collins <msc at freeswitch.org>

>
>
> 2009/8/11 Jo?o Mesquita <jmesquita at gmail.com>
>
>> Mike, the gender thing will eventually have to change code, I guess. I
>> have not yet looked at the say code, so I am just imagining here.
>>
>
> Are there gender differences between dialects of the same language?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Aug 11 20:43:09 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 Aug 2009 23:43:09 -0400
Subject: [Freeswitch-users] hangup_after_bridge=false only works if
	bypass_media=false
In-Reply-To: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
References: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
Message-ID: <C5AD471E-636F-427A-A0A9-9F894D799CEB@jerris.com>

Please post a bug for this on jira.freeswitch.org.

Mike

On Aug 11, 2009, at 2:29 PM, Jeremiah Johnson wrote:

> This is an integral part of my application.  I need to have  
> FreeSWITCH outside of the media path as well as be able to do  
> multiple bridges for the same "A" leg.
>
> /*WORKS*/
> <action application="set" data="hangup_after_bridge=false"/>
> <action application="bridge" data="sofia/gateway/${mygateway}/1$ 
> {client_one}"/>
> <action application="bridge" data="sofia/gateway/${mygateway}/1$ 
> {client_two}"/>
>
> /*DOES NOT WORK*/
> <action application="set" data="hangup_after_bridge=false"/>
> <action application="set" data="bypass_media=true"/>
> <action application="bridge" data="sofia/gateway/${mygateway}/1$ 
> {client_one}"/>
> <action application="bridge" data="sofia/gateway/${mygateway}/1$ 
> {client_two}"/>
>
> In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the  
> leg for client_one hangs up.



From charlieb at cot.net  Tue Aug 11 23:37:03 2009
From: charlieb at cot.net (Charles Boening)
Date: Tue, 11 Aug 2009 23:37:03 -0700
Subject: [Freeswitch-users] FS 1.0.4 official core dumping LUA streamFile
Message-ID: <4FB2938B89459C41860C4DB9B1821D6FB6657A2421@exchange.calore.local>

Greetings,

I have the following LUA script (at end of email) in a fresh FS 1.0.4 install.  I originally did an upgrade from one of the 1.0.4preX versions but when I came across this issue I went fresh just to make sure there wasn't an incompatibility with my previous config.

What I'm seeing is a seg fault and a core dump after playing a sound file.  I originally had a file I recorded but when I ran into this issue I figured I'd try an included sound file but that doesn't seem to make a bit of difference.

2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Starting test.lua
2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Caller [XXXXXXXXXX] connected
2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Pre streamFile
Segmentation fault (core dumped)


Any ideas?

Thanks,
Charlie



freeswitch.consoleLog("INFO", string.format("Starting test.lua\n"))
session:answer();
session:setHangupHook("session_hangup_hook")
calleridnumber = session:getVariable("caller_id_number")
calleridname = session:getVariable("caller_id_name")

if session:ready() then
   freeswitch.consoleLog("INFO", string.format("Caller [" .. calleridnumber .. "] connected\n"))
   freeswitch.consoleLog("INFO", string.format("Pre streamFile\n"))

   session:streamFile("conference/8000/conf-welcome.wav")

   freeswitch.consoleLog("INFO", string.format("Post streamFile.\n"))
end

function session_hangup_hook(status)
  freeswitch.consoleLog("INFO", "Session hangup: \n") --[[ .. status .. "\n") ]]--
  error()
end

session:hangup()



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From kirk.bateman at gmail.com  Wed Aug 12 01:34:16 2009
From: kirk.bateman at gmail.com (Kirk Bateman)
Date: Wed, 12 Aug 2009 09:34:16 +0100
Subject: [Freeswitch-users] VoiceMail transcription
In-Reply-To: <5a8712120908110930s74b84bf8x673ca1209fbae965@mail.gmail.com>
References: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
	<1249998014.20224.82.camel@dk-d820>
	<2bee4fc40908110717p43649678ue2b94ecb571bd09@mail.gmail.com>
	<5a8712120908110930s74b84bf8x673ca1209fbae965@mail.gmail.com>
Message-ID: <2bee4fc40908120134v5781a339ha7a31cc677b7f205@mail.gmail.com>

Not sure, but they do certainly have a reasonably large server farm for
doing processing :)

I note that sphinx4 I believe has a java example for doing dictation
transcription from an audio file (saw something on a sphinx forum or mailing
list while trawling the net).

I'm still investigating modifications to use pocketsphinx.

Regards

Kirk


2009/8/11 Jo?o Mesquita <jmesquita at gmail.com>

> I am sorry for the ignorance on the matter, but how does google voice does?
> Do they also have humans?
>
> jmesquita
>
>
> On Tue, Aug 11, 2009 at 11:17 AM, Kirk Bateman <kirk.bateman at gmail.com>wrote:
>
>> I'm still interested in getting pocketsphinx to attempt speech recognition
>> on an audio file.
>>
>> To be honest, most of the problem is that at 8Khz (mobile phone call
>> rate), speech detection is NOT very accurate, at 16Khz it IS significantly
>> better.
>>
>> I'm planning to have a play with the speechtools module and
>> mod_pocketsphinx etc to try and get an audio file parsed, spare time
>> permitting.
>>
>> Will let the list know if I get anywhere.
>>
>> Regards
>>
>> Kirk Bateman
>>
>>
>> 2009/8/11 David Knell <dave at 3c.co.uk>
>>
>> Hi Pete,
>>>
>>> I'm afraid that the answer's still the same: use a human.  Here's an
>>> article describing the state of the art:
>>> http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/
>>> - the links to previous stories at the bottom provide good background.
>>>
>>> --Dave
>>>
>>> > I apologize, I should have been more clear.  We will be using humans
>>> > to scan the translated results.  But we are looking for a system to
>>> > perform the "first pass" on the audio to hopefully help the human type
>>> > less.
>>> >
>>> > Although the question has been raised if it's faster to have a human
>>> > just transcribe the whole thing, or fix up what the computer spit out.
>>> > If you have any insights on this, that would be great.
>>> >
>>> > -pete
>>> >
>>> >         -------- Original Message --------
>>> >         Subject: Re: [Freeswitch-users] VoiceMail transcription
>>> >         From: David Knell <dave at 3c.co.uk>
>>> >         Date: Mon, August 10, 2009 11:51 am
>>> >         To: freeswitch-users at lists.freeswitch.org
>>> >
>>> >         Good evening Pete,
>>> >
>>> >         The only way to do this is, I'm afraid, to use a human. We use
>>> >         Amazon's
>>> >         Mechanical Turk to good effect.
>>> >
>>> >         Cheers --
>>> >
>>> >         Dave
>>> >
>>> >         > Good morning all,
>>> >         >
>>> >         > I realize this is slightly off the FS topic, but I am
>>> >         wondering if
>>> >         > anyone out there has experience with software packages
>>> >         designed for
>>> >         > the transcription of voicemails to text. I've used
>>> >         pocketsphinx with
>>> >         > FS to handle IVR menus, but now have the task of figuring
>>> >         out how to
>>> >         > convert recorded phone conversations (voicemails mostly) to
>>> >         text.
>>> >         >
>>> >         > This does not have to be a real-time process, I can store
>>> >         the audio
>>> >         > files and process them over time. This would need to be a
>>> >         software
>>> >         > (preferable open source) solution. ASPs like VoiceCloud
>>> >         would not
>>> >         > work for this application.
>>> >         >
>>> >         > Thanks for any help
>>> >         > -pete
>>> >         > _______________________________________________
>>> >         > FreeSWITCH-users mailing list
>>> >         > FreeSWITCH-users at lists.freeswitch.org
>>> >         >
>>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >         >
>>> >         UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >         > http://www.freeswitch.org
>>> >         --
>>> >         David Knell, Director, 3C Limited
>>> >         T: +44 20 3298 2000
>>> >         E: dave at 3c.co.uk
>>> >         W: http://www.3c.co.uk
>>> >
>>> >
>>> >         _______________________________________________
>>> >         FreeSWITCH-users mailing list
>>> >         FreeSWITCH-users at lists.freeswitch.org
>>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >         UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >         http://www.freeswitch.org
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> --
>>> David Knell, Director, 3C Limited
>>> T: +44 20 3298 2000
>>> E: dave at 3c.co.uk
>>> W: http://www.3c.co.uk
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bruce.mcalister at blueface.ie  Wed Aug 12 01:43:36 2009
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Wed, 12 Aug 2009 09:43:36 +0100
Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 &
	v1.0.4)
Message-ID: <4A8280B8.6050308@blueface.ie>

Hi All,

I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and 
1.0.4.

I am running on Solaris 10 Update 5 on x86 hardware (32-bit).

The build fails with:

--- snip ---
make: Fatal error: Command failed for target `all-recursive'
Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4
*** Error code 1
make: Fatal error: Command failed for target `all'
---

Looking back through the build I can see the following error:

--- snip ---
creating libfreeswitch.la
(cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la 
libfreeswitch.la)
/usr/bin/cc 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
-KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
-DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch 
freeswitch-switch.o  ./.libs/libfreeswitch.so 
-L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib 
/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib/.libs/libexpat.a 
/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libapr-1.a 
-lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp 
-L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt 
-lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl 
-lcurses -lsocket  -R/opt/freeswitch/lib -R/usr/sfw/lib
Undefined                       first referenced
  symbol                             in file
herror                              ./.libs/libfreeswitch.so
ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch
*** Error code 1
The following command caused the error:
`if test -z "" ; then echo /bin/bash 
/export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else 
echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` 
--tag=CC --mode=link /usr/bin/cc 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
-KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
-DPATH_MAX=2048 -g -v -Xc -xc99=all  -lm -R/opt/freeswitch/lib  -o 
freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib 
freeswitch-switch.o   libfreeswitch.la libs/apr/libapr-1.la 
libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl 
-lcrypto -lsocket -lnsl -ldl   -lcurses -lsocket
--- snip ---

Then a little above this error, there is the following warning that is 
displayed (I'm not sure if it is related):

--- snip ---
*** Warning: Linking the shared library libfreeswitch.la against the
*** static library libs/libedit/src/.libs/libedit.a is not portable!
--- snip ---

My configure line is as follows:

---
./configure --prefix=/opt/freeswitch
---

I have the complete configure and make output if anyone needs them.

Any help/pointers would be greatly appreciated.

Thanks
Bruce


From kevin at kgolding.co.uk  Wed Aug 12 01:47:20 2009
From: kevin at kgolding.co.uk (Kevin Golding)
Date: Wed, 12 Aug 2009 09:47:20 +0100
Subject: [Freeswitch-users] problem when adding more extension
In-Reply-To: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>
References: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>
Message-ID: <4A828198.6020000@kgolding.co.uk>

Hello,

I've just had the same problem. Solved it by adding the new extension to 
the default group.

i.e. In the /usr/local/freeswitch/conf/directory/default.xml file you 
need to add <user id="1050" type="pointer"> with one of the "group" 
blocks (e.g. after the line <group name="default">

Kevin

Tzury Bar Yochay wrote:
> Hi,
> 
> I wanted to add more extension to freeswitch.
> to add extension 1050 with password 1234 I did the following:
> 
> $ cd /usr/local/freeswitch/conf/directory/default
> 
> created 1050.xml having all '1000' strings replaced by '1050' by typing
> $ sed s/1000/1050/g < 1000.xml > 1050.xml
> 
> rescan and reload the xml by typing into the CLI
> 
> freeswitch at internal> sofia profile internal rescan reloadxml
> 
> However, when I tried to login with these credentials I got the
> following in the fs_cli:
> 
> 2009-08-12 06:35:01 [DEBUG] sofia_reg.c:1469 sofia_reg_parse_auth()
> SIP username 1050 does not match auth username
> 2009-08-12 06:35:01 [DEBUG] sofia_reg.c:869
> sofia_reg_handle_register() Send challenge for
> [1050 at server_address.net]
> 
> below are the content of 1000 and 1050 xml files
> 
> please advise.
> 
> 
> $ cat 1050.xml
> <include>
>  <user id="1050" mailbox="1050">
>    <params>
>      <param name="password" value="1234"/>
>      <param name="vm-password" value="1050"/>
>    </params>
>    <variables>
>      <variable name="toll_allow" value="domestic,international,local"/>
>      <variable name="accountcode" value="1050"/>
>      <variable name="user_context" value="default"/>
>      <variable name="effective_caller_id_name" value="Extension 1050"/>
>      <variable name="effective_caller_id_number" value="1050"/>
>      <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>      <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>      <variable name="callgroup" value="techsupport"/>
>    </variables>
>  </user>
> </include>
> 
> $ cat 1000.xml
> <include>
>  <user id="1000" mailbox="1000">
>    <params>
>      <param name="password" value="1234"/>
>      <param name="vm-password" value="1000"/>
>    </params>
>    <variables>
>      <variable name="toll_allow" value="domestic,international,local"/>
>      <variable name="accountcode" value="1000"/>
>      <variable name="user_context" value="default"/>
>      <variable name="effective_caller_id_name" value="Extension 1000"/>
>      <variable name="effective_caller_id_number" value="1000"/>
>      <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>      <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>      <variable name="callgroup" value="techsupport"/>
>    </variables>
>  </user>
> </include>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From maxim.tsvetov at gmail.com  Wed Aug 12 01:54:38 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Wed, 12 Aug 2009 01:54:38 -0700 (PDT)
Subject: [Freeswitch-users] answer command
In-Reply-To: <86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
Message-ID: <24931876.post@talk.nabble.com>


I've tried to use "answer" command from outbound event socket and it's
working, but 
the problem is that FS answering the call, but SIP Client (we tried this
with EyeBeam and CISCO 7960)
doesn't know that call was answered. So, as long as FS doesn't know what to
do with this number it then disconnects the call.

2009-08-12 11:25:07.599250 [NOTICE] mod_dptools.c:649 Channel
[sofia/internal/sip:1000 at 10.107.181.160:42840] has been answered
2009-08-12 11:25:07.599250 [NOTICE] switch_ivr_originate.c:2015 Channel
[sofia/internal/1003 at 10.107.249.12] has been answered
2009-08-12 11:25:07.614875 [ERR] switch_core_io.c:118
sofia/internal/sip:1000 at 10.107.181.160:42840 has no read codec.
2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:503 Hangup
sofia/internal/sip:1000 at 10.107.181.160:42840 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]
2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:1016 Hangup
sofia/internal/1003 at 10.107.249.12 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-12 11:25:07.630500 [NOTICE] switch_core_session.c:1086 Session 133
(sofia/internal/sip:1000 at 10.107.181.160:42840) Ended


Maybe there is the way to acknowledge SIP client that call was answered?

Regards,
Maxim Tsvetov

Diego Viola wrote:
> 
> I suggest that you learn the differences between mod_commands commands and
> mod_dptools applications, and also the interfaces where you can access and
> use them.
> 
> As said before, mod_dptools is accessible from dialplan, event socket
> outbound, etc. and mod_commands is accessible from the CLI, event socket
> (inbound/outbound), XML RPC, etc.
> 
> That's all described in the wiki I think.
> 
> Let us know if you have any questions =D.
> 
> On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola <diego.viola at gmail.com>
> wrote:
> 
>> Michael, you're welcome :).
>>
>> Milena, answer is a mod_dptools command, you can use it from the XML
>> dialplan or from the event socket outbound. mod_commands API are APIs
>> that
>> you execute from the socket, event socket inbound, etc. But you can also
>> execute them from event socket outbound using the "api" command.
>>
>> I hope that makes sense, correct me if I'm wrong =D.
>>
>> On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins
>> <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Tue, Aug 11, 2009 at 9:05 AM, Milena <testeador01 at gmail.com> wrote:
>>>
>>>>
>>>> Hello Brian,
>>>>
>>>> I wanna fix the wiki, but to make sure i got it right, does it only
>>>> work
>>>> on outbound event socket? or is there any other scenario where it would
>>>> work.
>>>>
>>>
>>> FYI, Diego Viola fixed the wiki. (Thanks Diego!)
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/answer-command-tp24912812p24931876.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From mike at jerris.com  Wed Aug 12 02:11:40 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 12 Aug 2009 05:11:40 -0400
Subject: [Freeswitch-users] FS 1.0.4 official core dumping LUA streamFile
In-Reply-To: <4FB2938B89459C41860C4DB9B1821D6FB6657A2421@exchange.calore.local>
References: <4FB2938B89459C41860C4DB9B1821D6FB6657A2421@exchange.calore.local>
Message-ID: <A9CD3A3C-923A-47FE-B9BC-DEAE701F3248@jerris.com>

If your seeing a segfault, please report it to jira.freeswitch.org  
with a backtrace and details of how to reproduce.

Mike

On Aug 12, 2009, at 2:37 AM, Charles Boening wrote:

> Greetings,
>
> I have the following LUA script (at end of email) in a fresh FS  
> 1.0.4 install.  I originally did an upgrade from one of the  
> 1.0.4preX versions but when I came across this issue I went fresh  
> just to make sure there wasn?t an incompatibility with my previous  
> config.
>
> What I?m seeing is a seg fault and a core dump after playing a sound  
> file.  I originally had a file I recorded but when I ran into this  
> issue I figured I?d try an included sound file but that doesn?t seem  
> to make a bit of difference.
>
> 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Starting  
> test.lua
> 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Caller  
> [XXXXXXXXXX] connected
> 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Pre streamFile
> Segmentation fault (core dumped)
>
>
> Any ideas?
>
> Thanks,
> Charlie
>
>
>
> freeswitch.consoleLog("INFO", string.format("Starting test.lua\n"))
> session:answer();
> session:setHangupHook("session_hangup_hook")
> calleridnumber = session:getVariable("caller_id_number")
> calleridname = session:getVariable("caller_id_name")
>
> if session:ready() then
>    freeswitch.consoleLog("INFO", string.format("Caller [" ..  
> calleridnumber .. "] connected\n"))
>    freeswitch.consoleLog("INFO", string.format("Pre streamFile\n"))
>
>    session:streamFile("conference/8000/conf-welcome.wav")
>
>    freeswitch.consoleLog("INFO", string.format("Post streamFile.\n"))
> end
>
> function session_hangup_hook(status)
>   freeswitch.consoleLog("INFO", "Session hangup: \n") --[[ ..  
> status .. "\n") ]]--
>   error()
> end
>
> session:hangup()
>
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From tzury.by at reguluslabs.com  Wed Aug 12 02:50:52 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Wed, 12 Aug 2009 12:50:52 +0300
Subject: [Freeswitch-users] problem when adding more extension
In-Reply-To: <4A828198.6020000@kgolding.co.uk>
References: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>
	<4A828198.6020000@kgolding.co.uk>
Message-ID: <10128ef10908120250y77917460g5c6b9cd1991fe074@mail.gmail.com>

still not working, I mean, I can initiate a call from 1060 to 1000 but
not from 1000 to 1060.
1060 is just an example. This applies to all new extension I have
added (beyond to the default 1000-1019).
as you can see below I added them all to <group name="support">

This is how the confs look like

/usr/local/freeswitch/conf/directory# cat default.xml
<include>
  <!--the domain or ip (the right hand side of the @ in the addr-->
  <domain name="$${domain}">
    <params>
      <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
    </params>

    <variables>
      <variable name="record_stereo" value="true"/>
      <variable name="default_gateway" value="$${default_provider}"/>
      <variable name="default_areacode" value="$${default_areacode}"/>
      <variable name="transfer_fallback_extension" value="operator"/>
    </variables>

    <groups>
      <group name="default">
	<users>
	  <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
	</users>
      </group>

      <group name="sales">
	<users>
	  <!--
	      type="pointer" is a pointer so you can have the
	      same user in multiple groups.  It basically means
	      to keep searching for the user in the directory.
	  -->
	  <user id="1000" type="pointer"/>
	  <user id="1001" type="pointer"/>
	  <user id="1002" type="pointer"/>
	  <user id="1003" type="pointer"/>
	  <user id="1004" type="pointer"/>
	</users>
      </group>

      <group name="billing">
	<users>
	  <user id="1005" type="pointer"/>
	  <user id="1006" type="pointer"/>
	  <user id="1007" type="pointer"/>
	  <user id="1008" type="pointer"/>
	  <user id="1009" type="pointer"/>
	</users>
      </group>

      <group name="support">
	<users>
	  <user id="1010" type="pointer"/>
	  <user id="1011" type="pointer"/>
	  <user id="1012" type="pointer"/>
	  <user id="1013" type="pointer"/>
          <user id="1014" type="pointer"/>
          <user id="1015" type="pointer"/>
          <user id="1016" type="pointer"/>
          <user id="1017" type="pointer"/>
          <user id="1018" type="pointer"/>
          <user id="1019" type="pointer"/>
          <user id="1020" type="pointer"/>
          <user id="1050" type="pointer"/>
          <user id="1051" type="pointer"/>
	  <user id="1052" type="pointer"/>
	  <user id="1053" type="pointer"/>
	  <user id="1054" type="pointer"/>
          <user id="1055" type="pointer"/>
          <user id="1056" type="pointer"/>
          <user id="1057" type="pointer"/>
          <user id="1058" type="pointer"/>
          <user id="1059" type="pointer"/>
          <user id="1060" type="pointer"/>
          <user id="1061" type="pointer"/>
          <user id="1062" type="pointer"/>
          <user id="1063" type="pointer"/>

	</users>
      </group>
    </groups>

  </domain>
</include>

and the xml files under directory/default

root at snoip-srv-001:/usr/local/freeswitch/conf/directory/default# ls -l
total 164
-rw-r--r-- 1 root root  750 2009-07-21 19:44 1000.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1001.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1002.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1003.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1004.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1005.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1006.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1007.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1008.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1009.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1010.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1011.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1012.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1013.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1014.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1015.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1016.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1017.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1018.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1019.xml
-rw-r--r-- 1 root root  750 2009-08-12 06:49 1020.xml
-rw-r--r-- 1 root root  750 2009-07-27 13:14 1050.xml
-rw-r--r-- 1 root root  750 2009-07-27 13:14 1051.xml
-rw-r--r-- 1 root root  750 2009-07-27 13:14 1052.xml
-rw-r--r-- 1 root root  750 2009-08-06 08:59 1053.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:56 1054.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:56 1055.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:56 1056.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:56 1057.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:57 1058.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:57 1059.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:57 1060.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:57 1061.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:57 1062.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:58 1063.xml
-rw-r--r-- 1 root root  750 2009-08-11 10:10 1064.xml
-rw-r--r-- 1 root root  750 2009-08-11 10:12 1065.xml
-rw-r--r-- 1 root root 5029 2009-07-20 23:47 brian.xml
-rw-r--r-- 1 root root  526 2009-07-20 23:47 default.xml
-rw-r--r-- 1 root root  905 2009-07-20 23:47 example.com.xml


From kevin at kgolding.co.uk  Wed Aug 12 03:31:01 2009
From: kevin at kgolding.co.uk (Kevin Golding)
Date: Wed, 12 Aug 2009 11:31:01 +0100
Subject: [Freeswitch-users] problem when adding more extension
In-Reply-To: <10128ef10908120250y77917460g5c6b9cd1991fe074@mail.gmail.com>
References: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>	<4A828198.6020000@kgolding.co.uk>
	<10128ef10908120250y77917460g5c6b9cd1991fe074@mail.gmail.com>
Message-ID: <4A8299E5.9050207@kgolding.co.uk>

Edit the line below as shown (in the dialplan/default.xml file)

Original line (about line 206)
<condition field="desination_number" expression="^(10[01][0-9])$">

Replacement line
<condition field="desination_number" expression="^(10[0-9][0-9])$">

This will allow extensions number 1000 to 1099.

Kevin

Tzury Bar Yochay wrote:
> still not working, I mean, I can initiate a call from 1060 to 1000 but
> not from 1000 to 1060.
> 1060 is just an example. This applies to all new extension I have
> added (beyond to the default 1000-1019).
> as you can see below I added them all to <group name="support">
> 
> This is how the confs look like
> 
> /usr/local/freeswitch/conf/directory# cat default.xml
> <include>
>   <!--the domain or ip (the right hand side of the @ in the addr-->
>   <domain name="$${domain}">
>     <params>
>       <param name="dial-string"
> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>     </params>
> 
>     <variables>
>       <variable name="record_stereo" value="true"/>
>       <variable name="default_gateway" value="$${default_provider}"/>
>       <variable name="default_areacode" value="$${default_areacode}"/>
>       <variable name="transfer_fallback_extension" value="operator"/>
>     </variables>
> 
>     <groups>
>       <group name="default">
> 	<users>
> 	  <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
> 	</users>
>       </group>
> 
>       <group name="sales">
> 	<users>
> 	  <!--
> 	      type="pointer" is a pointer so you can have the
> 	      same user in multiple groups.  It basically means
> 	      to keep searching for the user in the directory.
> 	  -->
> 	  <user id="1000" type="pointer"/>
> 	  <user id="1001" type="pointer"/>
> 	  <user id="1002" type="pointer"/>
> 	  <user id="1003" type="pointer"/>
> 	  <user id="1004" type="pointer"/>
> 	</users>
>       </group>
> 
>       <group name="billing">
> 	<users>
> 	  <user id="1005" type="pointer"/>
> 	  <user id="1006" type="pointer"/>
> 	  <user id="1007" type="pointer"/>
> 	  <user id="1008" type="pointer"/>
> 	  <user id="1009" type="pointer"/>
> 	</users>
>       </group>
> 
>       <group name="support">
> 	<users>
> 	  <user id="1010" type="pointer"/>
> 	  <user id="1011" type="pointer"/>
> 	  <user id="1012" type="pointer"/>
> 	  <user id="1013" type="pointer"/>
>           <user id="1014" type="pointer"/>
>           <user id="1015" type="pointer"/>
>           <user id="1016" type="pointer"/>
>           <user id="1017" type="pointer"/>
>           <user id="1018" type="pointer"/>
>           <user id="1019" type="pointer"/>
>           <user id="1020" type="pointer"/>
>           <user id="1050" type="pointer"/>
>           <user id="1051" type="pointer"/>
> 	  <user id="1052" type="pointer"/>
> 	  <user id="1053" type="pointer"/>
> 	  <user id="1054" type="pointer"/>
>           <user id="1055" type="pointer"/>
>           <user id="1056" type="pointer"/>
>           <user id="1057" type="pointer"/>
>           <user id="1058" type="pointer"/>
>           <user id="1059" type="pointer"/>
>           <user id="1060" type="pointer"/>
>           <user id="1061" type="pointer"/>
>           <user id="1062" type="pointer"/>
>           <user id="1063" type="pointer"/>
> 
> 	</users>
>       </group>
>     </groups>
> 
>   </domain>
> </include>
> 
> and the xml files under directory/default
> 
> root at snoip-srv-001:/usr/local/freeswitch/conf/directory/default# ls -l
> total 164
> -rw-r--r-- 1 root root  750 2009-07-21 19:44 1000.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1001.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1002.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1003.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1004.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1005.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1006.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1007.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1008.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1009.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1010.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1011.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1012.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1013.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1014.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1015.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1016.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1017.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1018.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1019.xml
> -rw-r--r-- 1 root root  750 2009-08-12 06:49 1020.xml
> -rw-r--r-- 1 root root  750 2009-07-27 13:14 1050.xml
> -rw-r--r-- 1 root root  750 2009-07-27 13:14 1051.xml
> -rw-r--r-- 1 root root  750 2009-07-27 13:14 1052.xml
> -rw-r--r-- 1 root root  750 2009-08-06 08:59 1053.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:56 1054.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:56 1055.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:56 1056.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:56 1057.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:57 1058.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:57 1059.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:57 1060.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:57 1061.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:57 1062.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:58 1063.xml
> -rw-r--r-- 1 root root  750 2009-08-11 10:10 1064.xml
> -rw-r--r-- 1 root root  750 2009-08-11 10:12 1065.xml
> -rw-r--r-- 1 root root 5029 2009-07-20 23:47 brian.xml
> -rw-r--r-- 1 root root  526 2009-07-20 23:47 default.xml
> -rw-r--r-- 1 root root  905 2009-07-20 23:47 example.com.xml
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From tzury.by at reguluslabs.com  Wed Aug 12 03:48:09 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Wed, 12 Aug 2009 13:48:09 +0300
Subject: [Freeswitch-users] problem when adding more extension
In-Reply-To: <4A8299E5.9050207@kgolding.co.uk>
References: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>
	<4A828198.6020000@kgolding.co.uk>
	<10128ef10908120250y77917460g5c6b9cd1991fe074@mail.gmail.com>
	<4A8299E5.9050207@kgolding.co.uk>
Message-ID: <10128ef10908120348m645994b6k5935f17010cbc79c@mail.gmail.com>

Thanks allot Kevin.
I felt it is about a missing configuration parameter


On Wed, Aug 12, 2009 at 1:31 PM, Kevin Golding<kevin at kgolding.co.uk> wrote:
> Edit the line below as shown (in the dialplan/default.xml file)
>
> Original line (about line 206)
> <condition field="desination_number" expression="^(10[01][0-9])$">
>
> Replacement line
> <condition field="desination_number" expression="^(10[0-9][0-9])$">
>
> This will allow extensions number 1000 to 1099.
>
> Kevin
>
> Tzury Bar Yochay wrote:
>> still not working, I mean, I can initiate a call from 1060 to 1000 but
>> not from 1000 to 1060.
>> 1060 is just an example. This applies to all new extension I have
>> added (beyond to the default 1000-1019).
>> as you can see below I added them all to <group name="support">
>>
>> This is how the confs look like
>>
>> /usr/local/freeswitch/conf/directory# cat default.xml
>> <include>
>> ? <!--the domain or ip (the right hand side of the @ in the addr-->
>> ? <domain name="$${domain}">
>> ? ? <params>
>> ? ? ? <param name="dial-string"
>> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>> ? ? </params>
>>
>> ? ? <variables>
>> ? ? ? <variable name="record_stereo" value="true"/>
>> ? ? ? <variable name="default_gateway" value="$${default_provider}"/>
>> ? ? ? <variable name="default_areacode" value="$${default_areacode}"/>
>> ? ? ? <variable name="transfer_fallback_extension" value="operator"/>
>> ? ? </variables>
>>
>> ? ? <groups>
>> ? ? ? <group name="default">
>> ? ? ? <users>
>> ? ? ? ? <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
>> ? ? ? </users>
>> ? ? ? </group>
>>
>> ? ? ? <group name="sales">
>> ? ? ? <users>
>> ? ? ? ? <!--
>> ? ? ? ? ? ? type="pointer" is a pointer so you can have the
>> ? ? ? ? ? ? same user in multiple groups. ?It basically means
>> ? ? ? ? ? ? to keep searching for the user in the directory.
>> ? ? ? ? -->
>> ? ? ? ? <user id="1000" type="pointer"/>
>> ? ? ? ? <user id="1001" type="pointer"/>
>> ? ? ? ? <user id="1002" type="pointer"/>
>> ? ? ? ? <user id="1003" type="pointer"/>
>> ? ? ? ? <user id="1004" type="pointer"/>
>> ? ? ? </users>
>> ? ? ? </group>
>>
>> ? ? ? <group name="billing">
>> ? ? ? <users>
>> ? ? ? ? <user id="1005" type="pointer"/>
>> ? ? ? ? <user id="1006" type="pointer"/>
>> ? ? ? ? <user id="1007" type="pointer"/>
>> ? ? ? ? <user id="1008" type="pointer"/>
>> ? ? ? ? <user id="1009" type="pointer"/>
>> ? ? ? </users>
>> ? ? ? </group>
>>
>> ? ? ? <group name="support">
>> ? ? ? <users>
>> ? ? ? ? <user id="1010" type="pointer"/>
>> ? ? ? ? <user id="1011" type="pointer"/>
>> ? ? ? ? <user id="1012" type="pointer"/>
>> ? ? ? ? <user id="1013" type="pointer"/>
>> ? ? ? ? ? <user id="1014" type="pointer"/>
>> ? ? ? ? ? <user id="1015" type="pointer"/>
>> ? ? ? ? ? <user id="1016" type="pointer"/>
>> ? ? ? ? ? <user id="1017" type="pointer"/>
>> ? ? ? ? ? <user id="1018" type="pointer"/>
>> ? ? ? ? ? <user id="1019" type="pointer"/>
>> ? ? ? ? ? <user id="1020" type="pointer"/>
>> ? ? ? ? ? <user id="1050" type="pointer"/>
>> ? ? ? ? ? <user id="1051" type="pointer"/>
>> ? ? ? ? <user id="1052" type="pointer"/>
>> ? ? ? ? <user id="1053" type="pointer"/>
>> ? ? ? ? <user id="1054" type="pointer"/>
>> ? ? ? ? ? <user id="1055" type="pointer"/>
>> ? ? ? ? ? <user id="1056" type="pointer"/>
>> ? ? ? ? ? <user id="1057" type="pointer"/>
>> ? ? ? ? ? <user id="1058" type="pointer"/>
>> ? ? ? ? ? <user id="1059" type="pointer"/>
>> ? ? ? ? ? <user id="1060" type="pointer"/>
>> ? ? ? ? ? <user id="1061" type="pointer"/>
>> ? ? ? ? ? <user id="1062" type="pointer"/>
>> ? ? ? ? ? <user id="1063" type="pointer"/>
>>
>> ? ? ? </users>
>> ? ? ? </group>
>> ? ? </groups>
>>
>> ? </domain>
>> </include>
>>
>> and the xml files under directory/default
>>
>> root at snoip-srv-001:/usr/local/freeswitch/conf/directory/default# ls -l
>> total 164
>> -rw-r--r-- 1 root root ?750 2009-07-21 19:44 1000.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1001.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1002.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1003.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1004.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1005.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1006.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1007.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1008.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1009.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1010.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1011.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1012.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1013.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1014.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1015.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1016.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1017.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1018.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1019.xml
>> -rw-r--r-- 1 root root ?750 2009-08-12 06:49 1020.xml
>> -rw-r--r-- 1 root root ?750 2009-07-27 13:14 1050.xml
>> -rw-r--r-- 1 root root ?750 2009-07-27 13:14 1051.xml
>> -rw-r--r-- 1 root root ?750 2009-07-27 13:14 1052.xml
>> -rw-r--r-- 1 root root ?750 2009-08-06 08:59 1053.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:56 1054.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:56 1055.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:56 1056.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:56 1057.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:57 1058.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:57 1059.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:57 1060.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:57 1061.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:57 1062.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:58 1063.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 10:10 1064.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 10:12 1065.xml
>> -rw-r--r-- 1 root root 5029 2009-07-20 23:47 brian.xml
>> -rw-r--r-- 1 root root ?526 2009-07-20 23:47 default.xml
>> -rw-r--r-- 1 root root ?905 2009-07-20 23:47 example.com.xml
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Tzury Bar Yochay

Regulus Labs ltd.
http://reguluslabs.com
+972 52 5133399


From dujinfang at gmail.com  Wed Aug 12 04:17:21 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 12 Aug 2009 19:17:21 +0800
Subject: [Freeswitch-users] answer command
In-Reply-To: <24931876.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
Message-ID: <B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>

It's not Eyebeam but FS hung up the call because it have nothing to do  
after answer.

You should either playback a sound, do the echo command, record, hold  
the call, bridge to another channel or transfer somewhere else.....

On Aug 12, 2009, at 4:54 PM, Maxim Tsvetov wrote:
>
> I've tried to use "answer" command from outbound event socket and it's
> working, but
> the problem is that FS answering the call, but SIP Client (we tried  
> this
> with EyeBeam and CISCO 7960)
> doesn't know that call was answered. So, as long as FS doesn't know  
> what to
> do with this number it then disconnects the call.
>
> 2009-08-12 11:25:07.599250 [NOTICE] mod_dptools.c:649 Channel
> [sofia/internal/sip:1000 at 10.107.181.160:42840] has been answered
> 2009-08-12 11:25:07.599250 [NOTICE] switch_ivr_originate.c:2015  
> Channel
> [sofia/internal/1003 at 10.107.249.12] has been answered
> 2009-08-12 11:25:07.614875 [ERR] switch_core_io.c:118
> sofia/internal/sip:1000 at 10.107.181.160:42840 has no read codec.
> 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:503 Hangup
> sofia/internal/sip:1000 at 10.107.181.160:42840 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
> 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:1016 Hangup
> sofia/internal/1003 at 10.107.249.12 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-08-12 11:25:07.630500 [NOTICE] switch_core_session.c:1086  
> Session 133
> (sofia/internal/sip:1000 at 10.107.181.160:42840) Ended
>
>
> Maybe there is the way to acknowledge SIP client that call was  
> answered?
>
> Regards,
> Maxim Tsvetov
>
> Diego Viola wrote:
>>
>> I suggest that you learn the differences between mod_commands  
>> commands and
>> mod_dptools applications, and also the interfaces where you can  
>> access and
>> use them.
>>
>> As said before, mod_dptools is accessible from dialplan, event socket
>> outbound, etc. and mod_commands is accessible from the CLI, event  
>> socket
>> (inbound/outbound), XML RPC, etc.
>>
>> That's all described in the wiki I think.
>>
>> Let us know if you have any questions =D.
>>
>> On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola <diego.viola at gmail.com>
>> wrote:
>>
>>> Michael, you're welcome :).
>>>
>>> Milena, answer is a mod_dptools command, you can use it from the XML
>>> dialplan or from the event socket outbound. mod_commands API are  
>>> APIs
>>> that
>>> you execute from the socket, event socket inbound, etc. But you  
>>> can also
>>> execute them from event socket outbound using the "api" command.
>>>
>>> I hope that makes sense, correct me if I'm wrong =D.
>>>
>>> On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins
>>> <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Tue, Aug 11, 2009 at 9:05 AM, Milena <testeador01 at gmail.com>  
>>>> wrote:
>>>>
>>>>>
>>>>> Hello Brian,
>>>>>
>>>>> I wanna fix the wiki, but to make sure i got it right, does it  
>>>>> only
>>>>> work
>>>>> on outbound event socket? or is there any other scenario where  
>>>>> it would
>>>>> work.
>>>>>
>>>>
>>>> FYI, Diego Viola fixed the wiki. (Thanks Diego!)
>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://www.nabble.com/answer-command-tp24912812p24931876.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dujinfang at gmail.com  Wed Aug 12 04:20:31 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 12 Aug 2009 19:20:31 +0800
Subject: [Freeswitch-users] problem when adding more extension
In-Reply-To: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>
References: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>
Message-ID: <93FED77F-903E-4B38-BD33-D43C7690F0C1@gmail.com>


On Aug 12, 2009, at 2:44 PM, Tzury Bar Yochay wrote:
> Hi,
>
> I wanted to add more extension to freeswitch.
> to add extension 1050 with password 1234 I did the following:
>
> $ cd /usr/local/freeswitch/conf/directory/default
>
> created 1050.xml having all '1000' strings replaced by '1050' by  
> typing
> $ sed s/1000/1050/g < 1000.xml > 1050.xml
>

just run reloadxml should be ok no need to rescan the profile

> rescan and reload the xml by typing into the CLI
>
> freeswitch at internal> sofia profile internal rescan reloadxml
>
> However, when I tried to login with these credentials I got the
> following in the fs_cli:
>
> 2009-08-12 06:35:01 [DEBUG] sofia_reg.c:1469 sofia_reg_parse_auth()
> SIP username 1050 does not match auth username
> 2009-08-12 06:35:01 [DEBUG] sofia_reg.c:869
> sofia_reg_handle_register() Send challenge for
> [1050 at server_address.net]
>
> below are the content of 1000 and 1050 xml files
>
> please advise.
>
>
> $ cat 1050.xml
> <include>
> <user id="1050" mailbox="1050">
>   <params>
>     <param name="password" value="1234"/>
>     <param name="vm-password" value="1050"/>
>   </params>
>   <variables>
>     <variable name="toll_allow" value="domestic,international,local"/>
>     <variable name="accountcode" value="1050"/>
>     <variable name="user_context" value="default"/>
>     <variable name="effective_caller_id_name" value="Extension 1050"/>
>     <variable name="effective_caller_id_number" value="1050"/>
>     <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>     <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>     <variable name="callgroup" value="techsupport"/>
>   </variables>
> </user>
> </include>
>
> $ cat 1000.xml
> <include>
> <user id="1000" mailbox="1000">
>   <params>
>     <param name="password" value="1234"/>
>     <param name="vm-password" value="1000"/>
>   </params>
>   <variables>
>     <variable name="toll_allow" value="domestic,international,local"/>
>     <variable name="accountcode" value="1000"/>
>     <variable name="user_context" value="default"/>
>     <variable name="effective_caller_id_name" value="Extension 1000"/>
>     <variable name="effective_caller_id_number" value="1000"/>
>     <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>     <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>     <variable name="callgroup" value="techsupport"/>
>   </variables>
> </user>
> </include>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From markmorreny at gmail.com  Wed Aug 12 04:26:38 2009
From: markmorreny at gmail.com (mark morreny)
Date: Wed, 12 Aug 2009 19:26:38 +0800
Subject: [Freeswitch-users] Fwd:  Fwd: Scheduler in module
In-Reply-To: <5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
Message-ID: <20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>

Hi,

In my LOAD_FUNCTION, I am trying to have freeswitch to flush out some data
every 10 s.  The following lines of code does not show any effect at all.

    switch_scheduler_task_thread_start();
    switch_scheduler_add_task(switch_epoch_time_now(NULL),
data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|SSHF_NO_DEL);


SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {

switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to flush
data  buffer...\n");


    task->runtime = switch_time_now() + 10;

}

Does anyone know how to get it to work?

Thanks,
Mark


---------- Forwarded message ----------
From: Brian West <brian at freeswitch.org>
Date: Mon, Aug 10, 2009 at 8:53 PM
Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
To: freeswitch-users at lists.freeswitch.org


switch_rtp.c has a simple one for the zrtp cache storing.

/b

On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:

> Re schedule is done in your callback, take a look at places that use
> these apis in the code for details.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From juanbackson at gmail.com  Wed Aug 12 05:01:59 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Wed, 12 Aug 2009 20:01:59 +0800
Subject: [Freeswitch-users] freeswitch time conversion
Message-ID: <27c25bc40908120501l19815773ne1494040ddcab111@mail.gmail.com>

Does anyone know how to take the epoch time in switch_event_t and convert it
into a format such as "Sat Jul 5 02:44:33 2009"?

Is there any existing facility that I can use for this purpose?

br,
JB
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From boonedox at gmail.com  Wed Aug 12 06:02:48 2009
From: boonedox at gmail.com (Jeremiah Johnson)
Date: Wed, 12 Aug 2009 07:02:48 -0600
Subject: [Freeswitch-users] hangup_after_bridge=false only works if
	bypass_media=false
In-Reply-To: <C5AD471E-636F-427A-A0A9-9F894D799CEB@jerris.com>
References: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
	<C5AD471E-636F-427A-A0A9-9F894D799CEB@jerris.com>
Message-ID: <c626f0a50908120602l155bd7f5w6357ff6d2303ae5@mail.gmail.com>

I posted it yesterday evening:  http://jira.freeswitch.org/browse/FSCORE-417

On Tue, Aug 11, 2009 at 9:43 PM, Michael Jerris <mike at jerris.com> wrote:

> Please post a bug for this on jira.freeswitch.org.
>
> Mike
>
> On Aug 11, 2009, at 2:29 PM, Jeremiah Johnson wrote:
>
> > This is an integral part of my application.  I need to have
> > FreeSWITCH outside of the media path as well as be able to do
> > multiple bridges for the same "A" leg.
> >
> > /*WORKS*/
> > <action application="set" data="hangup_after_bridge=false"/>
> > <action application="bridge" data="sofia/gateway/${mygateway}/1$
> > {client_one}"/>
> > <action application="bridge" data="sofia/gateway/${mygateway}/1$
> > {client_two}"/>
> >
> > /*DOES NOT WORK*/
> > <action application="set" data="hangup_after_bridge=false"/>
> > <action application="set" data="bypass_media=true"/>
> > <action application="bridge" data="sofia/gateway/${mygateway}/1$
> > {client_one}"/>
> > <action application="bridge" data="sofia/gateway/${mygateway}/1$
> > {client_two}"/>
> >
> > In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the
> > leg for client_one hangs up.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Wed Aug 12 06:29:31 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 12 Aug 2009 09:29:31 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
Message-ID: <367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>

David / Michael - thanks for your your replies. The SoftIVR example is
particularly useful. Must admit though - I was hoping not to have to
do any custom stuff at this stage.

It does appear there is no method to do this by staking bridge lines
so I will put an issue in jira to try and get loopback working with
bypass_media.

In the meantime I will also start looking to build a custom bridging
app. As I said though - not a road I wanted to go down.

Thanks for your help!


Phillip Jones

On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavo<mgg at giagnocavo.net> wrote:
> It's also simple enough to write a plugin in one of the scripting languages to add an app to do exactly what you want...
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Knell
> Sent: Tuesday, August 11, 2009 1:55 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Loopback and bypass_media
>
> Just to add my $0.02-worth (if you're feeling generous..) - I don't
> think that the dialplan is expressive enough to do what's needed here,
> and that's where the trouble's coming from. ?It's not enormously tricky
> to build a generic "dial this set of numbers according to these rules"
> service using something hanging off the event socket - there's a writeup
> here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
> be done on SoftIVR.
>
> To roll something similar yourself using the event socket, you'd need to
> map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
> and have some way of passing messages around between the threads
> handling the different call legs, assuming that you're using one thread
> per leg.
>
> --Dave
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From moises.silva at gmail.com  Wed Aug 12 07:17:29 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 12 Aug 2009 10:17:29 -0400
Subject: [Freeswitch-users] FW: Sangoma/FS...
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C52ECE61290@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED944C@cooper>
	<549CFEF87AEDE841A38E9D15EAB4C04C52ECE61290@cooper>
Message-ID: <c4d05cbe0908120717t26a9b7c2m3840fb3ab641b002@mail.gmail.com>

Hello Peter,
I'd appreciate if you can keep the discussion going in the freeswitch-users
mailing list, there are other people there that will benefit of the
discussion or even can help. Read my comments below.

On Wed, Aug 12, 2009 at 5:59 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Sorry for spamming you :) But I have some more results now. I?ve tried
> using another lab PBX with Q.SIG enabled, and when using that one I?m able
> to connect calls as I should. At least incoming to FS, outgoing seem to have
> some problems still.. So the problem for the PBX I used yesterday seems to
> be both related to Q.SIG (maybe) and the PBX itself (it does connect to
> other providers though, so I know the trunk works).
>
>
>
> Should I take some dumps from the PRI card to try to find out why it didn?t
> work with the first one, or is this ?as expected?, since they have Q.SIG
> enabled?
>
> I have no experience with Q.SIG, so I won't be able to help much. One thing
though, is that if I were you, I'd be using openzap with libpri support, is
that what you are using, or are you using the ISDN openzap stack?

As of the dumps, they may help, or not, but pastebin them anyways so I can
make an un-educated guess.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From rupa at rupa.com  Wed Aug 12 07:22:12 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 12 Aug 2009 09:22:12 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
Message-ID: <b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>

perhaps we need to add some syntax + logic to originate:

application="originate"
data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"

This would acomplish the equiv of

loopback/bar,loopback/yum  where bar and yum are then further expanded
in the dialplan as

sofia/foo/bar|sofia/baz/bar and sofia/foo/yum|sofia/baz/yum

except that the threads of execution are handled directly by
originate.  I'm not sure that is really the "solution" since each ()
group would still have to be a separate thread to run independently.

To me, loopback is the way to accomplish this issue (how I've done it
with the same requirements that you have) since all the hard work is
layered and works.  The problem is that you require bypass_media which
doesn't play nice with loopback.

Perhaps have an on answer hook that tries to enable bypass media
(re-invite) after the call is setup?

On Wed, Aug 12, 2009 at 8:29 AM, Phillip Jones<pjintheusa at gmail.com> wrote:
> David / Michael - thanks for your your replies. The SoftIVR example is
> particularly useful. Must admit though - I was hoping not to have to
> do any custom stuff at this stage.
>
> It does appear there is no method to do this by staking bridge lines
> so I will put an issue in jira to try and get loopback working with
> bypass_media.
>
> In the meantime I will also start looking to build a custom bridging
> app. As I said though - not a road I wanted to go down.
>
> Thanks for your help!
>
>
> Phillip Jones
>
> On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavo<mgg at giagnocavo.net> wrote:
>> It's also simple enough to write a plugin in one of the scripting languages to add an app to do exactly what you want...
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Knell
>> Sent: Tuesday, August 11, 2009 1:55 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Loopback and bypass_media
>>
>> Just to add my $0.02-worth (if you're feeling generous..) - I don't
>> think that the dialplan is expressive enough to do what's needed here,
>> and that's where the trouble's coming from. ?It's not enormously tricky
>> to build a generic "dial this set of numbers according to these rules"
>> service using something hanging off the event socket - there's a writeup
>> here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
>> be done on SoftIVR.
>>
>> To roll something similar yourself using the event socket, you'd need to
>> map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
>> and have some way of passing messages around between the threads
>> handling the different call legs, assuming that you're using one thread
>> per leg.
>>
>> --Dave
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From email.list.subscriber at gmail.com  Wed Aug 12 07:28:08 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Wed, 12 Aug 2009 10:28:08 -0400
Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9
	&	v1.0.4)
In-Reply-To: <4A8280B8.6050308@blueface.ie>
References: <4A8280B8.6050308@blueface.ie>
Message-ID: <4a82d138.1f588c0a.2915.4e3d@mx.google.com>

Hi Bruce,

I am having similar issues trying build freeswitch 1.0.4 on Solaris
x86 as well.  I sent some information over the mailing list, and I
received a response from Michal Bielicki (attached), stating he'd test
this and direct me to the steps to successfully build freeswitch.

Just an FYI in case you see his response.

Vladimir

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Bruce McAlister
Sent: Wednesday, August 12, 2009 4:44 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 &
v1.0.4)

Hi All,

I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and

1.0.4.

I am running on Solaris 10 Update 5 on x86 hardware (32-bit).

The build fails with:

--- snip ---
make: Fatal error: Command failed for target `all-recursive'
Current working directory
/export/home/user/packages/BUILD/freeswitch-1.0.4
*** Error code 1
make: Fatal error: Command failed for target `all'
---

Looking back through the build I can see the following error:

--- snip ---
creating libfreeswitch.la
(cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la 
libfreeswitch.la)
/usr/bin/cc 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/s
rc 
-KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
-DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch 
freeswitch-switch.o  ./.libs/libfreeswitch.so 
-L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/
expat/lib 
/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/ex
pat/lib/.libs/libexpat.a 
/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libap
r-1.a 
-lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp 
-L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt 
-lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl 
-lcurses -lsocket  -R/opt/freeswitch/lib -R/usr/sfw/lib
Undefined                       first referenced
  symbol                             in file
herror                              ./.libs/libfreeswitch.so
ld: fatal: Symbol referencing errors. No output written to
.libs/freeswitch
*** Error code 1
The following command caused the error:
`if test -z "" ; then echo /bin/bash 
/export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else 
echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` 
--tag=CC --mode=link /usr/bin/cc 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/s
rc 
-KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
-DPATH_MAX=2048 -g -v -Xc -xc99=all  -lm -R/opt/freeswitch/lib  -o 
freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib 
freeswitch-switch.o   libfreeswitch.la libs/apr/libapr-1.la 
libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl 
-lcrypto -lsocket -lnsl -ldl   -lcurses -lsocket
--- snip ---

Then a little above this error, there is the following warning that is

displayed (I'm not sure if it is related):

--- snip ---
*** Warning: Linking the shared library libfreeswitch.la against the
*** static library libs/libedit/src/.libs/libedit.a is not portable!
--- snip ---

My configure line is as follows:

---
./configure --prefix=/opt/freeswitch
---

I have the complete configure and make output if anyone needs them.

Any help/pointers would be greatly appreciated.

Thanks
Bruce

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
http://www.freeswitch.org
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From: "Michal Bielicki" <michal.bielicki at halo2.pl>
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
Date: Tue, 11 Aug 2009 05:33:06 -0400
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From peter.olsson at visionutveckling.se  Wed Aug 12 07:29:11 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 12 Aug 2009 16:29:11 +0200
Subject: [Freeswitch-users] Sangoma/FS...
In-Reply-To: <c4d05cbe0908120717t26a9b7c2m3840fb3ab641b002@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED944C@cooper>
	<549CFEF87AEDE841A38E9D15EAB4C04C52ECE61290@cooper>
	<c4d05cbe0908120717t26a9b7c2m3840fb3ab641b002@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C52ECE61309@cooper>

Of course ? no problem!

I?m not using libpri support now, I don?t think it?s ported for Windows (yet)?

I?ll try it out some more, and try to detect what?s going wrong...

/Peter

Fr?n: Moises Silva [mailto:moises.silva at gmail.com]
Skickat: den 12 augusti 2009 16:17
Till: freeswitch-users at lists.freeswitch.org
Kopia: Peter Olsson
?mne: Re: FW: Sangoma/FS...

Hello Peter,

I'd appreciate if you can keep the discussion going in the freeswitch-users mailing list, there are other people there that will benefit of the discussion or even can help. Read my comments below.

On Wed, Aug 12, 2009 at 5:59 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:

Sorry for spamming you :) But I have some more results now. I?ve tried using another lab PBX with Q.SIG enabled, and when using that one I?m able to connect calls as I should. At least incoming to FS, outgoing seem to have some problems still.. So the problem for the PBX I used yesterday seems to be both related to Q.SIG (maybe) and the PBX itself (it does connect to other providers though, so I know the trunk works).



Should I take some dumps from the PRI card to try to find out why it didn?t work with the first one, or is this ?as expected?, since they have Q.SIG enabled?
I have no experience with Q.SIG, so I won't be able to help much. One thing though, is that if I were you, I'd be using openzap with libpri support, is that what you are using, or are you using the ISDN openzap stack?

As of the dumps, they may help, or not, but pastebin them anyways so I can make an un-educated guess.

--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com<mailto:moy at sangoma.com>
!DSPAM:4a82cefe32931477278362!
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From mrene_lists at avgs.ca  Wed Aug 12 08:04:00 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 12 Aug 2009 11:04:00 -0400
Subject: [Freeswitch-users] freeswitch time conversion
In-Reply-To: <27c25bc40908120501l19815773ne1494040ddcab111@mail.gmail.com>
References: <27c25bc40908120501l19815773ne1494040ddcab111@mail.gmail.com>
Message-ID: <EECDBB6A-49A4-491C-8952-14901E2E50EC@avgs.ca>

Hi,


The standard C function is strftime.

FreeSWITCH has some wrapped ones:

switch_apr.h:SWITCH_DECLARE(switch_status_t) switch_strftime(char *s,  
switch_size_t *retsize, switch_size_t max, const char *format,  
switch_time_exp_t *tm);
switch_apr.h:SWITCH_DECLARE(switch_status_t)  
switch_strftime_nocheck(char *s, switch_size_t *retsize, switch_size_t  
max, const char *format, switch_time_exp_t *tm);
switch_core.h:SWITCH_DECLARE(switch_status_t) switch_strftime_tz(const  
char *tz, const char *format, char *date, size_t len, switch_time_t  
thetime);


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 12-Aug-09, at 8:01 AM, Juan Backson wrote:

> Does anyone know how to take the epoch time in switch_event_t and  
> convert it into a format such as "Sat Jul 5 02:44:33 2009"?
>
> Is there any existing facility that I can use for this purpose?
>
> br,
> JB
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mrene_lists at avgs.ca  Wed Aug 12 08:09:21 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 12 Aug 2009 11:09:21 -0400
Subject: [Freeswitch-users] Fwd:  Fwd: Scheduler in module
In-Reply-To: <20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
Message-ID: <3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>

Hi,

I did the same thing on my side....

API CALL [load(mod_skel)] output:
+OK

2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task 2  
data_flush (core) to run at 1250089698
2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889  
Successfully Loaded [mod_skel]
2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270 Adding  
API Function 'skel'
freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]  
mod_skel.c:120 starting to flush data  buffer...

Note that you don't need to start the thread manually, the core  
already has threads running for the scheduler.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 12-Aug-09, at 7:26 AM, mark morreny wrote:

> Hi,
>
> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out  
> some data every 10 s.  The following lines of code does not show any  
> effect at all.
>
>     switch_scheduler_task_thread_start();
>     switch_scheduler_add_task(switch_epoch_time_now(NULL),  
> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE| 
> SSHF_NO_DEL);
>
>
> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to  
> flush data  buffer...\n");
>
>
>     task->runtime = switch_time_now() + 10;
>
> }
>
> Does anyone know how to get it to work?
>
> Thanks,
> Mark
>
>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> Date: Mon, Aug 10, 2009 at 8:53 PM
> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
> To: freeswitch-users at lists.freeswitch.org
>
>
> switch_rtp.c has a simple one for the zrtp cache storing.
>
> /b
>
> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>
> > Re schedule is done in your callback, take a look at places that use
> > these apis in the code for details.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From charlieb at cot.net  Wed Aug 12 08:14:17 2009
From: charlieb at cot.net (Charles Boening)
Date: Wed, 12 Aug 2009 08:14:17 -0700
Subject: [Freeswitch-users] FS 1.0.4 official core dumping LUA streamFile
In-Reply-To: <A9CD3A3C-923A-47FE-B9BC-DEAE701F3248@jerris.com>
References: <4FB2938B89459C41860C4DB9B1821D6FB6657A2421@exchange.calore.local>
	<A9CD3A3C-923A-47FE-B9BC-DEAE701F3248@jerris.com>
Message-ID: <4FB2938B89459C41860C4DB9B1821D6FB6657A2427@exchange.calore.local>

Done.

Thanks,
Charlie


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Wednesday, August 12, 2009 2:12 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS 1.0.4 official core dumping LUA streamFile

If your seeing a segfault, please report it to jira.freeswitch.org with a backtrace and details of how to reproduce.

Mike

On Aug 12, 2009, at 2:37 AM, Charles Boening wrote:


Greetings,

I have the following LUA script (at end of email) in a fresh FS 1.0.4 install.  I originally did an upgrade from one of the 1.0.4preX versions but when I came across this issue I went fresh just to make sure there wasn't an incompatibility with my previous config.

What I'm seeing is a seg fault and a core dump after playing a sound file.  I originally had a file I recorded but when I ran into this issue I figured I'd try an included sound file but that doesn't seem to make a bit of difference.

2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Starting test.lua
2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Caller [XXXXXXXXXX] connected
2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Pre streamFile
Segmentation fault (core dumped)


Any ideas?

Thanks,
Charlie



freeswitch.consoleLog("INFO", string.format("Starting test.lua\n"))
session:answer();
session:setHangupHook("session_hangup_hook")
calleridnumber = session:getVariable("caller_id_number")
calleridname = session:getVariable("caller_id_name")

if session:ready() then
   freeswitch.consoleLog("INFO", string.format("Caller [" .. calleridnumber .. "] connected\n"))
   freeswitch.consoleLog("INFO", string.format("Pre streamFile\n"))

   session:streamFile("conference/8000/conf-welcome.wav")

   freeswitch.consoleLog("INFO", string.format("Post streamFile.\n"))
end

function session_hangup_hook(status)
  freeswitch.consoleLog("INFO", "Session hangup: \n") --[[ .. status .. "\n") ]]--
  error()
end

session:hangup()

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From pjintheusa at gmail.com  Wed Aug 12 08:22:08 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 12 Aug 2009 11:22:08 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
Message-ID: <367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>

Hi there,

>> application="originate" data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"

I agree. However, perhaps the ideal is not to specify the carriers at
this level, as carriers are added and removed fairly often as costings
change. So it would be nice to have some sort of proxy that resolves
to a list of carriers:

application="originate" data="sofia/MyCarriers/bar,sofia/MyCarriers/yum"

<MyCarriers>
<action application="carrier" data="sofia/foo"/>
<action application="carrier" data="sofia/baz"/>
<action application="carrier" data="sofia/etc"/>
</MyCarriers>


or something similar. This would achieve the same as loopback in this
use case but without dangers of looping? Complicated stuff though.

>>Perhaps have an on answer hook that tries to enable bypass media (re-invite) after the call is setup?

That's a good idea - I will look into that.


Thanks again.


Phillip

On Wed, Aug 12, 2009 at 10:22 AM, Rupa Schomaker<rupa at rupa.com> wrote:
> perhaps we need to add some syntax + logic to originate:
>
> application="originate"
> data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"
>
> This would acomplish the equiv of
>
> loopback/bar,loopback/yum ?where bar and yum are then further expanded
> in the dialplan as
>
> sofia/foo/bar|sofia/baz/bar and sofia/foo/yum|sofia/baz/yum
>
> except that the threads of execution are handled directly by
> originate. ?I'm not sure that is really the "solution" since each ()
> group would still have to be a separate thread to run independently.
>
> To me, loopback is the way to accomplish this issue (how I've done it
> with the same requirements that you have) since all the hard work is
> layered and works. ?The problem is that you require bypass_media which
> doesn't play nice with loopback.
>
> Perhaps have an on answer hook that tries to enable bypass media
> (re-invite) after the call is setup?
>
> On Wed, Aug 12, 2009 at 8:29 AM, Phillip Jones<pjintheusa at gmail.com> wrote:
>> David / Michael - thanks for your your replies. The SoftIVR example is
>> particularly useful. Must admit though - I was hoping not to have to
>> do any custom stuff at this stage.
>>
>> It does appear there is no method to do this by staking bridge lines
>> so I will put an issue in jira to try and get loopback working with
>> bypass_media.
>>
>> In the meantime I will also start looking to build a custom bridging
>> app. As I said though - not a road I wanted to go down.
>>
>> Thanks for your help!
>>
>>
>> Phillip Jones
>>
>> On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavo<mgg at giagnocavo.net> wrote:
>>> It's also simple enough to write a plugin in one of the scripting languages to add an app to do exactly what you want...
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Knell
>>> Sent: Tuesday, August 11, 2009 1:55 PM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Loopback and bypass_media
>>>
>>> Just to add my $0.02-worth (if you're feeling generous..) - I don't
>>> think that the dialplan is expressive enough to do what's needed here,
>>> and that's where the trouble's coming from. ?It's not enormously tricky
>>> to build a generic "dial this set of numbers according to these rules"
>>> service using something hanging off the event socket - there's a writeup
>>> here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
>>> be done on SoftIVR.
>>>
>>> To roll something similar yourself using the event socket, you'd need to
>>> map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
>>> and have some way of passing messages around between the threads
>>> handling the different call legs, assuming that you're using one thread
>>> per leg.
>>>
>>> --Dave
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From rupa at rupa.com  Wed Aug 12 09:21:58 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 12 Aug 2009 11:21:58 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
Message-ID: <b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>

On Wed, Aug 12, 2009 at 10:22 AM, Phillip Jones<pjintheusa at gmail.com> wrote:
> Hi there,
>
>>> application="originate" data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"
>
> I agree. However, perhaps the ideal is not to specify the carriers at
> this level, as carriers are added and removed fairly often as costings
> change. So it would be nice to have some sort of proxy that resolves
> to a list of carriers:
>
> application="originate" data="sofia/MyCarriers/bar,sofia/MyCarriers/yum"

> <MyCarriers>
> <action application="carrier" data="sofia/foo"/>
> <action application="carrier" data="sofia/baz"/>
> <action application="carrier" data="sofia/etc"/>
> </MyCarriers>
>
>
> or something similar. This would achieve the same as loopback in this
> use case but without dangers of looping? Complicated stuff though.

Well, that is all done by mod_lcr.  I was simplifying to narrow down
to just originate.

First we need to see if this is worth pursuing over fixing (modifying,
whatever) loopback to handle bypass media.  If it is, then I'll modify
mod_lcr to deal with the situation in question (comma or pipe sep list
of numbers to call.  mod_lcr would then group as appropriate).

Right now, my bridge is setup in a small javascript script that builds
the appropriate dialstring (using loopback for external calls, user/
for internal calls) and then when doing the loopback call to mod_lcr
to get the dialstring with all providers in the right order.

>>>Perhaps have an on answer hook that tries to enable bypass media (re-invite) after the call is setup?
>
> That's a good idea - I will look into that.
>
>
> Thanks again.
>
>
> Phillip

Let us know how it works for you...

-- 
-Rupa


From maxim.tsvetov at gmail.com  Wed Aug 12 09:49:20 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Wed, 12 Aug 2009 09:49:20 -0700 (PDT)
Subject: [Freeswitch-users] answer command
In-Reply-To: <B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
Message-ID: <24940548.post@talk.nabble.com>


I will try to paraphrase my question.
Is there any possibility to answer call  from CTI application and
synchronise answer with answer in SIP client?Maybe we can use SIP functions
in our CTI application instead of FS api commands?
I'm trying to find the way to make prototype of lineAnswer command in TAPI.


Seven Du wrote:
> 
> It's not Eyebeam but FS hung up the call because it have nothing to do  
> after answer.
> 
> You should either playback a sound, do the echo command, record, hold  
> the call, bridge to another channel or transfer somewhere else.....
> 
> On Aug 12, 2009, at 4:54 PM, Maxim Tsvetov wrote:
>>
>> I've tried to use "answer" command from outbound event socket and it's
>> working, but
>> the problem is that FS answering the call, but SIP Client (we tried  
>> this
>> with EyeBeam and CISCO 7960)
>> doesn't know that call was answered. So, as long as FS doesn't know  
>> what to
>> do with this number it then disconnects the call.
>>
>> 2009-08-12 11:25:07.599250 [NOTICE] mod_dptools.c:649 Channel
>> [sofia/internal/sip:1000 at 10.107.181.160:42840] has been answered
>> 2009-08-12 11:25:07.599250 [NOTICE] switch_ivr_originate.c:2015  
>> Channel
>> [sofia/internal/1003 at 10.107.249.12] has been answered
>> 2009-08-12 11:25:07.614875 [ERR] switch_core_io.c:118
>> sofia/internal/sip:1000 at 10.107.181.160:42840 has no read codec.
>> 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:503 Hangup
>> sofia/internal/sip:1000 at 10.107.181.160:42840 [CS_EXCHANGE_MEDIA]
>> [NORMAL_CLEARING]
>> 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:1016 Hangup
>> sofia/internal/1003 at 10.107.249.12 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-08-12 11:25:07.630500 [NOTICE] switch_core_session.c:1086  
>> Session 133
>> (sofia/internal/sip:1000 at 10.107.181.160:42840) Ended
>>
>>
>> Maybe there is the way to acknowledge SIP client that call was  
>> answered?
>>
>> Regards,
>> Maxim Tsvetov
>>
>> Diego Viola wrote:
>>>
>>> I suggest that you learn the differences between mod_commands  
>>> commands and
>>> mod_dptools applications, and also the interfaces where you can  
>>> access and
>>> use them.
>>>
>>> As said before, mod_dptools is accessible from dialplan, event socket
>>> outbound, etc. and mod_commands is accessible from the CLI, event  
>>> socket
>>> (inbound/outbound), XML RPC, etc.
>>>
>>> That's all described in the wiki I think.
>>>
>>> Let us know if you have any questions =D.
>>>
>>> On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola <diego.viola at gmail.com>
>>> wrote:
>>>
>>>> Michael, you're welcome :).
>>>>
>>>> Milena, answer is a mod_dptools command, you can use it from the XML
>>>> dialplan or from the event socket outbound. mod_commands API are  
>>>> APIs
>>>> that
>>>> you execute from the socket, event socket inbound, etc. But you  
>>>> can also
>>>> execute them from event socket outbound using the "api" command.
>>>>
>>>> I hope that makes sense, correct me if I'm wrong =D.
>>>>
>>>> On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins
>>>> <msc at freeswitch.org>wrote:
>>>>
>>>>>
>>>>>
>>>>> On Tue, Aug 11, 2009 at 9:05 AM, Milena <testeador01 at gmail.com>  
>>>>> wrote:
>>>>>
>>>>>>
>>>>>> Hello Brian,
>>>>>>
>>>>>> I wanna fix the wiki, but to make sure i got it right, does it  
>>>>>> only
>>>>>> work
>>>>>> on outbound event socket? or is there any other scenario where  
>>>>>> it would
>>>>>> work.
>>>>>>
>>>>>
>>>>> FYI, Diego Viola fixed the wiki. (Thanks Diego!)
>>>>> -MC
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> -- 
>> View this message in context:
>> http://www.nabble.com/answer-command-tp24912812p24931876.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/answer-command-tp24912812p24940548.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Wed Aug 12 09:57:01 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 11:57:01 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <24940548.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
Message-ID: <B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>

Well you can only truly answer an inbound call to FS... you can't  
force answer an outbound call.

/b

On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:

>
> I will try to paraphrase my question.
> Is there any possibility to answer call  from CTI application and
> synchronise answer with answer in SIP client?Maybe we can use SIP  
> functions
> in our CTI application instead of FS api commands?
> I'm trying to find the way to make prototype of lineAnswer command  
> in TAPI.



From juanbackson at gmail.com  Wed Aug 12 10:21:07 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 13 Aug 2009 01:21:07 +0800
Subject: [Freeswitch-users] random route selection
Message-ID: <27c25bc40908121021r6b108b1fye42ca4e3970ee93a@mail.gmail.com>

Hi,

I would like to implement a random route selection based on some arbitrary
percentage.

Does anyone know if there is any good way of doing that within freeswitch?

If there isn't any api that I can use, does freeswitch has any random
generator that I can be used for this purpose?

br,
JB
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From maxim.tsvetov at gmail.com  Wed Aug 12 10:38:22 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Wed, 12 Aug 2009 10:38:22 -0700 (PDT)
Subject: [Freeswitch-users] answer command
In-Reply-To: <B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
	<B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
Message-ID: <24941422.post@talk.nabble.com>


If I have two FS extensions A and B. I'm calling from A to B and want to
answer from B-side in my CTI application and to make SIP phone to be
synchronised to my CTI application. Is it possible to do it?


Brian West-3 wrote:
> 
> Well you can only truly answer an inbound call to FS... you can't  
> force answer an outbound call.
> 
> /b
> 
> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
> 
>>
>> I will try to paraphrase my question.
>> Is there any possibility to answer call  from CTI application and
>> synchronise answer with answer in SIP client?Maybe we can use SIP  
>> functions
>> in our CTI application instead of FS api commands?
>> I'm trying to find the way to make prototype of lineAnswer command  
>> in TAPI.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/answer-command-tp24912812p24941422.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From rupa at rupa.com  Wed Aug 12 10:42:56 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 12 Aug 2009 12:42:56 -0500
Subject: [Freeswitch-users] random route selection
In-Reply-To: <27c25bc40908121021r6b108b1fye42ca4e3970ee93a@mail.gmail.com>
References: <27c25bc40908121021r6b108b1fye42ca4e3970ee93a@mail.gmail.com>
Message-ID: <b39273530908121042v18f9d1c0ne8c551bd1b416dd2@mail.gmail.com>

mod_lcr will do random route selection if the rates are the same.  But
that gives an equal distribution.  There is no weighting/percentage
supported.

On Wed, Aug 12, 2009 at 12:21 PM, Juan Backson<juanbackson at gmail.com> wrote:
> Hi,
>
> I would like to implement a random route selection based on some arbitrary
> percentage.
>
> Does anyone know if there is any good way of doing that within freeswitch?
>
> If there isn't any api that I can use, does freeswitch has any random
> generator that I can be used for this purpose?
>
> br,
> JB
>



-- 
-Rupa


From mike at jerris.com  Wed Aug 12 10:50:13 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 12 Aug 2009 10:50:13 -0700
Subject: [Freeswitch-users] answer command
In-Reply-To: <24941422.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
	<B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
	<24941422.post@talk.nabble.com>
Message-ID: <6539128A-D7A1-4555-A3DB-60E94E461DB3@jerris.com>

Sip does not support this functionality.  The called device would have  
to support this via some other mechanism such as ctsa which I have  
seen recently someone was looking at for freeswitch.  So the first  
issue you must resolve is the called device needs to support some way  
to do this.

Mike

On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov <maxim.tsvetov at gmail.com>  
wrote:

>
> If I have two FS extensions A and B. I'm calling from A to B and  
> want to
> answer from B-side in my CTI application and to make SIP phone to be
> synchronised to my CTI application. Is it possible to do it?
>
>
> Brian West-3 wrote:
>>
>> Well you can only truly answer an inbound call to FS... you can't
>> force answer an outbound call.
>>
>> /b
>>
>> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
>>
>>>
>>> I will try to paraphrase my question.
>>> Is there any possibility to answer call  from CTI application and
>>> synchronise answer with answer in SIP client?Maybe we can use SIP
>>> functions
>>> in our CTI application instead of FS api commands?
>>> I'm trying to find the way to make prototype of lineAnswer command
>>> in TAPI.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://www.nabble.com/answer-command-tp24912812p24941422.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From alan at chandlerfamily.org.uk  Wed Aug 12 10:56:19 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Wed, 12 Aug 2009 18:56:19 +0100
Subject: [Freeswitch-users] Confused about conferences
Message-ID: <4A830243.8080703@chandlerfamily.org.uk>

I have been reading all the docs about conferences I can find and am 
getting somewhat confused.  What I am trying to do is set up a dialplan 
where I have subscribers with extensions in the 1xx range, and then to 
set an ability to have a series of conference rooms for each subscriber 
in the 21xx range where if the user enters is "own" conference he is the 
moderator, but if not he is just a normal user.

I want to be able for the moderator to do things like mute or kick people.

So dialplan would probably have something like this in it



     <extension name="user_conference">
       <condition field="destination_number" 
expression="^(2${caller_id_number})$">
         <action application="set" data="i_am_moderator=true">
	<action application="answer"/>
	<action application="conference" data="$1 at default+flags{moderator}"/>
       </condition>
       <condition field="destination_number" expression="^(21\d{2})^">
         <action application="set" data="i_am_moderator=false">
	<action application="answer"/>
	<action application="conference" data="$1 at default"/>
       </condition>
     </extension>
     <extension name="kick">
       <condition field="i_am_moderator" expression="true">
         <action application="play_and_get_digits" data="3 3 3 7000 # 
/ask-for-extension.wav /invalid.wav conf-user-id \d+" >

WHAT GOES HERE???

       </condition>
     </extension>


in the conference.conf.xml file, I would change the caller controls to 
include

   <caller-controls>
     <group name="somekeys">
        <control action="transfer" digits="9"   data="kick XML default"/>
     </group>
   </caller-controls>


My question (at the moment) is

In the WHAT GOES HERE place how do it

Kick extension ${conf-user-id}  (DOES IT REQUIRE A SCRIPT TO CALL THE 
Conference API?)
Re-Enter the moderator back into the conference
Re-Enter the ordinary user who happened to press 9 back into the conference

I am assuming I can't stop the non moderator getting the control - since 
all users get the same controls.


-- 
Alan Chandler
http://www.chandlerfamily.org.uk



From testeador01 at gmail.com  Wed Aug 12 10:59:08 2009
From: testeador01 at gmail.com (Milena)
Date: Wed, 12 Aug 2009 12:59:08 -0500
Subject: [Freeswitch-users] fs installation is broken in version: FreeSWITCH
	Version 1.0.trunk (14500M)
Message-ID: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>

I just did a rebootstrap on a fs box, it turned out the new revision has
this at the end of mod_sofia.h:

> char *sofia_glue_get_extra_headers(switch_channel_t *channel, const char
> *prefix);

<<<<<<< .mine

void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const
> *sip, const char *prefix);


> =======

void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const
> *sip, const char *prefix);

>>>>>>> .r14490


no wonder why it wouldn't compile :P
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From brian at freeswitch.org  Wed Aug 12 11:18:06 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 13:18:06 -0500
Subject: [Freeswitch-users] fs installation is broken in version:
	FreeSWITCH Version 1.0.trunk (14500M)
In-Reply-To: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
Message-ID: <B8137064-55DE-48C9-87C3-A9BFE6DF3B3E@freeswitch.org>

You have a merge conflict. svn revert sofia_glue.c

/b

On Aug 12, 2009, at 12:59 PM, Milena wrote:

> I just did a rebootstrap on a fs box, it turned out the new revision  
> has this at the end of mod_sofia.h:
>
> char *sofia_glue_get_extra_headers(switch_channel_t *channel, const  
> char *prefix);
> <<<<<<< .mine
> void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t  
> const *sip, const char *prefix);
>
> =======
> void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t  
> const *sip, const char *prefix);
> >>>>>>> .r14490
>
> no wonder why it wouldn't compile :P



From krice at suspicious.org  Wed Aug 12 11:20:42 2009
From: krice at suspicious.org (Ken Rice)
Date: Wed, 12 Aug 2009 13:20:42 -0500
Subject: [Freeswitch-users] random route selection
In-Reply-To: <27c25bc40908121021r6b108b1fye42ca4e3970ee93a@mail.gmail.com>
Message-ID: <C6A8722A.5AC41%krice@suspicious.org>

Write you a small C app to randomly return them based on the percentages...
Currently we do something similar to this but use a random round robin based
thing using a simple sql backend and doing a select order by random sort of
thing... 

Contact me off list if you need some profession help figuring this out

K



From: Juan Backson <juanbackson at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Thu, 13 Aug 2009 01:21:07 +0800
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] random route selection

Hi,

I would like to implement a random route selection based on some arbitrary
percentage.? 

Does anyone know if there is any good way of doing that within freeswitch?

If there isn't any api that I can use, does freeswitch has any random
generator that I can be used for this purpose?

br,
JB


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From mike at jerris.com  Wed Aug 12 11:22:41 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 12 Aug 2009 11:22:41 -0700
Subject: [Freeswitch-users] fs installation is broken in version:
	FreeSWITCH Version 1.0.trunk (14500M)
In-Reply-To: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
Message-ID: <D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com>

The M in the version number means modified.  You had local code  
changes that conflicted when you updated trunk.  Revert the changes to  
that file and it should be fine.

Mike

On Aug 12, 2009, at 10:59 AM, Milena <testeador01 at gmail.com> wrote:

> I just did a rebootstrap on a fs box, it turned out the new revision  
> has this at the end of mod_sofia.h:
>
> char *sofia_glue_get_extra_headers(switch_channel_t *channel, const  
> char *prefix);
> <<<<<<< .mine
> void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t  
> const *sip, const char *prefix);
>
> =======
> void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t  
> const *sip, const char *prefix);
> >>>>>>> .r14490
>
> no wonder why it wouldn't compile :P
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From alan at chandlerfamily.org.uk  Wed Aug 12 11:25:21 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Wed, 12 Aug 2009 19:25:21 +0100
Subject: [Freeswitch-users] The application "hash"
Message-ID: <4A830911.1060004@chandlerfamily.org.uk>

I was trying to explore the documentation for the application "hash" 
which is in the default dialplan.  Its vaguely obvious what its doing, 
but I wanted to be sure.

It appears to be listed as a application under dp_tools, but when I 
click on it I get taken to a page that talks about limit_hash rather 
than hash.

Is there any documentation for this?  Am I looking in the wring place?
-- 
Alan Chandler
http://www.chandlerfamily.org.uk



From testeador01 at gmail.com  Wed Aug 12 11:28:23 2009
From: testeador01 at gmail.com (Milena)
Date: Wed, 12 Aug 2009 13:28:23 -0500
Subject: [Freeswitch-users] fs installation is broken in version:
	FreeSWITCH Version 1.0.trunk (14500M)
In-Reply-To: <D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
	<D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com>
Message-ID: <c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>

It's modified because it wouldn't compile with those "<<<<<<" at the end of
the file

2009/8/12 Michael Jerris <mike at jerris.com>

> The M in the version number means modified.  You had local code
> changes that conflicted when you updated trunk.  Revert the changes to
> that file and it should be fine.
>
> Mike
>
> On Aug 12, 2009, at 10:59 AM, Milena <testeador01 at gmail.com> wrote:
>
> > I just did a rebootstrap on a fs box, it turned out the new revision
> > has this at the end of mod_sofia.h:
> >
> > char *sofia_glue_get_extra_headers(switch_channel_t *channel, const
> > char *prefix);
> > <<<<<<< .mine
> > void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t
> > const *sip, const char *prefix);
> >
> > =======
> > void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t
> > const *sip, const char *prefix);
> > >>>>>>> .r14490
> >
> > no wonder why it wouldn't compile :P
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Aug 12 11:29:23 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 13:29:23 -0500
Subject: [Freeswitch-users] The application "hash"
In-Reply-To: <4A830911.1060004@chandlerfamily.org.uk>
References: <4A830911.1060004@chandlerfamily.org.uk>
Message-ID: <87FBD426-C373-49CF-8E12-1539796F4146@freeswitch.org>

hash is just like db but its all in memory.. you can interchange db  
and hash.

/b

On Aug 12, 2009, at 1:25 PM, Alan Chandler wrote:

> I was trying to explore the documentation for the application "hash"
> which is in the default dialplan.  Its vaguely obvious what its doing,
> but I wanted to be sure.
>
> It appears to be listed as a application under dp_tools, but when I
> click on it I get taken to a page that talks about limit_hash rather
> than hash.
>
> Is there any documentation for this?  Am I looking in the wring place?
> -- 
> Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From testeador01 at gmail.com  Wed Aug 12 11:34:42 2009
From: testeador01 at gmail.com (Milena)
Date: Wed, 12 Aug 2009 13:34:42 -0500
Subject: [Freeswitch-users] fs installation is broken in version:
	FreeSWITCH Version 1.0.trunk (14500M)
In-Reply-To: <c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
	<D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com>
	<c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>
Message-ID: <c29b930b0908121134m50b1c8b2j3a91b57e71b70b10@mail.gmail.com>

Ok, done and fixed, thank you very much :)

2009/8/12 Milena <testeador01 at gmail.com>

> It's modified because it wouldn't compile with those "<<<<<<" at the end of
> the file
>
> 2009/8/12 Michael Jerris <mike at jerris.com>
>
> The M in the version number means modified.  You had local code
>> changes that conflicted when you updated trunk.  Revert the changes to
>> that file and it should be fine.
>>
>> Mike
>>
>> On Aug 12, 2009, at 10:59 AM, Milena <testeador01 at gmail.com> wrote:
>>
>> > I just did a rebootstrap on a fs box, it turned out the new revision
>> > has this at the end of mod_sofia.h:
>> >
>> > char *sofia_glue_get_extra_headers(switch_channel_t *channel, const
>> > char *prefix);
>> > <<<<<<< .mine
>> > void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t
>> > const *sip, const char *prefix);
>> >
>> > =======
>> > void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t
>> > const *sip, const char *prefix);
>> > >>>>>>> .r14490
>> >
>> > no wonder why it wouldn't compile :P
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From Christian.Jensen at Teligence.Net  Wed Aug 12 12:18:05 2009
From: Christian.Jensen at Teligence.Net (Christian Jensen)
Date: Wed, 12 Aug 2009 12:18:05 -0700
Subject: [Freeswitch-users] ClueCon Presentations - Where?
In-Reply-To: <c29b930b0908121134m50b1c8b2j3a91b57e71b70b10@mail.gmail.com>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com><D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com><c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>
	<c29b930b0908121134m50b1c8b2j3a91b57e71b70b10@mail.gmail.com>
Message-ID: <F571A97F57A2D946B229C8231805867D32AD4527@svanexcsvr01.alldir.net>

Hi there,

 

Does anyone have the URL for where I might find all the electronic
versions of the presentations made at ClueCon last week?

 

Thanks!

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From brian at freeswitch.org  Wed Aug 12 12:25:50 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 14:25:50 -0500
Subject: [Freeswitch-users] ClueCon Presentations - Where?
In-Reply-To: <F571A97F57A2D946B229C8231805867D32AD4527@svanexcsvr01.alldir.net>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com><D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com><c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>
	<c29b930b0908121134m50b1c8b2j3a91b57e71b70b10@mail.gmail.com>
	<F571A97F57A2D946B229C8231805867D32AD4527@svanexcsvr01.alldir.net>
Message-ID: <65BC9732-5836-48F4-83AC-B305F4D1028C@freeswitch.org>

They are all getting gathered up and put online...  
files.freeswitch.org/cluecon_2009 just keep an eye there some of the  
videos are up also.

/b

On Aug 12, 2009, at 2:18 PM, Christian Jensen wrote:

> Hi there,
>
> Does anyone have the URL for where I might find all the electronic  
> versions of the presentations made at ClueCon last week?
>
> Thanks!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jaugenstine at gmail.com  Wed Aug 12 12:57:28 2009
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Wed, 12 Aug 2009 12:57:28 -0700
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <1249685648.16901.34.camel@dk-d820>
References: <1249685648.16901.34.camel@dk-d820>
Message-ID: <207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>

I to would like to put my thanks on the table.  I have been going to
conferences for a very long time and often question the value of taking time
off to attend these venues.  When I was asked to attend by a client I was
very hesitant.  I am very pleased that I decided to attend.
Now the skeptical among you may say that I am only pleased because I won the
MacBook.  I cannot deny there is truth in that statement.  However, on
Thursday morning I was sitting in the conference room waiting for the first
presentation.  I was thinking that it had been a valuable three days.  I had
been able to connect with some clients at the conference, made some new
contacts, met people face to face that I had only met online, listened to
good presentations, learned some valuable new information, and lastly
received some insight from Anthony and Michael Jerris on fixing a bug that
had been plaguing me for sometime.
As I sat pondering, I clicked on the picture of the MacBook and thought the
only way the conference could end better was if I won the MacBook.  Never
did I actually think that would happen and the conference did end better.  I
will be much more motivated to consider attending next year, even knowing
lightening does not strike the same spot twice.

Jonathan Augenstine

On Fri, Aug 7, 2009 at 3:54 PM, David Knell <dave at 3c.co.uk> wrote:

> Just a quick note to say thanks to Cluecon's organisers for putting
> together such a useful, informative and packed three days.  I've come
> away with a head full of ideas, a bunch of new contacts and a collection
> of things to do; I'd thoroughly recommend that anyone interested in IP
> telephony blocks out the first week of August 2010, right now..!
>
> Cheers --
>
> Dave
>
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Aug 12 13:03:59 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 15:03:59 -0500
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
References: <1249685648.16901.34.camel@dk-d820>
	<207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
Message-ID: <74BD91EA-8D4B-44DD-9CA3-C0B3E3031E32@freeswitch.org>

Remember next year we'll have more Mac Book's to give away and iPod  
Touches with engraved sponsor logos on them.  :)

/b

On Aug 12, 2009, at 2:57 PM, jonathan augenstine wrote:

> I to would like to put my thanks on the table.  I have been going to  
> conferences for a very long time and often question the value of  
> taking time off to attend these venues.  When I was asked to attend  
> by a client I was very hesitant.  I am very pleased that I decided  
> to attend.
>
> Now the skeptical among you may say that I am only pleased because I  
> won the MacBook.  I cannot deny there is truth in that statement.   
> However, on Thursday morning I was sitting in the conference room  
> waiting for the first presentation.  I was thinking that it had been  
> a valuable three days.  I had been able to connect with some clients  
> at the conference, made some new contacts, met people face to face  
> that I had only met online, listened to good presentations, learned  
> some valuable new information, and lastly received some insight from  
> Anthony and Michael Jerris on fixing a bug that had been plaguing me  
> for sometime.
>
> As I sat pondering, I clicked on the picture of the MacBook and  
> thought the only way the conference could end better was if I won  
> the MacBook.  Never did I actually think that would happen and the  
> conference did end better.  I will be much more motivated to  
> consider attending next year, even knowing lightening does not  
> strike the same spot twice.
>
> Jonathan Augenstine



From Prometheus001 at gmx.net  Wed Aug 12 13:04:52 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 12 Aug 2009 22:04:52 +0200
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
Message-ID: <4A832064.8000908@gmx.net>

Hello,

anybody has a clue what this message means?
  [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from
PROGRESS_MEDIA to PROGRESS
What does VETO mean here?

Best regards
Peter


From brian at freeswitch.org  Wed Aug 12 13:12:22 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 15:12:22 -0500
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <1249685648.16901.34.camel@dk-d820>
References: <1249685648.16901.34.camel@dk-d820>
Message-ID: <378EC23A-D23A-4CFC-98DB-693946CF91E6@freeswitch.org>

Dave,
	Thanks,	Hope to see you there next year...

/b

On Aug 7, 2009, at 5:54 PM, David Knell wrote:

> Just a quick note to say thanks to Cluecon's organisers for putting
> together such a useful, informative and packed three days.  I've come
> away with a head full of ideas, a bunch of new contacts and a  
> collection
> of things to do; I'd thoroughly recommend that anyone interested in IP
> telephony blocks out the first week of August 2010, right now..!
>
> Cheers --
>
> Dave



From tina at a2unlimited.com  Wed Aug 12 12:55:35 2009
From: tina at a2unlimited.com (Tina Martinez)
Date: Wed, 12 Aug 2009 15:55:35 -0400
Subject: [Freeswitch-users] Question about sharing conference between servers
Message-ID: <55518.1250106935@a2unlimited.com>

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From terrymr at gmail.com  Wed Aug 12 13:25:56 2009
From: terrymr at gmail.com (Terry Moore-Read)
Date: Wed, 12 Aug 2009 13:25:56 -0700
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
References: <1249685648.16901.34.camel@dk-d820>
	<207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
Message-ID: <2d9dff7e0908121325g664dd580j9304d316ce4183ee@mail.gmail.com>

Macbook ... that's nothing,  I got $1500 worth of coffee :-)


On Wed, Aug 12, 2009 at 12:57 PM, jonathan
augenstine<jaugenstine at gmail.com> wrote:
> I to would like to put my thanks on the table. ?I have been going to
> conferences for a very long time and often question the value of taking time
> off to attend these venues. ?When I was asked to attend by a client I was
> very hesitant. ?I am very pleased that I decided to attend.
> Now the skeptical among you may say that I am only pleased because I won the
> MacBook. ?I cannot deny there is truth in that statement. ?However, on
> Thursday morning I was sitting in the conference room waiting for the first
> presentation. ?I was thinking that it had been a valuable three days. ?I had
> been able to connect with some clients at the conference, made some new
> contacts, met people face to face that I had only met online, listened to
> good presentations, learned some valuable new information, and lastly
> received some insight from Anthony and Michael Jerris on fixing a bug that
> had been plaguing me for sometime.
> As I sat pondering,?I clicked on the picture of the MacBook and thought the
> only way the conference could end better was if I won the MacBook. ?Never
> did I actually think that would happen and the conference did end better. ?I
> will be much more motivated to consider attending next year, even knowing
> lightening does not strike the same spot twice.
> Jonathan Augenstine
> On Fri, Aug 7, 2009 at 3:54 PM, David Knell <dave at 3c.co.uk> wrote:
>>
>> Just a quick note to say thanks to Cluecon's organisers for putting
>> together such a useful, informative and packed three days. ?I've come
>> away with a head full of ideas, a bunch of new contacts and a collection
>> of things to do; I'd thoroughly recommend that anyone interested in IP
>> telephony blocks out the first week of August 2010, right now..!
>>
>> Cheers --
>>
>> Dave
>>
>> --
>> David Knell, Director, 3C Limited
>> T: +44 20 3298 2000
>> E: dave at 3c.co.uk
>> W: http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Want to buy my photo's ? :
http://www.shutterstock.com/gallery.mhtml?id=309295&rid=309295


From moises.silva at gmail.com  Wed Aug 12 13:33:35 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 12 Aug 2009 16:33:35 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <4A832064.8000908@gmx.net>
References: <4A832064.8000908@gmx.net>
Message-ID: <c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>

On Wed, Aug 12, 2009 at 4:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> anybody has a clue what this message means?
>  [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from
> PROGRESS_MEDIA to PROGRESS
> What does VETO mean here?
>
> Best regards
> Peter
>

Means that state transition should not occur. The only thing that it would
cause that (I think) is a bug in the the openzap code.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From brian at freeswitch.org  Wed Aug 12 13:42:08 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 15:42:08 -0500
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <2d9dff7e0908121325g664dd580j9304d316ce4183ee@mail.gmail.com>
References: <1249685648.16901.34.camel@dk-d820>
	<207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
	<2d9dff7e0908121325g664dd580j9304d316ce4183ee@mail.gmail.com>
Message-ID: <FCF9FC37-0377-4947-BEC7-9264166574BF@freeswitch.org>

So you're the one that drank 16 gallons of coffee!  Good luck sleeping!

/b

On Aug 12, 2009, at 3:25 PM, Terry Moore-Read wrote:

> Macbook ... that's nothing,  I got $1500 worth of coffee :-)



From brian at freeswitch.org  Wed Aug 12 13:42:47 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 15:42:47 -0500
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
Message-ID: <15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>

Isn't progress_media already past progress in the state machine?  so  
the state machine can't move backwards in states right?

/b

On Aug 12, 2009, at 3:33 PM, Moises Silva wrote:

> On Wed, Aug 12, 2009 at 4:04 PM, Peter P GMX <Prometheus001 at gmx.net>  
> wrote:
> Hello,
>
> anybody has a clue what this message means?
>  [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from
> PROGRESS_MEDIA to PROGRESS
> What does VETO mean here?
>
> Best regards
> Peter
>
> Means that state transition should not occur. The only thing that it  
> would cause that (I think) is a bug in the the openzap code.
>

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From moises.silva at gmail.com  Wed Aug 12 13:57:37 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 12 Aug 2009 16:57:37 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
Message-ID: <c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>

Correct, so the question is why ozmod_libpri attempting to move from
progress_media to progress ... may be a delayed libpri event? or some crap
along those lines.

On Wed, Aug 12, 2009 at 4:42 PM, Brian West <brian at freeswitch.org> wrote:

> Isn't progress_media already past progress in the state machine?  so the
> state machine can't move backwards in states right?
> /b
>
> On Aug 12, 2009, at 3:33 PM, Moises Silva wrote:
>
> On Wed, Aug 12, 2009 at 4:04 PM, Peter P GMX <Prometheus001 at gmx.net>
> wrote:
>
>> Hello,
>>
>> anybody has a clue what this message means?
>>  [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from
>> PROGRESS_MEDIA to PROGRESS
>> What does VETO mean here?
>>
>> Best regards
>> Peter
>>
>
> Means that state transition should not occur. The only thing that it would
> cause that (I think) is a bug in the the openzap code.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From brian at freeswitch.org  Wed Aug 12 14:01:49 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 16:01:49 -0500
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
Message-ID: <90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>

Yes you can get a progress after you get a progress with media ... I  
have seen it.

/b

On Aug 12, 2009, at 3:57 PM, Moises Silva wrote:

> Correct, so the question is why ozmod_libpri attempting to move from  
> progress_media to progress ... may be a delayed libpri event? or  
> some crap along those lines.



From msc at freeswitch.org  Wed Aug 12 14:07:52 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Aug 2009 16:07:52 -0500
Subject: [Freeswitch-users] ClueCon Presentations - Where?
In-Reply-To: <65BC9732-5836-48F4-83AC-B305F4D1028C@freeswitch.org>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
	<D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com>
	<c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>
	<c29b930b0908121134m50b1c8b2j3a91b57e71b70b10@mail.gmail.com>
	<F571A97F57A2D946B229C8231805867D32AD4527@svanexcsvr01.alldir.net>
	<65BC9732-5836-48F4-83AC-B305F4D1028C@freeswitch.org>
Message-ID: <87f2f3b90908121407n77efcde2v374008b07af77e92@mail.gmail.com>

On Wed, Aug 12, 2009 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:

> They are all getting gathered up and put online...
> files.freeswitch.org/cluecon_2009 just keep an eye there some of the
> videos are up also.
> /b
>

FYI,

I've uploaded the first batch and they should get synched up on
files.freeswitch.org/cluecon_2009/presentations any time...
-MC
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From msc at freeswitch.org  Wed Aug 12 14:10:39 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Aug 2009 16:10:39 -0500
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
Message-ID: <87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>

On Wed, Aug 12, 2009 at 4:01 PM, Brian West <brian at freeswitch.org> wrote:

> Yes you can get a progress after you get a progress with media ... I
> have seen it.
>

Yes, you definitely can and I believe that some of the PRI specs suggest
that this is totally legal, even though it's kind of silly.
-MC
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From terrymr at gmail.com  Wed Aug 12 14:27:19 2009
From: terrymr at gmail.com (Terry Moore-Read)
Date: Wed, 12 Aug 2009 14:27:19 -0700
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <FCF9FC37-0377-4947-BEC7-9264166574BF@freeswitch.org>
References: <1249685648.16901.34.camel@dk-d820>
	<207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
	<2d9dff7e0908121325g664dd580j9304d316ce4183ee@mail.gmail.com>
	<FCF9FC37-0377-4947-BEC7-9264166574BF@freeswitch.org>
Message-ID: <2d9dff7e0908121427u33151adcj2bf9fb1968338b21@mail.gmail.com>

That's the trouble with a 8am conference in a town where the bars
close at 4am :-)

On Wed, Aug 12, 2009 at 1:42 PM, Brian West<brian at freeswitch.org> wrote:
> So you're the one that drank 16 gallons of coffee! ?Good luck sleeping!
>
> /b
>
> On Aug 12, 2009, at 3:25 PM, Terry Moore-Read wrote:
>
>> Macbook ... that's nothing, ?I got $1500 worth of coffee :-)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Want to buy my photo's ? :
http://www.shutterstock.com/gallery.mhtml?id=309295&rid=309295


From brian at freeswitch.org  Wed Aug 12 14:30:55 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 16:30:55 -0500
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <2d9dff7e0908121427u33151adcj2bf9fb1968338b21@mail.gmail.com>
References: <1249685648.16901.34.camel@dk-d820>
	<207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
	<2d9dff7e0908121325g664dd580j9304d316ce4183ee@mail.gmail.com>
	<FCF9FC37-0377-4947-BEC7-9264166574BF@freeswitch.org>
	<2d9dff7e0908121427u33151adcj2bf9fb1968338b21@mail.gmail.com>
Message-ID: <82A46A35-0CFF-4A5F-8D5A-CF013937ABB0@freeswitch.org>

And it didn't help we had an open bar two of the nights!

/b

On Aug 12, 2009, at 4:27 PM, Terry Moore-Read wrote:

> That's the trouble with a 8am conference in a town where the bars
> close at 4am :-)



From gmaruzz at celliax.org  Wed Aug 12 14:40:30 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 12 Aug 2009 23:40:30 +0200
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <82A46A35-0CFF-4A5F-8D5A-CF013937ABB0@freeswitch.org>
References: <1249685648.16901.34.camel@dk-d820>
	<207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com> 
	<2d9dff7e0908121325g664dd580j9304d316ce4183ee@mail.gmail.com> 
	<FCF9FC37-0377-4947-BEC7-9264166574BF@freeswitch.org>
	<2d9dff7e0908121427u33151adcj2bf9fb1968338b21@mail.gmail.com> 
	<82A46A35-0CFF-4A5F-8D5A-CF013937ABB0@freeswitch.org>
Message-ID: <7b197bef0908121440o57fde283l7cf1f7566781f730@mail.gmail.com>

it helped me!
oh... well, I helped myself!

-giovanni


On Wed, Aug 12, 2009 at 11:30 PM, Brian West<brian at freeswitch.org> wrote:
> And it didn't help we had an open bar two of the nights!
>
> /b
>
> On Aug 12, 2009, at 4:27 PM, Terry Moore-Read wrote:
>
>> That's the trouble with a 8am conference in a town where the bars
>> close at 4am :-)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From bjbrashier at gmail.com  Wed Aug 12 14:40:02 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Wed, 12 Aug 2009 14:40:02 -0700
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <4A830243.8080703@chandlerfamily.org.uk>
References: <4A830243.8080703@chandlerfamily.org.uk>
Message-ID: <7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>

You've thought through some of the difficult points, which is good. You're
right that the moderator can't have different controls (unless you're
controlling the conference yourself from outside, using, say, the event
socket).

Before I go further, I want to make sure I understand what you're proposing.
What you're essentially saying is that when the command to kick someone is
pressed the person should be transferred out of the conference, checked for
moderator status, asked whom to kick (if so), and then let back in while the
system kicks that person. The other commands would work similarly. Does this
sound like a correct summary?

If so, I think what you've got is mostly right, but I'm not sure it will
work, having not tried to do it that way myself (I went the event socket
route). In theory, it should look something like this:

In the scripts entering the conference, you'll need to save the current
conference number. Something like (in both cases):

        <action application="set" data="conf-id=$1"/>

"Extension" doesn't actually check for anything, so you'll need to check
for "kick" with a condition. Try this:

    <extension name="kick">
      <condition field="destination_number" expression="^kick$"/>
      <condition field="$(i_am_moderator)" expression="true">
        <action application="play_and_get_digits" data="3 3 3 7000 #
/ask-for-extension.wav /invalid.wav conf-user-id \d+" >

Followed by (I picked up the "kick" syntax from the wiki):

        <action application="conference" data="$(conf-id) kick
$(conf-user-id)"/>
        <action application="conference"
data="$(conf-id)@default+flags{mod}"/>
        <anti-action application="conference" data="$(conf-id)@default"/>
      </condition>
    </extension>

In theory, then, you've transferred out with the "9" key, you check the
moderator flag you've made, and do stuff based on the conf-id and the data
you enter.  Obviously, if this works, the moderator won't be able to hear
what's going on in the conference while he's entering stuff.

Bigger problem: the conf-user-id he enters has to be the member ID that FS
chose for the user he's trying to kick. That number will make sense if
you're following the system closely, but for someone who doesn't know FS
internals, it will be impossible to know unless you broadcast it to him
somehow. To my knowledge, there's no way to use any other identifier (like
caller-id) to kick them with.

BB

On Wed, Aug 12, 2009 at 10:56 AM, Alan Chandler
<alan at chandlerfamily.org.uk>wrote:

> I have been reading all the docs about conferences I can find and am
> getting somewhat confused.  What I am trying to do is set up a dialplan
> where I have subscribers with extensions in the 1xx range, and then to
> set an ability to have a series of conference rooms for each subscriber
> in the 21xx range where if the user enters is "own" conference he is the
> moderator, but if not he is just a normal user.
>
> I want to be able for the moderator to do things like mute or kick people.
>
> So dialplan would probably have something like this in it
>
>
>
>     <extension name="user_conference">
>       <condition field="destination_number"
> expression="^(2${caller_id_number})$">
>         <action application="set" data="i_am_moderator=true">
>        <action application="answer"/>
>        <action application="conference" data="$1 at default
> +flags{moderator}"/>
>       </condition>
>       <condition field="destination_number" expression="^(21\d{2})^">
>         <action application="set" data="i_am_moderator=false">
>        <action application="answer"/>
>        <action application="conference" data="$1 at default"/>
>       </condition>
>     </extension>
>     <extension name="kick">
>       <condition field="i_am_moderator" expression="true">
>         <action application="play_and_get_digits" data="3 3 3 7000 #
> /ask-for-extension.wav /invalid.wav conf-user-id \d+" >
>
> WHAT GOES HERE???
>
>       </condition>
>     </extension>
>
>
> in the conference.conf.xml file, I would change the caller controls to
> include
>
>   <caller-controls>
>     <group name="somekeys">
>        <control action="transfer" digits="9"   data="kick XML default"/>
>     </group>
>   </caller-controls>
>
>
> My question (at the moment) is
>
> In the WHAT GOES HERE place how do it
>
> Kick extension ${conf-user-id}  (DOES IT REQUIRE A SCRIPT TO CALL THE
> Conference API?)
> Re-Enter the moderator back into the conference
> Re-Enter the ordinary user who happened to press 9 back into the conference
>
> I am assuming I can't stop the non moderator getting the control - since
> all users get the same controls.
>
>
> --
> Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Aug 12 14:41:26 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Aug 2009 16:41:26 -0500
Subject: [Freeswitch-users] Question about sharing conference between
	servers
In-Reply-To: <55518.1250106935@a2unlimited.com>
References: <55518.1250106935@a2unlimited.com>
Message-ID: <87f2f3b90908121441p741e73cco4a947c0115722082@mail.gmail.com>

On Wed, Aug 12, 2009 at 2:55 PM, Tina Martinez <tina at a2unlimited.com> wrote:

> Hello,
>
> I have spent the past couple of weeks toying around with FS to evaluate the
> possibility of using it for a large scale conference server for our
> organization.  The plan is to have several FS servers initiate calls to
> participants and connect them together, but not have to transfer all of the
> calls to one server supporting the conference.  My problem is that I do not
> see a simple way to link the servers together.  Has anyone done anything
> like this, or know of a way to create a connection between servers without
> an actual caller on the line?  I started to go down a path of registering a
> soft-phone on each machine and place a call those extensions prior to
> initiating the conference call -- then connecting, but that feels like a
> kludge.
>
> Any thoughts, suggestions or guidance would be greatly appreciated.
> T


Tina,

Welcome to the FreeSWITCH community! I'm pretty sure that FS can do what you
need. My first question to you is this: do envision having some sort of call
control mechanism, like a script or something that knows which callers to
connect to and which ones to put into conferences, etc.? The reason I ask is
that controlling calls using the event-socket allows for great power and
flexibility.

As for "connecting" the FS servers you have several options. You can use
ACLs to allow the servers to call each other via SIP without needing
authentication. You could also use gateways to allow FS machines to do SIP
registrations with each other. (I suppose it depends on your exact needs,
but ACLs are pretty easy and clean.) Once you have FS machines able to dial
each other then it's pretty much a matter of configuring your dialplans to
route the calls. You can create conferences on the fly so that part is easy.
The tricky part will be controlling how to link the conferences together,
although FS has a handy API to add a conference member to an existing
conference by dialing out. (See
http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference)

Let's say you have two servers, FS_A and FS_B. Each server can dial the
other and each server has a conference running, named FS_A_Conf and
FS_B_Conf, respectively. You can have either conference dial the other
server. For example, from the FS_A command line:

conference FS_A_Conf dial
{originate_timeout=30}sofia/internal/3900 at FS_B.IP.Address 1111 FS_A_Conf

Where 3900 is an extension set up in FS_B's dialplan to route to FS_B_Conf
and "1111 FS_A_Conf" are the caller ID number and name that FS_B should
receive from FS_A.

You still need to figure out other things like if you want the ability to
"break apart" the two conferences after they've been connected together,
etc. This is definitely doable but you'll need to handle the call control
stuff some how.

HTH,
MC
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From bjbrashier at gmail.com  Wed Aug 12 15:18:48 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Wed, 12 Aug 2009 15:18:48 -0700
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>
References: <4A830243.8080703@chandlerfamily.org.uk>
	<7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>
Message-ID: <7bcfdd290908121518i42b8eebv6a96f20dea5c3078@mail.gmail.com>

Whoops.  All of my parens () should be curly braces {}. Wasn't paying
attention.

BB

On Wed, Aug 12, 2009 at 2:40 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> You've thought through some of the difficult points, which is good. You're
> right that the moderator can't have different controls (unless you're
> controlling the conference yourself from outside, using, say, the event
> socket).
>
> Before I go further, I want to make sure I understand what you're
> proposing. What you're essentially saying is that when the command to kick
> someone is pressed the person should be transferred out of the conference,
> checked for moderator status, asked whom to kick (if so), and then let back
> in while the system kicks that person. The other commands would work
> similarly. Does this sound like a correct summary?
>
> If so, I think what you've got is mostly right, but I'm not sure it will
> work, having not tried to do it that way myself (I went the event socket
> route). In theory, it should look something like this:
>
> In the scripts entering the conference, you'll need to save the current
> conference number. Something like (in both cases):
>
>         <action application="set" data="conf-id=$1"/>
>
> "Extension" doesn't actually check for anything, so you'll need to check
> for "kick" with a condition. Try this:
>
>     <extension name="kick">
>       <condition field="destination_number" expression="^kick$"/>
>       <condition field="$(i_am_moderator)" expression="true">
>         <action application="play_and_get_digits" data="3 3 3 7000 #
> /ask-for-extension.wav /invalid.wav conf-user-id \d+" >
>
> Followed by (I picked up the "kick" syntax from the wiki):
>
>         <action application="conference" data="$(conf-id) kick
> $(conf-user-id)"/>
>         <action application="conference"
> data="$(conf-id)@default+flags{mod}"/>
>         <anti-action application="conference" data="$(conf-id)@default"/>
>       </condition>
>     </extension>
>
> In theory, then, you've transferred out with the "9" key, you check the
> moderator flag you've made, and do stuff based on the conf-id and the data
> you enter.  Obviously, if this works, the moderator won't be able to hear
> what's going on in the conference while he's entering stuff.
>
> Bigger problem: the conf-user-id he enters has to be the member ID that FS
> chose for the user he's trying to kick. That number will make sense if
> you're following the system closely, but for someone who doesn't know FS
> internals, it will be impossible to know unless you broadcast it to him
> somehow. To my knowledge, there's no way to use any other identifier (like
> caller-id) to kick them with.
>
> BB
>
> On Wed, Aug 12, 2009 at 10:56 AM, Alan Chandler <
> alan at chandlerfamily.org.uk> wrote:
>
>> I have been reading all the docs about conferences I can find and am
>> getting somewhat confused.  What I am trying to do is set up a dialplan
>> where I have subscribers with extensions in the 1xx range, and then to
>> set an ability to have a series of conference rooms for each subscriber
>> in the 21xx range where if the user enters is "own" conference he is the
>> moderator, but if not he is just a normal user.
>>
>> I want to be able for the moderator to do things like mute or kick people.
>>
>> So dialplan would probably have something like this in it
>>
>>
>>
>>     <extension name="user_conference">
>>       <condition field="destination_number"
>> expression="^(2${caller_id_number})$">
>>         <action application="set" data="i_am_moderator=true">
>>        <action application="answer"/>
>>        <action application="conference" data="$1 at default
>> +flags{moderator}"/>
>>       </condition>
>>       <condition field="destination_number" expression="^(21\d{2})^">
>>         <action application="set" data="i_am_moderator=false">
>>        <action application="answer"/>
>>        <action application="conference" data="$1 at default"/>
>>       </condition>
>>     </extension>
>>     <extension name="kick">
>>       <condition field="i_am_moderator" expression="true">
>>         <action application="play_and_get_digits" data="3 3 3 7000 #
>> /ask-for-extension.wav /invalid.wav conf-user-id \d+" >
>>
>> WHAT GOES HERE???
>>
>>       </condition>
>>     </extension>
>>
>>
>> in the conference.conf.xml file, I would change the caller controls to
>> include
>>
>>   <caller-controls>
>>     <group name="somekeys">
>>        <control action="transfer" digits="9"   data="kick XML default"/>
>>     </group>
>>   </caller-controls>
>>
>>
>> My question (at the moment) is
>>
>> In the WHAT GOES HERE place how do it
>>
>> Kick extension ${conf-user-id}  (DOES IT REQUIRE A SCRIPT TO CALL THE
>> Conference API?)
>> Re-Enter the moderator back into the conference
>> Re-Enter the ordinary user who happened to press 9 back into the
>> conference
>>
>> I am assuming I can't stop the non moderator getting the control - since
>> all users get the same controls.
>>
>>
>> --
>> Alan Chandler
>> http://www.chandlerfamily.org.uk
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From moises.silva at gmail.com  Wed Aug 12 15:28:59 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 12 Aug 2009 18:28:59 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
	<87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
Message-ID: <c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>

then probably we should check the current state and ignore the libpri event
when already in progress with media.

On Wed, Aug 12, 2009 at 5:10 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 12, 2009 at 4:01 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Yes you can get a progress after you get a progress with media ... I
>> have seen it.
>>
>
> Yes, you definitely can and I believe that some of the PRI specs suggest
> that this is totally legal, even though it's kind of silly.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From alan at chandlerfamily.org.uk  Wed Aug 12 15:33:48 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Wed, 12 Aug 2009 23:33:48 +0100
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>
References: <4A830243.8080703@chandlerfamily.org.uk>
	<7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>
Message-ID: <4A83434C.3090505@chandlerfamily.org.uk>

Bradley Brashier wrote:
...
> Before I go further, I want to make sure I understand what you're 
> proposing. What you're essentially saying is that when the command to 
> kick someone is pressed the person should be transferred out of the 
> conference, checked for moderator status, asked whom to kick (if so), 
> and then let back in while the system kicks that person. The other 
> commands would work similarly. Does this sound like a correct summary?

Absolutely what I am trying to do.

...
>  
> Followed by (I picked up the "kick" syntax from the wiki):
>  
>         <action application="conference" data="$(conf-id) kick 
> $(conf-user-id)"/>

This is a significant new fact for me.  What you seem to be doing is 
calling the commands referenced in the conference api here

http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference

by using application="conference" and then the data string as the second 
part of the command.  Am I correct in the assumption that you can do this.


>  
> Bigger problem: the conf-user-id he enters has to be the member ID that 
> FS chose for the user he's trying to kick. That number will make sense 
> if you're following the system closely, but for someone who doesn't know 
> FS internals, it will be impossible to know unless you broadcast it to 
> him somehow. To my knowledge, there's no way to use any other identifier 
> (like caller-id) to kick them with.

If my assumption about the use of the API above is correct, then 
couldn't I do something like

<action application="set" data="member-list=${conference(${conf-id} 
list)}"/>
<action application="execute_extension" data="do-kick"/>
and the figure out a regular expression to get participant-id and 
caller-id out of the resultant string.

Or is this not how its done?




-- 
Alan Chandler
http://www.chandlerfamily.org.uk



From bjbrashier at gmail.com  Wed Aug 12 16:09:51 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Wed, 12 Aug 2009 16:09:51 -0700
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <4A83434C.3090505@chandlerfamily.org.uk>
References: <4A830243.8080703@chandlerfamily.org.uk>
	<7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com> 
	<4A83434C.3090505@chandlerfamily.org.uk>
Message-ID: <7bcfdd290908121609i5289fae6n40b1be48b0173ff0@mail.gmail.com>

>This is a significant new fact for me.  What you seem to be doing is
>calling the commands referenced in the conference api here
>
>http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
>
>by using application="conference" and then the data string as the second
>part of the command.  Am I correct in the assumption that you can do this.

I agree that that's what it looks like. What I don't know is if it works. I
got this example from the page
http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did
exactly what you're trying, and never tried using the API in this fashion.

><action application="set" data="member-list=${conference(${conf-id}
>list)}"/>
><action application="execute_extension" data="do-kick"/>
>and the figure out a regular expression to get participant-id and
>caller-id out of the resultant string.
>
>Or is this not how its done?

Even assuming that our understanding on the kick command is correct, and
there's an accessible API here, I don't believe the above would actually
populate your variable with the data that you want. I don't think it
actually issues the command with the "set" application. Instead, you'd need
to do something like

<action application="conference" data="${conf-id} list"/>

and then you'd have to capture the output somehow, which I also don't
believe is possible.

I'm hardly a master of XML, though, or of dialplans, so feel free to try it
if you want. Just make sure you're logging everything so you can watch it
all unfold.

BB
On Wed, Aug 12, 2009 at 3:33 PM, Alan Chandler
<alan at chandlerfamily.org.uk>wrote:

> Bradley Brashier wrote:
> ...
> > Before I go further, I want to make sure I understand what you're
> > proposing. What you're essentially saying is that when the command to
> > kick someone is pressed the person should be transferred out of the
> > conference, checked for moderator status, asked whom to kick (if so),
> > and then let back in while the system kicks that person. The other
> > commands would work similarly. Does this sound like a correct summary?
>
> Absolutely what I am trying to do.
>
> ...
> >
> > Followed by (I picked up the "kick" syntax from the wiki):
> >
> >         <action application="conference" data="$(conf-id) kick
> > $(conf-user-id)"/>
>
> This is a significant new fact for me.  What you seem to be doing is
> calling the commands referenced in the conference api here
>
> http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
>
> by using application="conference" and then the data string as the second
> part of the command.  Am I correct in the assumption that you can do this.
>
>
> >
> > Bigger problem: the conf-user-id he enters has to be the member ID that
> > FS chose for the user he's trying to kick. That number will make sense
> > if you're following the system closely, but for someone who doesn't know
> > FS internals, it will be impossible to know unless you broadcast it to
> > him somehow. To my knowledge, there's no way to use any other identifier
> > (like caller-id) to kick them with.
>
> If my assumption about the use of the API above is correct, then
> couldn't I do something like
>
> <action application="set" data="member-list=${conference(${conf-id}
> list)}"/>
> <action application="execute_extension" data="do-kick"/>
> and the figure out a regular expression to get participant-id and
> caller-id out of the resultant string.
>
> Or is this not how its done?
>
>
>
>
> --
>  Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tina at a2unlimited.com  Wed Aug 12 16:10:59 2009
From: tina at a2unlimited.com (Tina Martinez)
Date: Wed, 12 Aug 2009 19:10:59 -0400
Subject: [Freeswitch-users]  Question about sharing conference between
Message-ID: <41480.1250118659@a2unlimited.com>

Michael,

Thanks for the welcome, and for the response to my question.

The call control and dynamic setup of conferences I have working (pretty cool stuff).
The tricky part, as you said, is "linking" the servers together.

Basically, what I need to do is establish a connection that will not be dependent
on a live person being on the call.  And I would prefer to avoid having to
register actual phone extensions for every server -- and for every conference call.

I apologize if I'm slow, but I'm new to working an application like this.  

In your example, you stated, "3900 is an extension set up in FS_B's dialplan",
does this extension have to be a live person (or soft-phone connected using an
auto-answer mechanism)?  or can I setup something where there is not a phone
actually connected?

Right now I'm able to place a call from one machine to another, but I'm calling
an X-Lite phone on the second server.

Also, if it is possible to call a "virtual" extension, I have no problem
incorporating application logic that would clean-up the orphaned conferences on
all machines when a conference call is complete.  I'm more concerned with being
able to setup the "links" quickly and in an elegant fashion.

- T





From brian at freeswitch.org  Wed Aug 12 16:19:08 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 18:19:08 -0500
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
	<87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
	<c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>
Message-ID: <A681A304-5B94-49B8-BA25-E3089ED3F4A9@freeswitch.org>

Well you really can't ignore it... it happens with our ISDN stack  
too.   Thats what the VETO handles.

/b

On Aug 12, 2009, at 5:28 PM, Moises Silva wrote:

> then probably we should check the current state and ignore the  
> libpri event when already in progress with media.



From msc at freeswitch.org  Wed Aug 12 17:34:38 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Aug 2009 19:34:38 -0500
Subject: [Freeswitch-users] Question about sharing conference between
In-Reply-To: <41480.1250118659@a2unlimited.com>
References: <41480.1250118659@a2unlimited.com>
Message-ID: <87f2f3b90908121734r78b9830bw38a135c340a6a845@mail.gmail.com>

On Wed, Aug 12, 2009 at 6:10 PM, Tina Martinez <tina at a2unlimited.com> wrote:

> Michael,
>
> Thanks for the welcome, and for the response to my question.
>
> The call control and dynamic setup of conferences I have working (pretty
> cool stuff).
> The tricky part, as you said, is "linking" the servers together.
>
> Basically, what I need to do is establish a connection that will not be
> dependent
> on a live person being on the call.  And I would prefer to avoid having to
> register actual phone extensions for every server -- and for every
> conference call.
>
> I apologize if I'm slow, but I'm new to working an application like this.


No worries. :)


>
>
> In your example, you stated, "3900 is an extension set up in FS_B's
> dialplan",
> does this extension have to be a live person (or soft-phone connected using
> an
> auto-answer mechanism)?  or can I setup something where there is not a
> phone
> actually connected?
>

No actual telephone is needed. Here's an example dialplan snippet that you
could drop right into conf/dialplan/default/ in a new file. (I prefer to put
my own custom dialplan entries into a separate file instead of editing
default.xml)

<include>
  <extension name="Sample Conference at 3900">
    <condition field="destination_number" expression="^3900$">
      <action application="conference" data="Conf at FS_B"/>
    </condition>
  </extension>
</include>

Put the above into conf/dialplan/default/01_ConfB.xml on the "FS_B" server.
(You can make a similar file on the "FS_A" server or any other server if
you'd like.)

You will also need to create a "public" extension which will route the
inbound calls appropriately. (If that doesn't make any sense right now then
don't worry, just do it. :)

Put the following into a file named conf/dialplan/public/01_ConfB.xml:
<include>
  <extension name="Sample Conference at 3900">
    <condition field="destination_number" expression="^3900$">
      <action application="transfer" data="3900 XML default"/>
    </condition>
  </extension>
</include>

(Again, you can do the same on all of your servers - this will allow all
servers to receive calls and route them to x3900.)

Now that you've got 3900 set up you can test it. Press F6 (or type
"reloadxml") at the CLI for FS_B. Then, have a phone that is registered to
FS_B make a call to 3900. It should be alone in the conference.

Now you'll need to set up some sort of dialplan routing to call from FS_A to
FS_B, unless you have a SIP phone registered to FS_A that can dial a SIP
URI. The SIP URI is: sip:3900 at FS_B.IP.Address. For kicks, let's add a simple
dialplan extension on FS_A that allows you to dial "23900" to get to FS_B's
3900 extension. Put this into conf/dialplan/default/01_Dial_ConfB.xml on
FS_A:

<include>
  <extension name="Sample Conference at 3900">
    <condition field="destination_number" expression="^2(3900)$">
      <action application="bridge" data="sofia/internal/$1 at FS_B.IP.Address
"/>
    </condition>
  </extension>
</include>

Save, and do the reloadxml (or F6) thing on FS_A CLI. Now on FS_A you can
dial "23900" and it will ring right into 3900 on FS_B so that the phone
registered at FS_A is in the conference on FS_B. Got it?

Have fun tinkering and let us know how it all goes.
-MC



> Right now I'm able to place a call from one machine to another, but I'm
> calling
> an X-Lite phone on the second server.
>
> Also, if it is possible to call a "virtual" extension, I have no problem
> incorporating application logic that would clean-up the orphaned
> conferences on
> all machines when a conference call is complete.  I'm more concerned with
> being
> able to setup the "links" quickly and in an elegant fashion.
>
> - T
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From string01 at gmail.com  Wed Aug 12 17:42:02 2009
From: string01 at gmail.com (String Larson)
Date: Wed, 12 Aug 2009 19:42:02 -0500
Subject: [Freeswitch-users] Setting max inbound for UA
Message-ID: <34D23392-2847-41BA-81A1-EF173B2F5061@gmail.com>

Is there a way to limit the number of calls a UA can receive in the FS  
configs?

I'm doing some testing with XLite as the UA, and can not figure out  
how to keep line 2 from answering if line 1 is in use.

THanks.

-str


From krice at freeswitch.org  Wed Aug 12 19:22:49 2009
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 12 Aug 2009 21:22:49 -0500
Subject: [Freeswitch-users] Setting max inbound for UA
In-Reply-To: <34D23392-2847-41BA-81A1-EF173B2F5061@gmail.com>
Message-ID: <C6A8E329.5ACC0%krice@freeswitch.org>

Check out mod_limit... Other wise you'll have to look specifically at the UA
you are trying to use, some like polycom and sipura offer a way to disable
"call waiting"

Remember with SIP there is no such thing as a line, its a SESSION and you
can have as many sessions as the software allows (and most software doesn't
put sane limits based on CPU/RAM/Bandwidth etc)


> From: String Larson <string01 at gmail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 12 Aug 2009 19:42:02 -0500
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Setting max inbound for UA
> 
> Is there a way to limit the number of calls a UA can receive in the FS
> configs?
> 
> I'm doing some testing with XLite as the UA, and can not figure out
> how to keep line 2 from answering if line 1 is in use.
> 
> THanks.
> 
> -str
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jlenk at frontiernet.net  Wed Aug 12 21:11:33 2009
From: jlenk at frontiernet.net (Jeff Lenk)
Date: Wed, 12 Aug 2009 23:11:33 -0500 (CDT)
Subject: [Freeswitch-users] OpenZAP/Sangoma in Windows
In-Reply-To: <c4d05cbe0908110759j2c929796g1ce96f530b7ad681@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED9435@cooper>
	<c4d05cbe0908110759j2c929796g1ce96f530b7ad681@mail.gmail.com>
Message-ID: <1250136693271-3435444.post@n2.nabble.com>


I have been testing analog support under Windows with good results so far. I
am waiting on another driver fix to solve some small problems with the api.

I have been running this code for 6 weeks or so now under a home test
environment(light call traffic) and the reliabilty has been fine - no errors
or other abnormalites.

The openzap for windows code is very simular to the others now(with the new
LibSangoma) so we should be in pretty good shape.

I would love to hear more from you regarding your testing of PRI support
under windows.

-Jeff


Moises Silva wrote:
> 
> On Tue, Aug 11, 2009 at 2:16 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
> 
>> Hi, I'm trying to evaluate the OpenZAP/Sangoma-support in Windows, using
>> PRI E1 connections.
>>
> 
> Thanks for testing this :-)
> 
> I have been meaning to install FreeSWITCH on Windows but just could not
> find
> the time.
> 
> 
>> 1. Has anyone tested it in Windows at all? I know the build-files for
>> Visual Studio has only been checked in for a couple of months, so that's
>> why
>> I'm asking.
>>
> 
> The drivers have been tested quite well but not using FreeSWITCH.
> 
> 
>> 2. Does anyone have any directions how to configure the driver within
>> Windows? Should I use BitStream or HDLC, and how should the channel
>> groups
>> be configured?
>>
> 
> You should use HDLC for the D-channel and Bitstream for the B-Channels.
> Typically you would create 2 groups, one with channels 1-23 and the other
> with just channel 24. The first group would work in bitstream and
> TDM_CHAN_VOICE_API operational mode and the second in HDLC/API mode.
> 
> You can find me in #openzap, #freeswitch or #freeswitch-dev as "moy" if
> you
> have more questions.
> 
> -- 
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3
> Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/OpenZAP-Sangoma-in-Windows-tp3422060p3435444.html
Sent from the freeswitch-users mailing list archive at Nabble.com.


From diego.viola at gmail.com  Wed Aug 12 23:47:07 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Thu, 13 Aug 2009 02:47:07 -0400
Subject: [Freeswitch-users] ClueCon Presentations - Where?
In-Reply-To: <87f2f3b90908121407n77efcde2v374008b07af77e92@mail.gmail.com>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
	<D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com>
	<c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>
	<c29b930b0908121134m50b1c8b2j3a91b57e71b70b10@mail.gmail.com>
	<F571A97F57A2D946B229C8231805867D32AD4527@svanexcsvr01.alldir.net>
	<65BC9732-5836-48F4-83AC-B305F4D1028C@freeswitch.org>
	<87f2f3b90908121407n77efcde2v374008b07af77e92@mail.gmail.com>
Message-ID: <86a32abc0908122347y4f9a702dg7f11615597311ce9@mail.gmail.com>

Just seen Anthony presentation, very cool ;)

Everyone, watch it!

http://files.freeswitch.org/cluecon_2009/presentations/Day%2001%20Presentation%2002.Anthony%20Minessale.1500kbps.mp4

=D

On Wed, Aug 12, 2009 at 5:07 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 12, 2009 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:
>
>> They are all getting gathered up and put online...
>> files.freeswitch.org/cluecon_2009 just keep an eye there some of the
>> videos are up also.
>> /b
>>
>
> FYI,
>
> I've uploaded the first batch and they should get synched up on
> files.freeswitch.org/cluecon_2009/presentations any time...
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From alan at chandlerfamily.org.uk  Thu Aug 13 00:11:48 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Thu, 13 Aug 2009 08:11:48 +0100
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <7bcfdd290908121609i5289fae6n40b1be48b0173ff0@mail.gmail.com>
References: <4A830243.8080703@chandlerfamily.org.uk>	<7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>
	<4A83434C.3090505@chandlerfamily.org.uk>
	<7bcfdd290908121609i5289fae6n40b1be48b0173ff0@mail.gmail.com>
Message-ID: <4A83BCB4.2050604@chandlerfamily.org.uk>

Bradley Brashier wrote:
 >  I wrote:
>  >This is a significant new fact for me.  What you seem to be doing is
>  >calling the commands referenced in the conference api here
>  >
>  >http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
>  >
>  >by using application="conference" and then the data string as the second
>  >part of the command.  Am I correct in the assumption that you can do this.
>  
> I agree that that's what it looks like. What I don't know is if it 
> works. I got this example from the page 
> http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did 
> exactly what you're trying, and never tried using the API in this fashion.

I just found this - which I think helps

http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan

An API can be called from the dialplan but it is not recommended. Example:

  <extension name="Make API call from Dialplan">
    <condition field="destination_number" expression="^(999)$">
      <!-- next line calls hupall, so be careful! -->
      <action application="set" 
data="api_result=${hupall(normal_clearing)}"/>
    </condition>
  </extension>

Anyway - thanks for you help - I am going away to rethink that 
particular interface again.  Its getting so complicated that it might be 
better to copy the Javascript approach in the examples.



-- 
Alan Chandler
http://www.chandlerfamily.org.uk



From diego.viola at gmail.com  Thu Aug 13 01:03:24 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Thu, 13 Aug 2009 04:03:24 -0400
Subject: [Freeswitch-users] answer command
In-Reply-To: <6539128A-D7A1-4555-A3DB-60E94E461DB3@jerris.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
	<B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
	<24941422.post@talk.nabble.com>
	<6539128A-D7A1-4555-A3DB-60E94E461DB3@jerris.com>
Message-ID: <86a32abc0908130103h6c0093d4hcef03ceed2f2aa3@mail.gmail.com>

Hey Michael,

Just wondering something, I have found that you added
"conference_set_auto_outcall" on the dptools wiki, but I could not find that
function in the mod_dptools.c, shouldn't that be part of the mod_conference
wiki article? =D.

Best regards,

Diego

On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris <mike at jerris.com> wrote:

> Sip does not support this functionality.  The called device would have
> to support this via some other mechanism such as ctsa which I have
> seen recently someone was looking at for freeswitch.  So the first
> issue you must resolve is the called device needs to support some way
> to do this.
>
> Mike
>
> On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov <maxim.tsvetov at gmail.com>
> wrote:
>
> >
> > If I have two FS extensions A and B. I'm calling from A to B and
> > want to
> > answer from B-side in my CTI application and to make SIP phone to be
> > synchronised to my CTI application. Is it possible to do it?
> >
> >
> > Brian West-3 wrote:
> >>
> >> Well you can only truly answer an inbound call to FS... you can't
> >> force answer an outbound call.
> >>
> >> /b
> >>
> >> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
> >>
> >>>
> >>> I will try to paraphrase my question.
> >>> Is there any possibility to answer call  from CTI application and
> >>> synchronise answer with answer in SIP client?Maybe we can use SIP
> >>> functions
> >>> in our CTI application instead of FS api commands?
> >>> I'm trying to find the way to make prototype of lineAnswer command
> >>> in TAPI.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > --
> > View this message in context:
> http://www.nabble.com/answer-command-tp24912812p24941422.html
> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yivzhenko at mksat.net  Thu Aug 13 01:14:02 2009
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Thu, 13 Aug 2009 11:14:02 +0300
Subject: [Freeswitch-users] Grangstream Early Dial
Message-ID: <200908131114.02970.yivzhenko@mksat.net>

Hello all.

I want to use Grandstream Early Dial future.
How i can enable support 484 response?

I tried simply use
<action application="hangup" data="484"/>
and
<action application="respond" data="484"/>
on uncompleted extensions, 
but there is not work


Thanks.

Yuriy .



From tzury.by at reguluslabs.com  Thu Aug 13 01:40:55 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Thu, 13 Aug 2009 11:40:55 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
Message-ID: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>

Hi,

Googeling about this shows that FS aims to support this (in fact it supports
all 3: UDP/TCP/TLS).
Yet I could not find the way to configure FS in order to support that.
In fact, it does not work in my current install.
I have TLS configured and work, but could not make TCP works

thanks in advance

/tzury
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From mike at jerris.com  Thu Aug 13 06:10:57 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 13 Aug 2009 09:10:57 -0400
Subject: [Freeswitch-users] answer command
In-Reply-To: <86a32abc0908130103h6c0093d4hcef03ceed2f2aa3@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
	<B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
	<24941422.post@talk.nabble.com>
	<6539128A-D7A1-4555-A3DB-60E94E461DB3@jerris.com>
	<86a32abc0908130103h6c0093d4hcef03ceed2f2aa3@mail.gmail.com>
Message-ID: <EA91B26D-1BB9-40BC-9D94-6614F9890456@jerris.com>

It probably belongs there.  It's a wiki, feel free to fix it.  What  
does this have to do with this thread?

On Aug 13, 2009, at 4:03 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Hey Michael,
>
> Just wondering something, I have found that you added  
> "conference_set_auto_outcall" on the dptools wiki, but I could not  
> find that function in the mod_dptools.c, shouldn't that be part of  
> the mod_conference wiki article? =D.
>
> Best regards,
>
> Diego
>
> On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> Sip does not support this functionality.  The called device would have
> to support this via some other mechanism such as ctsa which I have
> seen recently someone was looking at for freeswitch.  So the first
> issue you must resolve is the called device needs to support some way
> to do this.
>
> Mike
>
> On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov <maxim.tsvetov at gmail.com>
> wrote:
>
> >
> > If I have two FS extensions A and B. I'm calling from A to B and
> > want to
> > answer from B-side in my CTI application and to make SIP phone to be
> > synchronised to my CTI application. Is it possible to do it?
> >
> >
> > Brian West-3 wrote:
> >>
> >> Well you can only truly answer an inbound call to FS... you can't
> >> force answer an outbound call.
> >>
> >> /b
> >>
> >> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
> >>
> >>>
> >>> I will try to paraphrase my question.
> >>> Is there any possibility to answer call  from CTI application and
> >>> synchronise answer with answer in SIP client?Maybe we can use SIP
> >>> functions
> >>> in our CTI application instead of FS api commands?
> >>> I'm trying to find the way to make prototype of lineAnswer command
> >>> in TAPI.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > --
> > View this message in context: http://www.nabble.com/answer-command-tp24912812p24941422.html
> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From brian at freeswitch.org  Thu Aug 13 06:45:58 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 13 Aug 2009 08:45:58 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
Message-ID: <2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>

It just works... to force TCP you append ;transport=tcp

In reality you should be using SRV records.

/b

On Aug 13, 2009, at 3:40 AM, Tzury Bar Yochay wrote:

> Hi,
>
> Googeling about this shows that FS aims to support this (in fact it  
> supports all 3: UDP/TCP/TLS).
> Yet I could not find the way to configure FS in order to support that.
> In fact, it does not work in my current install.
> I have TLS configured and work, but could not make TCP works
>
> thanks in advance
>
> /tzury



From irmatov at gmail.com  Thu Aug 13 02:35:05 2009
From: irmatov at gmail.com (Timur Irmatov)
Date: Thu, 13 Aug 2009 14:35:05 +0500
Subject: [Freeswitch-users] calling through same gateway with multiple
	registrations
Message-ID: <241d382f0908130235w460339baxcc39527c03b0724@mail.gmail.com>

Hi,


I am new to FreeSWITCH and need an advice.

All calls to PSTN from our server will go through single gateway,
which is a soft switch supporting SIP protocol. FreeSWITCH will need
to register with soft switch, but soft switch permits only single
active call (in either direction) per registration. So we will need 10
SIP accounts to allow 10 simultaneous connections.

Question is - how should I configure FreeSWITCH for this scenario? I
see two options:

1) Create 10 gateways with different registrations, use mod_limit to
route only one outgoing call per gateway;
2) Create 10 gateways with different registrations, use event socket
to route calls manually and monitor used lines (incoming and outgoing
calls through soft switch).

Are there any other possibilities? Corrections/ suggestions are very welcome.


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru


From brian at freeswitch.org  Thu Aug 13 06:47:18 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 13 Aug 2009 08:47:18 -0500
Subject: [Freeswitch-users] Grangstream Early Dial
In-Reply-To: <200908131114.02970.yivzhenko@mksat.net>
References: <200908131114.02970.yivzhenko@mksat.net>
Message-ID: <CF2F282D-7E28-4F19-ACD4-6692F54E5052@freeswitch.org>

I don't think we ever got this working correctly.  Can you do a trace  
of it working vs not working?

/b

On Aug 13, 2009, at 3:14 AM, Yuriy Ivzhenko wrote:

> Hello all.
>
> I want to use Grandstream Early Dial future.
> How i can enable support 484 response?
>
> I tried simply use
> <action application="hangup" data="484"/>
> and
> <action application="respond" data="484"/>
> on uncompleted extensions,
> but there is not work
>
>
> Thanks.
>
> Yuriy .



From tzury.by at reguluslabs.com  Thu Aug 13 07:04:18 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Thu, 13 Aug 2009 17:04:18 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
Message-ID: <10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>

> It just works... to force TCP you append ;transport=tcp
Hi Brian

In fact this is exactly what I did and I could not get it work.
I am using a console based application supplied by pjsip.org and when
trying to register i get some error messages saying 'invalid transport
"SIP/2.0/tcp"' and 'REGISTER (30669) has invalid Via'

using the very same client against iptel.org seems to work.

> In reality you should be using SRV records.
can you please elaborate a bit more about this?


I am dumping below the cli output.
thanks in advance for your time and attention


tport_wakeup(0x7fd82c2afaf0): events IN
tport_recv_event(0x7fd82c2afaf0)
tport_recv_iovec(0x7fd82c2afaf0) msg 0x7fd82c2a1830 from
(tcp/80.74.97.189:42634) has 472 bytes, veclen = 1
tport_deliver(0x7fd82c2afaf0): msg 0x7fd82c2a1830 (472 bytes) from
tcp/80.74.97.189:42634/sip next=(nil)
nta: received REGISTER sip:cheerfulsanity.net;transport=tcp SIP/2.0 (CSeq 30669)
nta: Via check: invalid transport "SIP/2.0/tcp" from 80.74.97.189:42634
nta: REGISTER (30669) has invalid Via
tport(0x7fd82c2afaf0): reset timer
tport(0x7fd82c2afaf0): set timer at 1800000 ms because idle since recv
tport_wakeup_pri(0x713dd0): events IN
tport_recv_event(0x713dd0)
tport_recv_iovec(0x713dd0) msg 0x7fd82c2a1830 from
(udp/67.23.5.142:5060) has 2 bytes, veclen = 1
tport_deliver(0x713dd0): bad msg 0x7fd82c2a1830 (2 bytes) from
udp/199.245.214.130:5060/sip next=(nil)
tport_wakeup_pri(0x713dd0): events IN
tport_recv_event(0x713dd0)
tport_recv_iovec(0x713dd0) msg 0x7fd82c2a1830 from
(udp/67.23.5.142:5060) has 2 bytes, veclen = 1
tport_deliver(0x713dd0): bad msg 0x7fd82c2a1830 (2 bytes) from
udp/199.245.214.130:5060/sip next=(nil)
tport_wakeup(0x7fd84027d7d0): events IN
tport_recv_event(0x7fd84027d7d0)


From brian at freeswitch.org  Thu Aug 13 07:11:03 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 13 Aug 2009 09:11:03 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
Message-ID: <43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>

I need to see the sip packet.  TCP should be uppercase I'm pretty sure.

/b

On Aug 13, 2009, at 9:04 AM, Tzury Bar Yochay wrote:

> nta: Via check: invalid transport "SIP/2.0/tcp" from  
> 80.74.97.189:42634
> nta: REGISTER (30669) has invalid Via



From csa at nowthor.com  Thu Aug 13 07:14:56 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 13 Aug 2009 10:14:56 -0400
Subject: [Freeswitch-users] bind_server_ip issue
Message-ID: <4A841FE0.90904@nowthor.com>

Hello!

First of all, I would like to express my thanks to all the developers of 
Freeswitch.

I am testing Freeswitch on a Debian machine with physical network 
interface with four virtual IP addresses. One of these IP addresses, 
aliased as eth0:3, has been created specifically for Freeswitch.  I then 
set bind_server_ip with the IP addresses associated with eth0:3. To my 
surprise, however, tow things happen more or less randomly: 1) in 
certain cases, Freeswitch binds to eth0:2 instead (with a different IP 
address); and in another, although Freeswitch binds initially to eth0:3, 
after a few hours it changes its mind and rebinds to eth0:2. Is this an 
issue with bind_server_ip or am I missing some configuration detail?

Thanks!

Carlos


From brian at freeswitch.org  Thu Aug 13 07:21:35 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 13 Aug 2009 09:21:35 -0500
Subject: [Freeswitch-users] bind_server_ip issue
In-Reply-To: <4A841FE0.90904@nowthor.com>
References: <4A841FE0.90904@nowthor.com>
Message-ID: <12E3506D-3F79-4987-9FA7-B40AE461B97A@freeswitch.org>

If you read the latest vars.xml I have clarified this:

   <!--
        THIS IS ONLY USED FOR DINGALING

        bind_server_ip

        Can be an ip address, a dns name, or "auto".
        This determines an ip address available on this host to bind.
        If you are separating RTP and SIP traffic, you will want to have
        use different addresses where this variable appears.
        Used by: dingaling.conf.xml
   -->
   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>


So you'll need to open up the sip profile in sip_profiles and set the  
bind ip to exactly what you want.

Thanks,
Brian


On Aug 13, 2009, at 9:14 AM, Carlos S. Antunes wrote:

> Hello!
>
> First of all, I would like to express my thanks to all the  
> developers of
> Freeswitch.
>
> I am testing Freeswitch on a Debian machine with physical network
> interface with four virtual IP addresses. One of these IP addresses,
> aliased as eth0:3, has been created specifically for Freeswitch.  I  
> then
> set bind_server_ip with the IP addresses associated with eth0:3. To my
> surprise, however, tow things happen more or less randomly: 1) in
> certain cases, Freeswitch binds to eth0:2 instead (with a different IP
> address); and in another, although Freeswitch binds initially to  
> eth0:3,
> after a few hours it changes its mind and rebinds to eth0:2. Is this  
> an
> issue with bind_server_ip or am I missing some configuration detail?
>
> Thanks!
>
> Carlos
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From tzury.by at reguluslabs.com  Thu Aug 13 07:37:18 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Thu, 13 Aug 2009 17:37:18 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
Message-ID: <10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>

> I need to see the sip packet.
dumped below

> TCP should be uppercase I'm pretty sure.
you mean the via should be Via: SIP/2.0/TCP right?

If so, then that would a bug in the client then.

#wireshark dumps:
000  00 21 29 66 15 9f 00 23  ae 64 af a6 08 00 45 00   .!)f...# .d....E.
0010  02 3e 80 c2 40 00 40 06  64 e6 0a 00 00 6d 43 17   .>.. at .@. d....mC.
0020  05 8e db f4 13 c4 8a 0e  91 93 f6 a7 50 06 80 18   ........ ....P...
0030  00 5c 55 42 00 00 01 01  08 0a 0e 94 11 e4 01 ba   .\UB.... ........
0040  bf 9d 52 45 47 49 53 54  45 52 20 73 69 70 3a 63   ..REGIST ER sip:c
0050  68 65 65 72 66 75 6c 73  61 6e 69 74 79 2e 6e 65   heerfuls anity.ne
0060  74 3b 74 72 61 6e 73 70  6f 72 74 3d 74 63 70 20   t;transp ort=tcp
0070  53 49 50 2f 32 2e 30 0d  0a 56 69 61 3a 20 53 49   SIP/2.0. .Via: SI
0080  50 2f 32 2e 30 2f 74 63  70 20 31 30 2e 30 2e 30   P/2.0/tc p 10.0.0
0090  2e 31 30 39 3a 35 36 33  30 38 3b 72 70 6f 72 74   .109:563 08;rport
00a0  3b 62 72 61 6e 63 68 3d  7a 39 68 47 34 62 4b 50   ;branch= z9hG4bKP
00b0  6a 69 43 34 39 35 5a 50  52 6d 32 39 39 65 4f 6e   jiC495ZP Rm299eOn
00c0  6e 32 4f 4b 34 79 6a 6b  64 37 32 4c 2d 41 73 76   n2OK4yjk d72L-Asv
00d0  67 0d 0a 52 6f 75 74 65  3a 20 3c 73 69 70 3a 63   g..Route : <sip:c
00e0  68 65 65 72 66 75 6c 73  61 6e 69 74 79 2e 6e 65   heerfuls anity.ne
00f0  74 3b 74 72 61 6e 73 70  6f 72 74 3d 74 63 70 3b   t;transp ort=tcp;
0100  6c 72 3e 0d 0a 4d 61 78  2d 46 6f 72 77 61 72 64   lr>..Max -Forward
0110  73 3a 20 37 30 0d 0a 46  72 6f 6d 3a 20 3c 73 69   s: 70..F rom: <si
0120  70 3a 31 30 36 30 40 63  68 65 65 72 66 75 6c 73   p:1060 at c heerfuls
0130  61 6e 69 74 79 2e 6e 65  74 3e 3b 74 61 67 3d 52   anity.ne t>;tag=R
0140  45 2e 58 78 5a 6e 4b 6a  71 66 32 5a 48 2d 38 4f   E.XxZnKj qf2ZH-8O
0150  58 37 30 71 70 73 78 5a  44 43 69 51 4a 37 65 0d   X70qpsxZ DCiQJ7e.
0160  0a 54 6f 3a 20 3c 73 69  70 3a 31 30 36 30 40 63   .To: <si p:1060 at c
0170  68 65 65 72 66 75 6c 73  61 6e 69 74 79 2e 6e 65   heerfuls anity.ne
0180  74 3e 0d 0a 43 61 6c 6c  2d 49 44 3a 20 5a 6e 61   t>..Call -ID: Zna
0190  30 61 56 68 72 2d 51 43  4a 4a 58 34 62 70 79 31   0aVhr-QC JJX4bpy1
01a0  4c 78 6b 62 4a 46 78 6a  50 4f 2d 4f 6d 0d 0a 43   LxkbJFxj PO-Om..C
01b0  53 65 71 3a 20 34 37 37  38 30 20 52 45 47 49 53   Seq: 477 80 REGIS
01c0  54 45 52 0d 0a 55 73 65  72 2d 41 67 65 6e 74 3a   TER..Use r-Agent:
01d0  20 50 4a 53 55 41 20 76  31 2e 33 2d 74 72 75 6e    PJSUA v 1.3-trun
01e0  6b 2f 69 36 38 36 2d 70  63 2d 6c 69 6e 75 78 2d   k/i686-p c-linux-
01f0  67 6e 75 0d 0a 43 6f 6e  74 61 63 74 3a 20 3c 73   gnu..Con tact: <s
0200  69 70 3a 31 30 36 30 40  31 30 2e 30 2e 30 2e 31   ip:1060@ 10.0.0.1
0210  30 39 3a 35 30 36 30 3b  74 72 61 6e 73 70 6f 72   09:5060; transpor
0220  74 3d 54 43 50 3e 0d 0a  45 78 70 69 72 65 73 3a   t=TCP>.. Expires:
0230  20 33 30 30 0d 0a 43 6f  6e 74 65 6e 74 2d 4c 65    300..Co ntent-Le
0240  6e 67 74 68 3a 20 20 30  0d 0a 0d 0a               ngth:  0 ....


From brian at freeswitch.org  Thu Aug 13 07:42:22 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 13 Aug 2009 09:42:22 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
Message-ID: <45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>


On Aug 13, 2009, at 9:37 AM, Tzury Bar Yochay wrote:

>> I need to see the sip packet.
> dumped below
>
>> TCP should be uppercase I'm pretty sure.
> you mean the via should be Via: SIP/2.0/TCP right?
>

Yep

> If so, then that would a bug in the client then.

Some things might accept it but sofia is usually strict about some of  
this stuff.

/b




From string01 at gmail.com  Thu Aug 13 07:43:09 2009
From: string01 at gmail.com (String Larson)
Date: Thu, 13 Aug 2009 09:43:09 -0500
Subject: [Freeswitch-users] Setting max inbound for UA
In-Reply-To: <C6A8E329.5ACC0%krice@freeswitch.org>
References: <C6A8E329.5ACC0%krice@freeswitch.org>
Message-ID: <D3C49EE3-E1F0-47FA-AF55-545DF0B92075@gmail.com>

Thanks Ken.
I'll look at mod_limit

The XLite softphone doesn't seem to have a switch for controlling it.
-str

On Aug 12, 2009, at 9:22 PM, Ken Rice wrote:

> Check out mod_limit... Other wise you'll have to look specifically  
> at the UA
> you are trying to use, some like polycom and sipura offer a way to  
> disable
> "call waiting"
>
> Remember with SIP there is no such thing as a line, its a SESSION  
> and you
> can have as many sessions as the software allows (and most software  
> doesn't
> put sane limits based on CPU/RAM/Bandwidth etc)
>
>
>> From: String Larson <string01 at gmail.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Wed, 12 Aug 2009 19:42:02 -0500
>> To: <freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] Setting max inbound for UA
>>
>> Is there a way to limit the number of calls a UA can receive in the  
>> FS
>> configs?
>>
>> I'm doing some testing with XLite as the UA, and can not figure out
>> how to keep line 2 from answering if line 1 is in use.
>>
>> THanks.
>>
>> -str
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From tina at a2unlimited.com  Thu Aug 13 07:43:01 2009
From: tina at a2unlimited.com (Tina Martinez)
Date: Thu, 13 Aug 2009 10:43:01 -0400
Subject: [Freeswitch-users]  Question about sharing conference between
Message-ID: <39253.1250174581@a2unlimited.com>

Thank you Michael,

I will tinker around with it and definitely follow-up with the results.


- T


From moises.silva at gmail.com  Thu Aug 13 08:02:05 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 13 Aug 2009 11:02:05 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <A681A304-5B94-49B8-BA25-E3089ED3F4A9@freeswitch.org>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
	<87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
	<c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>
	<A681A304-5B94-49B8-BA25-E3089ED3F4A9@freeswitch.org>
Message-ID: <c4d05cbe0908130802w6528ca34ma9d6e34ab9ce28c2@mail.gmail.com>

On Wed, Aug 12, 2009 at 7:19 PM, Brian West <brian at freeswitch.org> wrote:

> Well you really can't ignore it... it happens with our ISDN stack
> too.   Thats what the VETO handles.
>
> /b
>

You lost me. What do you mean we can't ignore it? the way I see it, sure we
can and we should.

Currently that warning comes from the on_ringing() callback which blindly
attempts to move the state of the zap channel to ZAP_CHANNEL_STATE_PROGRESS,
even when the state may be already ZAP_CHANNEL_STATE_PROGRESS_MEDIA (which
means on_proceed() was called first).

As I see it, the VETO warning is more an aid to the programmer so you
quickly realize your doing a useless state change, which should be fixed. In
this case, the fix is simply checking the state of the channel before trying
to move it to progress, and don't even try to move it if already in progress
with media.

 --
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From yivzhenko at mksat.net  Thu Aug 13 08:11:43 2009
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Thu, 13 Aug 2009 18:11:43 +0300
Subject: [Freeswitch-users] Grangstream Early Dial
In-Reply-To: <CF2F282D-7E28-4F19-ACD4-6692F54E5052@freeswitch.org>
References: <200908131114.02970.yivzhenko@mksat.net>
	<CF2F282D-7E28-4F19-ACD4-6692F54E5052@freeswitch.org>
Message-ID: <200908131811.43414.yivzhenko@mksat.net>

On Thursday 13 August 2009 16:47:18 Brian West wrote:
> I don't think we ever got this working correctly.  Can you do a trace
> of it working vs not working?

I can't do working trace, only not working

http://pastebin.freeswitch.org/9980

with dialplan action
 <action application="hangup" data="INVALID_NUMBER_FORMAT"/>



>
> /b
>
> On Aug 13, 2009, at 3:14 AM, Yuriy Ivzhenko wrote:
> > Hello all.
> >
> > I want to use Grandstream Early Dial future.
> > How i can enable support 484 response?
> >
> > I tried simply use
> > <action application="hangup" data="484"/>
> > and
> > <action application="respond" data="484"/>
> > on uncompleted extensions,
> > but there is not work
> >
> >
> > Thanks.
> >
> > Yuriy .
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From bjbrashier at gmail.com  Thu Aug 13 08:48:13 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 08:48:13 -0700
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <4A83BCB4.2050604@chandlerfamily.org.uk>
References: <4A830243.8080703@chandlerfamily.org.uk>
	<7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com> 
	<4A83434C.3090505@chandlerfamily.org.uk>
	<7bcfdd290908121609i5289fae6n40b1be48b0173ff0@mail.gmail.com> 
	<4A83BCB4.2050604@chandlerfamily.org.uk>
Message-ID: <7bcfdd290908130848t15592b50y68a6b0ee8808d095@mail.gmail.com>

So it sounds like "set" can work. But you'd still have to parse it. And even
then it's not recommended.

I have another couple of possible methods for you:
1) modification of mod_conference.
2) event socket.

If you modify mod_conference, you can probably do what you want, but it
obviously requires using C and modifying existing code.

If you use the event socket, you've got a bigger learning curve, perhaps,
but you can use a variety of languages, your code is separate (and therefore
easier to maintain), and you then know how the event socket works in case
you need to do something else later.
 Good luck with whatever you end up doing.

BB
On Thu, Aug 13, 2009 at 12:11 AM, Alan Chandler
<alan at chandlerfamily.org.uk>wrote:

> Bradley Brashier wrote:
>  >  I wrote:
> >  >This is a significant new fact for me.  What you seem to be doing is
> >  >calling the commands referenced in the conference api here
> >  >
> >  >http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
> >  >
> >  >by using application="conference" and then the data string as the
> second
> >  >part of the command.  Am I correct in the assumption that you can do
> this.
> >
> > I agree that that's what it looks like. What I don't know is if it
> > works. I got this example from the page
> > http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did
> > exactly what you're trying, and never tried using the API in this
> fashion.
>
> I just found this - which I think helps
>
> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
>
> An API can be called from the dialplan but it is not recommended. Example:
>
>  <extension name="Make API call from Dialplan">
>    <condition field="destination_number" expression="^(999)$">
>      <!-- next line calls hupall, so be careful! -->
>      <action application="set"
> data="api_result=${hupall(normal_clearing)}"/>
>    </condition>
>  </extension>
>
> Anyway - thanks for you help - I am going away to rethink that
> particular interface again.  Its getting so complicated that it might be
> better to copy the Javascript approach in the examples.
>
>
>
> --
>  Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Aug 13 09:06:03 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 13 Aug 2009 09:06:03 -0700
Subject: [Freeswitch-users] calling through same gateway with multiple
	registrations
In-Reply-To: <241d382f0908130235w460339baxcc39527c03b0724@mail.gmail.com>
References: <241d382f0908130235w460339baxcc39527c03b0724@mail.gmail.com>
Message-ID: <87f2f3b90908130906h6f40889erc33a6ef18deace32@mail.gmail.com>

On Thu, Aug 13, 2009 at 2:35 AM, Timur Irmatov <irmatov at gmail.com> wrote:

> Hi,
>
>
> I am new to FreeSWITCH and need an advice.
>
> All calls to PSTN from our server will go through single gateway,
> which is a soft switch supporting SIP protocol. FreeSWITCH will need
> to register with soft switch, but soft switch permits only single
> active call (in either direction) per registration. So we will need 10
> SIP accounts to allow 10 simultaneous connections.


Are you going to have incoming calls as well? If so, how does the
soft-switch handle two concurrent calls to the same number?


>
>
> Question is - how should I configure FreeSWITCH for this scenario? I
> see two options:
>
> 1) Create 10 gateways with different registrations, use mod_limit to
> route only one outgoing call per gateway;
> 2) Create 10 gateways with different registrations, use event socket
> to route calls manually and monitor used lines (incoming and outgoing
> calls through soft switch).
>
> Are there any other possibilities? Corrections/ suggestions are very
> welcome.


This seems like a serious defect in the soft-switch. I can understand if it
allows you to specify only one call per SIP registration, but to hard-code
that limit seems pretty silly. Can you find out more about the soft-switch
in question and see if that limitation is flexible?
-MC
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From bjbrashier at gmail.com  Thu Aug 13 09:37:11 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 09:37:11 -0700
Subject: [Freeswitch-users] Conference silence timeouts
Message-ID: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>

Hi all.

The solution to this one should be short.

My conference hangs up when there's 2+ users and silence for 5 sec or so.
I'm trying to find a parameter that changes that (I'd rather it be, say, 60
seconds).

I didn't see a parameter like this specific to conferences, so I looked
abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300
(the default), so I'm pretty sure that's not the problem. I also searched
through the mod_conference.c code and didn't see it, though I was only
skimming.

I'm not 100% convinced that this is limited to conferences, but I don't
currently have a way to test in a non-conference environment.

Anybody know how to increase the conference silence-hangup timeout?

BB
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From msc at freeswitch.org  Thu Aug 13 09:50:55 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 13 Aug 2009 11:50:55 -0500
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>
Message-ID: <87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>

Check out the 'waste' member flag. I think if at least one member has that
set then RTP will get sent out even during silence. Let us know if that
helps...

-MC

On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> Hi all.
>
> The solution to this one should be short.
>
> My conference hangs up when there's 2+ users and silence for 5 sec or so.
> I'm trying to find a parameter that changes that (I'd rather it be, say, 60
> seconds).
>
> I didn't see a parameter like this specific to conferences, so I looked
> abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300
> (the default), so I'm pretty sure that's not the problem. I also searched
> through the mod_conference.c code and didn't see it, though I was only
> skimming.
>
> I'm not 100% convinced that this is limited to conferences, but I don't
> currently have a way to test in a non-conference environment.
>
> Anybody know how to increase the conference silence-hangup timeout?
>
> BB
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bjbrashier at gmail.com  Thu Aug 13 10:47:48 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 10:47:48 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>
Message-ID: <7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>

I'm sure that would work, but I'm worried about it sucking up bandwidth,
especially since you'd need it on every caller (since otherwise the one
person who had it could hang up and you'd be back to square 1).

Any other ideas, or should I hunt through the code to try to override the
behavior manually?

BB

On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins <msc at freeswitch.org> wrote:

> Check out the 'waste' member flag. I think if at least one member has that
> set then RTP will get sent out even during silence. Let us know if that
> helps...
>
> -MC
>
>   On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <bjbrashier at gmail.com
> > wrote:
>
>>   Hi all.
>>
>> The solution to this one should be short.
>>
>> My conference hangs up when there's 2+ users and silence for 5 sec or so.
>> I'm trying to find a parameter that changes that (I'd rather it be, say, 60
>> seconds).
>>
>> I didn't see a parameter like this specific to conferences, so I looked
>> abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300
>> (the default), so I'm pretty sure that's not the problem. I also searched
>> through the mod_conference.c code and didn't see it, though I was only
>> skimming.
>>
>> I'm not 100% convinced that this is limited to conferences, but I don't
>> currently have a way to test in a non-conference environment.
>>
>> Anybody know how to increase the conference silence-hangup timeout?
>>
>> BB
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at cloudtel.com  Thu Aug 13 11:14:23 2009
From: chris at cloudtel.com (Chris Burns)
Date: Thu, 13 Aug 2009 11:14:23 -0700
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <7bcfdd290908130848t15592b50y68a6b0ee8808d095@mail.gmail.com>
References: <4A830243.8080703@chandlerfamily.org.uk>
	<7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>
	<4A83434C.3090505@chandlerfamily.org.uk>
	<7bcfdd290908121609i5289fae6n40b1be48b0173ff0@mail.gmail.com>
	<4A83BCB4.2050604@chandlerfamily.org.uk>
	<7bcfdd290908130848t15592b50y68a6b0ee8808d095@mail.gmail.com>
Message-ID: <e6ba88620908131114k7e48f529t8809bc9673639d0a@mail.gmail.com>

I couldn't imagine managing a conference without a GUI. I need to see who is
making noise so I can boot/mute em ;)

If I were you I would dive into ESL and make a simple web app to frontend
the conferences. There will surely be something in contrib to get you
started.

On Thu, Aug 13, 2009 at 8:48 AM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> So it sounds like "set" can work. But you'd still have to parse it. And
> even then it's not recommended.
>
> I have another couple of possible methods for you:
> 1) modification of mod_conference.
> 2) event socket.
>
> If you modify mod_conference, you can probably do what you want, but it
> obviously requires using C and modifying existing code.
>
> If you use the event socket, you've got a bigger learning curve, perhaps,
> but you can use a variety of languages, your code is separate (and therefore
> easier to maintain), and you then know how the event socket works in case
> you need to do something else later.
>  Good luck with whatever you end up doing.
>
> BB
> On Thu, Aug 13, 2009 at 12:11 AM, Alan Chandler <
> alan at chandlerfamily.org.uk> wrote:
>
>> Bradley Brashier wrote:
>>  >  I wrote:
>> >  >This is a significant new fact for me.  What you seem to be doing is
>> >  >calling the commands referenced in the conference api here
>> >  >
>> >  >http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
>> >  >
>> >  >by using application="conference" and then the data string as the
>> second
>> >  >part of the command.  Am I correct in the assumption that you can do
>> this.
>> >
>> > I agree that that's what it looks like. What I don't know is if it
>> > works. I got this example from the page
>> > http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did
>> > exactly what you're trying, and never tried using the API in this
>> fashion.
>>
>> I just found this - which I think helps
>>
>> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
>>
>> An API can be called from the dialplan but it is not recommended. Example:
>>
>>  <extension name="Make API call from Dialplan">
>>    <condition field="destination_number" expression="^(999)$">
>>      <!-- next line calls hupall, so be careful! -->
>>      <action application="set"
>> data="api_result=${hupall(normal_clearing)}"/>
>>    </condition>
>>  </extension>
>>
>> Anyway - thanks for you help - I am going away to rethink that
>> particular interface again.  Its getting so complicated that it might be
>> better to copy the Javascript approach in the examples.
>>
>>
>>
>> --
>>  Alan Chandler
>> http://www.chandlerfamily.org.uk
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Thu Aug 13 11:15:12 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Thu, 13 Aug 2009 14:15:12 -0400
Subject: [Freeswitch-users] answer command
In-Reply-To: <EA91B26D-1BB9-40BC-9D94-6614F9890456@jerris.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
	<B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
	<24941422.post@talk.nabble.com>
	<6539128A-D7A1-4555-A3DB-60E94E461DB3@jerris.com>
	<86a32abc0908130103h6c0093d4hcef03ceed2f2aa3@mail.gmail.com>
	<EA91B26D-1BB9-40BC-9D94-6614F9890456@jerris.com>
Message-ID: <86a32abc0908131115x4bbefcdfy3e3f84b8cb692a06@mail.gmail.com>

I was talking with Michael about fixing stuff in the wiki, so I just asked
to fix that also.

On Thu, Aug 13, 2009 at 9:10 AM, Michael Jerris <mike at jerris.com> wrote:

> It probably belongs there.  It's a wiki, feel free to fix it.  What does
> this have to do with this thread?
>
>
> On Aug 13, 2009, at 4:03 AM, Diego Viola <diego.viola at gmail.com> wrote:
>
> Hey Michael,
>
> Just wondering something, I have found that you added
> "conference_set_auto_outcall" on the dptools wiki, but I could not find that
> function in the mod_dptools.c, shouldn't that be part of the mod_conference
> wiki article? =D.
>
> Best regards,
>
> Diego
>
> On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris < <mike at jerris.com>
> mike at jerris.com> wrote:
>
>> Sip does not support this functionality.  The called device would have
>> to support this via some other mechanism such as ctsa which I have
>> seen recently someone was looking at for freeswitch.  So the first
>> issue you must resolve is the called device needs to support some way
>> to do this.
>>
>> Mike
>>
>> On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov < <maxim.tsvetov at gmail.com>
>> maxim.tsvetov at gmail.com>
>> wrote:
>>
>> >
>> > If I have two FS extensions A and B. I'm calling from A to B and
>> > want to
>> > answer from B-side in my CTI application and to make SIP phone to be
>> > synchronised to my CTI application. Is it possible to do it?
>> >
>> >
>> > Brian West-3 wrote:
>> >>
>> >> Well you can only truly answer an inbound call to FS... you can't
>> >> force answer an outbound call.
>> >>
>> >> /b
>> >>
>> >> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
>> >>
>> >>>
>> >>> I will try to paraphrase my question.
>> >>> Is there any possibility to answer call  from CTI application and
>> >>> synchronise answer with answer in SIP client?Maybe we can use SIP
>> >>> functions
>> >>> in our CTI application instead of FS api commands?
>> >>> I'm trying to find the way to make prototype of lineAnswer command
>> >>> in TAPI.
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> <http://www.freeswitch.org>http://www.freeswitch.org
>> >>
>> >>
>> >
>> > --
>> > View this message in context:
>> <http://www.nabble.com/answer-command-tp24912812p24941422.html>
>> http://www.nabble.com/answer-command-tp24912812p24941422.html
>> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Thu Aug 13 11:16:29 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Thu, 13 Aug 2009 14:16:29 -0400
Subject: [Freeswitch-users] answer command
In-Reply-To: <86a32abc0908131115x4bbefcdfy3e3f84b8cb692a06@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
	<B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
	<24941422.post@talk.nabble.com>
	<6539128A-D7A1-4555-A3DB-60E94E461DB3@jerris.com>
	<86a32abc0908130103h6c0093d4hcef03ceed2f2aa3@mail.gmail.com>
	<EA91B26D-1BB9-40BC-9D94-6614F9890456@jerris.com>
	<86a32abc0908131115x4bbefcdfy3e3f84b8cb692a06@mail.gmail.com>
Message-ID: <86a32abc0908131116g1fa3218egc146f519b93c7cd0@mail.gmail.com>

Err, I asked if that was wrong to fix it.

On Thu, Aug 13, 2009 at 2:15 PM, Diego Viola <diego.viola at gmail.com> wrote:

> I was talking with Michael about fixing stuff in the wiki, so I just asked
> to fix that also.
>
>
> On Thu, Aug 13, 2009 at 9:10 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> It probably belongs there.  It's a wiki, feel free to fix it.  What does
>> this have to do with this thread?
>>
>>
>> On Aug 13, 2009, at 4:03 AM, Diego Viola <diego.viola at gmail.com> wrote:
>>
>> Hey Michael,
>>
>> Just wondering something, I have found that you added
>> "conference_set_auto_outcall" on the dptools wiki, but I could not find that
>> function in the mod_dptools.c, shouldn't that be part of the mod_conference
>> wiki article? =D.
>>
>> Best regards,
>>
>> Diego
>>
>> On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris < <mike at jerris.com>
>> mike at jerris.com> wrote:
>>
>>> Sip does not support this functionality.  The called device would have
>>> to support this via some other mechanism such as ctsa which I have
>>> seen recently someone was looking at for freeswitch.  So the first
>>> issue you must resolve is the called device needs to support some way
>>> to do this.
>>>
>>> Mike
>>>
>>> On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov < <maxim.tsvetov at gmail.com>
>>> maxim.tsvetov at gmail.com>
>>> wrote:
>>>
>>> >
>>> > If I have two FS extensions A and B. I'm calling from A to B and
>>> > want to
>>> > answer from B-side in my CTI application and to make SIP phone to be
>>> > synchronised to my CTI application. Is it possible to do it?
>>> >
>>> >
>>> > Brian West-3 wrote:
>>> >>
>>> >> Well you can only truly answer an inbound call to FS... you can't
>>> >> force answer an outbound call.
>>> >>
>>> >> /b
>>> >>
>>> >> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
>>> >>
>>> >>>
>>> >>> I will try to paraphrase my question.
>>> >>> Is there any possibility to answer call  from CTI application and
>>> >>> synchronise answer with answer in SIP client?Maybe we can use SIP
>>> >>> functions
>>> >>> in our CTI application instead of FS api commands?
>>> >>> I'm trying to find the way to make prototype of lineAnswer command
>>> >>> in TAPI.
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> <http://www.freeswitch.org>http://www.freeswitch.org
>>> >>
>>> >>
>>> >
>>> > --
>>> > View this message in context:
>>> <http://www.nabble.com/answer-command-tp24912812p24941422.html>
>>> http://www.nabble.com/answer-command-tp24912812p24941422.html
>>> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From irmatov at gmail.com  Thu Aug 13 11:21:48 2009
From: irmatov at gmail.com (Timur Irmatov)
Date: Thu, 13 Aug 2009 23:21:48 +0500
Subject: [Freeswitch-users] calling through same gateway with multiple
	registrations
In-Reply-To: <87f2f3b90908130906h6f40889erc33a6ef18deace32@mail.gmail.com>
References: <241d382f0908130235w460339baxcc39527c03b0724@mail.gmail.com>
	<87f2f3b90908130906h6f40889erc33a6ef18deace32@mail.gmail.com>
Message-ID: <241d382f0908131121s66298225xa5fe49036b6ec5e@mail.gmail.com>

Thanks for responding, Michael!

On Thu, Aug 13, 2009 at 9:06 PM, Michael Collins<msc at freeswitch.org> wrote:
> Are you going to have incoming calls as well? If so, how does the
> soft-switch handle two concurrent calls to the same number?

Yes, we'll have incoming calls as well. I did not performed any tests
myself yet, but a guy working with this soft-switch says it just
rejects second call. So soft-switch treats each sip registration more
or less like single phone line. Soft-switch is Huawei SoftX3000, if
somebody experienced in it can prove me wrong, I'll be glad.. :)

>> Question is - how should I configure FreeSWITCH for this scenario? I
>> see two options:
>>
>> 1) Create 10 gateways with different registrations, use mod_limit to
>> route only one outgoing call per gateway;
>> 2) Create 10 gateways with different registrations, use event socket
>> to route calls manually and monitor used lines (incoming and outgoing
>> calls through soft switch).
>>
>> Are there any other possibilities? Corrections/ suggestions are very
>> welcome.
>
> This seems like a serious defect in the soft-switch. I can understand if it
> allows you to specify only one call per SIP registration, but to hard-code
> that limit seems pretty silly. Can you find out more about the soft-switch
> in question and see if that limitation is flexible?

Yes, that was my initial thought too - that this limit should be
configurable. But a guy working with it says it is not configurable,
and local support engineers from Huawei also confirm this. But, who
knows, may be they are all wrong.

-- 
Timur Irmatov, xmpp:irmatov at jabber.ru


From pjintheusa at gmail.com  Thu Aug 13 11:33:30 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 13 Aug 2009 14:33:30 -0400
Subject: [Freeswitch-users] mod_managed users?
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702C4A8E0C8@mse17be1.mse17.exchange.ms>
References: <6E8D2069C08AA84A83D336E996AE4C6702C4A8E02E@mse17be1.mse17.exchange.ms>
	<501641.90132.qm@web33503.mail.mud.yahoo.com>
	<6E8D2069C08AA84A83D336E996AE4C6702C4A8E0C8@mse17be1.mse17.exchange.ms>
Message-ID: <367751820908131133q20a724cdw3fa4ab1026e6f908@mail.gmail.com>

Hey Michael,

I am a little late to the party I know - but just want to say thanks
for your latest efforts.

I updated my dev environment with the latest managed mod and swapped
my app to the latest plugin architecture last night and all is working
well.

Love the dynamic loading of my dll into freeswitch - no more starting
and stopping freeswitch!

Also appreciate the f# example.

Thanks again.


Phillip Jones

On Wed, Jul 29, 2009 at 8:04 PM, Michael Giagnocavo<mgg at giagnocavo.net> wrote:
> Which directory ends with ; ?
>
>
>
> I?m not following ? if you want email me off list and we can work together
> on it .
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego
> Toro
> Sent: Wednesday, July 29, 2009 5:06 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_managed users?
>
>
>
> Hi,
>
>
>
> My dll's are loaded correctly, I have a trouble becouse the directory where
> are loaded it finish with ";" (whitout ""), i mean the default path assembly
> finish with ";"
>
>
>
> Is possible remove?the character ";"? ?
>
>
>
> thanks
>
>
>
> Diego
>
> --- On Wed, 7/29/09, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
>
> From: Michael Giagnocavo <mgg at giagnocavo.net>
> Subject: Re: [Freeswitch-users] mod_managed users?
> To: "freeswitch-users at lists.freeswitch.org"
> <freeswitch-users at lists.freeswitch.org>
> Date: Wednesday, July 29, 2009, 2:32 PM
>
> The error is that it just doesn?t find that alias to call the plugin.
> Assuming everything is spelled correctly, this probably means the DLL did
> not load.
>
>
>
> I just checked in a fix for dlls that don?t have both Api and App interfaces
> ? it would not load them at all. Try with it now and see if that?s the
> problem. If it is, I apologize.
>
>
>
> If it still doesn?t, paste the full log of when it loads your file.
>
>
>
> Thanks,
>
> Michael
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego
> Toro
> Sent: Wednesday, July 29, 2009 12:44 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_managed users?
>
>
>
> Hi Michael,
>
>
>
> I am working with lastest managed module, I have assemblies wich dependen of
> others assemblies (dll's), the past version it works fine, but I have now
> next the error message:
>
>
>
> EXECUTE sofia/internal/10510 at 192.168.27.10 managed(CIV_BPFSProcess)
> 2009-07-29 13:31:27.718750 [DEBUG] switch_cpp.cpp:1130 FreeSWITCH.Managed:
> attempting to run application 'CIV_BPFSProcess'.
> 2009-07-29 13:31:27.718750 [ERR] switch_cpp.cpp:1130 App plugin
> CIV_BPFSProcess not found.
> 2009-07-29 13:31:27.718750 [ERR] mod_managed.cpp:405 Application run failed
> for CIV_BPFSProcess (unknown module or exception).
>
>
>
> My declaration class? is:
> public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
>
> It has public method:
> public void Run(AppContext context).
>
>
>
> I have VS2008 on Windows
>
> Thanks
>
>
>
> Diego
>
> --- On Wed, 7/29/09, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
>
> From: Michael Giagnocavo <mgg at giagnocavo.net>
> Subject: Re: [Freeswitch-users] mod_managed users?
> To: "freeswitch-users at lists.freeswitch.org"
> <freeswitch-users at lists.freeswitch.org>
> Date: Wednesday, July 29, 2009, 1:46 AM
>
> Hi ?ukasz,
>
> ? ? ? ? Would you please send me the DLL offlist and I'll figure it out?
>
> ? ? ? ? The new session you create is the b-leg. The parameter it takes in
> originate is the a-leg. So you'd do:
>
> var session = new ManagedSession();
> session.Originate(context.Session, "sofia/default/1000",10);
>
> ? ? ? ? As to non-blocking, I'm quite sure it's possible, but I don't recall
> offhand which functions. This should be the same as in any other language
> for FreeSWITCH -- these functions are just passthrough from the FS C++ API.
>
> -Michael
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lukasz
> Zwierko
> Sent: Wednesday, July 29, 2009 12:13 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_managed users?
>
> Hi Michael ,
>
> thanks a lot for support on this.
>
>> As to the main problem of your DLL not working, can you send me the full
>> source code, or all the logging output from loading it? Try "managedreload
>> my.dll" to reload the DLL and see how it is registering them. It should
>> output something like "Registering API FullName with Aliases fullname,
>> shortname".
>>
>
> I'm just using the Demo.cs example, I compile it to dll undef VC#, not
> mono, maybe that is the difference?
> The output from the log is just as you stated "Registering API
> FullName with Aliases fullname, shortname". The difference between
> loading dll and csx is that, when loading csx all api and app classes
> are listed as registered, while with dll nothing is listed..
>
> Anyway, I have another question regarding usage of the CoreSession and
> ManagedSession object.
> Basically in my script I want to start new session and originate a call.
> So what I do is
>
> ManageSession session = new ManagedSession();
> session.Originate(session,"sofia/default/1000",10);
>
> And it works but I have some doubts. First thing is, why the first
> param of the Originate method is the CoreSession object? Can't it just
> use 'this'?? Or is there more to it?
>
> Second thing is the third parameter - timeout in seconds. Can't the
> call be started in non-blocking mode? I can start in a different
> thread of course, is that the intended behavior?
>
> Thanks for help,
>
> ?ukasz
>
>
> 2009/7/28 Michael Giagnocavo <mgg at giagnocavo.net>:
>> Hello Lukasz,
>>
>>? ? ? ? Thanks for testing mod_managed. I apologize for the problems you've
>> encountered, and I'll try to sort them out for you.
>>
>> A few things first:
>>
>>? ? ? ? - Scripting support: This is made to allow "true" scripts, as
>> invoked as an EXE - similar to the Lua and spidermonkey support. So, without
>> a Main(), it won't compile as an EXE. If you aren't using it as a script,
>> then an empty Main method will work fine.
>>
>>? ? ? ? - Entry points must be public for Mono. I'll update the demo code
>> to make sure that Main is public. This is a bug in Mono's lightweight code
>> generation -- it won't skip the JIT access checks.
>>
>> As to the main problem of your DLL not working, can you send me the full
>> source code, or all the logging output from loading it? Try "managedreload
>> my.dll" to reload the DLL and see how it is registering them. It should
>> output something like "Registering API FullName with Aliases fullname,
>> shortname".
>>
>> Thanks,
>> Michael
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lukasz
>> Zwierko
>> Sent: Tuesday, July 28, 2009 1:26 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] mod_managed users?
>>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> Hi,
>>
>> I've just tried new mod_managed under Win32 and I get a weird behavior.
>> I try the example below:
>>
>> public class DemoScript : IApiPlugin
>> {
>>? ? public void Execute(ApiContext context)
>>? ? {
>>? ? ? ? context.Stream.Write(string.Format("DemoScripts executed with
>> args '{0}' and event type {1}.",
>>? ? ? ? ? ? context.Arguments, context.Event == null ? "<none>" :
>> context.Event.GetEventType()));
>>? ? }
>>
>>? ? public void ExecuteBackground(ApiBackgroundContext context)
>>? ? {
>>? ? ? ? Log.WriteLine(LogLevel.Notice, "DemoScripts on a background
>> thread #({0}), with args '{1}'.",
>>? ? ? ? ? ? System.Threading.Thread.CurrentThread.ManagedThreadId,
>>? ? ? ? ? ? context.Arguments);
>>? ? }
>> }
>>
>> It's just like the ApiDemo from Demo.cs
>>
>> So When I copy DemoScript.csx to managed dir the console log is:
>>
>> "
>> freeswitch at Zwierko-laptop> Loading
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx from
>> domain DemoScripts.csx_3
>> 2009-07-28 20:57:16.710000 [INFO] switch_cpp.cpp:1130 Compiling
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx
>> 2009-07-28 20:57:16.970000 [ERR] switch_cpp.cpp:1130 There were 1 errors
>> compiling
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx.
>> 2009-07-28 20:57:16.970000 [ERR] switch_cpp.cpp:1130 CS5001: Program
>> 'c:\Users\Zwierko\AppData\Local\Temp\fc8jnlir.exe' does not contain a
>> static 'Main' method
>> suitable for an entry point
>> "
>>
>> Adding
>>
>> "
>>? ? public static void Main()
>>? ? {
>>? ? }
>> "
>>
>> solves the issue. Is this how it's supposed to work?
>>
>> Another strange thing is that when I compile this class to DLL (release)
>> it does not work at all...
>>
>> freeswitch at Zwierko-laptop> Loading
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\FSScripts.dll from
>> domain FSScripts.dll_5
>>
>>
>> freeswitch at Zwierko-laptop>
>>
>> freeswitch at Zwierko-laptop> managed DemoScript 111
>> API CALL [managed(DemoScript 111)] output:
>>
>> 2009-07-28 21:13:03.542000 [ERR] switch_cpp.cpp:1130 API plugin
>> DemoScript not found.
>> 2009-07-28 21:13:03.542000 [ERR] mod_managed.cpp:393 Execute failed for
>> DemoScript 111 (unknown module or exception).
>>
>> And another issue with scripts. I use script code as example:
>>
>> "
>> public class ScriptDemo
>> {
>>? ? static void Main()
>>? ? {
>>? ? ? ? switch (FreeSWITCH.Script.ContextType)
>>? ? ? ? {
>>? ? ? ? ? ? case ScriptContextType.Api:
>>? ? ? ? ? ? ? ? {
>>? ? ? ? ? ? ? ? ? ? var ctx = FreeSWITCH.Script.GetApiContext();
>>? ? ? ? ? ? ? ? ? ? ctx.Stream.Write("Script executing as API with args:
>> " + ctx.Arguments);
>>? ? ? ? ? ? ? ? ? ? break;
>>? ? ? ? ? ? ? ? }
>>? ? ? ? ? ? case ScriptContextType.ApiBackground:
>>? ? ? ? ? ? ? ? {
>>? ? ? ? ? ? ? ? ? ? var ctx = FreeSWITCH.Script.GetApiBackgroundContext();
>>? ? ? ? ? ? ? ? ? ? Log.WriteLine(LogLevel.Notice, "Executing as
>> APIBackground with args: " + ctx.Arguments);
>>? ? ? ? ? ? ? ? ? ? break;
>>? ? ? ? ? ? ? ? }
>>? ? ? ? ? ? case ScriptContextType.App:
>>? ? ? ? ? ? ? ? {
>>? ? ? ? ? ? ? ? ? ? var ctx = FreeSWITCH.Script.GetAppContext();
>>? ? ? ? ? ? ? ? ? ? Log.WriteLine(LogLevel.Notice, "Executing as App
>> with args: " + ctx.Arguments);
>>? ? ? ? ? ? ? ? ? ? break;
>>? ? ? ? ? ? ? ? }
>>? ? ? ? }
>>
>>? ? }
>> }
>> "
>>
>>
>> console log is as follows:
>>
>>
>> freeswitch at Zwierko-laptop> Loading
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx from
>> domain Script.csx_8
>> 2009-07-28 21:19:36.289000 [INFO] switch_cpp.cpp:1130 Compiling
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx
>> 2009-07-28 21:19:36.438000 [INFO] switch_cpp.cpp:1130 File
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx compiled
>> successfully.
>> 2009-07-28 21:19:36.451000 [ERR] switch_cpp.cpp:1130 Entry point:
>> ScriptDemo.Main is not public. This may cause errors with Mono.
>> 2009-07-28 21:19:36.458000 [NOTICE] switch_cpp.cpp:1130 Loaded App
>> Script.csx, aliases 'Script.csx', into domain Script.csx_8.
>> 2009-07-28 21:19:36.459000 [NOTICE] switch_cpp.cpp:1130 Loaded Api
>> Script.csx, aliases 'Script.csx', into domain Script.csx_8.
>> 2009-07-28 21:19:36.459000 [INFO] switch_cpp.cpp:1130 Finished loading
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx into
>> domain Script.csx_8.
>> managed ScriptDemo 111
>> 2009-07-28 21:19:53.452000 [ERR] switch_cpp.cpp:1130 API plugin
>> ScriptDemo not found.
>> API CALL [managed(ScriptDemo 111)] output:
>>
>> 2009-07-28 21:19:53.452000 [ERR] mod_managed.cpp:393 Execute failed for
>> ScriptDemo 111 (unknown module or exception).
>>
>>
>> Again, am I doing something wrong in here?
>>
>> Thanks, for any feedback
>>
>> Lukasz Zwierko
>>
>>
>>
>>
>>
>> Michael Giagnocavo wrote:
>>> Ah, that's embarrassing. I added them and tried building
>>> FreeSWITCH.Managed from svn and it worked fine now. (I'll kick off a new
>>> complete build in a minute.)
>>>
>>> -Michael
>>>
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego
>>> Toro
>>> Sent: Sunday, July 26, 2009 8:47 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] mod_managed users?
>>>
>>> Hi Michael,
>>>
>>> Thank you for your job with mod_managed, I get lastest version with
>>> mod_managed but the files PluginInterfaces.cs, PluginManager.cs and
>>> ScriptPluginManager.cs were not downloaded.
>>>
>>> Diego
>>>
>>>
>>> --- On Sun, 7/26/09, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
>>>
>>> From: Michael Giagnocavo <mgg at giagnocavo.net>
>>> Subject: Re: [Freeswitch-users] mod_managed users?
>>> To: "freeswitch-users at lists.freeswitch.org"
>>> <freeswitch-users at lists.freeswitch.org>
>>> Date: Sunday, July 26, 2009, 2:18 AM
>>> Hello,
>>>
>>>? ? ? ? ? ? ? ???I just checked in a new mod_managed. It breaks backwards
>>> compatibility, but adds scripting and reloading support.
>>>
>>>? ? ? ? ? ? ? ???I tested it on CentOS 5.3 x64 with Mono 2.4.2.2. Just
>>> make & make install seemed to take care of everything.
>>>
>>>? ? ? ? ? ? ? ???Let me know if you have better luck with this version.
>>>
>>> Thanks,
>>> Michael
>>>
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad
>>> Shahzad
>>> Sent: Saturday, July 18, 2009 6:58 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] mod_managed users?
>>>
>>> Sorry for replying late.
>>>
>>> I have tried mod_managed again on same machine (Lenovo 3000 N200), same
>>> OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249.
>>>
>>> It compiles correctly this time but gives following error upon "make
>>> install",
>>>
>>> =====================================================================
>>> making install mod_managed
>>> make[5]: *** No rule to make target
>>> `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'.
>>> Stop.
>>> make[4]: *** [install] Error 1
>>> make[3]: *** [mod_managed-install] Error 1
>>> make[2]: *** [install-recursive] Error 1
>>>
>>> =====================================================================
>>>
>>> Here is compilation log when executing "make", if it could of any help.
>>>
>>> =====================================================================
>>> making all mod_managed
>>> Compiling freeswitch_managed.cpp...
>>> g++ -I/usr/src/svn-src/freeswitch/src/include
>>> -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>> -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0
>>> -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o
>>> freeswitch_managed.cpp
>>> Compiling freeswitch_wrap.cpp...
>>> g++ -I/usr/src/svn-src/freeswitch/src/include
>>> -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>> -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0
>>> -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o
>>> freeswitch_wrap.cpp
>>> Demo.cs(58,14): warning CS0169: The private method
>>> `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used
>>> Compilation succeeded - 1 warning(s)
>>> Compiling mod_managed.cpp...
>>> /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++
>>> -I/usr/src/svn-src/freeswitch/src/include
>>> -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>> -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0
>>> -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo
>>> mod_managed.cpp
>>> libtool: compile:? g++ -I/usr/src/svn-src/freeswitch/src/include
>>> -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>> -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0
>>> -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp? -fPIC -DPIC
>>> -o .libs/mod_managed.o
>>> mod_managed.cpp: In function 'void InitManagedSession(ManagedSession*,
>>> char* (*)(void*, switch_input_type_t), void (*)())':
>>> mod_managed.cpp:97: warning: deprecated conversion from string constant
>>> to 'char*'
>>> libtool: compile:? g++ -I/usr/src/svn-src/freeswitch/src/include
>>> -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>> -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0
>>> -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o
>>> mod_managed.o >/dev/null 2>&1
>>> Creating mod_managed.la...
>>> cat: .libs/mod_managed.log: No such file or directory
>>>
>>> =====================================================================
>>>
>>> Thank you.
>>> On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad
>>> <shaheryarkh at googlemail.com<http://us.mc335.mail.yahoo.com/mc/compose?to=shaheryarkh at googlemail.com>>
>>> wrote:
>>> I tried to install mod_managed on ubuntu-9.04, mono framework version
>>> 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG
>>> related. Since i am not a expert in SWIG so i disabled this module. This
>>> happend long ago, i think FS svn revision 136xx.
>>>
>>> Let me try to compile it from latest FS revision and see if it works. I
>>> will let you know the results.
>>>
>>> Thank you.
>>>
>>> On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro
>>> <dftoro at yahoo.com<http://us.mc335.mail.yahoo.com/mc/compose?to=dftoro at yahoo.com>>
>>> wrote:
>>> Hey, I am here? :)
>>>
>>> I am working with mod_managed on Windows 2003 and Windows Vista with
>>> sucessfull.? I noted on user list the issue with LoadFile on Loader.cs when
>>> a assembly had reference to others assemblies, I change LoadFile by LoadFrom
>>> and the load is made fine.
>>>
>>> I use c# application and sqlserver 2005, using FS and mod_managed.
>>>
>>> Diego
>>>
>>> --- On Thu, 7/16/09, Michael Giagnocavo
>>> <mgg at giagnocavo.net<http://us.mc335.mail.yahoo.com/mc/compose?to=mgg at giagnocavo.net>>
>>> wrote:
>>>
>>> From: Michael Giagnocavo
>>> <mgg at giagnocavo.net<http://us.mc335.mail.yahoo.com/mc/compose?to=mgg at giagnocavo.net>>
>>> Subject: [Freeswitch-users] mod_managed users?
>>> To:
>>> "freeswitch-users at lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users at lists.freeswitch.org>"
>>> <freeswitch-users at lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users at lists.freeswitch.org>>
>>> Date: Thursday, July 16, 2009, 4:43 PM
>>>
>>> Hey, if there are any mod_managed users on this list, I'd love it if you
>>> were able to let me know. I'd like to get feedback, positive or negative, on
>>> what worked, what didn't, and how mod_managed can improve for you. Feel free
>>> to write on list or directly to me: mgg at
>>> giagnocavo.net<http://giagnocavo.net/>
>>>
>>> Thanks!
>>> -Michael
>>> -----Inline Attachment Follows-----
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org<http://www.freeswitch.org/>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org<http://www.freeswitch.org/>
>>>
>>>
>>> --
>>> Muhammad Shahzad
>>> -----------------------------------
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +92 334 422 40 88
>>> MSN:
>>> shari_786pk at hotmail.com<http://us.mc335.mail.yahoo.com/mc/compose?to=shari_786pk at hotmail.com>
>>> Email:
>>> shaheryarkh at googlemail.com<http://us.mc335.mail.yahoo.com/mc/compose?to=shaheryarkh at googlemail.com>
>>>
>>>
>>>
>>> --
>>> Muhammad Shahzad
>>> -----------------------------------
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +92 334 422 40 88
>>> MSN:
>>> shari_786pk at hotmail.com<http://us.mc335.mail.yahoo.com/mc/compose?to=shari_786pk at hotmail.com>
>>> Email:
>>> shaheryarkh at googlemail.com<http://us.mc335.mail.yahoo.com/mc/compose?to=shaheryarkh at googlemail.com>
>>>
>>> -----Inline Attachment Follows-----
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org<http://www.freeswitch.org/>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> -----BEGIN PGP SIGNATURE-----
>> Version: GnuPG v1.4.9 (MingW32)
>> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>>
>> iQEcBAEBAgAGBQJKb1CvAAoJED7LBosr0F2u6E4H/i0SVOJDrh4+3ex6nEDnVJQl
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>> pwTiGQACFdWWiWiYk/J09ZbRJR24S8zHxuETQK93/7fy53tgqW6o35hLxb3arOaH
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>> =LgK4
>> -----END PGP SIGNATURE-----
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From email.list.subscriber at gmail.com  Thu Aug 13 12:10:13 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Thu, 13 Aug 2009 15:10:13 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>
	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>
Message-ID: <4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>

Hi Michal,

Just checking in to see if you've been able to take a stab at this.

Thanks,
Vladimir


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michal Bielicki
Sent: Tuesday, August 11, 2009 5:33 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

I'll retst it later today and give you a link with instructions

Am 10.08.2009 um 20:14 schrieb vmorales:

> By "./compile" I was referring to "./configure"
>
> Vladimir
>
> -----Original Message-----
> From: vmorales [mailto:email.list.subscriber at gmail.com]
> Sent: Monday, August 10, 2009 11:49 AM
> To: 'freeswitch-users at lists.freeswitch.org'
> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> Thanks for the response(s):
>
> I ran the "./compile" script with a set PREFIX.  This took a few
> attempts with errors before it was able to complete error-free, as I
> had to install libtool.
>
> Since then, I have tried running 'make', 'gmake', and
> '/opt/gnu/bin/make', but each results with an error.  This is the
> error when running 'make' or 'gmake':
>
> <snip>
> make: Fatal error: Command failed for target `all-recursive'
> Current working directory /home/vmorales/freeswitch-1.0.4
> *** Error code 1
> make: Fatal error: Command failed for target `all'
> </snip>
>
>
> This is the error when running '/opt/gnu/bin/make':
>
> <snip>
> make[5]: *** [mod_amr.so] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_amr-all] Error 1
> make[2]: *** [all-recursive] Error 1
> Making all in build
> +-------- FreeSWITCH Build Complete -----------+
> + FreeSWITCH has been successfully built.      +
> + Install by running:                          +
> +                                              +
> +               /opt/gnu/bin/make install                   +
> +----------------------------------------------+
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> </snip>
>
>
> I re-untar'd before each compile/make attempt.  Let me know if this
is
> something that I can resolve.
>
> Vladimir
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Jerris
> Sent: Saturday, August 08, 2009 12:37 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> This is not currently a supported platform, it only builds on 64 bit
> right now I think on solaris.
>
> Mike
>
> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>
>> Hello,
>>
>> Does anyone have, or know where to get, a pre-compiled copy of
>> FreeSwitch for Solaris 10/x86?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
> http://www.freeswitch.org

Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. info at halokwadrat.de
e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
Hauptgesch?ftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 0000153539




From rswagoner at gmail.com  Thu Aug 13 12:34:05 2009
From: rswagoner at gmail.com (Ryan Wagoner)
Date: Thu, 13 Aug 2009 15:34:05 -0400
Subject: [Freeswitch-users] Sangoma A102 Overrun Issue
Message-ID: <7d86ddb90908131234t7aad15afk2fa3c85e46db5499@mail.gmail.com>

I've been trying to bridge FreeSWITCH with a Toshiba CIX using qsig
over a PRI. I have a Sangoma A102 card installed in a Dell PowerEdge
with CentOS 5.3. The issue I am having is no packets are being
transmitted back to FreeSWITCH. ifconfig w1g1 shows every frame
received as an overrun. I've tried a different server with CentOS 5.3
with the same issue. I have a support ticket in with Sangoma, but was
wondering if anybody had seen this before.

The T1 shows connected so I think I have the Toshiba configured
properly. From what I've read the overrun has to deal with the driver
not reading the data in time so maybe this is a CentOS 5.3 specific
issue. Any recommendations on alternative Linux distros known to work
with the Sangoma A102 card?

Thanks,
Ryan

[root at voip ~]# ifconfig w1g1
w1g1      Link encap:Point-to-Point Protocol
          UP POINTOPOINT RUNNING NOARP  MTU:80  Metric:1
          RX packets:22298 errors:0 dropped:0 overruns:274 frame:274
          TX packets:22298 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:100
          RX bytes:1783840 (1.7 MiB)  TX bytes:1783840 (1.7 MiB)
          Interrupt:169 Memory:f8e80000-f8e81fff


[root at voip ~]# wanrouter status

Devices currently active:
        wanpipe1


Wanpipe Config:

Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK | Baud rate |
wanpipe1    | N/A          | A101/1D/A102/2D/4/4D/8| 169 | 4       | 1
   | N/A | 0         |

Wanrouter Status:

Device name | Protocol | Station | Status        |
wanpipe1    | AFT TE1  | N/A     | Connected     |


[root at voip ~]# wanpipemon -i w1g1 -c Ta

***** w1g1: T1 Alarms (Framer) *****

ALOS:   OFF     | LOS:  OFF
RED:    OFF     | AIS:  OFF
RAI:    OFF     | OOF:  OFF

***** w1g1: T1 Alarms (LIU) *****

Short Circuit:  OFF
Open Circuit:   OFF
Loss of Signal: OFF


***** w1g1: T1 Performance Monitoring Counters *****

Line Code Violation     : 45
Bit Errors (CRC6/Ft/Fs) : 0
Out of Frame Errors     : 0


Rx Level        : > -2.5db


From msc at freeswitch.org  Thu Aug 13 12:44:35 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 13 Aug 2009 14:44:35 -0500
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>
Message-ID: <87f2f3b90908131244q4f157d6en2947ae5d7ed4e395@mail.gmail.com>

On Thu, Aug 13, 2009 at 12:47 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> I'm sure that would work, but I'm worried about it sucking up bandwidth,
> especially since you'd need it on every caller (since otherwise the one
> person who had it could hang up and you'd be back to square 1).
>
> Any other ideas, or should I hunt through the code to try to override the
> behavior manually?
>
> BB
>

Get a packet capture and debug trace of the symptom occurring. Put the PCAP
where we can download it and pastebin the debug log. We need to confirm who
the culprit is and what events precipitate the disconnect. I'd also be
curious to know if anyone else can reproduce these symptoms. BTW, which rev
of FS are you running?

Thanks,
MC
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From mike at jerris.com  Thu Aug 13 13:20:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 13 Aug 2009 16:20:32 -0400
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>
Message-ID: <BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>

My guess is that its the other end killing the call due to rtp  
timeouts, not us killing the call.  If you can confirm this the best  
method would be to get them not to do rtp timeouts.

On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:

> I'm sure that would work, but I'm worried about it sucking up  
> bandwidth, especially since you'd need it on every caller (since  
> otherwise the one person who had it could hang up and you'd be back  
> to square 1).
>
> Any other ideas, or should I hunt through the code to try to  
> override the behavior manually?
>
> BB
>
> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
> Check out the 'waste' member flag. I think if at least one member  
> has that set then RTP will get sent out even during silence. Let us  
> know if that helps...
>
> -MC
>
> On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <bjbrashier at gmail.com 
> > wrote:
> Hi all.
>
> The solution to this one should be short.
>
> My conference hangs up when there's 2+ users and silence for 5 sec  
> or so. I'm trying to find a parameter that changes that (I'd rather  
> it be, say, 60 seconds).
>
> I didn't see a parameter like this specific to conferences, so I  
> looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but  
> it's set to 300 (the default), so I'm pretty sure that's not the  
> problem. I also searched through the mod_conference.c code and  
> didn't see it, though I was only skimming.
>
> I'm not 100% convinced that this is limited to conferences, but I  
> don't currently have a way to test in a non-conference environment.
>
> Anybody know how to increase the conference silence-hangup timeout?
>
> BB
>
> _____
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From bjbrashier at gmail.com  Thu Aug 13 13:23:27 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 13:23:27 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <87f2f3b90908131244q4f157d6en2947ae5d7ed4e395@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com> 
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com> 
	<87f2f3b90908131244q4f157d6en2947ae5d7ed4e395@mail.gmail.com>
Message-ID: <7bcfdd290908131323r189f4585v72cf53e9a8b0321c@mail.gmail.com>

I'm currently running current trunk synched up Tues morning, but it was
happening in all of the versions I'd been using previous -- I first
downloaded around the end of May.

I'll look into getting you a PCap. I expected that this was a known thing
with a parameter somewhere, so I haven't looked into it too terribly far
myself, yet. I'm gonna try looking at the console outputs and logs myself,
first.

BB
On Thu, Aug 13, 2009 at 12:44 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
>  On Thu, Aug 13, 2009 at 12:47 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
>
>> I'm sure that would work, but I'm worried about it sucking up bandwidth,
>> especially since you'd need it on every caller (since otherwise the one
>> person who had it could hang up and you'd be back to square 1).
>>
>> Any other ideas, or should I hunt through the code to try to override the
>> behavior manually?
>>
>> BB
>>
>
> Get a packet capture and debug trace of the symptom occurring. Put the PCAP
> where we can download it and pastebin the debug log. We need to confirm who
> the culprit is and what events precipitate the disconnect. I'd also be
> curious to know if anyone else can reproduce these symptoms. BTW, which rev
> of FS are you running?
>
> Thanks,
> MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bjbrashier at gmail.com  Thu Aug 13 13:45:23 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 13:45:23 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com> 
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com> 
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
Message-ID: <7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com>

I had just thought of the exact same thing. I'm trying to test that now.
Thanks for your input.

BB

On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com> wrote:

> My guess is that its the other end killing the call due to rtp timeouts,
> not us killing the call.  If you can confirm this the best method would be
> to get them not to do rtp timeouts.
>  On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>
>  I'm sure that would work, but I'm worried about it sucking up bandwidth,
> especially since you'd need it on every caller (since otherwise the one
> person who had it could hang up and you'd be back to square 1).
>
> Any other ideas, or should I hunt through the code to try to override the
> behavior manually?
>
> BB
>
> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Check out the 'waste' member flag. I think if at least one member has that
>> set then RTP will get sent out even during silence. Let us know if that
>> helps...
>>
>> -MC
>>
>>   On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <
>> bjbrashier at gmail.com> wrote:
>>
>>>   Hi all.
>>>
>>> The solution to this one should be short.
>>>
>>> My conference hangs up when there's 2+ users and silence for 5 sec or so.
>>> I'm trying to find a parameter that changes that (I'd rather it be, say, 60
>>> seconds).
>>>
>>> I didn't see a parameter like this specific to conferences, so I looked
>>> abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300
>>> (the default), so I'm pretty sure that's not the problem. I also searched
>>> through the mod_conference.c code and didn't see it, though I was only
>>> skimming.
>>>
>>> I'm not 100% convinced that this is limited to conferences, but I don't
>>> currently have a way to test in a non-conference environment.
>>>
>>> Anybody know how to increase the conference silence-hangup timeout?
>>>
>>> BB
>>>
>>> _____
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bjbrashier at gmail.com  Thu Aug 13 14:48:55 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 14:48:55 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com> 
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com> 
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
	<7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com>
Message-ID: <7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com>

I took a closer look at the SIP messages on the console. From it, I
understand that it's not Freeswitch timing out, but rather FS is getting the
"BYE" msg from somewhere else. I've tested phones and tested calling without
going through the FS conference, though, and everything works fine. Then I
saw something else odd inside the BYE msg:

   Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"
So I Googled "RTP Broken Connection" and saw several sites talking about
AudioCodes gateways and Asterisk -- and our gateway is an AudioCodes. From
these sites it sounds like AudioCodes is rather aggressive in detecting RTP
breaks, and is interpreting the silence from FS as a break.

So now I'm looking into ways to maybe send "I'm still here" RTP packets from
FS or to tune the gateway to be less aggressive. I can't stop and get a
clean packet capture right now because I've got a bunch of testers working
on it today. I'll do that sometime when the system is less busy.

BB

On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> I had just thought of the exact same thing. I'm trying to test that now.
> Thanks for your input.
>
> BB
>
>   On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com> wrote:
>
>>   My guess is that its the other end killing the call due to rtp
>> timeouts, not us killing the call.  If you can confirm this the best method
>> would be to get them not to do rtp timeouts.
>>  On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>>
>>  I'm sure that would work, but I'm worried about it sucking up bandwidth,
>> especially since you'd need it on every caller (since otherwise the one
>> person who had it could hang up and you'd be back to square 1).
>>
>> Any other ideas, or should I hunt through the code to try to override the
>> behavior manually?
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Check out the 'waste' member flag. I think if at least one member has
>>> that set then RTP will get sent out even during silence. Let us know if that
>>> helps...
>>>
>>> -MC
>>>
>>>   On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <
>>> bjbrashier at gmail.com> wrote:
>>>
>>>>   Hi all.
>>>>
>>>> The solution to this one should be short.
>>>>
>>>> My conference hangs up when there's 2+ users and silence for 5 sec or
>>>> so. I'm trying to find a parameter that changes that (I'd rather it be,
>>>> say, 60 seconds).
>>>>
>>>> I didn't see a parameter like this specific to conferences, so I looked
>>>> abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300
>>>> (the default), so I'm pretty sure that's not the problem. I also searched
>>>> through the mod_conference.c code and didn't see it, though I was only
>>>> skimming.
>>>>
>>>> I'm not 100% convinced that this is limited to conferences, but I don't
>>>> currently have a way to test in a non-conference environment.
>>>>
>>>> Anybody know how to increase the conference silence-hangup timeout?
>>>>
>>>> BB
>>>>
>>>> _____
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From bjbrashier at gmail.com  Thu Aug 13 15:24:19 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 15:24:19 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com> 
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com> 
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
	<7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com> 
	<7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com>
Message-ID: <7bcfdd290908131524s10e209dcla7bebada79ecfa51@mail.gmail.com>

OK, I finally got a moment to do a packet capture and take a look at the
streams.  It became very clear very quickly that what happens is that during
silence the gateway still sends RTP packets to Freeswitch, but Freeswitch
doesn't send any back to the gateway. After 10s of this, the gateway says
"Oh, the RPT must be broken" and it hangs up.

We found a way to turn off this behavior in the gateway, and the good news
is that it did indeed fix the problem. But we'd rather not rely on that as a
long-term solution because then we can't detect and drop RTP streams that
really are broken.

So now I'm back to looking at Freeswitch to figure out how to send just a
single packet every second or so during silence. If anyone knows of a way to
do this, let me know, otherwise I'll get back to you if and when I find one.

BB

On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> I took a closer look at the SIP messages on the console. From it, I
> understand that it's not Freeswitch timing out, but rather FS is getting the
> "BYE" msg from somewhere else. I've tested phones and tested calling without
> going through the FS conference, though, and everything works fine. Then I
> saw something else odd inside the BYE msg:
>
>    Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"
> So I Googled "RTP Broken Connection" and saw several sites talking about
> AudioCodes gateways and Asterisk -- and our gateway is an AudioCodes. From
> these sites it sounds like AudioCodes is rather aggressive in detecting RTP
> breaks, and is interpreting the silence from FS as a break.
>
> So now I'm looking into ways to maybe send "I'm still here" RTP packets
> from FS or to tune the gateway to be less aggressive. I can't stop and get a
> clean packet capture right now because I've got a bunch of testers working
> on it today. I'll do that sometime when the system is less busy.
>
> BB
>
> On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
>
>> I had just thought of the exact same thing. I'm trying to test that now.
>> Thanks for your input.
>>
>> BB
>>
>>   On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com>wrote:
>>
>>>   My guess is that its the other end killing the call due to rtp
>>> timeouts, not us killing the call.  If you can confirm this the best method
>>> would be to get them not to do rtp timeouts.
>>>  On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>>>
>>>  I'm sure that would work, but I'm worried about it sucking up
>>> bandwidth, especially since you'd need it on every caller (since otherwise
>>> the one person who had it could hang up and you'd be back to square 1).
>>>
>>> Any other ideas, or should I hunt through the code to try to override the
>>> behavior manually?
>>>
>>> BB
>>>
>>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Check out the 'waste' member flag. I think if at least one member has
>>>> that set then RTP will get sent out even during silence. Let us know if that
>>>> helps...
>>>>
>>>> -MC
>>>>
>>>>   On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <
>>>> bjbrashier at gmail.com> wrote:
>>>>
>>>>>   Hi all.
>>>>>
>>>>> The solution to this one should be short.
>>>>>
>>>>> My conference hangs up when there's 2+ users and silence for 5 sec or
>>>>> so. I'm trying to find a parameter that changes that (I'd rather it be,
>>>>> say, 60 seconds).
>>>>>
>>>>> I didn't see a parameter like this specific to conferences, so I looked
>>>>> abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300
>>>>> (the default), so I'm pretty sure that's not the problem. I also searched
>>>>> through the mod_conference.c code and didn't see it, though I was only
>>>>> skimming.
>>>>>
>>>>> I'm not 100% convinced that this is limited to conferences, but I don't
>>>>> currently have a way to test in a non-conference environment.
>>>>>
>>>>> Anybody know how to increase the conference silence-hangup timeout?
>>>>>
>>>>> BB
>>>>>
>>>>> _____
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From pjintheusa at gmail.com  Thu Aug 13 15:59:19 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 13 Aug 2009 18:59:19 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
	<b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
Message-ID: <367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>

Rupa / all,

Just a quick follow up to this.

This is appears to a timing issue. If I try and do a "uuid_media off +
uuid"  in  "api_after_bridge"  it fails with "CHAN_NOT_IMPLEMENTED"
and  the call is dropped.

If appears to be trying to do a SIP reinvite on the loopback channel
which is of course just about to / has disappear/ed.

So I tried this, after the call is established, at the commend line, I
do "show calls" and using the uuid shown, type "uuid_media off  uuid".
The SIP REINVITE is issued and works.

I think the switch_ivr_nomedia function in switch_ivr_c is getting the
loopback uuid when it calls "other_uuid =
switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE)"

That's why the SIP REINVITE fails.

So... in api_after_bridge I issue:

"sched_api", "+3 none uuid_media off " + uuid. This calls the
switch_ivr_nomedia function 3 seconds after the calls bridge is
established.


And it works, Not nice - not scalable or production ready - but the
SIP-REINVITE is successful and at least now I understand what is going
on.

Make sense?

Thanks


Phil


On Wed, Aug 12, 2009 at 12:21 PM, Rupa Schomaker<rupa at rupa.com> wrote:
> On Wed, Aug 12, 2009 at 10:22 AM, Phillip Jones<pjintheusa at gmail.com> wrote:
>> Hi there,
>>
>>>> application="originate" data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"
>>
>> I agree. However, perhaps the ideal is not to specify the carriers at
>> this level, as carriers are added and removed fairly often as costings
>> change. So it would be nice to have some sort of proxy that resolves
>> to a list of carriers:
>>
>> application="originate" data="sofia/MyCarriers/bar,sofia/MyCarriers/yum"
>
>> <MyCarriers>
>> <action application="carrier" data="sofia/foo"/>
>> <action application="carrier" data="sofia/baz"/>
>> <action application="carrier" data="sofia/etc"/>
>> </MyCarriers>
>>
>>
>> or something similar. This would achieve the same as loopback in this
>> use case but without dangers of looping? Complicated stuff though.
>
> Well, that is all done by mod_lcr. ?I was simplifying to narrow down
> to just originate.
>
> First we need to see if this is worth pursuing over fixing (modifying,
> whatever) loopback to handle bypass media. ?If it is, then I'll modify
> mod_lcr to deal with the situation in question (comma or pipe sep list
> of numbers to call. ?mod_lcr would then group as appropriate).
>
> Right now, my bridge is setup in a small javascript script that builds
> the appropriate dialstring (using loopback for external calls, user/
> for internal calls) and then when doing the loopback call to mod_lcr
> to get the dialstring with all providers in the right order.
>
>>>>Perhaps have an on answer hook that tries to enable bypass media (re-invite) after the call is setup?
>>
>> That's a good idea - I will look into that.
>>
>>
>> Thanks again.
>>
>>
>> Phillip
>
> Let us know how it works for you...
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mrene_lists at avgs.ca  Thu Aug 13 16:54:08 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 13 Aug 2009 19:54:08 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
	<b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
	<367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>
Message-ID: <FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>

Hi All,

The reason it works when you wait 3 seconds is that mod_loopback bails  
out of the equation as soon as it detects a bridge.
It'll call switch_ivr_uuid_bridge() and you'll end up with 2 sofia  
channels.

Now the reason why you can't do uuid_media over a loopback channel is  
because it doesn't pass on SWITCH_MESSAGE_INDICATE_MEDIA and   
SWITCH_MESSAGE_INDICATE_NOMEDIA onto the underlying channel.
The handler for those two events require accessing channel variables  
on the both channels to get the ip+port of where the audio should go  
through, so that mod_sofia can send a re-invite.
Since mod_loopback is a completely different channel, it has its own  
channel variables, independent from mod_sofia (provided you have sofia  
channels on both side).  That's why even sofia<>loopback won't do  
bypass media.

On another note, mod_sofia will behave differently when it detects its  
being bridge with another sofia channel, providing optimizations when  
both call legs are SIP.

My personal opinion is not to use mod_loopback unless absolutely  
necessary, FreeSWITCH's core is very flexible and there's often a  
(better) way than using mod_loopback.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 13-Aug-09, at 6:59 PM, Phillip Jones wrote:

> Rupa / all,
>
> Just a quick follow up to this.
>
> This is appears to a timing issue. If I try and do a "uuid_media off +
> uuid"  in  "api_after_bridge"  it fails with "CHAN_NOT_IMPLEMENTED"
> and  the call is dropped.
>
> If appears to be trying to do a SIP reinvite on the loopback channel
> which is of course just about to / has disappear/ed.
>
> So I tried this, after the call is established, at the commend line, I
> do "show calls" and using the uuid shown, type "uuid_media off  uuid".
> The SIP REINVITE is issued and works.
>
> I think the switch_ivr_nomedia function in switch_ivr_c is getting the
> loopback uuid when it calls "other_uuid =
> switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE)"
>
> That's why the SIP REINVITE fails.
>
> So... in api_after_bridge I issue:
>
> "sched_api", "+3 none uuid_media off " + uuid. This calls the
> switch_ivr_nomedia function 3 seconds after the calls bridge is
> established.
>
>
> And it works, Not nice - not scalable or production ready - but the
> SIP-REINVITE is successful and at least now I understand what is going
> on.
>
> Make sense?
>
> Thanks
>
>
> Phil
>
>
> On Wed, Aug 12, 2009 at 12:21 PM, Rupa Schomaker<rupa at rupa.com> wrote:
>> On Wed, Aug 12, 2009 at 10:22 AM, Phillip  
>> Jones<pjintheusa at gmail.com> wrote:
>>> Hi there,
>>>
>>>>> application="originate" data="(sofia/foo/bar|sofia/baz/bar), 
>>>>> (sofia/foo/yum|sofia/baz/yum)"
>>>
>>> I agree. However, perhaps the ideal is not to specify the carriers  
>>> at
>>> this level, as carriers are added and removed fairly often as  
>>> costings
>>> change. So it would be nice to have some sort of proxy that resolves
>>> to a list of carriers:
>>>
>>> application="originate" data="sofia/MyCarriers/bar,sofia/ 
>>> MyCarriers/yum"
>>
>>> <MyCarriers>
>>> <action application="carrier" data="sofia/foo"/>
>>> <action application="carrier" data="sofia/baz"/>
>>> <action application="carrier" data="sofia/etc"/>
>>> </MyCarriers>
>>>
>>>
>>> or something similar. This would achieve the same as loopback in  
>>> this
>>> use case but without dangers of looping? Complicated stuff though.
>>
>> Well, that is all done by mod_lcr.  I was simplifying to narrow down
>> to just originate.
>>
>> First we need to see if this is worth pursuing over fixing  
>> (modifying,
>> whatever) loopback to handle bypass media.  If it is, then I'll  
>> modify
>> mod_lcr to deal with the situation in question (comma or pipe sep  
>> list
>> of numbers to call.  mod_lcr would then group as appropriate).
>>
>> Right now, my bridge is setup in a small javascript script that  
>> builds
>> the appropriate dialstring (using loopback for external calls, user/
>> for internal calls) and then when doing the loopback call to mod_lcr
>> to get the dialstring with all providers in the right order.
>>
>>>>> Perhaps have an on answer hook that tries to enable bypass media  
>>>>> (re-invite) after the call is setup?
>>>
>>> That's a good idea - I will look into that.
>>>
>>>
>>> Thanks again.
>>>
>>>
>>> Phillip
>>
>> Let us know how it works for you...
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mrene_lists at avgs.ca  Thu Aug 13 16:57:28 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 13 Aug 2009 19:57:28 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <c4d05cbe0908130802w6528ca34ma9d6e34ab9ce28c2@mail.gmail.com>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
	<87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
	<c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>
	<A681A304-5B94-49B8-BA25-E3089ED3F4A9@freeswitch.org>
	<c4d05cbe0908130802w6528ca34ma9d6e34ab9ce28c2@mail.gmail.com>
Message-ID: <EEE57DE2-9176-49FB-A986-A53275E148E3@avgs.ca>

It probably just VETO it so it avoid sending  
SWITCH_MESSAGE_INDICATE_PROGRESS again since the call is already  
making progress from the core's point of view?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 13-Aug-09, at 11:02 AM, Moises Silva wrote:

> On Wed, Aug 12, 2009 at 7:19 PM, Brian West <brian at freeswitch.org>  
> wrote:
> Well you really can't ignore it... it happens with our ISDN stack
> too.   Thats what the VETO handles.
>
> /b
>
> You lost me. What do you mean we can't ignore it? the way I see it,  
> sure we can and we should.
>
> Currently that warning comes from the on_ringing() callback which  
> blindly attempts to move the state of the zap channel to  
> ZAP_CHANNEL_STATE_PROGRESS, even when the state may be already  
> ZAP_CHANNEL_STATE_PROGRESS_MEDIA (which means on_proceed() was  
> called first).
>
> As I see it, the VETO warning is more an aid to the programmer so  
> you quickly realize your doing a useless state change, which should  
> be fixed. In this case, the fix is simply checking the state of the  
> channel before trying to move it to progress, and don't even try to  
> move it if already in progress with media.
>
>  --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON  
> L3R 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From moises.silva at gmail.com  Thu Aug 13 18:04:13 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 13 Aug 2009 21:04:13 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <EEE57DE2-9176-49FB-A986-A53275E148E3@avgs.ca>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
	<87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
	<c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>
	<A681A304-5B94-49B8-BA25-E3089ED3F4A9@freeswitch.org>
	<c4d05cbe0908130802w6528ca34ma9d6e34ab9ce28c2@mail.gmail.com>
	<EEE57DE2-9176-49FB-A986-A53275E148E3@avgs.ca>
Message-ID: <c4d05cbe0908131804g18cd923ewdc629e0e99ccb50b@mail.gmail.com>

Yes, agreed, but there is no point in sending a WARNING since is a normal
condition, therefore should not even try to change the state of the channel.

On Thu, Aug 13, 2009 at 7:57 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> It probably just VETO it so it avoid sending
> SWITCH_MESSAGE_INDICATE_PROGRESS
> again since the call is already making progress from the core's point of view?
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 13-Aug-09, at 11:02 AM, Moises Silva wrote:
>
> On Wed, Aug 12, 2009 at 7:19 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Well you really can't ignore it... it happens with our ISDN stack
>> too.   Thats what the VETO handles.
>>
>> /b
>>
>
> You lost me. What do you mean we can't ignore it? the way I see it, sure we
> can and we should.
>
> Currently that warning comes from the on_ringing() callback which blindly
> attempts to move the state of the zap channel to ZAP_CHANNEL_STATE_PROGRESS,
> even when the state may be already ZAP_CHANNEL_STATE_PROGRESS_MEDIA (which
> means on_proceed() was called first).
>
> As I see it, the VETO warning is more an aid to the programmer so you
> quickly realize your doing a useless state change, which should be fixed. In
> this case, the fix is simply checking the state of the channel before trying
> to move it to progress, and don't even try to move it if already in progress
> with media.
>
>  --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From Nick.Lemberger at lkfd.net  Thu Aug 13 18:24:35 2009
From: Nick.Lemberger at lkfd.net (Nick Lemberger)
Date: Thu, 13 Aug 2009 20:24:35 -0500
Subject: [Freeswitch-users] OpenSolaris Compile Error [gcc]
Message-ID: <4A847695.2C9A.00FE.0@lkfd.net>

64bit OpenSolaris w/ gcc-4.3.2

After a bootstrap and configure I get the following error when running make:


---snip---

Compiling src/switch_caller.c ...
cc1: warnings being treated as errors
src/switch_caller.c: In function 'switch_caller_profile_event_set_data':
src/switch_caller.c:299: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
src/switch_caller.c:301: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
src/switch_caller.c:303: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
src/switch_caller.c:305: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
src/switch_caller.c:307: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
src/switch_caller.c:309: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
src/switch_caller.c:311: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
make[2]: *** [libfreeswitch_la-switch_caller.lo] Error 1

---snip---

I get this error in both the source for 1.0.4 and last nights snapshot.  An suggestions or ideas?  There are no apparent errors during the bootstrap or configure processes.

Regards,
Nicholas Lemberger
Lakefield Communications




From pjintheusa at gmail.com  Thu Aug 13 20:48:17 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 13 Aug 2009 20:48:17 -0700
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
	<b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
	<367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>
	<FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>
Message-ID: <367751820908132048s2a39b0deu586aa3ecc01c1ac7@mail.gmail.com>

> The reason it works when you wait 3 seconds is that mod_loopback bails
> out of the equation as soon as it detects a bridge.
> It'll call switch_ivr_uuid_bridge() and you'll end up with 2 sofia
> channels.

Is there a hook that is fired when when that switch_ivr_uuid_bridge()
successfully executes? So the uuid_media off is called on the
appropriate sofia channels? Is "api_after_bridge" behaving correctly -
should that only be called on the sofia channels and not the loopback?
Is it being fired to early?



On Thu, Aug 13, 2009 at 4:54 PM, Mathieu Rene<mrene_lists at avgs.ca> wrote:
> Hi All,
>
> The reason it works when you wait 3 seconds is that mod_loopback bails
> out of the equation as soon as it detects a bridge.
> It'll call switch_ivr_uuid_bridge() and you'll end up with 2 sofia
> channels.
>
> Now the reason why you can't do uuid_media over a loopback channel is
> because it doesn't pass on SWITCH_MESSAGE_INDICATE_MEDIA and
> SWITCH_MESSAGE_INDICATE_NOMEDIA onto the underlying channel.
> The handler for those two events require accessing channel variables
> on the both channels to get the ip+port of where the audio should go
> through, so that mod_sofia can send a re-invite.
> Since mod_loopback is a completely different channel, it has its own
> channel variables, independent from mod_sofia (provided you have sofia
> channels on both side). ?That's why even sofia<>loopback won't do
> bypass media.
>
> On another note, mod_sofia will behave differently when it detects its
> being bridge with another sofia channel, providing optimizations when
> both call legs are SIP.
>
> My personal opinion is not to use mod_loopback unless absolutely
> necessary, FreeSWITCH's core is very flexible and there's often a
> (better) way than using mod_loopback.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 13-Aug-09, at 6:59 PM, Phillip Jones wrote:
>
>> Rupa / all,
>>
>> Just a quick follow up to this.
>>
>> This is appears to a timing issue. If I try and do a "uuid_media off +
>> uuid" ?in ?"api_after_bridge" ?it fails with "CHAN_NOT_IMPLEMENTED"
>> and ?the call is dropped.
>>
>> If appears to be trying to do a SIP reinvite on the loopback channel
>> which is of course just about to / has disappear/ed.
>>
>> So I tried this, after the call is established, at the commend line, I
>> do "show calls" and using the uuid shown, type "uuid_media off ?uuid".
>> The SIP REINVITE is issued and works.
>>
>> I think the switch_ivr_nomedia function in switch_ivr_c is getting the
>> loopback uuid when it calls "other_uuid =
>> switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE)"
>>
>> That's why the SIP REINVITE fails.
>>
>> So... in api_after_bridge I issue:
>>
>> "sched_api", "+3 none uuid_media off " + uuid. This calls the
>> switch_ivr_nomedia function 3 seconds after the calls bridge is
>> established.
>>
>>
>> And it works, Not nice - not scalable or production ready - but the
>> SIP-REINVITE is successful and at least now I understand what is going
>> on.
>>
>> Make sense?
>>
>> Thanks
>>
>>
>> Phil
>>
>>
>> On Wed, Aug 12, 2009 at 12:21 PM, Rupa Schomaker<rupa at rupa.com> wrote:
>>> On Wed, Aug 12, 2009 at 10:22 AM, Phillip
>>> Jones<pjintheusa at gmail.com> wrote:
>>>> Hi there,
>>>>
>>>>>> application="originate" data="(sofia/foo/bar|sofia/baz/bar),
>>>>>> (sofia/foo/yum|sofia/baz/yum)"
>>>>
>>>> I agree. However, perhaps the ideal is not to specify the carriers
>>>> at
>>>> this level, as carriers are added and removed fairly often as
>>>> costings
>>>> change. So it would be nice to have some sort of proxy that resolves
>>>> to a list of carriers:
>>>>
>>>> application="originate" data="sofia/MyCarriers/bar,sofia/
>>>> MyCarriers/yum"
>>>
>>>> <MyCarriers>
>>>> <action application="carrier" data="sofia/foo"/>
>>>> <action application="carrier" data="sofia/baz"/>
>>>> <action application="carrier" data="sofia/etc"/>
>>>> </MyCarriers>
>>>>
>>>>
>>>> or something similar. This would achieve the same as loopback in
>>>> this
>>>> use case but without dangers of looping? Complicated stuff though.
>>>
>>> Well, that is all done by mod_lcr. ?I was simplifying to narrow down
>>> to just originate.
>>>
>>> First we need to see if this is worth pursuing over fixing
>>> (modifying,
>>> whatever) loopback to handle bypass media. ?If it is, then I'll
>>> modify
>>> mod_lcr to deal with the situation in question (comma or pipe sep
>>> list
>>> of numbers to call. ?mod_lcr would then group as appropriate).
>>>
>>> Right now, my bridge is setup in a small javascript script that
>>> builds
>>> the appropriate dialstring (using loopback for external calls, user/
>>> for internal calls) and then when doing the loopback call to mod_lcr
>>> to get the dialstring with all providers in the right order.
>>>
>>>>>> Perhaps have an on answer hook that tries to enable bypass media
>>>>>> (re-invite) after the call is setup?
>>>>
>>>> That's a good idea - I will look into that.
>>>>
>>>>
>>>> Thanks again.
>>>>
>>>>
>>>> Phillip
>>>
>>> Let us know how it works for you...
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mike at jerris.com  Thu Aug 13 22:47:29 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 01:47:29 -0400
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908131524s10e209dcla7bebada79ecfa51@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
	<7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com>
	<7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com>
	<7bcfdd290908131524s10e209dcla7bebada79ecfa51@mail.gmail.com>
Message-ID: <DB70D4BE-CD40-4707-AD0C-333A6A0BA75E@jerris.com>

My suggestion is to use sip session timers not rtp timeouts as rtp is  
supposed to be discontinuous.  That being said, we have several  
settings to continuously send media, but then you are doing exactly  
what you said you didn't want to do.

Mike

On Aug 13, 2009, at 6:24 PM, Bradley Brashier wrote:

> OK, I finally got a moment to do a packet capture and take a look at  
> the streams.  It became very clear very quickly that what happens is  
> that during silence the gateway still sends RTP packets to  
> Freeswitch, but Freeswitch doesn't send any back to the gateway.  
> After 10s of this, the gateway says "Oh, the RPT must be broken" and  
> it hangs up.
>
> We found a way to turn off this behavior in the gateway, and the  
> good news is that it did indeed fix the problem. But we'd rather not  
> rely on that as a long-term solution because then we can't detect  
> and drop RTP streams that really are broken.
>
> So now I'm back to looking at Freeswitch to figure out how to send  
> just a single packet every second or so during silence. If anyone  
> knows of a way to do this, let me know, otherwise I'll get back to  
> you if and when I find one.
>
> BB
>
> On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier <bjbrashier at gmail.com 
> > wrote:
> I took a closer look at the SIP messages on the console. From it, I  
> understand that it's not Freeswitch timing out, but rather FS is  
> getting the "BYE" msg from somewhere else. I've tested phones and  
> tested calling without going through the FS conference, though, and  
> everything works fine. Then I saw something else odd inside the BYE  
> msg:
>
>    Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"
> So I Googled "RTP Broken Connection" and saw several sites talking  
> about AudioCodes gateways and Asterisk -- and our gateway is an  
> AudioCodes. From these sites it sounds like AudioCodes is rather  
> aggressive in detecting RTP breaks, and is interpreting the silence  
> from FS as a break.
>
> So now I'm looking into ways to maybe send "I'm still here" RTP  
> packets from FS or to tune the gateway to be less aggressive. I  
> can't stop and get a clean packet capture right now because I've got  
> a bunch of testers working on it today. I'll do that sometime when  
> the system is less busy.
>
> BB
>
> On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com 
> > wrote:
> I had just thought of the exact same thing. I'm trying to test that  
> now.  Thanks for your input.
>
> BB
>
> On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> My guess is that its the other end killing the call due to rtp  
> timeouts, not us killing the call.  If you can confirm this the best  
> method would be to get them not to do rtp timeouts.
>
> On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>
>> I'm sure that would work, but I'm worried about it sucking up  
>> bandwidth, especially since you'd need it on every caller (since  
>> otherwise the one person who had it could hang up and you'd be back  
>> to square 1).
>>
>> Any other ideas, or should I hunt through the code to try to  
>> override the behavior manually?
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins  
>> <msc at freeswitch.org> wrote:
>> Check out the 'waste' member flag. I think if at least one member  
>> has that set then RTP will get sent out even during silence. Let us  
>> know if that helps...
>>
>> -MC
>>
>> On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <bjbrashier at gmail.com 
>> > wrote:
>> Hi all.
>>
>> The solution to this one should be short.
>>
>> My conference hangs up when there's 2+ users and silence for 5 sec  
>> or so. I'm trying to find a parameter that changes that (I'd rather  
>> it be, say, 60 seconds).
>>
>> I didn't see a parameter like this specific to conferences, so I  
>> looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but  
>> it's set to 300 (the default), so I'm pretty sure that's not the  
>> problem. I also searched through the mod_conference.c code and  
>> didn't see it, though I was only skimming.
>>
>> I'm not 100% convinced that this is limited to conferences, but I  
>> don't currently have a way to test in a non-conference environment.
>>
>> Anybody know how to increase the conference silence-hangup timeout?
>>
>> BB
>>
>> _____
>

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From rupa at rupa.com  Thu Aug 13 23:33:55 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 14 Aug 2009 01:33:55 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
	<b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
	<367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>
	<FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>
Message-ID: <b39273530908132333y52279d64qe926eca47e15b5eb@mail.gmail.com>

On Thu, Aug 13, 2009 at 6:54 PM, Mathieu Rene<mrene_lists at avgs.ca> wrote:
> Hi All,
>
> The reason it works when you wait 3 seconds is that mod_loopback bails

[snip]

Thanks for that explanation.  It umm.. explains a lot. :)

> On another note, mod_sofia will behave differently when it detects its
> being bridge with another sofia channel, providing optimizations when
> both call legs are SIP.
>
> My personal opinion is not to use mod_loopback unless absolutely
> necessary, FreeSWITCH's core is very flexible and there's often a
> (better) way than using mod_loopback.

So, I think the temp solution is to use loopback+delayed no media.

but the real "solution" is to either drive the forked dialing logic
externally (event socket) or consider supporting groupings in the
bridge which.. umm...  is gonna be a pain and will need buy in from
from Tony and other core devs since that is a core (no pun intended)
piece of code that nearly everything uses.

I'm not sure I want to take a wack at it.

-- 
-Rupa


From bruce.mcalister at blueface.ie  Fri Aug 14 00:49:16 2009
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Fri, 14 Aug 2009 08:49:16 +0100
Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9	&
 v1.0.4)
In-Reply-To: <4a82d138.1f588c0a.2915.4e3d@mx.google.com>
References: <4A8280B8.6050308@blueface.ie>
	<4a82d138.1f588c0a.2915.4e3d@mx.google.com>
Message-ID: <4A8516FC.2030102@blueface.ie>

Hi All,

Anyone have any ideas about this?

Thanks
Bruce

vmorales wrote:
> Hi Bruce,
> 
> I am having similar issues trying build freeswitch 1.0.4 on Solaris
> x86 as well.  I sent some information over the mailing list, and I
> received a response from Michal Bielicki (attached), stating he'd test
> this and direct me to the steps to successfully build freeswitch.
> 
> Just an FYI in case you see his response.
> 
> Vladimir
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Bruce McAlister
> Sent: Wednesday, August 12, 2009 4:44 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 &
> v1.0.4)
> 
> Hi All,
> 
> I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and
> 
> 1.0.4.
> 
> I am running on Solaris 10 Update 5 on x86 hardware (32-bit).
> 
> The build fails with:
> 
> --- snip ---
> make: Fatal error: Command failed for target `all-recursive'
> Current working directory
> /export/home/user/packages/BUILD/freeswitch-1.0.4
> *** Error code 1
> make: Fatal error: Command failed for target `all'
> ---
> 
> Looking back through the build I can see the following error:
> 
> --- snip ---
> creating libfreeswitch.la
> (cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la 
> libfreeswitch.la)
> /usr/bin/cc 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/s
> rc 
> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
> -DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch 
> freeswitch-switch.o  ./.libs/libfreeswitch.so 
> -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/
> expat/lib 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/ex
> pat/lib/.libs/libexpat.a 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libap
> r-1.a 
> -lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp 
> -L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt 
> -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl 
> -lcurses -lsocket  -R/opt/freeswitch/lib -R/usr/sfw/lib
> Undefined                       first referenced
>   symbol                             in file
> herror                              ./.libs/libfreeswitch.so
> ld: fatal: Symbol referencing errors. No output written to
> .libs/freeswitch
> *** Error code 1
> The following command caused the error:
> `if test -z "" ; then echo /bin/bash 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else 
> echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` 
> --tag=CC --mode=link /usr/bin/cc 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/s
> rc 
> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
> -DPATH_MAX=2048 -g -v -Xc -xc99=all  -lm -R/opt/freeswitch/lib  -o 
> freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib 
> freeswitch-switch.o   libfreeswitch.la libs/apr/libapr-1.la 
> libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl 
> -lcrypto -lsocket -lnsl -ldl   -lcurses -lsocket
> --- snip ---
> 
> Then a little above this error, there is the following warning that is
> 
> displayed (I'm not sure if it is related):
> 
> --- snip ---
> *** Warning: Linking the shared library libfreeswitch.la against the
> *** static library libs/libedit/src/.libs/libedit.a is not portable!
> --- snip ---
> 
> My configure line is as follows:
> 
> ---
> ./configure --prefix=/opt/freeswitch
> ---
> 
> I have the complete configure and make output if anyone needs them.
> 
> Any help/pointers would be greatly appreciated.
> 
> Thanks
> Bruce
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From csa at nowthor.com  Fri Aug 14 05:14:29 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Fri, 14 Aug 2009 08:14:29 -0400
Subject: [Freeswitch-users] bind_server_ip issue
In-Reply-To: <12E3506D-3F79-4987-9FA7-B40AE461B97A@freeswitch.org>
References: <4A841FE0.90904@nowthor.com>
	<12E3506D-3F79-4987-9FA7-B40AE461B97A@freeswitch.org>
Message-ID: <4A855525.2080301@nowthor.com>

Hi Brian!

Thank you for your quick response. I ended up defining "local_ip_v4" at 
the top of vars.xml and set "bind_server_ip" to "local_ip_v4" as well. 
Is this the best way to go about selecting an IP address for FS to bind to?

In any case, even though it doesn't appear to affect operation, I am 
still getting this "IP change detected" somewhat periodically in my logs 
(running 1.0.4):

2009-08-14 01:30:21.215446 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 01:30:22.930551 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 02:00:21.226791 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 02:00:22.473798 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 02:10:21.255298 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 02:10:22.730414 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 02:20:21.279193 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 02:20:22.946537 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 02:30:21.297964 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 02:30:23.111418 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 02:40:21.333952 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 02:40:23.282789 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 02:50:21.362973 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 02:50:22.530569 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 03:20:21.409806 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 03:20:22.137788 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 03:30:21.414786 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 03:30:23.390800 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 03:50:21.422794 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 03:50:22.709792 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 04:00:21.422789 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 04:00:22.818539 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 05:00:21.2953 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 05:00:22.19405 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 05:10:21.29951 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 05:10:23.266294 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 05:50:21.98300 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 05:50:21.962418 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 06:00:21.125959 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 06:00:22.182413 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 06:50:21.238928 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 06:50:23.138401 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 07:00:21.246302 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 07:00:23.269782 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 08:00:21.373961 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 08:00:22.503561 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]

Thanks!

Carlos

Brian West wrote:
> If you read the latest vars.xml I have clarified this:
>
>    <!--
>         THIS IS ONLY USED FOR DINGALING
>
>         bind_server_ip
>
>         Can be an ip address, a dns name, or "auto".
>         This determines an ip address available on this host to bind.
>         If you are separating RTP and SIP traffic, you will want to have
>         use different addresses where this variable appears.
>         Used by: dingaling.conf.xml
>    -->
>    <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
>
>
> So you'll need to open up the sip profile in sip_profiles and set the  
> bind ip to exactly what you want.
>
> Thanks,
> Brian
>
>
> On Aug 13, 2009, at 9:14 AM, Carlos S. Antunes wrote:
>
>   
>> Hello!
>>
>> First of all, I would like to express my thanks to all the  
>> developers of
>> Freeswitch.
>>
>> I am testing Freeswitch on a Debian machine with physical network
>> interface with four virtual IP addresses. One of these IP addresses,
>> aliased as eth0:3, has been created specifically for Freeswitch.  I  
>> then
>> set bind_server_ip with the IP addresses associated with eth0:3. To my
>> surprise, however, tow things happen more or less randomly: 1) in
>> certain cases, Freeswitch binds to eth0:2 instead (with a different IP
>> address); and in another, although Freeswitch binds initially to  
>> eth0:3,
>> after a few hours it changes its mind and rebinds to eth0:2. Is this  
>> an
>> issue with bind_server_ip or am I missing some configuration detail?
>>
>> Thanks!
>>
>> Carlos
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
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From markmorreny at gmail.com  Fri Aug 14 05:32:53 2009
From: markmorreny at gmail.com (mark morreny)
Date: Fri, 14 Aug 2009 20:32:53 +0800
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
Message-ID: <20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>

Hi,

Thank you for your help.

I get that too, but the callback does not execute the second time.

When I do task->runtime = switch_time_now() + 10;, what does +10 mean?  Does
it mean 10 s or 10 mins?

Thanks,
Mark

On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
> I did the same thing on my side....
>  API CALL [load(mod_skel)] output:
> +OK
>
> 2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task 2
> data_flush (core) to run at 1250089698
> 2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889
> Successfully Loaded [mod_skel]
> 2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270 Adding API
> Function 'skel'
> freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]
> mod_skel.c:120 starting to flush data  buffer...
>
> Note that you don't need to start the thread manually, the core already has
> threads running for the scheduler.
>
>   Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
>  On 12-Aug-09, at 7:26 AM, mark morreny wrote:
>
> Hi,
>
> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out some data
> every 10 s.  The following lines of code does not show any effect at all.
>
>     switch_scheduler_task_thread_start();
>     switch_scheduler_add_task(switch_epoch_time_now(NULL),
> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|SSHF_NO_DEL);
>
>
> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to flush
> data  buffer...\n");
>
>
>     task->runtime = switch_time_now() + 10;
>
> }
>
> Does anyone know how to get it to work?
>
> Thanks,
> Mark
>
>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> Date: Mon, Aug 10, 2009 at 8:53 PM
> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
> To: freeswitch-users at lists.freeswitch.org
>
>
> switch_rtp.c has a simple one for the zrtp cache storing.
>
> /b
>
> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>
> > Re schedule is done in your callback, take a look at places that use
> > these apis in the code for details.
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Fri Aug 14 06:18:22 2009
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 14 Aug 2009 21:18:22 +0800
Subject: [Freeswitch-users] 403 forbidden on files.freeswitch.org
Message-ID: <DF4D5954-19C7-457F-9958-7DF20A1A00CA@gmail.com>

Got this:

Forbidden
You don't have permission to access /cluecon_2009/presentations/ 
Dale_Building_FreeSWITCH_App_Lua.pptx on this server.

Apache/2.2.3 (CentOS) Server at files-sync.freeswitch.org Port 80

$ ping files.freeswitch.org
PING filessync.freeswitch.netdna-cdn.com (69.174.57.101): 56 data bytes
64 bytes from 69.174.57.101: icmp_seq=0 ttl=46 time=224.900 ms
64 bytes from 69.174.57.101: icmp_seq=1 ttl=46 time=221.597 ms
^C

$ ping files-sync.freeswitch.org
PING www-01.freeswitch.org (216.82.231.69): 56 data bytes
64 bytes from 216.82.231.69: icmp_seq=0 ttl=44 time=271.147 ms
64 bytes from 216.82.231.69: icmp_seq=1 ttl=44 time=273.161 ms
^C



From william.suffill at gmail.com  Fri Aug 14 06:37:27 2009
From: william.suffill at gmail.com (William Suffill)
Date: Fri, 14 Aug 2009 09:37:27 -0400
Subject: [Freeswitch-users] 403 forbidden on files.freeswitch.org
In-Reply-To: <DF4D5954-19C7-457F-9958-7DF20A1A00CA@gmail.com>
References: <DF4D5954-19C7-457F-9958-7DF20A1A00CA@gmail.com>
Message-ID: <6b65470d0908140637h2bf8bf7ch6f682b97d3be66c5@mail.gmail.com>

Given that people were downloading directly from the source that feeds
the CDN (files.freeswitch.org is hosted by a  CDN) it will be some
time before  the files appear available again. It became a problem
with how much bandwidth was being used without using the CDN. Best to
just wait I know it's being worked on but going to need time to get
resolved.


From brian at freeswitch.org  Fri Aug 14 07:09:31 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 14 Aug 2009 09:09:31 -0500
Subject: [Freeswitch-users] 403 forbidden on files.freeswitch.org
In-Reply-To: <6b65470d0908140637h2bf8bf7ch6f682b97d3be66c5@mail.gmail.com>
References: <DF4D5954-19C7-457F-9958-7DF20A1A00CA@gmail.com>
	<6b65470d0908140637h2bf8bf7ch6f682b97d3be66c5@mail.gmail.com>
Message-ID: <1BDE3BFC-4282-4C88-B71F-9919E903CE91@freeswitch.org>

No I'm going to have to put them on torrent.  Too many people were  
watching them off of files-sync on their media center PC's and chewing  
bandwidth galore.

/b

On Aug 14, 2009, at 8:37 AM, William Suffill wrote:

> Given that people were downloading directly from the source that feeds
> the CDN (files.freeswitch.org is hosted by a  CDN) it will be some
> time before  the files appear available again. It became a problem
> with how much bandwidth was being used without using the CDN. Best to
> just wait I know it's being worked on but going to need time to get
> resolved.



From brian at freeswitch.org  Fri Aug 14 07:10:25 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 14 Aug 2009 09:10:25 -0500
Subject: [Freeswitch-users] bind_server_ip issue
In-Reply-To: <4A855525.2080301@nowthor.com>
References: <4A841FE0.90904@nowthor.com>
	<12E3506D-3F79-4987-9FA7-B40AE461B97A@freeswitch.org>
	<4A855525.2080301@nowthor.com>
Message-ID: <CBF2E5B2-4707-4822-9E79-B81D1D86FA83@freeswitch.org>

Don't use local_ip_v4 then... please hard code the param in the sofia  
profile because the value of local_ip_v4 can change.

/b

On Aug 14, 2009, at 7:14 AM, Carlos S. Antunes wrote:

> Hi Brian!
>
> Thank you for your quick response. I ended up defining "local_ip_v4"  
> at the top of vars.xml and set "bind_server_ip" to "local_ip_v4" as  
> well. Is this the best way to go about selecting an IP address for  
> FS to bind to?
>
> In any case, even though it doesn't appear to affect operation, I am  
> still getting this "IP change detected" somewhat periodically in my  
> logs (running 1.0.4):



From juanbackson at gmail.com  Fri Aug 14 09:02:24 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Sat, 15 Aug 2009 00:02:24 +0800
Subject: [Freeswitch-users] sip contact header question
Message-ID: <27c25bc40908140902k78d5bbe1naa75a21df21caa6e@mail.gmail.com>

Hi,

I would like to set outbound INVITE with customer Contact: field.  I did
sip_contact_user="2033334545", but it still did not work.  I am using 1.0.4
release.

Is this a known issue?

br,
JB
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From tina at a2unlimited.com  Fri Aug 14 09:20:45 2009
From: tina at a2unlimited.com (Tina Martinez)
Date: Fri, 14 Aug 2009 12:20:45 -0400
Subject: [Freeswitch-users]  Question about sharing conference between
Message-ID: <37033.1250266845@a2unlimited.com>

Michael,

Thanks again for bearing with my novice perspective on this.

I was able to achieve the link between two FS servers as intended.  However, I
was not able to setup a "new" dialplan file as you described.  I had to place the
script into the default.xml dialplan to get it to work.  Is there something I'm
supposed to do to get FS to look in the conf/dialplan/default ?

Also, in using the code as we discussed, I was also able to establish the link
between the two servers without having an actual soft-phone registered/connected
to the FS server (which is ideal for my situation).  I can create the link
between the servers, and then dial out to external phone numbers from the
respective servers and connect the participants.  Then, regardless of who
hangs-up, the conference between the servers remains in tact.  So far it works
very well.

- T


From gmaruzz at celliax.org  Fri Aug 14 10:43:27 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 14 Aug 2009 19:43:27 +0200
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness patch
Message-ID: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>

Hi FreeSWITCHers,

all the users of mod_skypiax are kindly requested to test the svn trunk 14519.

It contains a lot of changes meant to add stability and robustness,
toward a production environment.

Let me know how your feelings, and please add to the Jira any possible
bug/issue/etc.

Thanks to you all,

-giovanni



Sincerely,

Giovanni Maruzzelli

Cell : +39-347-2665618


From chris at maxpowersoft.com  Fri Aug 14 10:54:42 2009
From: chris at maxpowersoft.com (Chris Danielson)
Date: Fri, 14 Aug 2009 10:54:42 -0700
Subject: [Freeswitch-users] FreeSWITCH Console Released For iPhone/iPod
	Touch Users
Message-ID: <4A85A4E2.2030702@maxpowersoft.com>

Announcing the release of FreeSWITCH Console in the Apple Application 
Store.  The application is FREE and allows you to connect to a 
FreeSWITCH event socket layer module that is bound to an external 
interface.  Great for development purposes and general remote debugging.

Blog announcement:
http://www.chrisdanielson.com/2009/08/14/release-iphoneipod-touch-freeswitch-console/

iTunes Store Link:
http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221&mt=8 
<http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221&mt=8>

Kind Regards,
Chris Danielson


From mike at jerris.com  Fri Aug 14 11:05:01 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 14:05:01 -0400
Subject: [Freeswitch-users] Sangoma A102 Overrun Issue
In-Reply-To: <7d86ddb90908131234t7aad15afk2fa3c85e46db5499@mail.gmail.com>
References: <7d86ddb90908131234t7aad15afk2fa3c85e46db5499@mail.gmail.com>
Message-ID: <BC34F7E2-5873-48D0-A8CE-CB450C1D3ED6@jerris.com>

I know sangoma is working on their new pri stack, maybe it has proper  
qsig support.  The openzap isdn stack does not right now and I was  
under the impression that the libpri support was pretty limited, but  
have no direct knowledge there.

Mike

On Aug 13, 2009, at 3:34 PM, Ryan Wagoner wrote:

> I've been trying to bridge FreeSWITCH with a Toshiba CIX using qsig
> over a PRI. I have a Sangoma A102 card installed in a Dell PowerEdge
> with CentOS 5.3. The issue I am having is no packets are being
> transmitted back to FreeSWITCH. ifconfig w1g1 shows every frame
> received as an overrun. I've tried a different server with CentOS 5.3
> with the same issue. I have a support ticket in with Sangoma, but was
> wondering if anybody had seen this before.
>
> The T1 shows connected so I think I have the Toshiba configured
> properly. From what I've read the overrun has to deal with the driver
> not reading the data in time so maybe this is a CentOS 5.3 specific
> issue. Any recommendations on alternative Linux distros known to work
> with the Sangoma A102 card?
>
> Thanks,
> Ryan
>
> [root at voip ~]# ifconfig w1g1
> w1g1      Link encap:Point-to-Point Protocol
>          UP POINTOPOINT RUNNING NOARP  MTU:80  Metric:1
>          RX packets:22298 errors:0 dropped:0 overruns:274 frame:274
>          TX packets:22298 errors:0 dropped:0 overruns:0 carrier:0
>          collisions:0 txqueuelen:100
>          RX bytes:1783840 (1.7 MiB)  TX bytes:1783840 (1.7 MiB)
>          Interrupt:169 Memory:f8e80000-f8e81fff
>
>
> [root at voip ~]# wanrouter status
>
> Devices currently active:
>        wanpipe1
>
>
> Wanpipe Config:
>
> Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK |  
> Baud rate |
> wanpipe1    | N/A          | A101/1D/A102/2D/4/4D/8| 169 | 4       | 1
>   | N/A | 0         |
>
> Wanrouter Status:
>
> Device name | Protocol | Station | Status        |
> wanpipe1    | AFT TE1  | N/A     | Connected     |
>
>
> [root at voip ~]# wanpipemon -i w1g1 -c Ta
>
> ***** w1g1: T1 Alarms (Framer) *****
>
> ALOS:   OFF     | LOS:  OFF
> RED:    OFF     | AIS:  OFF
> RAI:    OFF     | OOF:  OFF
>
> ***** w1g1: T1 Alarms (LIU) *****
>
> Short Circuit:  OFF
> Open Circuit:   OFF
> Loss of Signal: OFF
>
>
> ***** w1g1: T1 Performance Monitoring Counters *****
>
> Line Code Violation     : 45
> Bit Errors (CRC6/Ft/Fs) : 0
> Out of Frame Errors     : 0
>
>
> Rx Level        : > -2.5db



From mike at jerris.com  Fri Aug 14 11:08:05 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 14:08:05 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <c4d05cbe0908131804g18cd923ewdc629e0e99ccb50b@mail.gmail.com>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
	<87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
	<c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>
	<A681A304-5B94-49B8-BA25-E3089ED3F4A9@freeswitch.org>
	<c4d05cbe0908130802w6528ca34ma9d6e34ab9ce28c2@mail.gmail.com>
	<EEE57DE2-9176-49FB-A986-A53275E148E3@avgs.ca>
	<c4d05cbe0908131804g18cd923ewdc629e0e99ccb50b@mail.gmail.com>
Message-ID: <3F9A015D-8583-4EB3-9211-C238628B6CE9@jerris.com>

Issue is we don't handle progress and progress media differently,  
maybe we should.  The message is however harmless, but annoying and  
should probably be dealt with a bit better.  Patches welcome.

Mike

On Aug 13, 2009, at 9:04 PM, Moises Silva wrote:

> Yes, agreed, but there is no point in sending a WARNING since is a  
> normal condition, therefore should not even try to change the state  
> of the channel.
>
> On Thu, Aug 13, 2009 at 7:57 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> It probably just VETO it so it avoid sending  
> SWITCH_MESSAGE_INDICATE_PROGRESS again since the call is already  
> making progress from the core's point of view?
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
> On 13-Aug-09, at 11:02 AM, Moises Silva wrote:
>
>> On Wed, Aug 12, 2009 at 7:19 PM, Brian West <brian at freeswitch.org>  
>> wrote:
>> Well you really can't ignore it... it happens with our ISDN stack
>> too.   Thats what the VETO handles.
>>
>> /b
>>
>> You lost me. What do you mean we can't ignore it? the way I see it,  
>> sure we can and we should.
>>
>> Currently that warning comes from the on_ringing() callback which  
>> blindly attempts to move the state of the zap channel to  
>> ZAP_CHANNEL_STATE_PROGRESS, even when the state may be already  
>> ZAP_CHANNEL_STATE_PROGRESS_MEDIA (which means on_proceed() was  
>> called first).
>>
>> As I see it, the VETO warning is more an aid to the programmer so  
>> you quickly realize your doing a useless state change, which should  
>> be fixed. In this case, the fix is simply checking the state of the  
>> channel before trying to move it to progress, and don't even try to  
>> move it if already in progress with media.
>>
>>  --
>> Moises Silva
>> Software Developer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham  
>> ON L3R 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>

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From mike at jerris.com  Fri Aug 14 11:11:25 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 14:11:25 -0400
Subject: [Freeswitch-users] OpenSolaris Compile Error [gcc]
In-Reply-To: <4A847695.2C9A.00FE.0@lkfd.net>
References: <4A847695.2C9A.00FE.0@lkfd.net>
Message-ID: <4E31892D-FC1B-4F8F-8D41-1D245D92699C@jerris.com>

It would appear that different versions of opensolaris / compiler /  
32/64 bit handle this totally differently.  I have tested this both 32  
and 64 on sun studio when I made the change and it works, seems gcc  
wants different format specifiers.  The patch that caused this is :

http://fisheye.freeswitch.org/changelog/FreeSWITCH?cs=14069

Anyone with access to 32 and 64 bit opensolaris with both sun cc and  
gcc build systems working I need a patch that works on all.

Mike



On Aug 13, 2009, at 9:24 PM, Nick Lemberger wrote:

> 64bit OpenSolaris w/ gcc-4.3.2
>
> After a bootstrap and configure I get the following error when  
> running make:
>
>
> ---snip---
>
> Compiling src/switch_caller.c ...
> cc1: warnings being treated as errors
> src/switch_caller.c: In function  
> 'switch_caller_profile_event_set_data':
> src/switch_caller.c:299: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> src/switch_caller.c:301: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> src/switch_caller.c:303: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> src/switch_caller.c:305: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> src/switch_caller.c:307: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> src/switch_caller.c:309: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> src/switch_caller.c:311: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> make[2]: *** [libfreeswitch_la-switch_caller.lo] Error 1
>
> ---snip---
>
> I get this error in both the source for 1.0.4 and last nights  
> snapshot.  An suggestions or ideas?  There are no apparent errors  
> during the bootstrap or configure processes.
>
> Regards,
> Nicholas Lemberger
> Lakefield Communications
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Fri Aug 14 11:13:47 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 14:13:47 -0400
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
	<20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
Message-ID: <EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>

thats in seconds.

Mike

On Aug 14, 2009, at 8:32 AM, mark morreny wrote:

> Hi,
>
> Thank you for your help.
>
> I get that too, but the callback does not execute the second time.
>
> When I do task->runtime = switch_time_now() + 10;, what does +10  
> mean?  Does it mean 10 s or 10 mins?
>
> Thanks,
> Mark
>
> On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> Hi,
>
> I did the same thing on my side....
>
> API CALL [load(mod_skel)] output:
> +OK
>
> 2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task  
> 2 data_flush (core) to run at 1250089698
> 2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889  
> Successfully Loaded [mod_skel]
> 2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270  
> Adding API Function 'skel'
> freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]  
> mod_skel.c:120 starting to flush data  buffer...
>
> Note that you don't need to start the thread manually, the core  
> already has threads running for the scheduler.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 12-Aug-09, at 7:26 AM, mark morreny wrote:
>
>> Hi,
>>
>> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out  
>> some data every 10 s.  The following lines of code does not show  
>> any effect at all.
>>
>>     switch_scheduler_task_thread_start();
>>     switch_scheduler_add_task(switch_epoch_time_now(NULL),  
>> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE| 
>> SSHF_NO_DEL);
>>
>>
>> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>>
>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting  
>> to flush data  buffer...\n");
>>
>>
>>     task->runtime = switch_time_now() + 10;
>>
>> }
>>
>> Does anyone know how to get it to work?
>>
>> Thanks,
>> Mark
>>
>>
>> ---------- Forwarded message ----------
>> From: Brian West <brian at freeswitch.org>
>> Date: Mon, Aug 10, 2009 at 8:53 PM
>> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
>> To: freeswitch-users at lists.freeswitch.org
>>
>>
>> switch_rtp.c has a simple one for the zrtp cache storing.
>>
>> /b
>>
>> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>>
>> > Re schedule is done in your callback, take a look at places that  
>> use
>> > these apis in the code for details.

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From mike at jerris.com  Fri Aug 14 11:20:31 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 14:20:31 -0400
Subject: [Freeswitch-users] sip contact header question
In-Reply-To: <27c25bc40908140902k78d5bbe1naa75a21df21caa6e@mail.gmail.com>
References: <27c25bc40908140902k78d5bbe1naa75a21df21caa6e@mail.gmail.com>
Message-ID: <543A09D4-B3A8-4BA0-8F5A-0C3C694EBB15@jerris.com>

How exactly are you setting the var?  It should be set on the b-leg,  
such as using [sip_contact_user=xxx] on the originate line.

Mike

On Aug 14, 2009, at 12:02 PM, Juan Backson wrote:

> Hi,
>
> I would like to set outbound INVITE with customer Contact: field.  I  
> did sip_contact_user="2033334545", but it still did not work.  I am  
> using 1.0.4 release.
>
> Is this a known issue?
>
> br,
> JB


From msc at freeswitch.org  Fri Aug 14 11:27:42 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 14 Aug 2009 13:27:42 -0500
Subject: [Freeswitch-users] Question about sharing conference between
In-Reply-To: <37033.1250266845@a2unlimited.com>
References: <37033.1250266845@a2unlimited.com>
Message-ID: <87f2f3b90908141127h5e216a06ta2ddd20c818fab9f@mail.gmail.com>

On Fri, Aug 14, 2009 at 11:20 AM, Tina Martinez <tina at a2unlimited.com>wrote:

> Michael,
>
> Thanks again for bearing with my novice perspective on this.
>
> I was able to achieve the link between two FS servers as intended.
>  However, I
> was not able to setup a "new" dialplan file as you described.  I had to
> place the
> script into the default.xml dialplan to get it to work.  Is there something
> I'm
> supposed to do to get FS to look in the conf/dialplan/default ?


Look at the sample files that are already in conf/dialplan/default/ and
conf/dialplan/public/ to see what they should look like. Just remember to
name the files with leading digits, so you have something like this:
conf/dialplan/default/01_My_Custom_Extensions.xml
conf/dialplan/public/01_More_Public_Extensions.xml

The files must be .xml files and preferable should have the <include> and
</include> tags at the beginning and end of the files. The other thing you
can do is just make a copy of one of the files that's already in
conf/dialplan/default/ or conf/dialplan/public/ and edit it.


>
>
> Also, in using the code as we discussed, I was also able to establish the
> link
> between the two servers without having an actual soft-phone
> registered/connected
> to the FS server (which is ideal for my situation).  I can create the link
> between the servers, and then dial out to external phone numbers from the
> respective servers and connect the participants.  Then, regardless of who
> hangs-up, the conference between the servers remains in tact.  So far it
> works
> very well.
>

Glad to hear it!
-MC


>
> - T
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From bjbrashier at gmail.com  Fri Aug 14 11:41:41 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Fri, 14 Aug 2009 11:41:41 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <DB70D4BE-CD40-4707-AD0C-333A6A0BA75E@jerris.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com> 
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com> 
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
	<7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com> 
	<7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com> 
	<7bcfdd290908131524s10e209dcla7bebada79ecfa51@mail.gmail.com> 
	<DB70D4BE-CD40-4707-AD0C-333A6A0BA75E@jerris.com>
Message-ID: <7bcfdd290908141141w1372a10cw158f332e8bf13763@mail.gmail.com>

I didn't see any SIP session timers in the wiki. Since I'm already using the
event socket for control, my current plan is to use sched_api to play a file
with a short (20ms?) clip of silence, capture the play_file event and use it
to reschule another one for a couple of seconds later.

I'll let you know what happens.

BB

On Thu, Aug 13, 2009 at 10:47 PM, Michael Jerris <mike at jerris.com> wrote:

> My suggestion is to use sip session timers not rtp timeouts as rtp is
> supposed to be discontinuous.  That being said, we have several settings to
> continuously send media, but then you are doing exactly what you said you
> didn't want to do.
> Mike
>
>  On Aug 13, 2009, at 6:24 PM, Bradley Brashier wrote:
>
>  OK, I finally got a moment to do a packet capture and take a look at the
> streams.  It became very clear very quickly that what happens is that during
> silence the gateway still sends RTP packets to Freeswitch, but Freeswitch
> doesn't send any back to the gateway. After 10s of this, the gateway says
> "Oh, the RPT must be broken" and it hangs up.
>
> We found a way to turn off this behavior in the gateway, and the good news
> is that it did indeed fix the problem. But we'd rather not rely on that as a
> long-term solution because then we can't detect and drop RTP streams that
> really are broken.
>
> So now I'm back to looking at Freeswitch to figure out how to send just a
> single packet every second or so during silence. If anyone knows of a way to
> do this, let me know, otherwise I'll get back to you if and when I find one.
>
> BB
>
> On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
>
>> I took a closer look at the SIP messages on the console. From it, I
>> understand that it's not Freeswitch timing out, but rather FS is getting the
>> "BYE" msg from somewhere else. I've tested phones and tested calling without
>> going through the FS conference, though, and everything works fine. Then I
>> saw something else odd inside the BYE msg:
>>
>>    Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"
>> So I Googled "RTP Broken Connection" and saw several sites talking about
>> AudioCodes gateways and Asterisk -- and our gateway is an AudioCodes. From
>> these sites it sounds like AudioCodes is rather aggressive in detecting RTP
>> breaks, and is interpreting the silence from FS as a break.
>>
>> So now I'm looking into ways to maybe send "I'm still here" RTP packets
>> from FS or to tune the gateway to be less aggressive. I can't stop and get a
>> clean packet capture right now because I've got a bunch of testers working
>> on it today. I'll do that sometime when the system is less busy.
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
>>
>>> I had just thought of the exact same thing. I'm trying to test that now.
>>> Thanks for your input.
>>>
>>> BB
>>>
>>>   On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com>wrote:
>>>
>>>>   My guess is that its the other end killing the call due to rtp
>>>> timeouts, not us killing the call.  If you can confirm this the best method
>>>> would be to get them not to do rtp timeouts.
>>>>  On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>>>>
>>>>  I'm sure that would work, but I'm worried about it sucking up
>>>> bandwidth, especially since you'd need it on every caller (since otherwise
>>>> the one person who had it could hang up and you'd be back to square 1).
>>>>
>>>> Any other ideas, or should I hunt through the code to try to override
>>>> the behavior manually?
>>>>
>>>> BB
>>>>
>>>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Check out the 'waste' member flag. I think if at least one member has
>>>>> that set then RTP will get sent out even during silence. Let us know if that
>>>>> helps...
>>>>>
>>>>> -MC
>>>>>
>>>>>   On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <
>>>>> bjbrashier at gmail.com> wrote:
>>>>>
>>>>>>   Hi all.
>>>>>>
>>>>>> The solution to this one should be short.
>>>>>>
>>>>>> My conference hangs up when there's 2+ users and silence for 5 sec or
>>>>>> so. I'm trying to find a parameter that changes that (I'd rather it be,
>>>>>> say, 60 seconds).
>>>>>>
>>>>>> I didn't see a parameter like this specific to conferences, so I
>>>>>> looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set
>>>>>> to 300 (the default), so I'm pretty sure that's not the problem. I also
>>>>>> searched through the mod_conference.c code and didn't see it, though I was
>>>>>> only skimming.
>>>>>>
>>>>>> I'm not 100% convinced that this is limited to conferences, but I
>>>>>> don't currently have a way to test in a non-conference environment.
>>>>>>
>>>>>> Anybody know how to increase the conference silence-hangup timeout?
>>>>>>
>>>>>> BB
>>>>>>
>>>>>> _____
>>>>>
>>>>>
>>>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From csa at nowthor.com  Fri Aug 14 11:52:25 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Fri, 14 Aug 2009 14:52:25 -0400
Subject: [Freeswitch-users] bind_server_ip issue
In-Reply-To: <CBF2E5B2-4707-4822-9E79-B81D1D86FA83@freeswitch.org>
References: <4A841FE0.90904@nowthor.com>	<12E3506D-3F79-4987-9FA7-B40AE461B97A@freeswitch.org>	<4A855525.2080301@nowthor.com>
	<CBF2E5B2-4707-4822-9E79-B81D1D86FA83@freeswitch.org>
Message-ID: <4A85B269.3070009@nowthor.com>

Thanks, Brian. If you don't mind my asking, what is the ultimate purpose 
of local_ip_v4, then?

Brian West wrote:
> Don't use local_ip_v4 then... please hard code the param in the sofia  
> profile because the value of local_ip_v4 can change.
>
> /b
>
> On Aug 14, 2009, at 7:14 AM, Carlos S. Antunes wrote:
>
>   
>> Hi Brian!
>>
>> Thank you for your quick response. I ended up defining "local_ip_v4"  
>> at the top of vars.xml and set "bind_server_ip" to "local_ip_v4" as  
>> well. Is this the best way to go about selecting an IP address for  
>> FS to bind to?
>>
>> In any case, even though it doesn't appear to affect operation, I am  
>> still getting this "IP change detected" somewhat periodically in my  
>> logs (running 1.0.4):
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
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From lon at kickasspixels.com  Fri Aug 14 12:00:37 2009
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 14 Aug 2009 12:00:37 -0700
Subject: [Freeswitch-users] Offline or background ASR
Message-ID: <5d3e0dc60908141200s70b7af5r76cbc75ddf3a615@mail.gmail.com>

Hi there,
Has anyone found a good solution for processing recordings/voicemail with
automatic speech recognition software?

I am researching numerous solutions, both open source and commercial, but
was wondering if I missed anything obvious.

Lon Baker
Kickass Pixels
-
(office) +1-415-287-0973
(mobile) +1-415-279-5019
(skype) lonbaker
-
http://kickasspixels.com
http://twitter.com/kickasspixels
http://www.linkedin.com/in/lonbaker
--
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From mike at jerris.com  Fri Aug 14 12:17:35 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 15:17:35 -0400
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908141141w1372a10cw158f332e8bf13763@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
	<7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com>
	<7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com>
	<7bcfdd290908131524s10e209dcla7bebada79ecfa51@mail.gmail.com>
	<DB70D4BE-CD40-4707-AD0C-333A6A0BA75E@jerris.com>
	<7bcfdd290908141141w1372a10cw158f332e8bf13763@mail.gmail.com>
Message-ID: <4649090E-BBA8-4677-B8C5-6AF3D6F4B891@jerris.com>

That sounds horrible.  There are settings both in sip/rtp and in  
conference to do this already.

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec
http://wiki.freeswitch.org/wiki/VAD_and_CNG

Mike

On Aug 14, 2009, at 2:41 PM, Bradley Brashier wrote:

> I didn't see any SIP session timers in the wiki. Since I'm already  
> using the event socket for control, my current plan is to use  
> sched_api to play a file with a short (20ms?) clip of silence,  
> capture the play_file event and use it to reschule another one for a  
> couple of seconds later.
>
> I'll let you know what happens.
>
> BB
>
> On Thu, Aug 13, 2009 at 10:47 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> My suggestion is to use sip session timers not rtp timeouts as rtp  
> is supposed to be discontinuous.  That being said, we have several  
> settings to continuously send media, but then you are doing exactly  
> what you said you didn't want to do.
>
> Mike
>
> On Aug 13, 2009, at 6:24 PM, Bradley Brashier wrote:
>
>> OK, I finally got a moment to do a packet capture and take a look  
>> at the streams.  It became very clear very quickly that what  
>> happens is that during silence the gateway still sends RTP packets  
>> to Freeswitch, but Freeswitch doesn't send any back to the gateway.  
>> After 10s of this, the gateway says "Oh, the RPT must be broken"  
>> and it hangs up.
>>
>> We found a way to turn off this behavior in the gateway, and the  
>> good news is that it did indeed fix the problem. But we'd rather  
>> not rely on that as a long-term solution because then we can't  
>> detect and drop RTP streams that really are broken.
>>
>> So now I'm back to looking at Freeswitch to figure out how to send  
>> just a single packet every second or so during silence. If anyone  
>> knows of a way to do this, let me know, otherwise I'll get back to  
>> you if and when I find one.
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier <bjbrashier at gmail.com 
>> > wrote:
>> I took a closer look at the SIP messages on the console. From it, I  
>> understand that it's not Freeswitch timing out, but rather FS is  
>> getting the "BYE" msg from somewhere else. I've tested phones and  
>> tested calling without going through the FS conference, though, and  
>> everything works fine. Then I saw something else odd inside the BYE  
>> msg:
>>
>>    Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"
>> So I Googled "RTP Broken Connection" and saw several sites talking  
>> about AudioCodes gateways and Asterisk -- and our gateway is an  
>> AudioCodes. From these sites it sounds like AudioCodes is rather  
>> aggressive in detecting RTP breaks, and is interpreting the silence  
>> from FS as a break.
>>
>> So now I'm looking into ways to maybe send "I'm still here" RTP  
>> packets from FS or to tune the gateway to be less aggressive. I  
>> can't stop and get a clean packet capture right now because I've  
>> got a bunch of testers working on it today. I'll do that sometime  
>> when the system is less busy.
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com 
>> > wrote:
>> I had just thought of the exact same thing. I'm trying to test that  
>> now.  Thanks for your input.
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com>  
>> wrote:
>> My guess is that its the other end killing the call due to rtp  
>> timeouts, not us killing the call.  If you can confirm this the  
>> best method would be to get them not to do rtp timeouts.
>>
>> On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>>
>>> I'm sure that would work, but I'm worried about it sucking up  
>>> bandwidth, especially since you'd need it on every caller (since  
>>> otherwise the one person who had it could hang up and you'd be  
>>> back to square 1).
>>>
>>> Any other ideas, or should I hunt through the code to try to  
>>> override the behavior manually?
>>>
>>> BB
>>>
>>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins  
>>> <msc at freeswitch.org> wrote:
>>> Check out the 'waste' member flag. I think if at least one member  
>>> has that set then RTP will get sent out even during silence. Let  
>>> us know if that helps...
>>>
>>> -MC
>>>
>>> On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <bjbrashier at gmail.com 
>>> > wrote:
>>> Hi all.
>>>
>>> The solution to this one should be short.
>>>
>>> My conference hangs up when there's 2+ users and silence for 5 sec  
>>> or so. I'm trying to find a parameter that changes that (I'd  
>>> rather it be, say, 60 seconds).
>>>
>>> I didn't see a parameter like this specific to conferences, so I  
>>> looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but  
>>> it's set to 300 (the default), so I'm pretty sure that's not the  
>>> problem. I also searched through the mod_conference.c code and  
>>> didn't see it, though I was only skimming.
>>>
>>> I'm not 100% convinced that this is limited to conferences, but I  
>>> don't currently have a way to test in a non-conference environment.
>>>
>>> Anybody know how to increase the conference silence-hangup timeout?
>>>
>>> BB
>>>
>>> _____
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From pjintheusa at gmail.com  Fri Aug 14 12:17:39 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 14 Aug 2009 15:17:39 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <b39273530908132333y52279d64qe926eca47e15b5eb@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
	<b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
	<367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>
	<FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>
	<b39273530908132333y52279d64qe926eca47e15b5eb@mail.gmail.com>
Message-ID: <367751820908141217g2facd41av25f65d6321967d74@mail.gmail.com>

Hi Rupa,

What about my suggestion above introduce a "api_after_bridge" event
that fires when the switch_ivr_uuid_bridge() bridges to the two sofia
channels that Mathieu mentioned?

Is that suggestion just way off the mark? If possible that would allow
me to move forward - although I agree that supporting groupings of
carriers is would be the most elegant solution.

Let me know if I am just talking rubbish re the api_after_bridge" event.

Thanks!


Phillip Jones


On Fri, Aug 14, 2009 at 2:33 AM, Rupa Schomaker<rupa at rupa.com> wrote:
> On Thu, Aug 13, 2009 at 6:54 PM, Mathieu Rene<mrene_lists at avgs.ca> wrote:
>> Hi All,
>>
>> The reason it works when you wait 3 seconds is that mod_loopback bails
>
> [snip]
>
> Thanks for that explanation. ?It umm.. explains a lot. :)
>
>> On another note, mod_sofia will behave differently when it detects its
>> being bridge with another sofia channel, providing optimizations when
>> both call legs are SIP.
>>
>> My personal opinion is not to use mod_loopback unless absolutely
>> necessary, FreeSWITCH's core is very flexible and there's often a
>> (better) way than using mod_loopback.
>
> So, I think the temp solution is to use loopback+delayed no media.
>
> but the real "solution" is to either drive the forked dialing logic
> externally (event socket) or consider supporting groupings in the
> bridge which.. umm... ?is gonna be a pain and will need buy in from
> from Tony and other core devs since that is a core (no pun intended)
> piece of code that nearly everything uses.
>
> I'm not sure I want to take a wack at it.
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From bjbrashier at gmail.com  Fri Aug 14 12:44:54 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Fri, 14 Aug 2009 12:44:54 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <4649090E-BBA8-4677-B8C5-6AF3D6F4B891@jerris.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com> 
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com> 
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
	<7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com> 
	<7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com> 
	<7bcfdd290908131524s10e209dcla7bebada79ecfa51@mail.gmail.com> 
	<DB70D4BE-CD40-4707-AD0C-333A6A0BA75E@jerris.com>
	<7bcfdd290908141141w1372a10cw158f332e8bf13763@mail.gmail.com> 
	<4649090E-BBA8-4677-B8C5-6AF3D6F4B891@jerris.com>
Message-ID: <7bcfdd290908141244k7e469966y2c69f665f15df1df@mail.gmail.com>

All right, I'm confused.

The RTP timeout parameters have no documentation, and from the names, I'd
have guessed that they hang up after a specified amount of time, not send
some other signal.

The session-timeout timer talks about calls expiring, and sending another
SIP invite, which I don't think is what I need. It also says it shouldn't be
set to less than 30 minutes.

I've been poking through RFC 4028, which the enable-timer parameter
mentions, but I'm not sure those will be helpful, either, because I've got
the SIP and RTP traffic going on different paths, and it's the RTP that's
having problems. Still, maybe I can use one of the timers to do something
different, but then I'm not sure how that's any better than using the event
socket the way I planned.

Clearly I'm missing something.

BB

On Fri, Aug 14, 2009 at 12:17 PM, Michael Jerris <mike at jerris.com> wrote:

> That sounds horrible.  There are settings both in sip/rtp and in conference
> to do this already.
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec
>
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec
> http://wiki.freeswitch.org/wiki/VAD_and_CNG
>
> Mike
>
>  On Aug 14, 2009, at 2:41 PM, Bradley Brashier wrote:
>
>  I didn't see any SIP session timers in the wiki. Since I'm already using
> the event socket for control, my current plan is to use sched_api to play a
> file with a short (20ms?) clip of silence, capture the play_file event and
> use it to reschule another one for a couple of seconds later.
>
> I'll let you know what happens.
>
> BB
>
> On Thu, Aug 13, 2009 at 10:47 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> My suggestion is to use sip session timers not rtp timeouts as rtp is
>> supposed to be discontinuous.  That being said, we have several settings to
>> continuously send media, but then you are doing exactly what you said you
>> didn't want to do.
>> Mike
>>
>>  On Aug 13, 2009, at 6:24 PM, Bradley Brashier wrote:
>>
>>  OK, I finally got a moment to do a packet capture and take a look at the
>> streams.  It became very clear very quickly that what happens is that during
>> silence the gateway still sends RTP packets to Freeswitch, but Freeswitch
>> doesn't send any back to the gateway. After 10s of this, the gateway says
>> "Oh, the RPT must be broken" and it hangs up.
>>
>> We found a way to turn off this behavior in the gateway, and the good news
>> is that it did indeed fix the problem. But we'd rather not rely on that as a
>> long-term solution because then we can't detect and drop RTP streams that
>> really are broken.
>>
>> So now I'm back to looking at Freeswitch to figure out how to send just a
>> single packet every second or so during silence. If anyone knows of a way to
>> do this, let me know, otherwise I'll get back to you if and when I find one.
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
>>
>>> I took a closer look at the SIP messages on the console. From it, I
>>> understand that it's not Freeswitch timing out, but rather FS is getting the
>>> "BYE" msg from somewhere else. I've tested phones and tested calling without
>>> going through the FS conference, though, and everything works fine. Then I
>>> saw something else odd inside the BYE msg:
>>>
>>>    Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"
>>> So I Googled "RTP Broken Connection" and saw several sites talking about
>>> AudioCodes gateways and Asterisk -- and our gateway is an AudioCodes. From
>>> these sites it sounds like AudioCodes is rather aggressive in detecting RTP
>>> breaks, and is interpreting the silence from FS as a break.
>>>
>>> So now I'm looking into ways to maybe send "I'm still here" RTP packets
>>> from FS or to tune the gateway to be less aggressive. I can't stop and get a
>>> clean packet capture right now because I've got a bunch of testers working
>>> on it today. I'll do that sometime when the system is less busy.
>>>
>>> BB
>>>
>>> On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
>>>
>>>> I had just thought of the exact same thing. I'm trying to test that
>>>> now.  Thanks for your input.
>>>>
>>>> BB
>>>>
>>>>   On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com>wrote:
>>>>
>>>>>   My guess is that its the other end killing the call due to rtp
>>>>> timeouts, not us killing the call.  If you can confirm this the best method
>>>>> would be to get them not to do rtp timeouts.
>>>>>  On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>>>>>
>>>>>  I'm sure that would work, but I'm worried about it sucking up
>>>>> bandwidth, especially since you'd need it on every caller (since otherwise
>>>>> the one person who had it could hang up and you'd be back to square 1).
>>>>>
>>>>> Any other ideas, or should I hunt through the code to try to override
>>>>> the behavior manually?
>>>>>
>>>>> BB
>>>>>
>>>>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> Check out the 'waste' member flag. I think if at least one member has
>>>>>> that set then RTP will get sent out even during silence. Let us know if that
>>>>>> helps...
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>   On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <
>>>>>> bjbrashier at gmail.com> wrote:
>>>>>>
>>>>>>>   Hi all.
>>>>>>>
>>>>>>> The solution to this one should be short.
>>>>>>>
>>>>>>> My conference hangs up when there's 2+ users and silence for 5 sec or
>>>>>>> so. I'm trying to find a parameter that changes that (I'd rather it be,
>>>>>>> say, 60 seconds).
>>>>>>>
>>>>>>> I didn't see a parameter like this specific to conferences, so I
>>>>>>> looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set
>>>>>>> to 300 (the default), so I'm pretty sure that's not the problem. I also
>>>>>>> searched through the mod_conference.c code and didn't see it, though I was
>>>>>>> only skimming.
>>>>>>>
>>>>>>> I'm not 100% convinced that this is limited to conferences, but I
>>>>>>> don't currently have a way to test in a non-conference environment.
>>>>>>>
>>>>>>> Anybody know how to increase the conference silence-hangup timeout?
>>>>>>>
>>>>>>> BB
>>>>>>>
>>>>>>> _____
>>>>>>
>>>>>>
>>>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Fri Aug 14 14:02:06 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 14 Aug 2009 23:02:06 +0200
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness patch
In-Reply-To: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
Message-ID: <7b197bef0908141402k15d4144bt87a287e07c51b5c6@mail.gmail.com>

svn 14521: skypiax: compiles on windoz, not yet tested (on windoz)



On Fri, Aug 14, 2009 at 7:43 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Hi FreeSWITCHers,
>
> all the users of mod_skypiax are kindly requested to test the svn trunk 14519.
>
> It contains a lot of changes meant to add stability and robustness,
> toward a production environment.
>
> Let me know how your feelings, and please add to the Jira any possible
> bug/issue/etc.
>
> Thanks to you all,
>
> -giovanni
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
>
> Cell : +39-347-2665618
>


From marc at kasteris.com  Fri Aug 14 15:27:30 2009
From: marc at kasteris.com (Marc Orenberg)
Date: Fri, 14 Aug 2009 15:27:30 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH crashes when bridging two calls in
	python script
Message-ID: <492335.3738.qm@web50806.mail.re2.yahoo.com>

Hello, 

I'm trying to play a prompt to the B-leg of a bridged call in Python. 
I place the call to the B-leg, play the prompt, and then bridge it with the A-Leg, but then  FreeSWITCH crashes when the call is completed.

Here's the code I'm using:

def bridge_call_with_prompt(session,carrier,caller_id,phonenum,promptfile,recordfile):
  try:
    sessiondata = "{ignore_early_media=true,origination_caller_id_number=" + str(caller_id) + "}sofia/gateway/" + carrier + "/" + phonenum
    new_session =  Session(sessiondata)
    if (not new_session.ready()): return(False)
    if (recordfile <> ""): new_session.streamFile(promptfile)
    new_session.execute("set","call_timeout=60")
    new_session.execute("set","continue_on_fail=true")
    new_session.execute("set","hangup_after_bridge=false")
    new_session.execute("set", "ringback=%(2000,4000,440.0,480.0)")
    if (not new_session.ready()): return(False)
    if (not session.ready()): return(False)
    bridge(session,new_session)
    new_session.hangup()
  except:
    report_exception()
    return(False)
  return(True)

Does anybody know how I can stop it from crashing, and/or another way to go about this?   If I simply do:

session.execute("bridge","sofia/gateway/" + carrier + "/" + phonenum)


it won't cause FreeSWITCH to crash, but then I can't play my prompt.

Thanks,
Marc
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From markmorreny at gmail.com  Fri Aug 14 19:19:09 2009
From: markmorreny at gmail.com (mark morreny)
Date: Sat, 15 Aug 2009 10:19:09 +0800
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
	<20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
	<EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>
Message-ID: <20ad6b920908141919u1b2aa1c4jacc75cd4896d72b0@mail.gmail.com>

Hi Michael,

The following code was executed once, but not after the next 10 s.

SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to flush
data  buffer...\n");

    task->runtime = switch_time_now() + 10;
}

Any suggestion why?


Thanks,
Mark
On Sat, Aug 15, 2009 at 2:13 AM, Michael Jerris <mike at jerris.com> wrote:

> thats in seconds.
> Mike
>
>  On Aug 14, 2009, at 8:32 AM, mark morreny wrote:
>
>  Hi,
>
> Thank you for your help.
>
> I get that too, but the callback does not execute the second time.
>
> When I do task->runtime = switch_time_now() + 10;, what does +10 mean?
> Does it mean 10 s or 10 mins?
>
> Thanks,
> Mark
>
> On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene <mrene_lists at avgs.ca>wrote:
>
>> Hi,
>> I did the same thing on my side....
>>  API CALL [load(mod_skel)] output:
>> +OK
>>
>> 2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task 2
>> data_flush (core) to run at 1250089698
>> 2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889
>> Successfully Loaded [mod_skel]
>> 2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270 Adding API
>> Function 'skel'
>> freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]
>> mod_skel.c:120 starting to flush data  buffer...
>>
>> Note that you don't need to start the thread manually, the core already
>> has threads running for the scheduler.
>>
>>   Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>>  On 12-Aug-09, at 7:26 AM, mark morreny wrote:
>>
>> Hi,
>>
>> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out some data
>> every 10 s.  The following lines of code does not show any effect at all.
>>
>>     switch_scheduler_task_thread_start();
>>     switch_scheduler_add_task(switch_epoch_time_now(NULL),
>> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|SSHF_NO_DEL);
>>
>>
>> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>>
>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to flush
>> data  buffer...\n");
>>
>>
>>     task->runtime = switch_time_now() + 10;
>>
>> }
>>
>> Does anyone know how to get it to work?
>>
>> Thanks,
>> Mark
>>
>>
>> ---------- Forwarded message ----------
>> From: Brian West <brian at freeswitch.org>
>> Date: Mon, Aug 10, 2009 at 8:53 PM
>> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
>> To: freeswitch-users at lists.freeswitch.org
>>
>>
>> switch_rtp.c has a simple one for the zrtp cache storing.
>>
>> /b
>>
>> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>>
>> > Re schedule is done in your callback, take a look at places that use
>> > these apis in the code for details.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Fri Aug 14 20:16:47 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 23:16:47 -0400
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <20ad6b920908141919u1b2aa1c4jacc75cd4896d72b0@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
	<20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
	<EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>
	<20ad6b920908141919u1b2aa1c4jacc75cd4896d72b0@mail.gmail.com>
Message-ID: <EF4B4642-E586-42E0-8D40-D9FCB7A93DD2@jerris.com>

  task->runtime = switch_epoch_time_now(NULL) + 10;

On Aug 14, 2009, at 10:19 PM, mark morreny wrote:

> Hi Michael,
>
> The following code was executed once, but not after the next 10 s.
>
> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to  
> flush data  buffer...\n");
>
>     task->runtime = switch_time_now() + 10;
> }
>
> Any suggestion why?
>
>
> Thanks,
> Mark
> On Sat, Aug 15, 2009 at 2:13 AM, Michael Jerris <mike at jerris.com>  
> wrote:
> thats in seconds.
>
> Mike
>
> On Aug 14, 2009, at 8:32 AM, mark morreny wrote:
>
>> Hi,
>>
>> Thank you for your help.
>>
>> I get that too, but the callback does not execute the second time.
>>
>> When I do task->runtime = switch_time_now() + 10;, what does +10  
>> mean?  Does it mean 10 s or 10 mins?
>>
>> Thanks,
>> Mark
>>
>> On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene  
>> <mrene_lists at avgs.ca> wrote:
>> Hi,
>>
>> I did the same thing on my side....
>>
>> API CALL [load(mod_skel)] output:
>> +OK
>>
>> 2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task  
>> 2 data_flush (core) to run at 1250089698
>> 2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889  
>> Successfully Loaded [mod_skel]
>> 2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270  
>> Adding API Function 'skel'
>> freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]  
>> mod_skel.c:120 starting to flush data  buffer...
>>
>> Note that you don't need to start the thread manually, the core  
>> already has threads running for the scheduler.
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 12-Aug-09, at 7:26 AM, mark morreny wrote:
>>
>>> Hi,
>>>
>>> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out  
>>> some data every 10 s.  The following lines of code does not show  
>>> any effect at all.
>>>
>>>     switch_scheduler_task_thread_start();
>>>     switch_scheduler_add_task(switch_epoch_time_now(NULL),  
>>> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE| 
>>> SSHF_NO_DEL);
>>>
>>>
>>> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>>>
>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting  
>>> to flush data  buffer...\n");
>>>
>>>
>>>     task->runtime = switch_time_now() + 10;
>>>
>>> }
>>>
>>> Does anyone know how to get it to work?
>>>
>>> Thanks,
>>> Mark
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Brian West <brian at freeswitch.org>
>>> Date: Mon, Aug 10, 2009 at 8:53 PM
>>> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
>>> To: freeswitch-users at lists.freeswitch.org
>>>
>>>
>>> switch_rtp.c has a simple one for the zrtp cache storing.
>>>
>>> /b
>>>
>>> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>>>
>>> > Re schedule is done in your callback, take a look at places that  
>>> use
>>> > these apis in the code for details.
>
>

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From mike at jerris.com  Fri Aug 14 20:17:44 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 23:17:44 -0400
Subject: [Freeswitch-users] FreeSWITCH crashes when bridging two calls
	in python script
In-Reply-To: <492335.3738.qm@web50806.mail.re2.yahoo.com>
References: <492335.3738.qm@web50806.mail.re2.yahoo.com>
Message-ID: <D1D6EEEA-C269-4E6F-A055-373D37CB1B10@jerris.com>

Please open a bug on jira.freeswitch.org

mike

On Aug 14, 2009, at 6:27 PM, Marc Orenberg wrote:

> Hello,
>
> I'm trying to play a prompt to the B-leg of a bridged call in Python.
> I place the call to the B-leg, play the prompt, and then bridge it  
> with the A-Leg, but then  FreeSWITCH crashes when the call is  
> completed.
>
> Here's the code I'm using:
>
> def  
> bridge_call_with_prompt 
> (session,carrier,caller_id,phonenum,promptfile,recordfile):
>   try:
>     sessiondata =  
> "{ignore_early_media=true,origination_caller_id_number=" +  
> str(caller_id) + "}sofia/gateway/" + carrier + "/" + phonenum
>     new_session =  Session(sessiondata)
>     if (not new_session.ready()): return(False)
>     if (recordfile <> ""): new_session.streamFile(promptfile)
>     new_session.execute("set","call_timeout=60")
>     new_session.execute("set","continue_on_fail=true")
>     new_session.execute("set","hangup_after_bridge=false")
>     new_session.execute("set", "ringback=%(2000,4000,440.0,480.0)")
>     if (not new_session.ready()): return(False)
>     if (not session.ready()): return(False)
>     bridge(session,new_session)
>     new_session.hangup()
>   except:
>     report_exception()
>     return(False)
>   return(True)
>
> Does anybody know how I can stop it from crashing, and/or another  
> way to go about this?   If I simply do:
>
>  	session.execute("bridge","sofia/gateway/" + carrier + "/" +  
> phonenum)
>
> it won't cause FreeSWITCH to crash, but then I can't play my prompt.
>
> Thanks,
> Marc

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From mrene_lists at avgs.ca  Fri Aug 14 20:20:07 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 14 Aug 2009 23:20:07 -0400
Subject: [Freeswitch-users] FreeSWITCH crashes when bridging two calls
	in python script
In-Reply-To: <492335.3738.qm@web50806.mail.re2.yahoo.com>
References: <492335.3738.qm@web50806.mail.re2.yahoo.com>
Message-ID: <2932436E-C699-4090-A658-BE8F17D255F3@avgs.ca>

Hi,

Can you provide us with a backtrace of the crash? you can open a bug  
report on http://jira.freeswitch.org/

Also, if you want to play a prompt file and wait for a dtmf to accept  
the call, there are variables called group_confirm_file and  
group_confirm_key that will make the core take care of that for you.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 14-Aug-09, at 6:27 PM, Marc Orenberg wrote:

> Hello,
>
> I'm trying to play a prompt to the B-leg of a bridged call in Python.
> I place the call to the B-leg, play the prompt, and then bridge it  
> with the A-Leg, but then  FreeSWITCH crashes when the call is  
> completed.
>
> Here's the code I'm using:
>
> def  
> bridge_call_with_prompt 
> (session,carrier,caller_id,phonenum,promptfile,recordfile):
>   try:
>     sessiondata =  
> "{ignore_early_media=true,origination_caller_id_number=" +  
> str(caller_id) + "}sofia/gateway/" + carrier + "/" + phonenum
>     new_session =  Session(sessiondata)
>     if (not new_session.ready()): return(False)
>     if (recordfile <> ""): new_session.streamFile(promptfile)
>     new_session.execute("set","call_timeout=60")
>     new_session.execute("set","continue_on_fail=true")
>     new_session.execute("set","hangup_after_bridge=false")
>     new_session.execute("set", "ringback=%(2000,4000,440.0,480.0)")
>     if (not new_session.ready()): return(False)
>     if (not session.ready()): return(False)
>     bridge(session,new_session)
>     new_session.hangup()
>   except:
>     report_exception()
>     return(False)
>   return(True)
>
> Does anybody know how I can stop it from crashing, and/or another  
> way to go about this?   If I simply do:
>
>  	session.execute("bridge","sofia/gateway/" + carrier + "/" +  
> phonenum)
>
> it won't cause FreeSWITCH to crash, but then I can't play my prompt.
>
> Thanks,
> Marc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mrene_lists at avgs.ca  Fri Aug 14 20:26:24 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 14 Aug 2009 23:26:24 -0400
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <EF4B4642-E586-42E0-8D40-D9FCB7A93DD2@jerris.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
	<20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
	<EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>
	<20ad6b920908141919u1b2aa1c4jacc75cd4896d72b0@mail.gmail.com>
	<EF4B4642-E586-42E0-8D40-D9FCB7A93DD2@jerris.com>
Message-ID: <FD54C936-D80E-4F96-8502-740ADDD1225C@avgs.ca>

Because switch_time_now() is in microseconds.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 14-Aug-09, at 11:16 PM, Michael Jerris wrote:

>  task->runtime = switch_epoch_time_now(NULL) + 10;
>
> On Aug 14, 2009, at 10:19 PM, mark morreny wrote:
>
>> Hi Michael,
>>
>> The following code was executed once, but not after the next 10 s.
>>
>> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting  
>> to flush data  buffer...\n");
>>
>>     task->runtime = switch_time_now() + 10;
>> }
>>
>> Any suggestion why?
>>
>>
>> Thanks,
>> Mark
>> On Sat, Aug 15, 2009 at 2:13 AM, Michael Jerris <mike at jerris.com>  
>> wrote:
>> thats in seconds.
>>
>> Mike
>>
>> On Aug 14, 2009, at 8:32 AM, mark morreny wrote:
>>
>>> Hi,
>>>
>>> Thank you for your help.
>>>
>>> I get that too, but the callback does not execute the second time.
>>>
>>> When I do task->runtime = switch_time_now() + 10;, what does +10  
>>> mean?  Does it mean 10 s or 10 mins?
>>>
>>> Thanks,
>>> Mark
>>>
>>> On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene  
>>> <mrene_lists at avgs.ca> wrote:
>>> Hi,
>>>
>>> I did the same thing on my side....
>>>
>>> API CALL [load(mod_skel)] output:
>>> +OK
>>>
>>> 2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added  
>>> task 2 data_flush (core) to run at 1250089698
>>> 2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889  
>>> Successfully Loaded [mod_skel]
>>> 2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270  
>>> Adding API Function 'skel'
>>> freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]  
>>> mod_skel.c:120 starting to flush data  buffer...
>>>
>>> Note that you don't need to start the thread manually, the core  
>>> already has threads running for the scheduler.
>>>
>>> Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> mrene at avgs.ca
>>>
>>>
>>>
>>>
>>> On 12-Aug-09, at 7:26 AM, mark morreny wrote:
>>>
>>>> Hi,
>>>>
>>>> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out  
>>>> some data every 10 s.  The following lines of code does not show  
>>>> any effect at all.
>>>>
>>>>     switch_scheduler_task_thread_start();
>>>>     switch_scheduler_add_task(switch_epoch_time_now(NULL),  
>>>> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE| 
>>>> SSHF_NO_DEL);
>>>>
>>>>
>>>> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>>>>
>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting  
>>>> to flush data  buffer...\n");
>>>>
>>>>
>>>>     task->runtime = switch_time_now() + 10;
>>>>
>>>> }
>>>>
>>>> Does anyone know how to get it to work?
>>>>
>>>> Thanks,
>>>> Mark
>>>>
>>>>
>>>> ---------- Forwarded message ----------
>>>> From: Brian West <brian at freeswitch.org>
>>>> Date: Mon, Aug 10, 2009 at 8:53 PM
>>>> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
>>>> To: freeswitch-users at lists.freeswitch.org
>>>>
>>>>
>>>> switch_rtp.c has a simple one for the zrtp cache storing.
>>>>
>>>> /b
>>>>
>>>> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>>>>
>>>> > Re schedule is done in your callback, take a look at places  
>>>> that use
>>>> > these apis in the code for details.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From hyppolite72 at yahoo.com  Sat Aug 15 07:17:23 2009
From: hyppolite72 at yahoo.com (Jean-Marc Hyppolite)
Date: Sat, 15 Aug 2009 07:17:23 -0700 (PDT)
Subject: [Freeswitch-users] Question about an ESL function
Message-ID: <788835.43200.qm@web35604.mail.mud.yahoo.com>

Hello,
?
I would like to know the purpose of the ESL function named recvEventTimed.
?
Thank you for your help.


      __________________________________________________________________
Be smarter than spam. See how smart SpamGuard is at giving junk email the boot with the All-new Yahoo! Mail.  Click on Options in Mail and switch to New Mail today or register for free at http://mail.yahoo.ca
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From dujinfang at gmail.com  Sat Aug 15 08:41:29 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 15 Aug 2009 23:41:29 +0800
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness patch
In-Reply-To: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
Message-ID: <38D6F955-9003-4862-ABA9-A79778508A46@gmail.com>

Great works. I tested and reported results in jira.

And as I noticed you removed the sequential line hunting methods.  
Though I don't use that I think someone else may need that. Think  
about the guy want skypeout accounts in a round robin manner, others  
might use that in a priority manner, that means sequential search. So  
it maybe a good idea to add it back. At the same time I think "ANY" is  
not a good name for that method, so named it "SEQ" maybe better. Then  
"ANY" can be removed later by announcing here or keep there for  
backwards compatibility.

Thank you very much for merging in the sk list with statistic patch,  
and for the other two features, I found it's a little hard to split  
codes, so, can make two jira, and upload one patch file?

Two features are:

continue load on fail: make sure the module continue load even it  
failed to talk to a skype instance
auto skype user: get the user name by the returned CURRENTUSERHANDLE  
other than  from the config xml, for easier config.

Thanks.

-7-

On Aug 15, 2009, at 1:43 AM, Giovanni Maruzzelli wrote:
> Hi FreeSWITCHers,
>
> all the users of mod_skypiax are kindly requested to test the svn  
> trunk 14519.
>
> It contains a lot of changes meant to add stability and robustness,
> toward a production environment.
>
> Let me know how your feelings, and please add to the Jira any possible
> bug/issue/etc.
>
> Thanks to you all,
>
> -giovanni
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
>
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From gmaruzz at celliax.org  Sat Aug 15 09:02:47 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 15 Aug 2009 18:02:47 +0200
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness 	patch
In-Reply-To: <38D6F955-9003-4862-ABA9-A79778508A46@gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com> 
	<38D6F955-9003-4862-ABA9-A79778508A46@gmail.com>
Message-ID: <7b197bef0908150902x18d4a68am7fb47dc76f781565@mail.gmail.com>

On Sat, Aug 15, 2009 at 5:41 PM, Seven Du<dujinfang at gmail.com> wrote:
> And as I noticed you removed the sequential line hunting methods.

Because was broken. So, I aliased it to the RR.
If you think it can be useful, add a Jira for it

> Thank you very much for merging in the sk list with statistic patch,
Thanks to you for sending the patch! I've only added the callflow of
the skype client to it

>
> Two features are:
>
> continue load on fail: make sure the module continue load even it
> failed to talk to a skype instance

mmmmh, I'm too conservative for this one: I prefer that if you
configured a skype instance, you expect it to work, so the module must
fail if there is not such instance

> auto skype user: get the user name by the returned CURRENTUSERHANDLE
> other than ?from the config xml, for easier config.

the username returned by CURRENTUSERHANDLE is checked against the
config file because is the only way you can associate interface_name
with its related Skype client instance on Windoz (no multiple X
servers there).

Thanks a lot for all your efforts!!!

-giovanni




>
> On Aug 15, 2009, at 1:43 AM, Giovanni Maruzzelli wrote:
>> Hi FreeSWITCHers,
>>
>> all the users of mod_skypiax are kindly requested to test the svn
>> trunk 14519.
>>
>> It contains a lot of changes meant to add stability and robustness,
>> toward a production environment.
>>
>> Let me know how your feelings, and please add to the Jira any possible
>> bug/issue/etc.
>>
>> Thanks to you all,
>>
>> -giovanni
>>
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>>
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dujinfang at gmail.com  Sat Aug 15 09:41:19 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 16 Aug 2009 00:41:19 +0800
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness 	patch
In-Reply-To: <7b197bef0908150902x18d4a68am7fb47dc76f781565@mail.gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
	<38D6F955-9003-4862-ABA9-A79778508A46@gmail.com>
	<7b197bef0908150902x18d4a68am7fb47dc76f781565@mail.gmail.com>
Message-ID: <EABE5B4A-4572-400E-BA60-0512CA329982@gmail.com>


On Aug 16, 2009, at 12:02 AM, Giovanni Maruzzelli wrote:
> On Sat, Aug 15, 2009 at 5:41 PM, Seven Du<dujinfang at gmail.com> wrote:
>> And as I noticed you removed the sequential line hunting methods.
>
> Because was broken. So, I aliased it to the RR.
> If you think it can be useful, add a Jira for it
>

Ok, as you think it's broken, better to leave it as is.
>> Thank you very much for merging in the sk list with statistic patch,
> Thanks to you for sending the patch! I've only added the callflow of
> the skype client to it
>
I think maybe you merged in an old version of the patch, the last  
should be skype_status_new1.diff, which I added one more statistic  
line and the unsigned long should be uint32_t I think. however, I can  
add another patch if you think it's useful. Oh, one more, it also set  
channel name to skypiax/RR/sk_1/other_skype_name for easy check in log. 
(As I also would like to see sofia/external/xxxx to be sofia/gateway/ 
gw/xxxx :) ).

>>
>> Two features are:
>>
>> continue load on fail: make sure the module continue load even it
>> failed to talk to a skype instance
>
> mmmmh, I'm too conservative for this one: I prefer that if you
> configured a skype instance, you expect it to work, so the module must
> fail if there is not such instance
>

One of the reason I think it's useful is one can configure to load  
everything on server boot time.  I run two skypiax servers, one can  
start 20 instances in batch without any problem but the other only  
starts 50%, then I need manually start them over and over until I  
confirmed all works with client or skypiax_auth. The two servers are  
not in the same datacenter but all have public ip. Once it started  
working we never never experienced a skype instance stoped working.  
But I experienced that kill a skype instance immediately caused  
skypiax core dump.

Sure it might cause other bugs even it is configured to false by  
default. I gona merge in my branch in case others using that.

>> auto skype user: get the user name by the returned CURRENTUSERHANDLE
>> other than  from the config xml, for easier config.
>
> the username returned by CURRENTUSERHANDLE is checked against the
> config file because is the only way you can associate interface_name
> with its related Skype client instance on Windoz (no multiple X
> servers there).
>

It by default disabled so I guess nothing will break.

> Thanks a lot for all your efforts!!!
>
> -giovanni
>
>
>
>
>>
>> On Aug 15, 2009, at 1:43 AM, Giovanni Maruzzelli wrote:
>>> Hi FreeSWITCHers,
>>>
>>> all the users of mod_skypiax are kindly requested to test the svn
>>> trunk 14519.
>>>
>>> It contains a lot of changes meant to add stability and robustness,
>>> toward a production environment.
>>>
>>> Let me know how your feelings, and please add to the Jira any  
>>> possible
>>> bug/issue/etc.
>>>
>>> Thanks to you all,
>>>
>>> -giovanni
>>>
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>>
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From peder at networkoblivion.com  Sat Aug 15 11:13:25 2009
From: peder at networkoblivion.com (Peder)
Date: Sat, 15 Aug 2009 13:13:25 -0500
Subject: [Freeswitch-users] ClueCon2009 Torrents
Message-ID: <018d01ca1dd4$14c97620$3e5c6260$@com>

I created torrents for all of the presentations from ClueCon 2009 and have
them seeded on 2-3 machines.  If you want the torrents, email me off list.

Peder




From gabe at gundy.org  Sat Aug 15 12:33:00 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 15 Aug 2009 13:33:00 -0600
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <018d01ca1dd4$14c97620$3e5c6260$@com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
Message-ID: <903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>

On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com> wrote:
> If you want the torrents, email me off list.

Why off list?  Isn't the point of torrents to have more people sharing
in the load?

Gabe


From scott.torr.fs at letterboxes.org  Sat Aug 15 10:31:32 2009
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Sun, 16 Aug 2009 03:31:32 +1000
Subject: [Freeswitch-users] Audio only in one direction when calling FS from
	Skype
Message-ID: <1250357492.23345.1330000179@webmail.messagingengine.com>

Freeswitch-users,

I'm very new to Freeswitch and have installed the following in VMware
Server 2.0.1


ubuntu-8.04.3-server-i386.iso (udate/upgrade)
skype-debian_2.0.0.72-1_i386.deb
FreeSWITCH Version 1.0.trunk (14492)
mod_skypiax


Using the FS wiki to install/setup and making small changes to default
XML configs as required.


(phone)--(SIP ATA)--(FS)--(MS-Skype client | Skype 'online number' via
PSTN)


The MS-Skype client is on the same local network as FS.


I can make calls from the phone to the Skype Client and this works OK.
Audio path OK: phone<-->FS<-->Skype


But ,if I call from the MS-Skype client to the phone I hear no audio
from the MS-Skype client.  

Audio path: phone<-->FS-->Skype

Likewise a call from a Skype 'online number' can hear for example the
default 5000 ivr but the DTMF tones from the PSTN phone are not detected
by FS when <action application="start_dtmf" /> is used.



Any suggestions or pointers in the right direction would be much
appreciated. 

Most likely I have overlook something very obvious to others.


Thanks in advance,
Scott Torr 







From manoj.joshi.13jan at gmail.com  Sat Aug 15 12:33:41 2009
From: manoj.joshi.13jan at gmail.com (Manu)
Date: Sun, 16 Aug 2009 01:03:41 +0530
Subject: [Freeswitch-users] mod nibblebill question
Message-ID: <26474d4e0908151233s1376ab15h74d2ac32b58593bf@mail.gmail.com>

Hello,

If we use heartbeat option on in nibblebill.conf.xml does that mean ODBC
database table will be updated every microsecond or any other interval we
set?

If this is so and there are many users (Lets say 500 users) are connected to
FS wouldn't it create locking issues in DB?

Regards,

Manoj
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From jmesquita at gmail.com  Sat Aug 15 13:34:05 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 15 Aug 2009 17:34:05 -0300
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
Message-ID: <5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>

I am interested and would also seed to the community

On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com> wrote:
>> If you want the torrents, email me off list.
>
> Why off list?  Isn't the point of torrents to have more people sharing
> in the load?
>
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From jmesquita at gmail.com  Sat Aug 15 13:49:35 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 15 Aug 2009 17:49:35 -0300
Subject: [Freeswitch-users] Question about an ESL function
In-Reply-To: <788835.43200.qm@web35604.mail.mud.yahoo.com>
References: <788835.43200.qm@web35604.mail.mud.yahoo.com>
Message-ID: <5a8712120908151349p7f584ab3wa07f679d04b7c06f@mail.gmail.com>

Recv will lock the calling thread until it gets an event or gets
disconnected while recvtimed will return controll when event received
or timer expires. Whatever comes fisrt.

On 8/15/09, Jean-Marc Hyppolite <hyppolite72 at yahoo.com> wrote:
> Hello,
>
> I would like to know the purpose of the ESL function named recvEventTimed.
>
> Thank you for your help.
>
>
>       __________________________________________________________________
> Be smarter than spam. See how smart SpamGuard is at giving junk email the
> boot with the All-new Yahoo! Mail.  Click on Options in Mail and switch to
> New Mail today or register for free at http://mail.yahoo.ca

-- 
Sent from my mobile device


From mike at jerris.com  Sat Aug 15 14:20:11 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 15 Aug 2009 17:20:11 -0400
Subject: [Freeswitch-users] mod nibblebill question
In-Reply-To: <26474d4e0908151233s1376ab15h74d2ac32b58593bf@mail.gmail.com>
References: <26474d4e0908151233s1376ab15h74d2ac32b58593bf@mail.gmail.com>
Message-ID: <4E77075F-4EF0-4CD0-A36A-D68DA8F438EC@jerris.com>

increments are in seconds, not microseconds.  In IMS for example I  
think it defaults to 20 or 30 second nibbles, depending on your  
tolerances and billing increments something much larger may even make  
sense.  Doing billing in sub second increments doesn't make a lot of  
sense to me.  Remember that this is just keeping track of available  
credit so if there are multiple calls at the same time you won't go  
over balance.  Everything is still reconciled at hang up, so if you  
have a bit too much reserved from your nibble the worst that could  
happen is it could cut off calls a little too early when multiple  
calls are in progress on the same account.

Mike

On Aug 15, 2009, at 3:33 PM, Manu wrote:

> Hello,
>
> If we use heartbeat option on in nibblebill.conf.xml does that mean  
> ODBC database table will be updated every microsecond or any other  
> interval we set?
>
> If this is so and there are many users (Lets say 500 users) are  
> connected to FS wouldn't it create locking issues in DB?
>
> Regards,
>
> Manoj


From mike at jerris.com  Sat Aug 15 14:20:59 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 15 Aug 2009 17:20:59 -0400
Subject: [Freeswitch-users] Audio only in one direction when calling FS
	from Skype
In-Reply-To: <1250357492.23345.1330000179@webmail.messagingengine.com>
References: <1250357492.23345.1330000179@webmail.messagingengine.com>
Message-ID: <0031C8A1-BCE9-445F-8DE0-177F24D576F5@jerris.com>

Make sure your external ip addresses and local networks are configured  
properly.

Mike

On Aug 15, 2009, at 1:31 PM, Scott Torr wrote:

> Freeswitch-users,
>
> I'm very new to Freeswitch and have installed the following in VMware
> Server 2.0.1
>
>
> ubuntu-8.04.3-server-i386.iso (udate/upgrade)
> skype-debian_2.0.0.72-1_i386.deb
> FreeSWITCH Version 1.0.trunk (14492)
> mod_skypiax
>
>
> Using the FS wiki to install/setup and making small changes to default
> XML configs as required.
>
>
> (phone)--(SIP ATA)--(FS)--(MS-Skype client | Skype 'online number' via
> PSTN)
>
>
> The MS-Skype client is on the same local network as FS.
>
>
> I can make calls from the phone to the Skype Client and this works OK.
> Audio path OK: phone<-->FS<-->Skype
>
>
> But ,if I call from the MS-Skype client to the phone I hear no audio
> from the MS-Skype client.
>
> Audio path: phone<-->FS-->Skype
>
> Likewise a call from a Skype 'online number' can hear for example the
> default 5000 ivr but the DTMF tones from the PSTN phone are not  
> detected
> by FS when <action application="start_dtmf" /> is used.
>
>
>
> Any suggestions or pointers in the right direction would be much
> appreciated.
>
> Most likely I have overlook something very obvious to others.
>
>
> Thanks in advance,
> Scott Torr
>



From hyppolite72 at yahoo.com  Sat Aug 15 14:39:32 2009
From: hyppolite72 at yahoo.com (Jean-Marc Hyppolite)
Date: Sat, 15 Aug 2009 14:39:32 -0700 (PDT)
Subject: [Freeswitch-users] Question about an ESL function
In-Reply-To: <5a8712120908151349p7f584ab3wa07f679d04b7c06f@mail.gmail.com>
Message-ID: <82725.8785.qm@web35601.mail.mud.yahoo.com>

Thanks a lot.

--- On Sat, 8/15/09, Jo?o Mesquita <jmesquita at gmail.com> wrote:


From: Jo?o Mesquita <jmesquita at gmail.com>
Subject: Re: [Freeswitch-users] Question about an ESL function
To: freeswitch-users at lists.freeswitch.org
Received: Saturday, August 15, 2009, 4:49 PM


Recv will lock the calling thread until it gets an event or gets
disconnected while recvtimed will return controll when event received
or timer expires. Whatever comes fisrt.

On 8/15/09, Jean-Marc Hyppolite <hyppolite72 at yahoo.com> wrote:
> Hello,
>
> I would like to know the purpose of the ESL function named recvEventTimed.
>
> Thank you for your help.
>
>
>? ? ???__________________________________________________________________
> Be smarter than spam. See how smart SpamGuard is at giving junk email the
> boot with the All-new Yahoo! Mail.? Click on Options in Mail and switch to
> New Mail today or register for free at http://mail.yahoo.ca

-- 
Sent from my mobile device

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      __________________________________________________________________
Ask a question on any topic and get answers from real people. Go to Yahoo! Answers and share what you know at http://ca.answers.yahoo.com
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From jaybinks at gmail.com  Sat Aug 15 14:43:20 2009
From: jaybinks at gmail.com (Jay Binks)
Date: Sun, 16 Aug 2009 07:43:20 +1000
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
Message-ID: <D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>

I'd also seed such a torrent.

Please send the link :)



On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:

> I am interested and would also seed to the community
>
> On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
>> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>  
>> wrote:
>>> If you want the torrents, email me off list.
>>
>> Why off list?  Isn't the point of torrents to have more people  
>> sharing
>> in the load?
>>
>> Gabe
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
> -- 
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From diego.viola at gmail.com  Sat Aug 15 14:47:28 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 15 Aug 2009 17:47:28 -0400
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
Message-ID: <86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>

Upload the torrent files in http://files.freeswitch.org ;)

On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com> wrote:

> I'd also seed such a torrent.
>
> Please send the link :)
>
>
>
> On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:
>
> > I am interested and would also seed to the community
> >
> > On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
> >> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>
> >> wrote:
> >>> If you want the torrents, email me off list.
> >>
> >> Why off list?  Isn't the point of torrents to have more people
> >> sharing
> >> in the load?
> >>
> >> Gabe
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >
> > --
> > Sent from my mobile device
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peder at networkoblivion.com  Sat Aug 15 15:34:32 2009
From: peder at networkoblivion.com (Peder)
Date: Sat, 15 Aug 2009 17:34:32 -0500
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
	<86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
Message-ID: <023901ca1df8$8f404810$adc0d830$@com>

I don?t have access to do that or I would.  That?s why I offered to email
them to whoever wants them.    I did send them to Brian earlier, but he must
have some sort of life outside of FreeSWITCH  because he hasn?t put them
there yet.  ;-)

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego
Viola
Sent: Saturday, August 15, 2009 4:47 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] ClueCon2009 Torrents

 

Upload the torrent files in http://files.freeswitch.org ;)

On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com> wrote:

I'd also seed such a torrent.

Please send the link :)




On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:

> I am interested and would also seed to the community
>
> On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
>> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>
>> wrote:
>>> If you want the torrents, email me off list.
>>
>> Why off list?  Isn't the point of torrents to have more people
>> sharing
>> in the load?
>>
>> Gabe
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>
> --
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From jmesquita at gmail.com  Sat Aug 15 15:37:37 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 15 Aug 2009 19:37:37 -0300
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <023901ca1df8$8f404810$adc0d830$@com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
	<86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
	<023901ca1df8$8f404810$adc0d830$@com>
Message-ID: <5a8712120908151537t50ddc42ej1b47718b6f4e2d79@mail.gmail.com>

I am already seeding from here.

jmesquita

On Sat, Aug 15, 2009 at 7:34 PM, Peder <peder at networkoblivion.com> wrote:

>  I don?t have access to do that or I would.  That?s why I offered to email
> them to whoever wants them.    I did send them to Brian earlier, but he must
> have some sort of life outside of FreeSWITCH  because he hasn?t put them
> there yet.  ;-)
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
> *Sent:* Saturday, August 15, 2009 4:47 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] ClueCon2009 Torrents
>
>
>
> Upload the torrent files in http://files.freeswitch.org ;)
>
> On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com> wrote:
>
> I'd also seed such a torrent.
>
> Please send the link :)
>
>
>
>
> On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:
>
> > I am interested and would also seed to the community
> >
> > On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
> >> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>
> >> wrote:
> >>> If you want the torrents, email me off list.
> >>
> >> Why off list?  Isn't the point of torrents to have more people
> >> sharing
> >> in the load?
> >>
> >> Gabe
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >
> > --
> > Sent from my mobile device
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From terrymr at gmail.com  Sat Aug 15 16:29:28 2009
From: terrymr at gmail.com (Terry Moore-Read)
Date: Sat, 15 Aug 2009 16:29:28 -0700
Subject: [Freeswitch-users] BLF and Openzap
Message-ID: <2d9dff7e0908151629o1a5a16fan8179048c1371fa67@mail.gmail.com>

Is it possible to have a sip phone show blf status for a phone which
is connected to an openzap port ?


From kjv at ken-ton.com  Sat Aug 15 17:30:54 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Sat, 15 Aug 2009 20:30:54 -0400
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <5a8712120908151537t50ddc42ej1b47718b6f4e2d79@mail.gmail.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
	<86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
	<023901ca1df8$8f404810$adc0d830$@com>
	<5a8712120908151537t50ddc42ej1b47718b6f4e2d79@mail.gmail.com>
Message-ID: <9136B992-B844-420C-B8C0-4224F09C8C36@ken-ton.com>


Torrent philez are 5mall...  I'd think it woud be a simple matter to  
attach them to the e-mail you're sending to the list eh?
I'd also be interested in seeding them...


Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Aug 15, 2009, at 6:37 PM, Jo?o Mesquita wrote:

> I am already seeding from here.
>
> jmesquita
>
> On Sat, Aug 15, 2009 at 7:34 PM, Peder <peder at networkoblivion.com>  
> wrote:
> I don?t have access to do that or I would.  That?s why I offered to  
> email them to whoever wants them.    I did send them to Brian  
> earlier, but he must have some sort of life outside of FreeSWITCH   
> because he hasn?t put them there yet.  ;-)
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Diego Viola
> Sent: Saturday, August 15, 2009 4:47 PM
>
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ClueCon2009 Torrents
>
>
> Upload the torrent files in http://files.freeswitch.org ;)
>
> On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com> wrote:
>
> I'd also seed such a torrent.
>
> Please send the link :)
>
>
>
>
> On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:
>
> > I am interested and would also seed to the community
> >
> > On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
> >> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>
> >> wrote:
> >>> If you want the torrents, email me off list.
> >>
> >> Why off list?  Isn't the point of torrents to have more people
> >> sharing
> >> in the load?
> >>
> >> Gabe
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >
> > --
> > Sent from my mobile device
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nicolas at medularis.com  Sat Aug 15 18:28:15 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Sat, 15 Aug 2009 21:28:15 -0400
Subject: [Freeswitch-users] Not receiving DTMF
Message-ID: <1b46b4e80908151828l2fc3b893rf39b203aba23c745@mail.gmail.com>

Hi, I'm trying to get dtmf input, but I'm not getting anything.

What I discovered though, is that my provider is at fault, since when I
switched to another voip provider, everything started to work beautifully.

My question is: since my provider is not doing RC2833 dtmf (even though they
say they do), is there another way to get dtmf to work? I'm doing everything
in a javascript file, so I tried doing:

- session.setVariable("dtmf_type","info");

Also:

- session.setVariable("dtmf_type","rfc2833");

And:

- session.execute("start_dtmf");

But none worked. The voip provider that works would be ideal, but the calls
are twice as expensive, hence besides testing, I wouldn't use them for a
real case scenario. BTW, anyone know of a good quality VoIP provider with
low rates for termination to Santiago, Chile and Chilean cell phones?

Thanks!


Nicolas
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From jason at jasonjgw.net  Sat Aug 15 18:47:18 2009
From: jason at jasonjgw.net (Jason White)
Date: Sun, 16 Aug 2009 11:47:18 +1000
Subject: [Freeswitch-users] Not receiving DTMF
In-Reply-To: <1b46b4e80908151828l2fc3b893rf39b203aba23c745@mail.gmail.com>
References: <1b46b4e80908151828l2fc3b893rf39b203aba23c745@mail.gmail.com>
Message-ID: <20090816014718.GA25066@jdc.jasonjgw.net>

Nicolas Brenner <nicolas at medularis.com> wrote:
> My question is: since my provider is not doing RC2833 dtmf (even though they
> say they do), is there another way to get dtmf to work? 

You can try info and tone detection just in case one of those is being used,
but it appears from your message that you have attempted both of these
already.

I think it's time to find a more reliable provider, or try to persuade your
current provider to fix it (the latter is probably a waste of time and effort,
however).



From dujinfang at gmail.com  Sat Aug 15 18:51:13 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 16 Aug 2009 09:51:13 +0800
Subject: [Freeswitch-users] opal build error
Message-ID: <5BBB829D-BBFC-4A2D-A18F-A6D92F631EA2@gmail.com>

Hi,

According to wiki it still in development status, but should compile  
right? Any idea about this? thanks.

make

In file included from mod_opal.cpp:25:
mod_opal.h:151: error: conflicting return type specified for ?virtual  
OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)?
/usr/include/opal/opal/localep.h:267: error:   overriding ?virtual  
ptlib_virtual_function_changed_or_removed******  
OpalLocalEndPoint::CreateConnection(OpalCall&, void*)?
mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&,  
FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,  
switch_channel_t*)?:
mod_opal.cpp:564: error: no matching function for call to  
?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)?
/usr/include/opal/opal/localep.h:290: note: candidates are:  
OpalLocalConnection::OpalLocalConnection(OpalCall&,  
OpalLocalEndPoint&, void*, unsigned int,  
OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h: 
276: note:                  
OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
mod_opal.cpp: In member function ?switch_status_t  
FSConnection::receive_message(switch_core_session_message_t*)?:
mod_opal.cpp:1037: error: ?SWITCH_CHANNEL_SESSION_LOG? was not  
declared in this scope
make[1]: *** [mod_opal.lo] Error 1
make: *** [all] Error 1

From mitul at enterux.com  Sat Aug 15 20:28:38 2009
From: mitul at enterux.com (Mitul Limbani)
Date: Sun, 16 Aug 2009 08:58:38 +0530
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
	<86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
Message-ID: <87677B2E-6B68-43D0-A718-01CF5D08F1E2@enterux.com>

I would be glad to offer mirror service to Cluecon 2009 videos :)

Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com

On 16-Aug-2009, at 3:17 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Upload the torrent files in http://files.freeswitch.org ;)
>
> On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com> wrote:
> I'd also seed such a torrent.
>
> Please send the link :)
>
>
>
> On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:
>
> > I am interested and would also seed to the community
> >
> > On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
> >> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>
> >> wrote:
> >>> If you want the torrents, email me off list.
> >>
> >> Why off list?  Isn't the point of torrents to have more people
> >> sharing
> >> in the load?
> >>
> >> Gabe
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >
> > --
> > Sent from my mobile device
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From markmorreny at gmail.com  Sat Aug 15 20:44:08 2009
From: markmorreny at gmail.com (mark morreny)
Date: Sun, 16 Aug 2009 11:44:08 +0800
Subject: [Freeswitch-users] obtain dialplan content from switch_extension_t
Message-ID: <20ad6b920908152044r3b549875tc650e02262bf80d4@mail.gmail.com>

Hi,

After I do some operation on switch_extension_t, basically adding some
actions into it, I would like to display its content in the form of:

<extension name="xxx">
   <condition>
   <action>xxxx</action>
 </condition>

</extension>

Is that doable?  If so,  is there any existing functional call that I can
use?

Thanks,
Mark
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From mrene_lists at avgs.ca  Sat Aug 15 20:50:39 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sat, 15 Aug 2009 23:50:39 -0400
Subject: [Freeswitch-users] obtain dialplan content from
	switch_extension_t
In-Reply-To: <20ad6b920908152044r3b549875tc650e02262bf80d4@mail.gmail.com>
References: <20ad6b920908152044r3b549875tc650e02262bf80d4@mail.gmail.com>
Message-ID: <D17834AF-2569-488F-AFCE-9048678860C0@avgs.ca>

Hi,

caller_extension->applications is a linked list so you can loop  
through it and print the app name & params, but the structure doesn't  
contain any information about the conditions that were matched as this  
is a mod_xml_dialplan feature.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 15-Aug-09, at 11:44 PM, mark morreny wrote:

> Hi,
>
> After I do some operation on switch_extension_t, basically adding  
> some actions into it, I would like to display its content in the  
> form of:
>
> <extension name="xxx">
>    <condition>
>    <action>xxxx</action>
>  </condition>
>
> </extension>
>
> Is that doable?  If so,  is there any existing functional call that  
> I can use?
>
> Thanks,
> Mark
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From peter.olsson at visionutveckling.se  Sat Aug 15 23:20:07 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 16 Aug 2009 08:20:07 +0200
Subject: [Freeswitch-users] opal build error
In-Reply-To: <5BBB829D-BBFC-4A2D-A18F-A6D92F631EA2@gmail.com>
Message-ID: <C6AD71B7.16ED9%peter.olsson@visionutveckling.se>

Make sure to do a complete rebuild. And also read the comments in jira MODOPAL-10.

/Peter


On 09-08-16 03.51, "Seven Du" <dujinfang at gmail.com> wrote:

Hi,

According to wiki it still in development status, but should compile
right? Any idea about this? thanks.

make

In file included from mod_opal.cpp:25:
mod_opal.h:151: error: conflicting return type specified for 'virtual
OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)'
/usr/include/opal/opal/localep.h:267: error:   overriding 'virtual
ptlib_virtual_function_changed_or_removed******
OpalLocalEndPoint::CreateConnection(OpalCall&, void*)'
mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall&,
FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
switch_channel_t*)':
mod_opal.cpp:564: error: no matching function for call to
'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)'
/usr/include/opal/opal/localep.h:290: note: candidates are:
OpalLocalConnection::OpalLocalConnection(OpalCall&,
OpalLocalEndPoint&, void*, unsigned int,
OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h:
276: note:
OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
mod_opal.cpp: In member function 'switch_status_t
FSConnection::receive_message(switch_core_session_message_t*)':
mod_opal.cpp:1037: error: 'SWITCH_CHANNEL_SESSION_LOG' was not
declared in this scope
make[1]: *** [mod_opal.lo] Error 1
make: *** [all] Error 1
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4a8767f232931167913993!





From dujinfang at gmail.com  Sat Aug 15 23:47:04 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 16 Aug 2009 14:47:04 +0800
Subject: [Freeswitch-users] opal build error
In-Reply-To: <C6AD71B7.16ED9%peter.olsson@visionutveckling.se>
References: <C6AD71B7.16ED9%peter.olsson@visionutveckling.se>
Message-ID: <33340D04-0986-4F5B-B8FF-4DFB59E10F60@gmail.com>

Thanks, will try later.

On Aug 16, 2009, at 2:20 PM, Peter Olsson wrote:
> Make sure to do a complete rebuild. And also read the comments in  
> jira MODOPAL-10.
>
> /Peter
>
>
> On 09-08-16 03.51, "Seven Du" <dujinfang at gmail.com> wrote:
>
> Hi,
>
> According to wiki it still in development status, but should compile
> right? Any idea about this? thanks.
>
> make
>
> In file included from mod_opal.cpp:25:
> mod_opal.h:151: error: conflicting return type specified for 'virtual
> OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)'
> /usr/include/opal/opal/localep.h:267: error:   overriding 'virtual
> ptlib_virtual_function_changed_or_removed******
> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)'
> mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall&,
> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
> switch_channel_t*)':
> mod_opal.cpp:564: error: no matching function for call to
> 'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&,  
> NULL)'
> /usr/include/opal/opal/localep.h:290: note: candidates are:
> OpalLocalConnection::OpalLocalConnection(OpalCall&,
> OpalLocalEndPoint&, void*, unsigned int,
> OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h:
> 276: note:
> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
> mod_opal.cpp: In member function 'switch_status_t
> FSConnection::receive_message(switch_core_session_message_t*)':
> mod_opal.cpp:1037: error: 'SWITCH_CHANNEL_SESSION_LOG' was not
> declared in this scope
> make[1]: *** [mod_opal.lo] Error 1
> make: *** [all] Error 1
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4a8767f232931167913993!
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From scott.torr.fs at letterboxes.org  Sun Aug 16 00:08:10 2009
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Sun, 16 Aug 2009 17:08:10 +1000
Subject: [Freeswitch-users] Audio only in one direction when calling FS
 from Skype
In-Reply-To: <0031C8A1-BCE9-445F-8DE0-177F24D576F5@jerris.com>
References: <1250357492.23345.1330000179@webmail.messagingengine.com>
	<0031C8A1-BCE9-445F-8DE0-177F24D576F5@jerris.com>
Message-ID: <1250406490.9719.1330052905@webmail.messagingengine.com>


I have used a MS-Skype client on the local area network to receive Skype
calls and do not have any audio continuity problems, and would conclude
that there is not an issue with the firewall if this is working
correctly.

To the best of my knowledge the external ip addresses and local networks
are configured properly in FS.


When I use the (linux Skpye client)<=>(mod_skypiax)<=>(FS) to recieve
Skype calls however I cannot hear the incoming audio.

This problem only occurs with an incoming Skype call (ie cannot hear
their audio, they can hear mine).

If I make an outgoing call from FS to Skype there are no audio problem.

So this is an asymmetrical problem.

I have used wireshark and can see UDP packets going in both directions
on the local area network.
So the incoming audio data is hitting the (linux Skype client).


How can I investigate where the data is being dropped?

Is it between:

(linux Skype client)-->(mod_skypiax)
or
(mod_skypiax)-->(FS)
 

I need some advice on what debug commands to use to see where the audio
is being dropped or some clues as to how to pinpoint where the problem
is occurring.


For example is (mod_skypiax) getting the audio data from (linux Skype
client)?

How do investigate that?
How do I see that?
Are there channel IO byte counters?

The machine is using snd_dummy, has no vmware sound card.

It is strange the problem is only in one direction?



Thanks,
Scott Torr



On Sat, 15 Aug 2009 17:20 -0400, "Michael Jerris" <mike at jerris.com>
wrote:
> Make sure your external ip addresses and local networks are configured  
> properly.
> 
> Mike
> 
> On Aug 15, 2009, at 1:31 PM, Scott Torr wrote:
> 
> > Freeswitch-users,
> >
> > I'm very new to Freeswitch and have installed the following in VMware
> > Server 2.0.1
> >
> >
> > ubuntu-8.04.3-server-i386.iso (udate/upgrade)
> > skype-debian_2.0.0.72-1_i386.deb
> > FreeSWITCH Version 1.0.trunk (14492)
> > mod_skypiax
> >
> >
> > Using the FS wiki to install/setup and making small changes to default
> > XML configs as required.
> >
> >
> > (phone)--(SIP ATA)--(FS)--(MS-Skype client | Skype 'online number' via
> > PSTN)
> >
> >
> > The MS-Skype client is on the same local network as FS.
> >
> >
> > I can make calls from the phone to the Skype Client and this works OK.
> > Audio path OK: phone<-->FS<-->Skype
> >
> >
> > But ,if I call from the MS-Skype client to the phone I hear no audio
> > from the MS-Skype client.
> >
> > Audio path: phone<-->FS-->Skype
> >
> > Likewise a call from a Skype 'online number' can hear for example the
> > default 5000 ivr but the DTMF tones from the PSTN phone are not  
> > detected
> > by FS when <action application="start_dtmf" /> is used.
> >
> >
> >
> > Any suggestions or pointers in the right direction would be much
> > appreciated.
> >
> > Most likely I have overlook something very obvious to others.
> >
> >
> > Thanks in advance,
> > Scott Torr
> >
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From manoj.joshi.13jan at gmail.com  Sun Aug 16 01:05:23 2009
From: manoj.joshi.13jan at gmail.com (Manu)
Date: Sun, 16 Aug 2009 13:35:23 +0530
Subject: [Freeswitch-users] mod nibblebill question
In-Reply-To: <4E77075F-4EF0-4CD0-A36A-D68DA8F438EC@jerris.com>
References: <26474d4e0908151233s1376ab15h74d2ac32b58593bf@mail.gmail.com>
	<4E77075F-4EF0-4CD0-A36A-D68DA8F438EC@jerris.com>
Message-ID: <26474d4e0908160105j3fc44d73s50e42f0940b420c7@mail.gmail.com>

Thank you for the reply Michael. Let me ask your opinion on another related
matter also.

I am using a MS SQL Database (which will be there on a remote server). I
wish to keep ...

1- User database in SQL.
2- Dialplans in SQL.
3- CDR logged in SQL.
4- I also require to cut the call in real time when credit is over.

I want to deply this to get good performance for 500 calls simultaneous. I
see in documents that using http responses i can fetch data from my web
server. I figure i can return number of gateways and other dial plan
parameters this way. In the same HTTP request i can also return "call Rates"
for the called destination (which i can use in nibble)

Is there any other efficiant apprroach you can suggest?

Regards,

Manoj





On Sun, Aug 16, 2009 at 2:50 AM, Michael Jerris <mike at jerris.com> wrote:

> increments are in seconds, not microseconds.  In IMS for example I
> think it defaults to 20 or 30 second nibbles, depending on your
> tolerances and billing increments something much larger may even make
> sense.  Doing billing in sub second increments doesn't make a lot of
> sense to me.  Remember that this is just keeping track of available
> credit so if there are multiple calls at the same time you won't go
> over balance.  Everything is still reconciled at hang up, so if you
> have a bit too much reserved from your nibble the worst that could
> happen is it could cut off calls a little too early when multiple
> calls are in progress on the same account.
>
> Mike
>
> On Aug 15, 2009, at 3:33 PM, Manu wrote:
>
> > Hello,
> >
> > If we use heartbeat option on in nibblebill.conf.xml does that mean
> > ODBC database table will be updated every microsecond or any other
> > interval we set?
> >
> > If this is so and there are many users (Lets say 500 users) are
> > connected to FS wouldn't it create locking issues in DB?
> >
> > Regards,
> >
> > Manoj
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mgg at giagnocavo.net  Sun Aug 16 01:54:59 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Sun, 16 Aug 2009 04:54:59 -0400
Subject: [Freeswitch-users] mod nibblebill question
In-Reply-To: <26474d4e0908160105j3fc44d73s50e42f0940b420c7@mail.gmail.com>
References: <26474d4e0908151233s1376ab15h74d2ac32b58593bf@mail.gmail.com>
	<4E77075F-4EF0-4CD0-A36A-D68DA8F438EC@jerris.com>
	<26474d4e0908160105j3fc44d73s50e42f0940b420c7@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702D79D3B73@mse17be1.mse17.exchange.ms>

Just an in-general performance hint: Get your setup running, then use the SQL Profiler to generate a tuning trace, and run the "Database Engine Tuning Advisor" with it. That way it'll tell you which indexes you need (or need to drop) to optimize the database side of performance.

500 simultaneous calls isn't much though. Even with 30 second nibblebillings, that's only 16 updates/sec - and your calls/sec is probably lower than that. That's not too much work for any average server these days, even doing dialplan/cdr/billing against remote SQL.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Manu
Sent: Sunday, August 16, 2009 2:05 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod nibblebill question

Thank you for the reply Michael. Let me ask your opinion on another related matter also.

I am using a MS SQL Database (which will be there on a remote server). I wish to keep ...

1- User database in SQL.
2- Dialplans in SQL.
3- CDR logged in SQL.
4- I also require to cut the call in real time when credit is over.

I want to deply this to get good performance for 500 calls simultaneous. I see in documents that using http responses i can fetch data from my web server. I figure i can return number of gateways and other dial plan parameters this way. In the same HTTP request i can also return "call Rates" for the called destination (which i can use in nibble)

Is there any other efficiant apprroach you can suggest?

Regards,

Manoj





On Sun, Aug 16, 2009 at 2:50 AM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
increments are in seconds, not microseconds.  In IMS for example I
think it defaults to 20 or 30 second nibbles, depending on your
tolerances and billing increments something much larger may even make
sense.  Doing billing in sub second increments doesn't make a lot of
sense to me.  Remember that this is just keeping track of available
credit so if there are multiple calls at the same time you won't go
over balance.  Everything is still reconciled at hang up, so if you
have a bit too much reserved from your nibble the worst that could
happen is it could cut off calls a little too early when multiple
calls are in progress on the same account.

Mike

On Aug 15, 2009, at 3:33 PM, Manu wrote:

> Hello,
>
> If we use heartbeat option on in nibblebill.conf.xml does that mean
> ODBC database table will be updated every microsecond or any other
> interval we set?
>
> If this is so and there are many users (Lets say 500 users) are
> connected to FS wouldn't it create locking issues in DB?
>
> Regards,
>
> Manoj
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org<http://www.freeswitch.org/>

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From raffaele.p.guidi at gmail.com  Sun Aug 16 02:43:27 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sun, 16 Aug 2009 11:43:27 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I
	have a little problem
Message-ID: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>

I had the sweet surprise to find the installer packaged with FreePBX...
really great! Why it has not been advertised as it deserves? It worked like
a breeze once launched, with the automatic configuration and all of that.,
Only thing: once stopped I cannot get it to load sofia profiles anymore -
issueing sofia status doesn't show anything. I had to copy internal.xml and
default.xml from a previous installation and everything got to work again -
but no FreePBX anymore :( I'm sure I'm missing something important.
Can you give me a hint? Should sofia profiles be served by curl or
something?

Thanks,
   Raffaele
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From woodydickson at gmail.com  Sun Aug 16 04:24:50 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Sun, 16 Aug 2009 19:24:50 +0800
Subject: [Freeswitch-users] how to set different action for different cause
	code
Message-ID: <a44688a0908160424n3569cef1vad8fcdc68cbfb96e@mail.gmail.com>

Hello,

I find hangup_hook, but I would like to define different actions for
different hangup codes.  Is there anyway to do that?

Woody
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From tzury.by at reguluslabs.com  Sun Aug 16 04:25:07 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Sun, 16 Aug 2009 14:25:07 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
Message-ID: <10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>

Hi Brian,

Just for your information here is a mail I got from a colleague of
mine which I consider as an experienced freeswitch integrator

> "... TCP used to work, and I had about 20-30 client phones connecting with it.
> About 3 months ago I did an upgrade and TCP no longer worked.? I was in panic mode, so I just ended up walking my
> clients through reconfiguring their phones to use UDP.
>
> It is definitely a bug in Freeswitch.? I can help you work up a jira bug report if you like.
> The bug happened somewhere between svn version 9800ish and 13223.
> 13223 is the version I'm running on my systems now, older, but stable for me..."

It would be just great if you can confirm the TCP functionality in the
latest release of FreeSwitch


/t


From asannucci at gmail.com  Sun Aug 16 05:07:18 2009
From: asannucci at gmail.com (bakko)
Date: Sun, 16 Aug 2009 14:07:18 +0200
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com><2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org><10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com><43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org><10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com><45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
Message-ID: <2278A0310B024C82B239E2FA9D6D2B77@voztovoice>

I'm just tryng TCP now and work (latest trunk)

I done this test with eyebeam phone registeres to TCP port and call other ip 
phone (registered over UDB port).

The audio work fine in both directions.

My configuration:

FS --> internet --> NAT --> Phones

In the internal profile:

Call-ID:        OTIyNzRlNTQxNTk5NzBhYzMzYzM1YmQyNTgzNTA3MTk.
User:           1004 at nydomain.com
Contact:        "User" 
<sip:1004 at 87.12.101.59:13035;transport=TCP;rinstance=f3fd252ea20dd34a>
Agent:          eyeBeam release 1102u stamp 52345
Status:         Registered(TCP)(unknown) EXP(2009-08-16 08:59:38)
Host:           mydomain.com
IP:             XXX:XXX.XXX.XXX
Port:           13035
Auth-User:      1004
Auth-Realm:     mydomain.com





From carlos.talbot at gmail.com  Sun Aug 16 07:04:30 2009
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Sun, 16 Aug 2009 09:04:30 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
Message-ID: <5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>

When you configure FreePBX for the first time it wipes out the sip_profiles
directory. If you follow the FreePBX shortcut on your desktop it'll mention
this on the last screen of the configuration. You might see something such
as the following below. If you plan to use FreePBX you'll need to define
trunk groups, trunks, etc in order to have the sip_profiles directory
populated.
regards,

Carlos


Incompatible ConfigurationWARNING: THE FOLLOWING FILES WILL BE DELETED!

   - D:/FreeSWITCH/conf/sip_profiles/external.xml
   - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
   - D:/FreeSWITCH/conf/sip_profiles/internal.xml


On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> I had the sweet surprise to find the installer packaged with FreePBX...
> really great! Why it has not been advertised as it deserves? It worked like
> a breeze once launched, with the automatic configuration and all of that.,
> Only thing: once stopped I cannot get it to load sofia profiles anymore -
> issueing sofia status doesn't show anything. I had to copy internal.xml and
> default.xml from a previous installation and everything got to work again -
> but no FreePBX anymore :( I'm sure I'm missing something important.
> Can you give me a hint? Should sofia profiles be served by curl or
> something?
>
> Thanks,
>    Raffaele
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sun Aug 16 07:10:41 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Aug 2009 09:10:41 -0500
Subject: [Freeswitch-users] BLF and Openzap
In-Reply-To: <2d9dff7e0908151629o1a5a16fan8179048c1371fa67@mail.gmail.com>
References: <2d9dff7e0908151629o1a5a16fan8179048c1371fa67@mail.gmail.com>
Message-ID: <C9ABED9B-675E-4A30-AFD5-54A2E52C91CA@freeswitch.org>

Yes you sure can... setting the presence_id to exten at domain and have  
the phone sub to that you'll have the light come on when its in use.

/b

On Aug 15, 2009, at 6:29 PM, Terry Moore-Read wrote:

> Is it possible to have a sip phone show blf status for a phone which
> is connected to an openzap port ?



From brian at freeswitch.org  Sun Aug 16 07:11:20 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Aug 2009 09:11:20 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
Message-ID: <01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>

Works fine here.

/b

On Aug 16, 2009, at 6:25 AM, Tzury Bar Yochay wrote:

> Hi Brian,
>
> Just for your information here is a mail I got from a colleague of
> mine which I consider as an experienced freeswitch integrator
>
>> "... TCP used to work, and I had about 20-30 client phones  
>> connecting with it.
>> About 3 months ago I did an upgrade and TCP no longer worked.  I  
>> was in panic mode, so I just ended up walking my
>> clients through reconfiguring their phones to use UDP.
>>
>> It is definitely a bug in Freeswitch.  I can help you work up a  
>> jira bug report if you like.
>> The bug happened somewhere between svn version 9800ish and 13223.
>> 13223 is the version I'm running on my systems now, older, but  
>> stable for me..."
>
> It would be just great if you can confirm the TCP functionality in the
> latest release of FreeSwitch
>
>
> /t
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sun Aug 16 07:11:56 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Aug 2009 09:11:56 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
Message-ID: <5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>

chances are mod_sofia isn't loaded.

/b

On Aug 16, 2009, at 4:43 AM, Raffaele P. Guidi wrote:

> Can you give me a hint? Should sofia profiles be served by curl or  
> something?
>
> Thanks,
>    Raffaele



From grevenx at me.com  Sun Aug 16 07:55:54 2009
From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=)
Date: Sun, 16 Aug 2009 16:55:54 +0200
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <87677B2E-6B68-43D0-A718-01CF5D08F1E2@enterux.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
	<86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
	<87677B2E-6B68-43D0-A718-01CF5D08F1E2@enterux.com>
Message-ID: <8FA1E6A3-1C07-4127-B25D-B91BA6FF2029@me.com>

Can we get the torrents added to a mail on this list soon please?

Best regards,
Even Andr?

On 16. aug.. 2009, at 05.28, Mitul Limbani wrote:

> I would be glad to offer mirror service to Cluecon 2009 videos :)
>
> Thanks & Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions Pvt. Ltd.,
> The Enterprise Linux Company (r),
> http://www.enterux.com
> http://www.entVoice.com
>
> On 16-Aug-2009, at 3:17 AM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Upload the torrent files in http://files.freeswitch.org ;)
>>
>> On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com>  
>> wrote:
>> I'd also seed such a torrent.
>>
>> Please send the link :)
>>
>>
>>
>> On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:
>>
>> > I am interested and would also seed to the community
>> >
>> > On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
>> >> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>
>> >> wrote:
>> >>> If you want the torrents, email me off list.
>> >>
>> >> Why off list?  Isn't the point of torrents to have more people
>> >> sharing
>> >> in the load?
>> >>
>> >> Gabe
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>> freeswitch-
>> >> users
>> >> http://www.freeswitch.org
>> >>
>> >
>> > --
>> > Sent from my mobile device
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kokoska.rokoska at post.cz  Sun Aug 16 09:41:08 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Sun, 16 Aug 2009 18:41:08 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
Message-ID: <4A8836A4.20402@post.cz>


Hi all,

I have just upgarded my (a little bit old :-) FreeSWITCH to current
trunk and now i have troubles with IP addresses in SIP & SDP, while
before upgrade everything work fine...

My FreeSWITCH is behind NAT and all relevant sip profiles have set:

<param name="ext-rtp-ip" value="1.2.3.4"/>
<param name="ext-sip-ip" value="1.2.3.4"/>

But when I send INVITE thru the profile to the gateway associated with
it i see local IP in "Via" header and in SDP "m" line too.
BTW: The "Contact" header is populated with "external" IP (1.2.3.4)

Could you, please, point me how could I modify IP in Via and in SDP?
Thank you.

Best regards,

kokoska.rokoska



From brian at freeswitch.org  Sun Aug 16 10:10:40 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Aug 2009 12:10:40 -0500
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A8836A4.20402@post.cz>
References: <4A8836A4.20402@post.cz>
Message-ID: <6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>

You have to set the local-network-acl on the profile so it knows when  
or not to fix up the VIA
/b

On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:

>
> Hi all,
>
> I have just upgarded my (a little bit old :-) FreeSWITCH to current
> trunk and now i have troubles with IP addresses in SIP & SDP, while
> before upgrade everything work fine...
>
> My FreeSWITCH is behind NAT and all relevant sip profiles have set:
>
> <param name="ext-rtp-ip" value="1.2.3.4"/>
> <param name="ext-sip-ip" value="1.2.3.4"/>
>
> But when I send INVITE thru the profile to the gateway associated with
> it i see local IP in "Via" header and in SDP "m" line too.
> BTW: The "Contact" header is populated with "external" IP (1.2.3.4)



From tayeb.meftah at gmail.com  Sun Aug 16 11:25:11 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 16 Aug 2009 18:25:11 +0000
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
 but I have a little problem
In-Reply-To: <5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>
Message-ID: <4A884F07.3040200@gmail.com>

hello,
could you give me the installer Link to test it?
thanks
meftah tayeb
DelphiWorld at irc.freenode.net

    * Call Me (SIP) <sip:voipworld at sip.voipuser.org>
    * Call me (skype) <skype:tayeb.meftah>
    * Call Me (PSTN) <callto:0021348526972>

Brian West wrote:
> chances are mod_sofia isn't loaded.
>
> /b
>
> On Aug 16, 2009, at 4:43 AM, Raffaele P. Guidi wrote:
>
>   
>> Can you give me a hint? Should sofia profiles be served by curl or  
>> something?
>>
>> Thanks,
>>    Raffaele
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4339 (20090816) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4339 (20090816) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From carlos.talbot at gmail.com  Sun Aug 16 11:35:58 2009
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Sun, 16 Aug 2009 13:35:58 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <4A884F07.3040200@gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>
	<4A884F07.3040200@gmail.com>
Message-ID: <5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>

http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe

On Sun, Aug 16, 2009 at 1:25 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

>  hello,
> could you give me the installer Link to test it?
> thanks
> meftah tayeb
> DelphiWorld at irc.freenode.net
>
>    - Call Me (SIP)
>    - Call me (skype)
>    - Call Me (PSTN) <callto:0021348526972>
>
> Brian West wrote:
>
> chances are mod_sofia isn't loaded.
>
> /b
>
> On Aug 16, 2009, at 4:43 AM, Raffaele P. Guidi wrote:
>
>
>
>  Can you give me a hint? Should sofia profiles be served by curl or
> something?
>
> Thanks,
>    Raffaele
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4339 (20090816) __________
>
> The message was checked by ESET NOD32 Antivirus.
> http://www.eset.com
>
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4339 (20090816) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From terrymr at gmail.com  Sun Aug 16 11:47:00 2009
From: terrymr at gmail.com (Terry Moore-Read)
Date: Sun, 16 Aug 2009 11:47:00 -0700
Subject: [Freeswitch-users] BLF and Openzap
In-Reply-To: <C9ABED9B-675E-4A30-AFD5-54A2E52C91CA@freeswitch.org>
References: <2d9dff7e0908151629o1a5a16fan8179048c1371fa67@mail.gmail.com>
	<C9ABED9B-675E-4A30-AFD5-54A2E52C91CA@freeswitch.org>
Message-ID: <2d9dff7e0908161147g1a4f08feu30bc33d20f1a9811@mail.gmail.com>

On Sun, Aug 16, 2009 at 7:10 AM, Brian West<brian at freeswitch.org> wrote:
> Yes you sure can... setting the presence_id to exten at domain and have

Thats what I thought ... but where should I be setting it ?

> the phone sub to that you'll have the light come on when its in use.
>
> /b
>
> On Aug 15, 2009, at 6:29 PM, Terry Moore-Read wrote:
>
>> Is it possible to have a sip phone show blf status for a phone which
>> is connected to an openzap port ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Want to buy my photo's ? :
http://www.shutterstock.com/gallery.mhtml?id=309295&rid=309295


From krice at freeswitch.org  Sun Aug 16 11:49:22 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 16 Aug 2009 13:49:22 -0500
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <8FA1E6A3-1C07-4127-B25D-B91BA6FF2029@me.com>
Message-ID: <C6ADBEE3.5AFB7%krice@freeswitch.org>

Lets see if this works... There are a few seeders out there...



From: Even Andr? Fiskvik <grevenx at me.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Sun, 16 Aug 2009 16:55:54 +0200
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] ClueCon2009 Torrents

Can we get the torrents added to a mail on this list soon please?

Best regards,
Even Andr?

On 16. aug.. 2009, at 05.28, Mitul Limbani wrote:

> I would be glad to offer mirror service to Cluecon 2009 videos :)
> 
> Thanks & Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions Pvt. Ltd.,
> The Enterprise Linux Company (r),
> http://www.enterux.com
> http://www.entVoice.com
> 
> On 16-Aug-2009, at 3:17 AM, Diego Viola <diego.viola at gmail.com> wrote:
> 
>> Upload the torrent files in  <http://files.freeswitch.org>
>> http://files.freeswitch.org ;)
>> 
>> On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks < <mailto:jaybinks at gmail.com>
>> jaybinks at gmail.com> wrote:
>>  
>>> I'd also seed such a torrent.
>>>  
>>>  Please send the link :)
>>>  
>>> 
>>>  
>>>  
>>>  On 16/08/2009, at 6:34, Jo?o Mesquita < <mailto:jmesquita at gmail.com>
>>> jmesquita at gmail.com> wrote:
>>>  
>>>>  > I am interested and would also seed to the community
>>>>  >
>>>>  > On 8/15/09, Gabriel Gunderson < <mailto:gabe at gundy.org> gabe at gundy.org>
>>>> wrote:
>>>>>  >> On Sat, Aug 15, 2009 at 12:13 PM, Peder<
>>>>> <mailto:peder at networkoblivion.com> peder at networkoblivion.com>
>>>>>  >> wrote:
>>>>>>  >>> If you want the torrents, email me off list.
>>>>>  >>
>>>>>  >> Why off list?  Isn't the point of torrents to have more people
>>>>>  >> sharing
>>>>>  >> in the load?
>>>>>  >>
>>>>>  >> Gabe
>>>>>  >>
>>>>>  >> _______________________________________________
>>>>>  >> FreeSWITCH-users mailing list
>>>>>  >>  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  >>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>  >> UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch->
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>  >> users
>>>>>  >>  <http://www.freeswitch.org> http://www.freeswitch.org
>>>>>  >>
>>>>  >
>>>>  > --
>>>>  > Sent from my mobile device
>>>>  >
>>>>  > _______________________________________________
>>>>  > FreeSWITCH-users mailing list
>>>>  >  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  >  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>  > UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch->
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>  > users
>>>>  >  <http://www.freeswitch.org> http://www.freeswitch.org
>>>  
>>>  _______________________________________________
>>>  FreeSWITCH-users mailing list
>>>   <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>   <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>  UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>   <http://www.freeswitch.org> http://www.freeswitch.org
>>>  
>>> 
>>>  
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org

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From kokoska.rokoska at post.cz  Sun Aug 16 12:40:06 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Sun, 16 Aug 2009 21:40:06 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>
References: <4A8836A4.20402@post.cz>
	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>
Message-ID: <4A886096.3050900@post.cz>


Thank you very much, Brian, for your help!

Now are all IPs correct :-) (BTW: In my previous post I mean "c" line in
SDP, not "m" line). Thanks once more!

Well, signaling works fine, but I still have troubles with RTPs:
I can see with ngrep RTPs going to FreeSWITCH from both legs (a, b) but
there are not outgoing RTPs from FreeSWITCH to any of the call legs.
And, of course, caller and callee hear nothing :-)

Could you, please, help me to solve this? Thanks.

Best regards,

kokoska.rokoska




Brian West napsal(a):
> You have to set the local-network-acl on the profile so it knows when  
> or not to fix up the VIA
> /b
> 
> On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:
> 
>> Hi all,
>>
>> I have just upgarded my (a little bit old :-) FreeSWITCH to current
>> trunk and now i have troubles with IP addresses in SIP & SDP, while
>> before upgrade everything work fine...
>>
>> My FreeSWITCH is behind NAT and all relevant sip profiles have set:
>>
>> <param name="ext-rtp-ip" value="1.2.3.4"/>
>> <param name="ext-sip-ip" value="1.2.3.4"/>
>>
>> But when I send INVITE thru the profile to the gateway associated with
>> it i see local IP in "Via" header and in SDP "m" line too.
>> BTW: The "Contact" header is populated with "external" IP (1.2.3.4)
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From shaheryarkh at googlemail.com  Sun Aug 16 13:33:29 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Mon, 17 Aug 2009 02:33:29 +0600
Subject: [Freeswitch-users] Is there any freeswitch show version command
Message-ID: <b16156850908161333m3b1f0bb0nd71836cf505ef3ec@mail.gmail.com>

Hi,

How can we check the version / svn revision of a running FS instance?

I know, this is kind of a stupid question, but i sometimes run into
situation where i don't know or don't have access to FS source, nor i can
restart it to get its version string.

Thank you.


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From mrene_lists at avgs.ca  Sun Aug 16 13:35:06 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 16 Aug 2009 16:35:06 -0400
Subject: [Freeswitch-users] Is there any freeswitch show version command
In-Reply-To: <b16156850908161333m3b1f0bb0nd71836cf505ef3ec@mail.gmail.com>
References: <b16156850908161333m3b1f0bb0nd71836cf505ef3ec@mail.gmail.com>
Message-ID: <496E941D-748E-4166-A78D-036193EFBADF@avgs.ca>

Hi,

type:
version

at the CLI, it'll tell you.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 16-Aug-09, at 4:33 PM, Muhammad Shahzad wrote:

> Hi,
>
> How can we check the version / svn revision of a running FS instance?
>
> I know, this is kind of a stupid question, but i sometimes run into  
> situation where i don't know or don't have access to FS source, nor  
> i can restart it to get its version string.
>
> Thank you.
>
>
> -- 
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From shaheryarkh at googlemail.com  Sun Aug 16 13:40:49 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Mon, 17 Aug 2009 02:40:49 +0600
Subject: [Freeswitch-users] Is there any freeswitch show version command
In-Reply-To: <496E941D-748E-4166-A78D-036193EFBADF@avgs.ca>
References: <b16156850908161333m3b1f0bb0nd71836cf505ef3ec@mail.gmail.com>
	<496E941D-748E-4166-A78D-036193EFBADF@avgs.ca>
Message-ID: <b16156850908161340i3c382dcfv1f483ea9a2327db4@mail.gmail.com>

thanks.

On Mon, Aug 17, 2009 at 2:35 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> type: version
> at the CLI, it'll tell you.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 16-Aug-09, at 4:33 PM, Muhammad Shahzad wrote:
>
> Hi,
>
> How can we check the version / svn revision of a running FS instance?
>
> I know, this is kind of a stupid question, but i sometimes run into
> situation where i don't know or don't have access to FS source, nor i can
> restart it to get its version string.
>
> Thank you.
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From brian at freeswitch.org  Sun Aug 16 14:20:58 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Aug 2009 16:20:58 -0500
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A886096.3050900@post.cz>
References: <4A8836A4.20402@post.cz>
	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>
	<4A886096.3050900@post.cz>
Message-ID: <81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>

Ok update in a few I now recall why we had that && ! 
sofia_test_pflag(tech_pvt->profile, PFLAG_AUTO_NAT) in there.

/b

On Aug 16, 2009, at 2:40 PM, kokoska rokoska wrote:

>
> Thank you very much, Brian, for your help!
>
> Now are all IPs correct :-) (BTW: In my previous post I mean "c"  
> line in
> SDP, not "m" line). Thanks once more!
>
> Well, signaling works fine, but I still have troubles with RTPs:
> I can see with ngrep RTPs going to FreeSWITCH from both legs (a, b)  
> but
> there are not outgoing RTPs from FreeSWITCH to any of the call legs.
> And, of course, caller and callee hear nothing :-)
>
> Could you, please, help me to solve this? Thanks.
>
> Best regards,
>
> kokoska.rokoska
>
>
>
>
> Brian West napsal(a):
>> You have to set the local-network-acl on the profile so it knows when
>> or not to fix up the VIA
>> /b
>>
>> On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:
>>
>>> Hi all,
>>>
>>> I have just upgarded my (a little bit old :-) FreeSWITCH to current
>>> trunk and now i have troubles with IP addresses in SIP & SDP, while
>>> before upgrade everything work fine...
>>>
>>> My FreeSWITCH is behind NAT and all relevant sip profiles have set:
>>>
>>> <param name="ext-rtp-ip" value="1.2.3.4"/>
>>> <param name="ext-sip-ip" value="1.2.3.4"/>
>>>
>>> But when I send INVITE thru the profile to the gateway associated  
>>> with
>>> it i see local IP in "Via" header and in SDP "m" line too.
>>> BTW: The "Contact" header is populated with "external" IP (1.2.3.4)
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From tayeb.meftah at gmail.com  Sun Aug 16 15:02:15 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 16 Aug 2009 22:02:15 +0000
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
 but I have a little problem
In-Reply-To: <5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>	<4A884F07.3040200@gmail.com>
	<5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>
Message-ID: <4A8881E7.8080509@gmail.com>

hello
i'm rewriting this executable file in MSI format
i can use:

    * MakeMSI
    * WIX (Windows installer XML)

what you like?
WIX is open source, but MakeMSI i'm not sur
for web server, i'm replacing WAMP with a great web server called 
uniform server
this server have a conventional Unix Path configuration, for example, 
web files can by stored in /www/ and apache is in /usr/local/apache2
this facilitate the deploiment of applications
any suggestion?
thanks
Meftah Tayeb
Carlos Talbot wrote:
> http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe
>
> On Sun, Aug 16, 2009 at 1:25 PM, Meftah Tayeb <tayeb.meftah at gmail.com 
> <mailto:tayeb.meftah at gmail.com>> wrote:
>
>     hello,
>     could you give me the installer Link to test it?
>     thanks
>     meftah tayeb
>     DelphiWorld at irc.freenode.net <mailto:DelphiWorld at irc.freenode.net>
>
>         * Call Me (SIP)
>         * Call me (skype)
>         * Call Me (PSTN) <callto:0021348526972>
>
>     Brian West wrote:
>>     chances are mod_sofia isn't loaded.
>>
>>     /b
>>
>>     On Aug 16, 2009, at 4:43 AM, Raffaele P. Guidi wrote:
>>
>>       
>>>     Can you give me a hint? Should sofia profiles be served by curl or  
>>>     something?
>>>
>>>     Thanks,
>>>        Raffaele
>>>         
>>     _______________________________________________ FreeSWITCH-users
>>     mailing list FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>>     __________ Information from ESET NOD32 Antivirus, version of virus signature database 4339 (20090816) __________
>>
>>     The message was checked by ESET NOD32 Antivirus.
>>
>>     http://www.eset.com
>>
>>
>>       
>
>
>
>     __________ Information from ESET NOD32 Antivirus, version of virus
>     signature database 4339 (20090816) __________
>
>     The message was checked by ESET NOD32 Antivirus.
>
>     http://www.eset.com
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4340 (20090816) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4340 (20090816) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From brian at freeswitch.org  Sun Aug 16 15:04:36 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Aug 2009 17:04:36 -0500
Subject: [Freeswitch-users] BLF and Openzap
In-Reply-To: <2d9dff7e0908161147g1a4f08feu30bc33d20f1a9811@mail.gmail.com>
References: <2d9dff7e0908151629o1a5a16fan8179048c1371fa67@mail.gmail.com>
	<C9ABED9B-675E-4A30-AFD5-54A2E52C91CA@freeswitch.org>
	<2d9dff7e0908161147g1a4f08feu30bc33d20f1a9811@mail.gmail.com>
Message-ID: <B92E348F-8320-4CA8-94F0-7FC541CB1F13@freeswitch.org>

Just as a variable on the session.

/b

On Aug 16, 2009, at 1:47 PM, Terry Moore-Read wrote:

> Thats what I thought ... but where should I be setting it ?



From kjv at ken-ton.com  Sun Aug 16 10:43:58 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Sun, 16 Aug 2009 13:43:58 -0400
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <8FA1E6A3-1C07-4127-B25D-B91BA6FF2029@me.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
	<86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
	<87677B2E-6B68-43D0-A718-01CF5D08F1E2@enterux.com>
	<8FA1E6A3-1C07-4127-B25D-B91BA6FF2029@me.com>
Message-ID: <D5CF3ACC-A7E0-47EF-871D-D5013C250398@ken-ton.com>


Let's see if this flies onto the list...




Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Aug 16, 2009, at 10:55 AM, Even Andr? Fiskvik wrote:

> Can we get the torrents added to a mail on this list soon please?
>
> Best regards,
> Even Andr?
>
> On 16. aug.. 2009, at 05.28, Mitul Limbani wrote:
>
>> I would be glad to offer mirror service to Cluecon 2009 videos :)
>>
>> Thanks & Regards,
>> Mitul Limbani,
>> Founder & CEO,
>> Enterux Solutions Pvt. Ltd.,
>> The Enterprise Linux Company (r),
>> http://www.enterux.com
>> http://www.entVoice.com
>>
>> On 16-Aug-2009, at 3:17 AM, Diego Viola <diego.viola at gmail.com>  
>> wrote:
>>
>>> Upload the torrent files in http://files.freeswitch.org ;)
>>>
>>> On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com>  
>>> wrote:
>>> I'd also seed such a torrent.
>>>
>>> Please send the link :)
>>>
>>>
>>>
>>> On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:
>>>
>>> > I am interested and would also seed to the community
>>> >
>>> > On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
>>> >> On Sat, Aug 15, 2009 at 12:13 PM,  
>>> Peder<peder at networkoblivion.com>
>>> >> wrote:
>>> >>> If you want the torrents, email me off list.
>>> >>
>>> >> Why off list?  Isn't the point of torrents to have more people
>>> >> sharing
>>> >> in the load?
>>> >>
>>> >> Gabe
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> > --
>>> > Sent from my mobile device
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>> freeswitch-
>>> > users
>>> > http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Sun Aug 16 16:04:33 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 16 Aug 2009 19:04:33 -0400
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <4A8881E7.8080509@gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>	<4A884F07.3040200@gmail.com>
	<5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>
	<4A8881E7.8080509@gmail.com>
Message-ID: <D3325007-BA3D-4B65-B533-AEC2BF7AB447@jerris.com>

I think sticking with standard WAMP is preferable.  What is the  
advantage to creating yet another installer over the one that we have  
already done and maintained?

Mike

On Aug 16, 2009, at 6:02 PM, Meftah Tayeb wrote:

> hello
> i'm rewriting this executable file in MSI format
> i can use:
> MakeMSI
> WIX (Windows installer XML)
> what you like?
> WIX is open source, but MakeMSI i'm not sur
> for web server, i'm replacing WAMP with a great web server called  
> uniform server
> this server have a conventional Unix Path configuration, for  
> example, web files can by stored in /www/ and apache is in /usr/ 
> local/apache2
> this facilitate the deploiment of applications
> any suggestion?
> thanks
> Meftah Tayeb
> Carlos Talbot wrote:
>>
>> http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe
>>

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From mattdfong at gmail.com  Sun Aug 16 16:35:23 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Sun, 16 Aug 2009 16:35:23 -0700
Subject: [Freeswitch-users] Maxmium Number of Concurrent Sessions on EIDE
	Drive
Message-ID: <4256bf830908161635l4b748a85v8ea5032842a06de4@mail.gmail.com>

I'm interested in doing some testing on the accuracy of mod_vmd (and
mod_amd) but wanted to see if anyone could provide some guidelines on the
maximum number of concurrent sessions I can record audio files to disk with
a typical EIDE drive under 64-bit linux without overloading my system.
Also, since I only need to record the incoming audio, would it be suggested
I use the api command uuid_record or session:record? Is there a way to only
record inbound audio with session:record? Thanks.

--matt
hello hunter corp.
http://www.hellohunter.com
hosted dialer & voice broadcasting
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From mattdfong at gmail.com  Sun Aug 16 21:26:14 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Sun, 16 Aug 2009 21:26:14 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
Message-ID: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>

I tried emailed Eric, seeking advice on this, but his email (the one in the
source code) is bouncing email (invalid user), so thought I would ask here
instead. If anyone has eric's new email address, I'd be interesting in it.

I did some tests with mod_vmd this afternoon, but I'm only finding about 33%
of the voice mail beeps and did have 1 false-positive in my test of 7 voice
mail machines. I've recorded the audio of the session in .wav files that
were both successful and not, as a comparison. I can upload the .wav files
if they would be useful.

mod_vmd works great for voicemails of Skype Users, and kall8.com, but has
issues dealing with mobile phone carriers.

sprint - not successful
tmobile - not successful
verizon - not successful
panasonic home answering machine system - not successful

kall8 - SUCCESS
skype - SUCCESS

I'm wondering if you can recommend a simple fix, like changing some of the
constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c source
file, or if better success requires more complex analysis.  Do you have any
recommendations on how this might be done? Listening to the .wav's
its apparent the beeps are not as loud for the mobile phone carriers as they
are with skype and kall8. Any guidance would be greatly appreciated.

--matt
hello hunter
http://www.hellohunter.com
voice broadcasting & hosted dialer
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From kokoska.rokoska at post.cz  Sun Aug 16 23:08:51 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Mon, 17 Aug 2009 08:08:51 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>
	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>
Message-ID: <4A88F3F3.3080007@post.cz>


Thank you very much, Brian, for your help!

I have just upgraded to curent trunk, but all remains the same - RTPs
from both endpoints arrives to the FreeSWITCH but no RTPs are send back
from FS (using ngrep I can't see even single RTP packet comming from FS).

I think it is due to my misconfiguration, but I have no idea where.

I have trid to blindly modify followed params to (I hope) all possible
values combination (including manual ones :-), but with no luck:

<param name='apply-nat-acl' value='rfc1918.auto'/>
<param name='local-network-acl' value='localnet.auto'/>
<param name='disable-rtp-auto-adjust' value='false'/>
<param name='rtp-autofix-timing' value='false'/>


Best regards,

kokoska.rokoska


Brian West napsal(a):
> Ok update in a few I now recall why we had that && ! 
> sofia_test_pflag(tech_pvt->profile, PFLAG_AUTO_NAT) in there.
> 
> /b
> 
> On Aug 16, 2009, at 2:40 PM, kokoska rokoska wrote:
> 
>> Thank you very much, Brian, for your help!
>>
>> Now are all IPs correct :-) (BTW: In my previous post I mean "c"  
>> line in
>> SDP, not "m" line). Thanks once more!
>>
>> Well, signaling works fine, but I still have troubles with RTPs:
>> I can see with ngrep RTPs going to FreeSWITCH from both legs (a, b)  
>> but
>> there are not outgoing RTPs from FreeSWITCH to any of the call legs.
>> And, of course, caller and callee hear nothing :-)
>>
>> Could you, please, help me to solve this? Thanks.
>>
>> Best regards,
>>
>> kokoska.rokoska
>>
>>
>>
>>
>> Brian West napsal(a):
>>> You have to set the local-network-acl on the profile so it knows when
>>> or not to fix up the VIA
>>> /b
>>>
>>> On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:
>>>
>>>> Hi all,
>>>>
>>>> I have just upgarded my (a little bit old :-) FreeSWITCH to current
>>>> trunk and now i have troubles with IP addresses in SIP & SDP, while
>>>> before upgrade everything work fine...
>>>>
>>>> My FreeSWITCH is behind NAT and all relevant sip profiles have set:
>>>>
>>>> <param name="ext-rtp-ip" value="1.2.3.4"/>
>>>> <param name="ext-sip-ip" value="1.2.3.4"/>
>>>>
>>>> But when I send INVITE thru the profile to the gateway associated  
>>>> with
>>>> it i see local IP in "Via" header and in SDP "m" line too.
>>>> BTW: The "Contact" header is populated with "external" IP (1.2.3.4)
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From kokoska.rokoska at post.cz  Sun Aug 16 23:29:12 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Mon, 17 Aug 2009 08:29:12 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A88F3F3.3080007@post.cz>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>
	<4A88F3F3.3080007@post.cz>
Message-ID: <4A88F8B8.8070003@post.cz>


After some testing I found following:

1. When I try to bridge call between endpoints, there are always no RTPs
from FreeSWITCH (like I wrote before).
2. When I just answer call on FS and play something localy (i.e. music)
everything works fine - FS sends RTPs like expected.

All tested with various endpoints (Linksys, Snom, Nokia, Asterisk :-)
and from public IPs, private IPs at foreign LAN, private IPs at the same
LAN as FreeSWITCH is (yes, I have separate profile for phones in the
same LAN). And I found no difference.

Any hint is really appreciated :-)

Best regards,

kokoska.rokoska


kokoska rokoska napsal(a):
> Thank you very much, Brian, for your help!
> 
> I have just upgraded to curent trunk, but all remains the same - RTPs
> from both endpoints arrives to the FreeSWITCH but no RTPs are send back
> from FS (using ngrep I can't see even single RTP packet comming from FS).
> 
> I think it is due to my misconfiguration, but I have no idea where.
> 
> I have trid to blindly modify followed params to (I hope) all possible
> values combination (including manual ones :-), but with no luck:
> 
> <param name='apply-nat-acl' value='rfc1918.auto'/>
> <param name='local-network-acl' value='localnet.auto'/>
> <param name='disable-rtp-auto-adjust' value='false'/>
> <param name='rtp-autofix-timing' value='false'/>
> 
> 
> Best regards,
> 
> kokoska.rokoska
> 
> 
> Brian West napsal(a):
>> Ok update in a few I now recall why we had that && ! 
>> sofia_test_pflag(tech_pvt->profile, PFLAG_AUTO_NAT) in there.
>>
>> /b
>>
>> On Aug 16, 2009, at 2:40 PM, kokoska rokoska wrote:
>>
>>> Thank you very much, Brian, for your help!
>>>
>>> Now are all IPs correct :-) (BTW: In my previous post I mean "c"  
>>> line in
>>> SDP, not "m" line). Thanks once more!
>>>
>>> Well, signaling works fine, but I still have troubles with RTPs:
>>> I can see with ngrep RTPs going to FreeSWITCH from both legs (a, b)  
>>> but
>>> there are not outgoing RTPs from FreeSWITCH to any of the call legs.
>>> And, of course, caller and callee hear nothing :-)
>>>
>>> Could you, please, help me to solve this? Thanks.
>>>
>>> Best regards,
>>>
>>> kokoska.rokoska
>>>
>>>
>>>
>>>
>>> Brian West napsal(a):
>>>> You have to set the local-network-acl on the profile so it knows when
>>>> or not to fix up the VIA
>>>> /b
>>>>
>>>> On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:
>>>>
>>>>> Hi all,
>>>>>
>>>>> I have just upgarded my (a little bit old :-) FreeSWITCH to current
>>>>> trunk and now i have troubles with IP addresses in SIP & SDP, while
>>>>> before upgrade everything work fine...
>>>>>
>>>>> My FreeSWITCH is behind NAT and all relevant sip profiles have set:
>>>>>
>>>>> <param name="ext-rtp-ip" value="1.2.3.4"/>
>>>>> <param name="ext-sip-ip" value="1.2.3.4"/>
>>>>>
>>>>> But when I send INVITE thru the profile to the gateway associated  
>>>>> with
>>>>> it i see local IP in "Via" header and in SDP "m" line too.
>>>>> BTW: The "Contact" header is populated with "external" IP (1.2.3.4)
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From tzury.by at reguluslabs.com  Mon Aug 17 00:14:55 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 17 Aug 2009 10:14:55 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
Message-ID: <10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>

Brian/Bakko,

Would you please tell me which softphone are you using?
As you know, my own one is not working and when I tried tcp with xlite
(providing transport=tls) I see in wireshark that it is still
transporting it over udp(!)

thanks allot in advance,
Tzury

On Sun, Aug 16, 2009 at 5:11 PM, Brian West <brian at freeswitch.org> wrote:
>
> Works fine here.
>
> /b
>
> On Aug 16, 2009, at 6:25 AM, Tzury Bar Yochay wrote:
>
> > Hi Brian,
> >
> > Just for your information here is a mail I got from a colleague of
> > mine which I consider as an experienced freeswitch integrator
> >
> >> "... TCP used to work, and I had about 20-30 client phones
> >> connecting with it.
> >> About 3 months ago I did an upgrade and TCP no longer worked. ?I
> >> was in panic mode, so I just ended up walking my
> >> clients through reconfiguring their phones to use UDP.
> >>
> >> It is definitely a bug in Freeswitch. ?I can help you work up a
> >> jira bug report if you like.
> >> The bug happened somewhere between svn version 9800ish and 13223.
> >> 13223 is the version I'm running on my systems now, older, but
> >> stable for me..."
> >
> > It would be just great if you can confirm the TCP functionality in the
> > latest release of FreeSwitch
> >
> >
> > /t
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



--
Tzury Bar Yochay

Regulus Labs ltd.
http://reguluslabs.com
+972 52 5133399


From jason at jasonjgw.net  Mon Aug 17 00:34:18 2009
From: jason at jasonjgw.net (Jason White)
Date: Mon, 17 Aug 2009 17:34:18 +1000
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
Message-ID: <20090817073418.GA5443@jdc.jasonjgw.net>

Tzury Bar Yochay <tzury.by at reguluslabs.com> wrote:
> Brian/Bakko,
> 
> Would you please tell me which softphone are you using?
> As you know, my own one is not working and when I tried tcp with xlite
> (providing transport=tls) I see in wireshark that it is still
> transporting it over udp(!)

I've successfully used TLS with FreeSWITCH at both ends (yes, that's with
FreeSWITCH itself as the softphone).

Snom 320 phones are known to work as well if you set up SRV records for the
TLS.

I haven't tried other softphones because, basically, FreeSWITCH is a better
phone than anything else I can find.

To debug this, try wireshark or tshark to find out whether your softphone is
trying to connect over the TLS port at all.



From tzury.by at reguluslabs.com  Mon Aug 17 00:38:47 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 17 Aug 2009 10:38:47 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <20090817073418.GA5443@jdc.jasonjgw.net>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<20090817073418.GA5443@jdc.jasonjgw.net>
Message-ID: <10128ef10908170038l2574727p45285d78347803f5@mail.gmail.com>

> I've successfully used TLS with FreeSWITCH at both ends (yes, that's with
> FreeSWITCH itself as the softphone).

Well, as I said at the beginning of this thread, TLS works fine for me.
The problem is when using TCP (transport=tcp and not transport=tls)


From jason at jasonjgw.net  Mon Aug 17 01:04:49 2009
From: jason at jasonjgw.net (Jason White)
Date: Mon, 17 Aug 2009 18:04:49 +1000
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170038l2574727p45285d78347803f5@mail.gmail.com>
References: <2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<20090817073418.GA5443@jdc.jasonjgw.net>
	<10128ef10908170038l2574727p45285d78347803f5@mail.gmail.com>
Message-ID: <20090817080449.GA9172@jdc.jasonjgw.net>

Tzury Bar Yochay <tzury.by at reguluslabs.com> wrote:
 
> Well, as I said at the beginning of this thread, TLS works fine for me.
> The problem is when using TCP (transport=tcp and not transport=tls)

I'm not sure whether that's supposed to use TLS. I suspect not.



From tzury.by at reguluslabs.com  Mon Aug 17 01:23:11 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 17 Aug 2009 11:23:11 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <20090817080449.GA9172@jdc.jasonjgw.net>
References: <2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<20090817073418.GA5443@jdc.jasonjgw.net>
	<10128ef10908170038l2574727p45285d78347803f5@mail.gmail.com>
	<20090817080449.GA9172@jdc.jasonjgw.net>
Message-ID: <10128ef10908170123o52aa2a06p80094006948e66df@mail.gmail.com>

> I'm not sure whether that's supposed to use TLS. I suspect not.


Jason,

I think I confused you with this TLS/TCP thing.
For the sake of clarification, I am talking about TCP and _not_ about TLS.
That is simply transporting the signaling packets over TCP instead of UDP.
No TLS should be involved at this stage. It is simply communication layer
matter.
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From bruce.mcalister at blueface.ie  Mon Aug 17 01:27:30 2009
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Mon, 17 Aug 2009 09:27:30 +0100
Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 &
 v1.0.4)
In-Reply-To: <4A8280B8.6050308@blueface.ie>
References: <4A8280B8.6050308@blueface.ie>
Message-ID: <4A891472.5060302@blueface.ie>

Hi All,

Shall I log a JIRA for this issue?

Thanks
Bruce

Bruce McAlister wrote:
> Hi All,
> 
> I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and 
> 1.0.4.
> 
> I am running on Solaris 10 Update 5 on x86 hardware (32-bit).
> 
> The build fails with:
> 
> --- snip ---
> make: Fatal error: Command failed for target `all-recursive'
> Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4
> *** Error code 1
> make: Fatal error: Command failed for target `all'
> ---
> 
> Looking back through the build I can see the following error:
> 
> --- snip ---
> creating libfreeswitch.la
> (cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la 
> libfreeswitch.la)
> /usr/bin/cc 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
> -DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch 
> freeswitch-switch.o  ./.libs/libfreeswitch.so 
> -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib/.libs/libexpat.a 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libapr-1.a 
> -lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp 
> -L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt 
> -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl 
> -lcurses -lsocket  -R/opt/freeswitch/lib -R/usr/sfw/lib
> Undefined                       first referenced
>   symbol                             in file
> herror                              ./.libs/libfreeswitch.so
> ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch
> *** Error code 1
> The following command caused the error:
> `if test -z "" ; then echo /bin/bash 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else 
> echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` 
> --tag=CC --mode=link /usr/bin/cc 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
> -DPATH_MAX=2048 -g -v -Xc -xc99=all  -lm -R/opt/freeswitch/lib  -o 
> freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib 
> freeswitch-switch.o   libfreeswitch.la libs/apr/libapr-1.la 
> libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl 
> -lcrypto -lsocket -lnsl -ldl   -lcurses -lsocket
> --- snip ---
> 
> Then a little above this error, there is the following warning that is 
> displayed (I'm not sure if it is related):
> 
> --- snip ---
> *** Warning: Linking the shared library libfreeswitch.la against the
> *** static library libs/libedit/src/.libs/libedit.a is not portable!
> --- snip ---
> 
> My configure line is as follows:
> 
> ---
> ./configure --prefix=/opt/freeswitch
> ---
> 
> I have the complete configure and make output if anyone needs them.
> 
> Any help/pointers would be greatly appreciated.
> 
> Thanks
> Bruce
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 


From jason at jasonjgw.net  Mon Aug 17 01:49:23 2009
From: jason at jasonjgw.net (Jason White)
Date: Mon, 17 Aug 2009 18:49:23 +1000
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170123o52aa2a06p80094006948e66df@mail.gmail.com>
References: <43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<20090817073418.GA5443@jdc.jasonjgw.net>
	<10128ef10908170038l2574727p45285d78347803f5@mail.gmail.com>
	<20090817080449.GA9172@jdc.jasonjgw.net>
	<10128ef10908170123o52aa2a06p80094006948e66df@mail.gmail.com>
Message-ID: <20090817084923.GA15420@jdc.jasonjgw.net>

Tzury Bar Yochay <tzury.by at reguluslabs.com> wrote:
 
> I think I confused you with this TLS/TCP thing.
> For the sake of clarification, I am talking about TCP and _not_ about TLS.
> That is simply transporting the signaling packets over TCP instead of UDP.
> No TLS should be involved at this stage. It is simply communication layer
> matter.

I can confirm that transport=tcp works fine here from one FreeSWITCH to
another. (I'm doing it over IPv6, but that shouldn't make any difference).



From tayeb.meftah at gmail.com  Mon Aug 17 02:29:03 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Mon, 17 Aug 2009 09:29:03 +0000
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
 but I have a little problem
In-Reply-To: <D3325007-BA3D-4B65-B533-AEC2BF7AB447@jerris.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>	<4A884F07.3040200@gmail.com>	<5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>	<4A8881E7.8080509@gmail.com>
	<D3325007-BA3D-4B65-B533-AEC2BF7AB447@jerris.com>
Message-ID: <4A8922DF.1080008@gmail.com>

hi MikeJ,
i prefer creating MSI file that is easy to mintin$
unstid of using inno setup or advanced installer (not free), we can use 
WIX (Windows installer XML) that is a open source one
we can create a customised MSI that fully install mor features, 
including Sounds / MOH/...
and we can edit XML files easyly to let users chouse each module to 
install during setup, for example conferencing, voice mail, Event Socket 
and ...
also the WAMP server is not fully stable
i prefer Uniform Server, that mintin Full Compatibility with Unix like 
platform including for example, perl files that need this path: /usr/...
apache2 is installed in /usr/local/apache2, mySQL: /usr/local/mysql and 
php is in /usr/local/php
and perl is in /usr/bin
php my admin is in /etc/phpmyadmin
and mor featurs, including multiple instance, virtual hosting and full 
management using control pannel
this will help enterprise users that need that in windows unstid of WAMP
     ---------------    
if you don't want to mintin it, jive me a chance to mintin it
      ---------------     
thanks


    Contact Me


    * SIP <sip:voipworld at sip.voipuser.org>
    * INUM <callto:+883510009102769>
    * PSTN <callto:0021348526972>
    * Email <mailto:tayeb.meftah at gmail.com>


Michael Jerris wrote:
> I think sticking with standard WAMP is preferable.  What is the 
> advantage to creating yet another installer over the one that we have 
> already done and maintained?
>
> Mike
>
> On Aug 16, 2009, at 6:02 PM, Meftah Tayeb wrote:
>
>> hello
>> i'm rewriting this executable file in MSI format
>> i can use:
>>
>>     * MakeMSI
>>     * WIX (Windows installer XML)
>>
>> what you like?
>> WIX is open source, but MakeMSI i'm not sur
>> for web server, i'm replacing WAMP with a great web server called 
>> uniform server
>> this server have a conventional Unix Path configuration, for example, 
>> web files can by stored in /www/ and apache is in /usr/local/apache2
>> this facilitate the deploiment of applications
>> any suggestion?
>> thanks
>> Meftah Tayeb
>> Carlos Talbot wrote:
>>> http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe
>>>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus 
> signature database 4340 (20090816) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4340 (20090816) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4340 (20090816) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From enno.egbert at googlemail.com  Mon Aug 17 05:10:33 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Mon, 17 Aug 2009 05:10:33 -0700 (PDT)
Subject: [Freeswitch-users]  SIPGATE Problem
Message-ID: <25005636.post@talk.nabble.com>


Hi,

i have a problem using my freeswitch with a sipgate account. The gateway
entry is o.k.. Freeswitch try a call, sent an invite and sipgate answer with
"403" (forbidden). I guess the string is incorrect.

I modify the "effective_caller_id_number" to accountnumber at sipgate.de but
the trace show the strting from: sip:accountnumber at sipgate.de@my-ip-address

How can i modify the string to sip:accountnumber at sipgate.de without my
ip-address.


-- 
View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25005636.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From asannucci at gmail.com  Mon Aug 17 05:18:24 2009
From: asannucci at gmail.com (bakko)
Date: Mon, 17 Aug 2009 14:18:24 +0200
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com><2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org><10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com><43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org><10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com><45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org><10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com><01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
Message-ID: <C62C40FE5A6547719969BCD0653D1A4E@voztovoice>

Y make my tests with eyebeam.

I thing X-lite dont't support TCP transport.

BR


From chris.chen2004 at gmail.com  Mon Aug 17 05:37:47 2009
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Mon, 17 Aug 2009 08:37:47 -0400
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <25005636.post@talk.nabble.com>
References: <25005636.post@talk.nabble.com>
Message-ID: <507898380908170537r4c68871akc13b2b8bd7129af4@mail.gmail.com>

Hi, could you please check the destination number in your dial string? If it
is the right format, one of the reasons could be that number is not in
service
when you get "403" response from the SIP gateway.

Thanks,

Chris

On Mon, Aug 17, 2009 at 8:10 AM, NOx-WHV <enno.egbert at googlemail.com> wrote:

>
> Hi,
>
> i have a problem using my freeswitch with a sipgate account. The gateway
> entry is o.k.. Freeswitch try a call, sent an invite and sipgate answer
> with
> "403" (forbidden). I guess the string is incorrect.
>
> I modify the "effective_caller_id_number" to accountnumber at sipgate.de but
> the trace show the strting from: sip:accountnumber at sipgate.de<sip%3Aaccountnumber at sipgate.de>
> @my-ip-address
>
> How can i modify the string to sip:accountnumber at sipgate.de<sip%3Aaccountnumber at sipgate.de>without my
> ip-address.
>
>
> --
> View this message in context:
> http://www.nabble.com/SIPGATE-Problem-tp25005636p25005636.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Aug 17 06:09:28 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 08:09:28 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com><2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org><10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com><43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org><10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com><45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org><10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com><01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
Message-ID: <3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>

They make you pay for TCP which is weird since the RFC says TCP is  
required.

/b

On Aug 17, 2009, at 7:18 AM, bakko wrote:

> Y make my tests with eyebeam.
>
> I thing X-lite dont't support TCP transport.
>
> BR



From brian at freeswitch.org  Mon Aug 17 06:11:05 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 08:11:05 -0500
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <25005636.post@talk.nabble.com>
References: <25005636.post@talk.nabble.com>
Message-ID: <672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>

You need to set from-domain on the gateway.  And set the  
effective_caller_id_number to just the  number not the number at host.

/b

On Aug 17, 2009, at 7:10 AM, NOx-WHV wrote:

>
> Hi,
>
> i have a problem using my freeswitch with a sipgate account. The  
> gateway
> entry is o.k.. Freeswitch try a call, sent an invite and sipgate  
> answer with
> "403" (forbidden). I guess the string is incorrect.
>
> I modify the "effective_caller_id_number" to  
> accountnumber at sipgate.de but
> the trace show the strting from: sip:accountnumber at sipgate.de@my-ip-address
>
> How can i modify the string to sip:accountnumber at sipgate.de without my
> ip-address.

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From brian at freeswitch.org  Mon Aug 17 06:11:43 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 08:11:43 -0500
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A88F8B8.8070003@post.cz>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>
	<4A88F3F3.3080007@post.cz> <4A88F8B8.8070003@post.cz>
Message-ID: <797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>

Update, Math refactored the functions that caused some of the problems.

/b

On Aug 17, 2009, at 1:29 AM, kokoska rokoska wrote:

>
> After some testing I found following:
>
> 1. When I try to bridge call between endpoints, there are always no  
> RTPs
> from FreeSWITCH (like I wrote before).
> 2. When I just answer call on FS and play something localy (i.e.  
> music)
> everything works fine - FS sends RTPs like expected.
>
> All tested with various endpoints (Linksys, Snom, Nokia, Asterisk :-)
> and from public IPs, private IPs at foreign LAN, private IPs at the  
> same
> LAN as FreeSWITCH is (yes, I have separate profile for phones in the
> same LAN). And I found no difference.
>
> Any hint is really appreciated :-)
>
> Best regards,
>
> kokoska.rokoska



From enno.egbert at googlemail.com  Mon Aug 17 06:36:17 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Mon, 17 Aug 2009 06:36:17 -0700 (PDT)
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
References: <25005636.post@talk.nabble.com>
	<672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
Message-ID: <25006858.post@talk.nabble.com>


Hi,

i have just taken some pictures. 

http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG 
http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG 
http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG 

the third picture is taken by a softphone that works. 

To Brian: 
If i set the effective_caller_id_name to 2395805 without @sipgate.de it?s
the same problem, because the freeswitch set in the from fielt:
2395805 at 139.13.37.160. And i can?t find the right parameter to change
"139.13.37.160" to "sipgate.de"

Which is the right parameter in the dialplan?

<action application=.....?????>

Thanks for your help





Brian West-3 wrote:
> 
> You need to set from-domain on the gateway.  And set the  
> effective_caller_id_number to just the  number not the number at host.
> 
> /b
> 
> On Aug 17, 2009, at 7:10 AM, NOx-WHV wrote:
> 
>>
>> Hi,
>>
>> i have a problem using my freeswitch with a sipgate account. The  
>> gateway
>> entry is o.k.. Freeswitch try a call, sent an invite and sipgate  
>> answer with
>> "403" (forbidden). I guess the string is incorrect.
>>
>> I modify the "effective_caller_id_number" to  
>> accountnumber at sipgate.de but
>> the trace show the strting from:
>> sip:accountnumber at sipgate.de@my-ip-address
>>
>> How can i modify the string to sip:accountnumber at sipgate.de without my
>> ip-address.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25006858.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Mon Aug 17 06:52:46 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 08:52:46 -0500
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <25006858.post@talk.nabble.com>
References: <25005636.post@talk.nabble.com>
	<672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
	<25006858.post@talk.nabble.com>
Message-ID: <11D5AA56-1609-4D2C-B5D1-8AAF4A575309@freeswitch.org>

Well if you pay attention I told you in the last email... set the  
param from-domain on the gateway to sipgate.de

/b

On Aug 17, 2009, at 8:36 AM, NOx-WHV wrote:

>
> Hi,
>
> i have just taken some pictures.
>
> http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
> http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG
> http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG
>
> the third picture is taken by a softphone that works.
>
> To Brian:
> If i set the effective_caller_id_name to 2395805 without @sipgate.de  
> it?s
> the same problem, because the freeswitch set in the from fielt:
> 2395805 at 139.13.37.160. And i can?t find the right parameter to change
> "139.13.37.160" to "sipgate.de"
>
> Which is the right parameter in the dialplan?
>
> <action application=.....?????>
>
> Thanks for your help
>

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From tzury.by at reguluslabs.com  Mon Aug 17 06:53:30 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 17 Aug 2009 16:53:30 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
Message-ID: <10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>

> They make you pay for TCP which is weird since the RFC says TCP is
> required.
>
> /b

my conclusion is then:
we are living in a sad world where there is no a production ready,
cross platform, fully compliant, decent and open source sip client


From brian at freeswitch.org  Mon Aug 17 06:57:48 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 08:57:48 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
Message-ID: <CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>

FreeSWITCH works very well as a client :P

/b

On Aug 17, 2009, at 8:53 AM, Tzury Bar Yochay wrote:

> my conclusion is then:
> we are living in a sad world where there is no a production ready,
> cross platform, fully compliant, decent and open source sip client



From brian at freeswitch.org  Mon Aug 17 06:57:48 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 08:57:48 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
Message-ID: <CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>

FreeSWITCH works very well as a client :P

/b

On Aug 17, 2009, at 8:53 AM, Tzury Bar Yochay wrote:

> my conclusion is then:
> we are living in a sad world where there is no a production ready,
> cross platform, fully compliant, decent and open source sip client



From tzury.by at reguluslabs.com  Mon Aug 17 07:06:49 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 17 Aug 2009 17:06:49 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
	<CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>
Message-ID: <10128ef10908170706p414098faq5d9d8024ba0cb3ef@mail.gmail.com>

> FreeSWITCH works very well as a client :P
I am currently porting it into iPhone and Symbian, I am almost done ;-)

anyway, seriously now, can one point to a wiki page about this?
How do I do that?
I would need 3 server instances to place a call, right?


From kokoska.rokoska at post.cz  Mon Aug 17 07:09:25 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Mon, 17 Aug 2009 16:09:25 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>	<4A88F3F3.3080007@post.cz>
	<4A88F8B8.8070003@post.cz>
	<797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>
Message-ID: <4A896495.5080009@post.cz>




Brian West napsal(a):
> Update, Math refactored the functions that caused some of the problems.
> 
> /b
> 

Still no luck - no RTPs from FreeSWITCH.
For sure I'll make fresh svn checkout in a minute and let you know if it
helps...

Thank you very much for your help!

Best regards,

kokoska.rokoska



> On Aug 17, 2009, at 1:29 AM, kokoska rokoska wrote:
> 
>> After some testing I found following:
>>
>> 1. When I try to bridge call between endpoints, there are always no  
>> RTPs
>> from FreeSWITCH (like I wrote before).
>> 2. When I just answer call on FS and play something localy (i.e.  
>> music)
>> everything works fine - FS sends RTPs like expected.
>>
>> All tested with various endpoints (Linksys, Snom, Nokia, Asterisk :-)
>> and from public IPs, private IPs at foreign LAN, private IPs at the  
>> same
>> LAN as FreeSWITCH is (yes, I have separate profile for phones in the
>> same LAN). And I found no difference.
>>
>> Any hint is really appreciated :-)
>>
>> Best regards,
>>
>> kokoska.rokoska
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From brian at freeswitch.org  Mon Aug 17 07:10:39 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 09:10:39 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170706p414098faq5d9d8024ba0cb3ef@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
	<CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>
	<10128ef10908170706p414098faq5d9d8024ba0cb3ef@mail.gmail.com>
Message-ID: <9B65BFD4-C93F-4E26-82F1-AF8D687DCF52@freeswitch.org>

Well if you append ;transport=tcp on the bridge lines it will use TCP .

/b

On Aug 17, 2009, at 9:06 AM, Tzury Bar Yochay wrote:

>> FreeSWITCH works very well as a client :P
> I am currently porting it into iPhone and Symbian, I am almost  
> done ;-)
>
> anyway, seriously now, can one point to a wiki page about this?
> How do I do that?
> I would need 3 server instances to place a call, right?



From brian at freeswitch.org  Mon Aug 17 07:11:01 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 09:11:01 -0500
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A896495.5080009@post.cz>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>	<4A88F3F3.3080007@post.cz>
	<4A88F8B8.8070003@post.cz>
	<797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>
	<4A896495.5080009@post.cz>
Message-ID: <22F6C640-1662-4F13-B25C-F816DC9F40E7@freeswitch.org>

if all else fails get me access to the machine please.

/b

On Aug 17, 2009, at 9:09 AM, kokoska rokoska wrote:

> Still no luck - no RTPs from FreeSWITCH.
> For sure I'll make fresh svn checkout in a minute and let you know  
> if it
> helps...
>
> Thank you very much for your help!
>
> Best regards,
>
> kokoska.rokoska



From mike at jerris.com  Mon Aug 17 08:24:05 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 17 Aug 2009 11:24:05 -0400
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <4A8922DF.1080008@gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>	<4A884F07.3040200@gmail.com>	<5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>	<4A8881E7.8080509@gmail.com>
	<D3325007-BA3D-4B65-B533-AEC2BF7AB447@jerris.com>
	<4A8922DF.1080008@gmail.com>
Message-ID: <8181A0DB-67EE-464C-9A69-30BA0719C24C@jerris.com>


On Aug 17, 2009, at 5:29 AM, Meftah Tayeb wrote:

> hi MikeJ,
> i prefer creating MSI file that is easy to mintin$

How is this useful vs. something that is already maintained?

> unstid of using inno setup or advanced installer (not free), we can  
> use WIX (Windows installer XML) that is a open source one

All the tools being used are free, however all are not open source.

> we can create a customised MSI that fully install mor features,  
> including Sounds / MOH/...

This installer already does all that and more.

> and we can edit XML files easyly to let users chouse each module to  
> install during setup, for example conferencing, voice mail, Event  
> Socket and ...
> also the WAMP server is not fully stable
> i prefer Uniform Server, that mintin Full Compatibility with Unix  
> like platform including for example, perl files that need this  
> path: /usr/...
> apache2 is installed in /usr/local/apache2, mySQL: /usr/local/mysql  
> and php is in /usr/local/php
> and perl is in /usr/bin
> php my admin is in /etc/phpmyadmin
> and mor featurs, including multiple instance, virtual hosting and  
> full management using control pannel
> this will help enterprise users that need that in windows unstid of  
> WAMP
>      ---------------
> if you don't want to mintin it, jive me a chance to mintin it
>       ---------------
> thanks

I just don't want to split efforts here, every different way we do  
this is yet another way we have to support.  I see no compelling  
argument here other than personal preference.  If I am missing  
anything someone please chime in and correct me.

Mike

> Contact Me
>
>
> SIP
> INUM
> PSTN
> Email
>
> Michael Jerris wrote:
>>
>> I think sticking with standard WAMP is preferable.  What is the  
>> advantage to creating yet another installer over the one that we  
>> have already done and maintained?
>>
>> Mike
>>
>> On Aug 16, 2009, at 6:02 PM, Meftah Tayeb wrote:
>>
>>> hello
>>> i'm rewriting this executable file in MSI format
>>> i can use:
>>> MakeMSI
>>> WIX (Windows installer XML)
>>> what you like?
>>> WIX is open source, but MakeMSI i'm not sur
>>> for web server, i'm replacing WAMP with a great web server called  
>>> uniform server
>>> this server have a conventional Unix Path configuration, for  
>>> example, web files can by stored in /www/ and apache is in /usr/ 
>>> local/apache2
>>> this facilitate the deploiment of applications
>>> any suggestion?
>>> thanks
>>> Meftah Tayeb
>>> Carlos Talbot wrote:
>>>>
>>>> http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe
>>>>
>>
>>

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From msc at freeswitch.org  Mon Aug 17 08:59:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Aug 2009 08:59:31 -0700
Subject: [Freeswitch-users] how to set different action for different
	cause code
In-Reply-To: <a44688a0908160424n3569cef1vad8fcdc68cbfb96e@mail.gmail.com>
References: <a44688a0908160424n3569cef1vad8fcdc68cbfb96e@mail.gmail.com>
Message-ID: <87f2f3b90908170859p52259449q913b2ddab224709f@mail.gmail.com>

On Sun, Aug 16, 2009 at 4:24 AM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hello,
>
> I find hangup_hook, but I would like to define different actions for
> different hangup codes.  Is there anyway to do that?
>
>

I can think of at least two ways you could do this: one that uses only the
dialplan and one that uses a script. If you don't mind using a scripting
language then you can make it very clean:

<action application="set" data="api_hangup_hook=luarun chancleanup.lua
${hangup_cause}"/>

Then have your Lua script handle all the if-then-else or case stuff.

Question: are you trying to transfer the a-leg to some other destination if
the b-leg hangup is a specific cause, or are you just doing some external
cleanup stuff? Just curious...

-MC


> Woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kokoska.rokoska at post.cz  Mon Aug 17 09:46:55 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Mon, 17 Aug 2009 18:46:55 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <22F6C640-1662-4F13-B25C-F816DC9F40E7@freeswitch.org>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>	<4A88F3F3.3080007@post.cz>	<4A88F8B8.8070003@post.cz>	<797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>	<4A896495.5080009@post.cz>
	<22F6C640-1662-4F13-B25C-F816DC9F40E7@freeswitch.org>
Message-ID: <4A89897F.9060607@post.cz>




Brian West napsal(a):
> if all else fails get me access to the machine please.
> 
> /b
> 

I'm sorry, but even after fresh svn checkout all goes wrong :-)

If you be so kind to look at that machine, I'll be very glad.
I still think it will be some trivial, stupid, misconfiguration, but
can't find what I did wrong. Especially if all worked fine before FS
update...
What public ssh key should I install (shinzon, freeswitch) or should I
get another one?

Thanks once more for your help!

Best regards,

kokoska.rokoska


> On Aug 17, 2009, at 9:09 AM, kokoska rokoska wrote:
> 
>> Still no luck - no RTPs from FreeSWITCH.
>> For sure I'll make fresh svn checkout in a minute and let you know  
>> if it
>> helps...
>>
>> Thank you very much for your help!
>>
>> Best regards,
>>
>> kokoska.rokoska
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From brian at freeswitch.org  Mon Aug 17 09:54:41 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 11:54:41 -0500
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A89897F.9060607@post.cz>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>	<4A88F3F3.3080007@post.cz>	<4A88F8B8.8070003@post.cz>	<797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>	<4A896495.5080009@post.cz>
	<22F6C640-1662-4F13-B25C-F816DC9F40E7@freeswitch.org>
	<4A89897F.9060607@post.cz>
Message-ID: <45DDEB13-3FB7-4AAC-A3A5-48272360C0F5@freeswitch.org>

shinzon.pub and get on IRC and msg bkw_ and i'll take a look.

/b

On Aug 17, 2009, at 11:46 AM, kokoska rokoska wrote:

>
> I'm sorry, but even after fresh svn checkout all goes wrong :-)
>
> If you be so kind to look at that machine, I'll be very glad.
> I still think it will be some trivial, stupid, misconfiguration, but
> can't find what I did wrong. Especially if all worked fine before FS
> update...
> What public ssh key should I install (shinzon, freeswitch) or should I
> get another one?
>
> Thanks once more for your help!
>
> Best regards,
>
> kokoska.rokoska



From kokoska.rokoska at post.cz  Mon Aug 17 10:11:03 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Mon, 17 Aug 2009 19:11:03 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <45DDEB13-3FB7-4AAC-A3A5-48272360C0F5@freeswitch.org>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>	<4A88F3F3.3080007@post.cz>	<4A88F8B8.8070003@post.cz>	<797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>	<4A896495.5080009@post.cz>	<22F6C640-1662-4F13-B25C-F816DC9F40E7@freeswitch.org>	<4A89897F.9060607@post.cz>
	<45DDEB13-3FB7-4AAC-A3A5-48272360C0F5@freeswitch.org>
Message-ID: <4A898F27.40806@post.cz>




Brian West napsal(a):
> shinzon.pub and get on IRC and msg bkw_ and i'll take a look.
> 
> /b
> 

OK, I do it.
But wait few minutes, please:

1. I should asleep my chidren :-)
2. Before a while I discovered that if I disable IPv6 networking in my
CentOS, the RTPs from FreeSWITCH works great (FS stays runnig untouched)
=> it looks like it may be network layer/config (iptables, ip6tables,
roueting etc.) error not related to FS.

I'll be back in an hour or so and let you know what is current status.

Many thanks, Brian, for your valuable help.

Best regards,

kokoska.rokoska


> On Aug 17, 2009, at 11:46 AM, kokoska rokoska wrote:
> 
>> I'm sorry, but even after fresh svn checkout all goes wrong :-)
>>
>> If you be so kind to look at that machine, I'll be very glad.
>> I still think it will be some trivial, stupid, misconfiguration, but
>> can't find what I did wrong. Especially if all worked fine before FS
>> update...
>> What public ssh key should I install (shinzon, freeswitch) or should I
>> get another one?
>>
>> Thanks once more for your help!
>>
>> Best regards,
>>
>> kokoska.rokoska
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From kokoska.rokoska at post.cz  Mon Aug 17 12:15:43 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Mon, 17 Aug 2009 21:15:43 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A898F27.40806@post.cz>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>	<4A88F3F3.3080007@post.cz>	<4A88F8B8.8070003@post.cz>	<797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>	<4A896495.5080009@post.cz>	<22F6C640-1662-4F13-B25C-F816DC9F40E7@freeswitch.org>	<4A89897F.9060607@post.cz>	<45DDEB13-3FB7-4AAC-A3A5-48272360C0F5@freeswitch.org>
	<4A898F27.40806@post.cz>
Message-ID: <4A89AC5F.3020900@post.cz>


Let me apologize to waste your time, Brian (and others too).

I'm still not 100% sure where problem is, but I'm nearly sure it is not
related to FreeSWITCH.

I found similar problems with SMB - with IPv6 active some clients (even
with only IPv4 stack) can't connect to server. Disabling IPv6 solves the
issue => it has to be network problem...

Thank you very very much for your help!

Best regards,

kokoska.rokoska



kokoska rokoska napsal(a):
> 
> 
> Brian West napsal(a):
>> shinzon.pub and get on IRC and msg bkw_ and i'll take a look.
>>
>> /b
>>
> 
> OK, I do it.
> But wait few minutes, please:
> 
> 1. I should asleep my chidren :-)
> 2. Before a while I discovered that if I disable IPv6 networking in my
> CentOS, the RTPs from FreeSWITCH works great (FS stays runnig untouched)
> => it looks like it may be network layer/config (iptables, ip6tables,
> roueting etc.) error not related to FS.
> 
> I'll be back in an hour or so and let you know what is current status.
> 
> Many thanks, Brian, for your valuable help.
> 
> Best regards,
> 
> kokoska.rokoska
> 
> 
>> On Aug 17, 2009, at 11:46 AM, kokoska rokoska wrote:
>>
>>> I'm sorry, but even after fresh svn checkout all goes wrong :-)
>>>
>>> If you be so kind to look at that machine, I'll be very glad.
>>> I still think it will be some trivial, stupid, misconfiguration, but
>>> can't find what I did wrong. Especially if all worked fine before FS
>>> update...
>>> What public ssh key should I install (shinzon, freeswitch) or should I
>>> get another one?
>>>
>>> Thanks once more for your help!
>>>
>>> Best regards,
>>>
>>> kokoska.rokoska
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 



From testa at voicetechnology.com.br  Mon Aug 17 13:04:07 2009
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Mon, 17 Aug 2009 17:04:07 -0300
Subject: [Freeswitch-users] JAVA ESL
In-Reply-To: <881036.91387.qm@web35603.mail.mud.yahoo.com>
References: <191c3a030907301800n221cab6cmfbc27b97e292bd85@mail.gmail.com>
	<881036.91387.qm@web35603.mail.mud.yahoo.com>
Message-ID: <9cb0e15e0908171304q1a47e358s9cbaac00fc284cfb@mail.gmail.com>

I found this same issue on my machine.
If you could compile a esl_wrap.o then you have to generate a libesl.so with
a cmd like this:
g++ -shared esl_wrap.o -o libesl.so
Then in your code, do something like this:
/* Test.java */
import org.freeswitch.esl.*;

class Test
{
  public static void main(String[] args)
  {
    System.loadLibrary("esl");
    System.out.println("hello");
  }
}



On Thu, Jul 30, 2009 at 10:31 PM, Jean-Marc Hyppolite <hyppolite72 at yahoo.com
> wrote:

> Thank you Anthony.
>
> --- On *Thu, 7/30/09, Anthony Minessale <anthony.minessale at gmail.com>*wrote:
>
>
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] JAVA ESL
> To: freeswitch-users at lists.freeswitch.org
> Received: Thursday, July 30, 2009, 9:00 PM
>
>
> it might be a build issue, I was not exactly sure how to build it etc.
> so it may need some help from a java expert
>
> I wrote all of that with swig and never was able to test it.
>
>
> On Thu, Jul 30, 2009 at 7:40 PM, Jean-Marc Hyppolite <
> hyppolite72 at yahoo.com<http://ca.mc356.mail.yahoo.com/mc/compose?to=hyppolite72 at yahoo.com>
> > wrote:
>
>>   Hello,
>>
>> I built libesl and JAVA mod. (make and make javamod). But when I try to
>> run a JAVA script with the following code
>>
>> ESLconnection connection = new ESLconnection("127.0.0.1", "9000", "");
>> ESLevent events = connection.getInfo();
>> System.out.println(events.toString());
>>
>> I get the following error message:
>>
>> Exception in thread "main" java.lang.UnsatisfiedLinkError:
>> /usr/lib/libesl.so: /usr/lib/libesl.so: undefined symbol:
>> __gxx_personality_v0
>>         at java.lang.ClassLoader$NativeLibrary.load(Native Method)
>>         at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778)
>>         at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703)
>>         at java.lang.Runtime.loadLibrary0(Runtime.java:823)
>>         at java.lang.System.loadLibrary(System.java:1030)
>>         at ivr.IVRServer.<clinit>(IVRServer.java:18)
>>
>> Any help would be appreciated.
>>
>> Thanks.
>>
>>
>> ------------------------------
>> The new Internet Explorer? 8 - Faster, safer, easier. Optimized for Yahoo!
>> *Get it Now for Free!* <http://downloads.yahoo.com/ca/internetexplorer/>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com<http://ca.mc356.mail.yahoo.com/mc/compose?to=MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<http://ca.mc356.mail.yahoo.com/mc/compose?to=PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>    ------------------------------
>
> *Yahoo! Canada Toolbar :* Search from anywhere on the web and bookmark
> your favourite sites. Download it now! <http://ca.toolbar.yahoo.com/>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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From mike at jerris.com  Mon Aug 17 13:18:20 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 17 Aug 2009 16:18:20 -0400
Subject: [Freeswitch-users] JAVA ESL
In-Reply-To: <9cb0e15e0908171304q1a47e358s9cbaac00fc284cfb@mail.gmail.com>
References: <191c3a030907301800n221cab6cmfbc27b97e292bd85@mail.gmail.com>
	<881036.91387.qm@web35603.mail.mud.yahoo.com>
	<9cb0e15e0908171304q1a47e358s9cbaac00fc284cfb@mail.gmail.com>
Message-ID: <F0B83E91-DE86-486B-82AB-C3BF9DF3783D@jerris.com>

can someone post a patch to that makefile to jira.freeswitch.org please.

Mike

On Aug 17, 2009, at 4:04 PM, Fernando Testa wrote:

> I found this same issue on my machine.
> If you could compile a esl_wrap.o then you have to generate a  
> libesl.so with a cmd like this:
> g++ -shared esl_wrap.o -o libesl.so
> Then in your code, do something like this:
> /* Test.java */
> import org.freeswitch.esl.*;
>
> class Test
> {
>   public static void main(String[] args)
>   {
>     System.loadLibrary("esl");
>     System.out.println("hello");
>   }
> }
>
>
>
> On Thu, Jul 30, 2009 at 10:31 PM, Jean-Marc Hyppolite <hyppolite72 at yahoo.com 
> > wrote:
> Thank you Anthony.
>
> --- On Thu, 7/30/09, Anthony Minessale <anthony.minessale at gmail.com>  
> wrote:
>
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] JAVA ESL
> To: freeswitch-users at lists.freeswitch.org
> Received: Thursday, July 30, 2009, 9:00 PM
>
>
> it might be a build issue, I was not exactly sure how to build it etc.
> so it may need some help from a java expert
>
> I wrote all of that with swig and never was able to test it.
>
>
> On Thu, Jul 30, 2009 at 7:40 PM, Jean-Marc Hyppolite <hyppolite72 at yahoo.com 
> > wrote:
> Hello,
>
> I built libesl and JAVA mod. (make and make javamod). But when I try  
> to run a JAVA script with the following code
>
> ESLconnection connection = new ESLconnection("127.0.0.1", "9000", "");
> ESLevent events = connection.getInfo();
> System.out.println(events.toString());
>
> I get the following error message:
>
> Exception in thread "main" java.lang.UnsatisfiedLinkError: /usr/lib/ 
> libesl.so: /usr/lib/libesl.so: undefined symbol: __gxx_personality_v0
>         at java.lang.ClassLoader$NativeLibrary.load(Native Method)
>         at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778)
>         at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703)
>         at java.lang.Runtime.loadLibrary0(Runtime.java:823)
>         at java.lang.System.loadLibrary(System.java:1030)
>         at ivr.IVRServer.<clinit>(IVRServer.java:18)
>
> Any help would be appreciated.
>
> Thanks.
>
>
> The new Internet Explorer? 8 - Faster, safer, easier. Optimized for  
> Yahoo! Get it Now for Free!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> -----Inline Attachment Follows-----
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>  Yahoo! Canada Toolbar : Search from anywhere on the web and  
> bookmark your favourite sites. Download it now!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Fernando Gregianin Testa
> Voice Technology Ltda
> +55 11 35882166
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From larclap at yahoo.com  Mon Aug 17 13:18:44 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Mon, 17 Aug 2009 13:18:44 -0700
Subject: [Freeswitch-users] Eavesdrop getting killed after being answered
Message-ID: <005c01ca1f77$eb8e6ef0$c2ab4cd0$@com>

I used to be able to dial 88+extension to eavesdrop, but now it is killed
right after the call is answered by the extension. Can anyone tell me what I
have done wrong? 

 

I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7
10:39:21 EDT 2009 i686 i686 i386 GNU/Linux.

 

http://pastebin.freeswitch.org/10025

 

Thanks Lars

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From fax at virgintechnologies.com  Mon Aug 17 13:30:51 2009
From: fax at virgintechnologies.com (Justin Miller)
Date: Mon, 17 Aug 2009 20:30:51 +0000
Subject: [Freeswitch-users] G729 transcoding workaround
Message-ID: <W4098424883208801250541051@webmail30>

To overcome the G729 transcoding issue with voicemail, I'm using an Audiocodes Mediant 1000 for transcoding. Our SIP trunk provider and all of our phones use G729 exclusively. 
When a call needs to go to voicemail, the call is bridged to the M1000, which transcodes to G711, and returns the call to Freeswitch on another port (5090). This seems to be working well. 

I'm now working on getting the IVR working so that I can start using the PBX functionality of Freeswitch for our office. When a call needs to hit the IVR, it's bridged to the M1000 the same as voicemail. I hear the custom menu I've created, and I can choose options successfully, except that the call is now operating in G711. When I bridge to a phone, I get the transcoding error. One option would be to transcode through the M1000 yet again, but this would take up 2 more DSP sessions in the M1000, and I would be running a total of 6 call legs in Freeswitch for this one call. 

Is there a way to end the transcoded call legs, and bridge to the phone from the original call leg? This would free up the M1000, and just seems like a better way to do things. 

Thank you 
Justin

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From eric.des.courtis at gmail.com  Mon Aug 17 10:29:30 2009
From: eric.des.courtis at gmail.com (Eric des Courtis)
Date: Mon, 17 Aug 2009 13:29:30 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
Message-ID: <539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>

Matt,

You must first capture the audio beeps and verify that they are sine
waves. If not, simply tweaking the algorithm will not give you better
results.

It might be possible to use FFT and I would be happy to help you
implement such a solution but keep in mind FFT is very very demanding
on the hardware. Ideally what you want to find out is what functions
was use to generate the beep in the first place so that it can be
detected. Is it two sines waves like in DTMF? Or something more
complex?

Anyway my email is eric.des.courtis at benbria.ca.

Cheers.

Eric des Courtis

On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong at gmail.com> wrote:
> I tried emailed Eric, seeking advice on this, but his email (the one in the
> source code) is bouncing email (invalid user), so thought I would ask here
> instead. If anyone has eric's new email address, I'd be interesting in it.
>
> I did some tests with mod_vmd this afternoon, but I'm?only finding about 33%
> of the voice mail beeps and did have 1 false-positive in my test of 7?voice
> mail?machines. I've recorded the audio of the session in .wav files that
> were both successful and not, as a comparison. I can upload the .wav files
> if they would be useful.
> mod_vmd works great for voicemails of Skype Users, and?kall8.com, but has
> issues dealing with mobile phone carriers.
> sprint - not successful
> tmobile - not successful
> verizon - not successful
> panasonic home answering machine system - not successful
> kall8 - SUCCESS
> skype - SUCCESS
> I'm wondering if you can recommend a simple fix, like changing some of the
> constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c source
> file, or if better success requires more complex analysis. ?Do you have any
> recommendations on how this might be done??Listening to the .wav's
> its?apparent?the beeps are not as loud for the mobile phone carriers as they
> are with skype and kall8. Any guidance would be greatly appreciated.
> --matt
> hello hunter
> http://www.hellohunter.com
> voice broadcasting & hosted dialer
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From eric.des.courtis at gmail.com  Mon Aug 17 14:15:57 2009
From: eric.des.courtis at gmail.com (Eric des Courtis)
Date: Mon, 17 Aug 2009 17:15:57 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
Message-ID: <539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>

Matt,

Okay the good news is vmd should be able to handle these cases. The
bad news is for whatever reason they are not getting detected at the
moment.

vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
MAX_FREQ to 1450 and play with MIN_AMPL if that still doesn't help.

The following seem to use the same beep:

vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise

You can try to play with these values:

POINTS 32
VALID 22
MAX_CHIRP 22

If that doesn't work let me know I will try to improve the algorithm
to detect the providers.

Cheers!

Eric des Courtis



On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong at gmail.com> wrote:
> Hi Eric,
> Thanks for the response. I had tried emailing you @brenbria.com and the
> email had bounced, thanks for responding to my mail.
> If you'd be interested I .zipped up my sample voicemail beeps
> at?http://bandcon.hellohunter.com/vmd_wav.zip
> I'm relatively new to telephony, but can you point me in the right direction
> for figuring out if the beeps are sinewaves. About as far as I've come with
> audio is being able to open the .wav files in audacity. Any website
> ?recommendations I can read? Thanks so much.
> --matt
>
> On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
> <eric.des.courtis at gmail.com> wrote:
>>
>> Matt,
>>
>> You must first capture the audio beeps and verify that they are sine
>> waves. If not, simply tweaking the algorithm will not give you better
>> results.
>>
>> It might be possible to use FFT and I would be happy to help you
>> implement such a solution but keep in mind FFT is very very demanding
>> on the hardware. Ideally what you want to find out is what functions
>> was use to generate the beep in the first place so that it can be
>> detected. Is it two sines waves like in DTMF? Or something more
>> complex?
>>
>> Anyway my email is eric.des.courtis at benbria.ca.
>>
>> Cheers.
>>
>> Eric des Courtis
>>
>> On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong at gmail.com> wrote:
>> > I tried emailed Eric, seeking advice on this, but his email (the one in
>> > the
>> > source code) is bouncing email (invalid user), so thought I would ask
>> > here
>> > instead. If anyone has eric's new email address, I'd be interesting in
>> > it.
>> >
>> > I did some tests with mod_vmd this afternoon, but I'm?only finding about
>> > 33%
>> > of the voice mail beeps and did have 1 false-positive in my test of
>> > 7?voice
>> > mail?machines. I've recorded the audio of the session in .wav files that
>> > were both successful and not, as a comparison. I can upload the .wav
>> > files
>> > if they would be useful.
>> > mod_vmd works great for voicemails of Skype Users, and?kall8.com, but
>> > has
>> > issues dealing with mobile phone carriers.
>> > sprint - not successful
>> > tmobile - not successful
>> > verizon - not successful
>> > panasonic home answering machine system - not successful
>> > kall8 - SUCCESS
>> > skype - SUCCESS
>> > I'm wondering if you can recommend a simple fix, like changing some of
>> > the
>> > constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c source
>> > file, or if better success requires more complex analysis. ?Do you have
>> > any
>> > recommendations on how this might be done??Listening to the .wav's
>> > its?apparent?the beeps are not as loud for the mobile phone carriers as
>> > they
>> > are with skype and kall8. Any guidance would be greatly appreciated.
>> > --matt
>> > hello hunter
>> > http://www.hellohunter.com
>> > voice broadcasting & hosted dialer
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>
>


From msc at freeswitch.org  Mon Aug 17 14:22:07 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Aug 2009 14:22:07 -0700
Subject: [Freeswitch-users] Eavesdrop getting killed after being answered
In-Reply-To: <005c01ca1f77$eb8e6ef0$c2ab4cd0$@com>
References: <005c01ca1f77$eb8e6ef0$c2ab4cd0$@com>
Message-ID: <87f2f3b90908171422x1990be2fm9c3d01a7f36d58e5@mail.gmail.com>

On Mon, Aug 17, 2009 at 1:18 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  I used to be able to dial 88+extension to eavesdrop, but now it is killed
> right after the call is answered by the extension. Can anyone tell me what I
> have done wrong?
>
>
>
> I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7
> 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux.
>
>
>
> http://pastebin.freeswitch.org/10025
>
>
>
Hmm, something isn't right. Here's a snippet from a successful eavesdrop:
http://pastebin.freeswitch.org/10027

As you can see, your log shows no media bug operation like mine does.

Can you do another "make current" just to make 100% certain that your build
environment isn't corrupted?
-MC


> Thanks Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From testa at voicetechnology.com.br  Mon Aug 17 14:35:41 2009
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Mon, 17 Aug 2009 18:35:41 -0300
Subject: [Freeswitch-users] JAVA ESL
In-Reply-To: <F0B83E91-DE86-486B-82AB-C3BF9DF3783D@jerris.com>
References: <191c3a030907301800n221cab6cmfbc27b97e292bd85@mail.gmail.com>
	<881036.91387.qm@web35603.mail.mud.yahoo.com>
	<9cb0e15e0908171304q1a47e358s9cbaac00fc284cfb@mail.gmail.com>
	<F0B83E91-DE86-486B-82AB-C3BF9DF3783D@jerris.com>
Message-ID: <9cb0e15e0908171435o54c7379fye4a3016d9b7c59b6@mail.gmail.com>

Done!
Check out patch at http://jira.freeswitch.org/browse/FSBUILD-185

Testa

On Mon, Aug 17, 2009 at 5:18 PM, Michael Jerris <mike at jerris.com> wrote:

> can someone post a patch to that makefile to jira.freeswitch.org please.
> Mike
>
> On Aug 17, 2009, at 4:04 PM, Fernando Testa wrote:
>
> I found this same issue on my machine.
> If you could compile a esl_wrap.o then you have to generate a libesl.so
> with a cmd like this:
> g++ -shared esl_wrap.o -o libesl.so
> Then in your code, do something like this:
> /* Test.java */
> import org.freeswitch.esl.*;
>
> class Test
> {
>   public static void main(String[] args)
>   {
>     System.loadLibrary("esl");
>     System.out.println("hello");
>   }
> }
>
>
>
> On Thu, Jul 30, 2009 at 10:31 PM, Jean-Marc Hyppolite <
> hyppolite72 at yahoo.com> wrote:
>
>>  Thank you Anthony.
>>
>> --- On *Thu, 7/30/09, Anthony Minessale <anthony.minessale at gmail.com>*wrote:
>>
>>
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Subject: Re: [Freeswitch-users] JAVA ESL
>> To: freeswitch-users at lists.freeswitch.org
>> Received: Thursday, July 30, 2009, 9:00 PM
>>
>>
>> it might be a build issue, I was not exactly sure how to build it etc.
>> so it may need some help from a java expert
>>
>> I wrote all of that with swig and never was able to test it.
>>
>>
>> On Thu, Jul 30, 2009 at 7:40 PM, Jean-Marc Hyppolite <
>> hyppolite72 at yahoo.com<http://ca.mc356.mail.yahoo.com/mc/compose?to=hyppolite72 at yahoo.com>
>> > wrote:
>>
>>>   Hello,
>>>
>>> I built libesl and JAVA mod. (make and make javamod). But when I try to
>>> run a JAVA script with the following code
>>>
>>> ESLconnection connection = new ESLconnection("127.0.0.1", "9000", "");
>>> ESLevent events = connection.getInfo();
>>> System.out.println(events.toString());
>>>
>>> I get the following error message:
>>>
>>> Exception in thread "main" java.lang.UnsatisfiedLinkError:
>>> /usr/lib/libesl.so: /usr/lib/libesl.so: undefined symbol:
>>> __gxx_personality_v0
>>>         at java.lang.ClassLoader$NativeLibrary.load(Native Method)
>>>         at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778)
>>>         at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703)
>>>         at java.lang.Runtime.loadLibrary0(Runtime.java:823)
>>>         at java.lang.System.loadLibrary(System.java:1030)
>>>         at ivr.IVRServer.<clinit>(IVRServer.java:18)
>>>
>>> Any help would be appreciated.
>>>
>>> Thanks.
>>>
>>>
>>> ------------------------------
>>> The new Internet Explorer? 8 - Faster, safer, easier. Optimized for
>>> Yahoo! *Get it Now for Free!*<http://downloads.yahoo.com/ca/internetexplorer/>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com<http://ca.mc356.mail.yahoo.com/mc/compose?to=MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<http://ca.mc356.mail.yahoo.com/mc/compose?to=PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> -----Inline Attachment Follows-----
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>   ------------------------------
>>
>> *Yahoo! Canada Toolbar :* Search from anywhere on the web and bookmark
>> your favourite sites. Download it now! <http://ca.toolbar.yahoo.com/>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Fernando Gregianin Testa
> Voice Technology Ltda
> +55 11 35882166
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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From telles-listas at devel-it.com.br  Mon Aug 17 14:36:03 2009
From: telles-listas at devel-it.com.br (Rodrigo P. Telles)
Date: Mon, 17 Aug 2009 18:36:03 -0300
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <378EC23A-D23A-4CFC-98DB-693946CF91E6@freeswitch.org>
References: <1249685648.16901.34.camel@dk-d820>
	<378EC23A-D23A-4CFC-98DB-693946CF91E6@freeswitch.org>
Message-ID: <4A89CD43.8030306@devel-it.com.br>

Hi FS Team,

Thanks for your great work at Cluecon, we came back to Brazil with some 
good ideas (+ Snom 360 + Sangoma B600) ;-)

See you again next year!

Rodrigo Telles
Devel-IT


From mike at jerris.com  Mon Aug 17 15:05:13 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 17 Aug 2009 18:05:13 -0400
Subject: [Freeswitch-users] JAVA ESL
In-Reply-To: <9cb0e15e0908171435o54c7379fye4a3016d9b7c59b6@mail.gmail.com>
References: <191c3a030907301800n221cab6cmfbc27b97e292bd85@mail.gmail.com>
	<881036.91387.qm@web35603.mail.mud.yahoo.com>
	<9cb0e15e0908171304q1a47e358s9cbaac00fc284cfb@mail.gmail.com>
	<F0B83E91-DE86-486B-82AB-C3BF9DF3783D@jerris.com>
	<9cb0e15e0908171435o54c7379fye4a3016d9b7c59b6@mail.gmail.com>
Message-ID: <06FE817A-222B-45FA-9A5F-B5BB891436F0@jerris.com>

Thanks, I'll get that merged in and clear from this we need to start  
using proper autoconf checks for java.

mike

On Aug 17, 2009, at 5:35 PM, Fernando Testa wrote:

> Done!
> Check out patch at http://jira.freeswitch.org/browse/FSBUILD-185
>
> Testa
>
> On Mon, Aug 17, 2009 at 5:18 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> can someone post a patch to that makefile to jira.freeswitch.org  
> please.
>
> Mike
>
> On Aug 17, 2009, at 4:04 PM, Fernando Testa wrote:
>
>> I found this same issue on my machine.
>> If you could compile a esl_wrap.o then you have to generate a  
>> libesl.so with a cmd like this:
>> g++ -shared esl_wrap.o -o libesl.so
>> Then in your code, do something like this:
>> /* Test.java */
>> import org.freeswitch.esl.*;
>>
>> class Test
>> {
>>   public static void main(String[] args)
>>   {
>>     System.loadLibrary("esl");
>>     System.out.println("hello");
>>   }
>> }
>>
>>
>>
>> On Thu, Jul 30, 2009 at 10:31 PM, Jean-Marc Hyppolite <hyppolite72 at yahoo.com 
>> > wrote:
>> Thank you Anthony.
>>
>> --- On Thu, 7/30/09, Anthony Minessale  
>> <anthony.minessale at gmail.com> wrote:
>>
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Subject: Re: [Freeswitch-users] JAVA ESL
>> To: freeswitch-users at lists.freeswitch.org
>> Received: Thursday, July 30, 2009, 9:00 PM
>>
>>
>> it might be a build issue, I was not exactly sure how to build it  
>> etc.
>> so it may need some help from a java expert
>>
>> I wrote all of that with swig and never was able to test it.
>>
>>
>> On Thu, Jul 30, 2009 at 7:40 PM, Jean-Marc Hyppolite <hyppolite72 at yahoo.com 
>> > wrote:
>> Hello,
>>
>> I built libesl and JAVA mod. (make and make javamod). But when I  
>> try to run a JAVA script with the following code
>>
>> ESLconnection connection = new ESLconnection("127.0.0.1", "9000",  
>> "");
>> ESLevent events = connection.getInfo();
>> System.out.println(events.toString());
>>
>> I get the following error message:
>>
>> Exception in thread "main" java.lang.UnsatisfiedLinkError: /usr/lib/ 
>> libesl.so: /usr/lib/libesl.so: undefined symbol: __gxx_personality_v0
>>         at java.lang.ClassLoader$NativeLibrary.load(Native Method)
>>         at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778)
>>         at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703)
>>         at java.lang.Runtime.loadLibrary0(Runtime.java:823)
>>         at java.lang.System.loadLibrary(System.java:1030)
>>         at ivr.IVRServer.<clinit>(IVRServer.java:18)
>>
>> Any help would be appreciated.
>>
>> Thanks.
>>
>>
>> The new Internet Explorer? 8 - Faster, safer, easier. Optimized for  
>> Yahoo! Get it Now for Free!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>
>> -----Inline Attachment Follows-----
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>  Yahoo! Canada Toolbar : Search from anywhere on the web and  
>> bookmark your favourite sites. Download it now!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Fernando Gregianin Testa
>> Voice Technology Ltda
>> +55 11 35882166
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Fernando Gregianin Testa
> Voice Technology Ltda
> +55 11 35882166
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From larclap at yahoo.com  Mon Aug 17 17:45:34 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Mon, 17 Aug 2009 17:45:34 -0700
Subject: [Freeswitch-users] Eavesdrop getting killed after being answered
In-Reply-To: <87f2f3b90908171422x1990be2fm9c3d01a7f36d58e5@mail.gmail.com>
References: <005c01ca1f77$eb8e6ef0$c2ab4cd0$@com>
	<87f2f3b90908171422x1990be2fm9c3d01a7f36d58e5@mail.gmail.com>
Message-ID: <00ac01ca1f9d$32428ff0$96c7afd0$@com>

Thanks, Michael, it's working at 14548.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, August 17, 2009 2:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Eavesdrop getting killed after being
answered

 

 

On Mon, Aug 17, 2009 at 1:18 PM, Lars Zeb <larclap at yahoo.com> wrote:

I used to be able to dial 88+extension to eavesdrop, but now it is killed
right after the call is answered by the extension. Can anyone tell me what I
have done wrong? 

 

I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7
10:39:21 EDT 2009 i686 i686 i386 GNU/Linux.

 

http://pastebin.freeswitch.org/10025

 

Hmm, something isn't right. Here's a snippet from a successful eavesdrop:
http://pastebin.freeswitch.org/10027

As you can see, your log shows no media bug operation like mine does.

Can you do another "make current" just to make 100% certain that your build
environment isn't corrupted?
-MC
 

Thanks Lars


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From intralanman at freeswitch.org  Mon Aug 17 18:27:12 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Mon, 17 Aug 2009 21:27:12 -0400
Subject: [Freeswitch-users] G729 transcoding workaround
In-Reply-To: <W4098424883208801250541051@webmail30>
References: <W4098424883208801250541051@webmail30>
Message-ID: <66B5BC8D-496B-4B7A-9C6F-DD430F97336D@freeswitch.org>

On Aug 17, 2009, at 4:30 PM, Justin Miller wrote:
> Is there a way to end the transcoded call legs, and bridge to the  
> phone from the original call leg?  This would free up the M1000, and  
> just seems like a better way to do things.


You might consider using a REFER if your endpoints support it.

Check out the "deflect" app


Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     

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From max.bridgewater at gmail.com  Mon Aug 17 21:59:22 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Tue, 18 Aug 2009 00:59:22 -0400
Subject: [Freeswitch-users] Setting WAV File as rinback
Message-ID: <c55d46950908172159w40232399l385ee9902869aa18@mail.gmail.com>

Hi,

I'm trying to have an audio as ringback (WAV, 8khz, mono) when originating a
call. Unfortunately, it doesn't seem to work; the RINGING teletone is being
used instead of my audio. I have debug level enabled on Freeswitch but i
don't find anything suspicious.

is there anything specific i should be taking into consideration?


Thanks,
max.
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From wangdq.no1 at gmail.com  Tue Aug 18 02:24:35 2009
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Tue, 18 Aug 2009 17:24:35 +0800
Subject: [Freeswitch-users] Setting WAV File as rinback
In-Reply-To: <c55d46950908172159w40232399l385ee9902869aa18@mail.gmail.com>
References: <c55d46950908172159w40232399l385ee9902869aa18@mail.gmail.com>
Message-ID: <ecd05ce30908180224k30472c44mebb7f54fba21b384@mail.gmail.com>

set ring_back = " your ring file "


2009/8/18 Max Bridgewater <max.bridgewater at gmail.com>

> Hi,
>
> I'm trying to have an audio as ringback (WAV, 8khz, mono) when originating
> a call. Unfortunately, it doesn't seem to work; the RINGING teletone is
> being used instead of my audio. I have debug level enabled on Freeswitch but
> i don't find anything suspicious.
>
> is there anything specific i should be taking into consideration?
>
>
> Thanks,
> max.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From enno.egbert at googlemail.com  Tue Aug 18 04:34:53 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Tue, 18 Aug 2009 04:34:53 -0700 (PDT)
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <11D5AA56-1609-4D2C-B5D1-8AAF4A575309@freeswitch.org>
References: <25005636.post@talk.nabble.com>
	<672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
	<25006858.post@talk.nabble.com>
	<11D5AA56-1609-4D2C-B5D1-8AAF4A575309@freeswitch.org>
Message-ID: <25023035.post@talk.nabble.com>


Hi,

sorry - i have to ask again. This is my entry in
..../conf/sip_profiles/external/ :

<include>
<gateway name="sipgate.de">


 

</gateway>
</include>

and it?s still the same... from: 2395805 at 139.13.37.160

Do you have another tip?

Thanks 

NOx






Brian West-3 wrote:
> 
> Well if you pay attention I told you in the last email... set the  
> param from-domain on the gateway to sipgate.de
> 
> /b
> 
> On Aug 17, 2009, at 8:36 AM, NOx-WHV wrote:
> 
>>
>> Hi,
>>
>> i have just taken some pictures.
>>
>> http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
>> http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG
>> http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG
>>
>> the third picture is taken by a softphone that works.
>>
>> To Brian:
>> If i set the effective_caller_id_name to 2395805 without @sipgate.de  
>> it?s
>> the same problem, because the freeswitch set in the from fielt:
>> 2395805 at 139.13.37.160. And i can?t find the right parameter to change
>> "139.13.37.160" to "sipgate.de"
>>
>> Which is the right parameter in the dialplan?
>>
>> <action application=.....?????>
>>
>> Thanks for your help
>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25023035.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From enno.egbert at googlemail.com  Tue Aug 18 04:36:04 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Tue, 18 Aug 2009 04:36:04 -0700 (PDT)
Subject: [Freeswitch-users] SIPGATE Problem
Message-ID: <25023035.post@talk.nabble.com>


Hi,

sorry - i have to ask again. This is my entry in
..../conf/sip_profiles/external/ :

include
gateway name="sipgate.de"
param name="username" value="2395805"/
param name="from-domain" value="sipgate.de"/
param name="password" value="abcde"/ 
param name="proxy" value="sipgate.de"/
/gateway
/include

and it?s still the same... from: 2395805 at 139.13.37.160

Do you have another tip?

Thanks 

NOx






Brian West-3 wrote:
> 
> Well if you pay attention I told you in the last email... set the  
> param from-domain on the gateway to sipgate.de
> 
> /b
> 
> On Aug 17, 2009, at 8:36 AM, NOx-WHV wrote:
> 
>>
>> Hi,
>>
>> i have just taken some pictures.
>>
>> http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
>> http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG
>> http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG
>>
>> the third picture is taken by a softphone that works.
>>
>> To Brian:
>> If i set the effective_caller_id_name to 2395805 without @sipgate.de  
>> it?s
>> the same problem, because the freeswitch set in the from fielt:
>> 2395805 at 139.13.37.160. And i can?t find the right parameter to change
>> "139.13.37.160" to "sipgate.de"
>>
>> Which is the right parameter in the dialplan?
>>
>> <action application=.....?????>
>>
>> Thanks for your help
>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25023035.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From mayamatakeshi at gmail.com  Tue Aug 18 04:53:24 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 18 Aug 2009 20:53:24 +0900
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <25023035.post@talk.nabble.com>
References: <25005636.post@talk.nabble.com>
	<672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
	<25006858.post@talk.nabble.com>
	<11D5AA56-1609-4D2C-B5D1-8AAF4A575309@freeswitch.org>
	<25023035.post@talk.nabble.com>
Message-ID: <15b9404e0908180453r59cb95bbn55f3ab9721a298d2@mail.gmail.com>

On Tue, Aug 18, 2009 at 8:34 PM, NOx-WHV<enno.egbert at googlemail.com> wrote:
>
> Hi,
>
> sorry - i have to ask again. This is my entry in
> ..../conf/sip_profiles/external/ :
>
> <include>
> <gateway name="sipgate.de">
>
>
>
>
> </gateway>
> </include>
>
> and it?s still the same... from: 2395805 at 139.13.37.160
>
> Do you have another tip?

I don't see where you are setting the param from-domain to sipgate.de
as Brian told you to do. I can only see you are setting the name of
the gateway to sipgate.de.
I never used this, but in the samples that come with FS we can see this:

<param name="from-domain" value="asterlink.com"/>

>
> Brian West-3 wrote:
>>
>> Well if you pay attention I told you in the last email... set the
>> param from-domain on the gateway to sipgate.de
>>
>> /b
>>
>> On Aug 17, 2009, at 8:36 AM, NOx-WHV wrote:
>>
>>>
>>> Hi,
>>>
>>> i have just taken some pictures.
>>>
>>> http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
>>> http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG
>>> http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG
>>>
>>> the third picture is taken by a softphone that works.
>>>
>>> To Brian:
>>> If i set the effective_caller_id_name to 2395805 without @sipgate.de
>>> it?s
>>> the same problem, because the freeswitch set in the from fielt:
>>> 2395805 at 139.13.37.160. And i can?t find the right parameter to change
>>> "139.13.37.160" to "sipgate.de"
>>>
>>> Which is the right parameter in the dialplan?
>>>
>>> <action application=.....?????>
>>>
>>> Thanks for your help
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> --
> View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25023035.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From shiyanov at gmail.com  Tue Aug 18 05:54:04 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Tue, 18 Aug 2009 16:54:04 +0400
Subject: [Freeswitch-users] "mute" channel programmatically with
	mod_event_socket
Message-ID: <d211f54a0908180554s4b5324cfx39e641d44307b173@mail.gmail.com>

Hello all!

I'm trying to implement "mute" feature with mod_event_socket: I want
programmatically mute/unmute some channel in a call.. And I don't see any
other ways except to use conference room with special rule "mute".
Anybody knows the better way?


Thanks,
Artem
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From enno.egbert at googlemail.com  Tue Aug 18 06:10:09 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Tue, 18 Aug 2009 06:10:09 -0700 (PDT)
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <15b9404e0908180453r59cb95bbn55f3ab9721a298d2@mail.gmail.com>
References: <25005636.post@talk.nabble.com>
	<672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
	<25006858.post@talk.nabble.com>
	<11D5AA56-1609-4D2C-B5D1-8AAF4A575309@freeswitch.org>
	<25023035.post@talk.nabble.com>
	<15b9404e0908180453r59cb95bbn55f3ab9721a298d2@mail.gmail.com>
Message-ID: <25024536.post@talk.nabble.com>


Hi,

this is the text without brackets:

include 
gateway name="sipgate.de" 
param name="username" value="2395805"/ 
param name="from-domain" value="sipgate.de"/ 
param name="password" value="abcde"/ 
param name="proxy" value="sipgate.de"/ 
/gateway 
/include 

I don?t know, why you can?t see the lines between gateway and /gateway.

NOx

mayamatakeshi wrote:
> 
> On Tue, Aug 18, 2009 at 8:34 PM, NOx-WHV<enno.egbert at googlemail.com>
> wrote:
>>
>> Hi,
>>
>> sorry - i have to ask again. This is my entry in
>> ..../conf/sip_profiles/external/ :
>>
>> <include>
>> <gateway name="sipgate.de">
>>
>>
>>
>>
>> </gateway>
>> </include>
>>
>> and it?s still the same... from: 2395805 at 139.13.37.160
>>
>> Do you have another tip?
> 
> I don't see where you are setting the param from-domain to sipgate.de
> as Brian told you to do. I can only see you are setting the name of
> the gateway to sipgate.de.
> I never used this, but in the samples that come with FS we can see this:
> 
> 
> 
>>
>> Brian West-3 wrote:
>>>
>>> Well if you pay attention I told you in the last email... set the
>>> param from-domain on the gateway to sipgate.de
>>>
>>> /b
>>>
>>> On Aug 17, 2009, at 8:36 AM, NOx-WHV wrote:
>>>
>>>>
>>>> Hi,
>>>>
>>>> i have just taken some pictures.
>>>>
>>>> http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
>>>> http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG
>>>> http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG
>>>>
>>>> the third picture is taken by a softphone that works.
>>>>
>>>> To Brian:
>>>> If i set the effective_caller_id_name to 2395805 without @sipgate.de
>>>> it?s
>>>> the same problem, because the freeswitch set in the from fielt:
>>>> 2395805 at 139.13.37.160. And i can?t find the right parameter to change
>>>> "139.13.37.160" to "sipgate.de"
>>>>
>>>> Which is the right parameter in the dialplan?
>>>>
>>>> <action application=.....?????>
>>>>
>>>> Thanks for your help
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> --
>> View this message in context:
>> http://www.nabble.com/SIPGATE-Problem-tp25005636p25023035.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25024536.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From mayamatakeshi at gmail.com  Tue Aug 18 07:03:43 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 18 Aug 2009 23:03:43 +0900
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <25024536.post@talk.nabble.com>
References: <25005636.post@talk.nabble.com>
	<672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
	<25006858.post@talk.nabble.com>
	<11D5AA56-1609-4D2C-B5D1-8AAF4A575309@freeswitch.org>
	<25023035.post@talk.nabble.com>
	<15b9404e0908180453r59cb95bbn55f3ab9721a298d2@mail.gmail.com>
	<25024536.post@talk.nabble.com>
Message-ID: <15b9404e0908180703s77a4877ek2cafd976ffc8a2f4@mail.gmail.com>

On Tue, Aug 18, 2009 at 10:10 PM, NOx-WHV<enno.egbert at googlemail.com> wrote:
>
> Hi,
>
> this is the text without brackets:
>
> include
> gateway name="sipgate.de"
> param name="username" value="2395805"/
> param name="from-domain" value="sipgate.de"/
> param name="password" value="abcde"/
> param name="proxy" value="sipgate.de"/
> /gateway
> /include
>
> I don?t know, why you can?t see the lines between gateway and /gateway.

I think my email client (gmail) doesn't like the brackets.

I've tried the above settings in my machine and it works.
I tried originating a call from the CLI:
originate sofia/gateway/sipgate.de/1234 &park

And I got this:
From: "FreeSWITCH" <sip:2395805 at sipgate.de;transport=udp>;


From tayeb.meftah at gmail.com  Tue Aug 18 07:49:11 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 18 Aug 2009 14:49:11 +0000
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
 but I have a little problem
In-Reply-To: <8181A0DB-67EE-464C-9A69-30BA0719C24C@jerris.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>	<4A884F07.3040200@gmail.com>	<5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>	<4A8881E7.8080509@gmail.com>	<D3325007-BA3D-4B65-B533-AEC2BF7AB447@jerris.com>	<4A8922DF.1080008@gmail.com>
	<8181A0DB-67EE-464C-9A69-30BA0719C24C@jerris.com>
Message-ID: <4A8ABF67.8040608@gmail.com>

hi MikeJ,
yes, is allready maintained with the latest APACHE/PHP/MySQL/Perl 
software in uniform server / sourceforge
i can provide you with any informations, if you want to help you about 
using Windows Installer XML
please just add a customisation to the installer and ignore Advanced 
installer
some users want to install FreePBX3 in IIS
we can provide Virtual directory installation optionaly, unstid of 
uniform server
in a custom dialog, we chouse if we use IIS or Uniform Server
http://www.wixedit.sourceforge.net/
thanks,
meftah tayeb
Michael Jerris wrote:
>
> On Aug 17, 2009, at 5:29 AM, Meftah Tayeb wrote:
>
>> hi MikeJ,
>> i prefer creating MSI file that is easy to mintin$
>
> How is this useful vs. something that is already maintained?
>
>> unstid of using inno setup or advanced installer (not free), we can 
>> use WIX (Windows installer XML) that is a open source one
>
> All the tools being used are free, however all are not open source.
>
>> we can create a customised MSI that fully install mor features, 
>> including Sounds / MOH/...
>
> This installer already does all that and more.
>
>> and we can edit XML files easyly to let users chouse each module to 
>> install during setup, for example conferencing, voice mail, Event 
>> Socket and ...
>> also the WAMP server is not fully stable
>> i prefer Uniform Server, that mintin Full Compatibility with Unix 
>> like platform including for example, perl files that need this path: 
>> /usr/...
>> apache2 is installed in /usr/local/apache2, mySQL: /usr/local/mysql 
>> and php is in /usr/local/php
>> and perl is in /usr/bin
>> php my admin is in /etc/phpmyadmin
>> and mor featurs, including multiple instance, virtual hosting and 
>> full management using control pannel
>> this will help enterprise users that need that in windows unstid of WAMP
>>      ---------------    
>> if you don't want to mintin it, jive me a chance to mintin it
>>       ---------------     
>> thanks
>
> I just don't want to split efforts here, every different way we do 
> this is yet another way we have to support.  I see no compelling 
> argument here other than personal preference.  If I am missing 
> anything someone please chime in and correct me.
>
> Mike
>
>>
>>     Contact Me
>>
>>
>>     * SIP <sip:voipworld at sip.voipuser.org>
>>     * INUM <callto:+883510009102769>
>>     * PSTN <callto:0021348526972>
>>     * Email <mailto:tayeb.meftah at gmail.com>
>>
>>
>> Michael Jerris wrote:
>>> I think sticking with standard WAMP is preferable.  What is the 
>>> advantage to creating yet another installer over the one that we 
>>> have already done and maintained?
>>>
>>> Mike
>>>
>>> On Aug 16, 2009, at 6:02 PM, Meftah Tayeb wrote:
>>>
>>>> hello
>>>> i'm rewriting this executable file in MSI format
>>>> i can use:
>>>>
>>>>     * MakeMSI
>>>>     * WIX (Windows installer XML)
>>>>
>>>> what you like?
>>>> WIX is open source, but MakeMSI i'm not sur
>>>> for web server, i'm replacing WAMP with a great web server called 
>>>> uniform server
>>>> this server have a conventional Unix Path configuration, for 
>>>> example, web files can by stored in /www/ and apache is in 
>>>> /usr/local/apache2
>>>> this facilitate the deploiment of applications
>>>> any suggestion?
>>>> thanks
>>>> Meftah Tayeb
>>>> Carlos Talbot wrote:
>>>>> http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe
>>>>>
>>>
>>>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus 
> signature database 4345 (20090818) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4345 (20090818) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   



__________ Information from ESET NOD32 Antivirus, version of virus signature database 4345 (20090818) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




From juanbackson at gmail.com  Tue Aug 18 08:04:31 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Tue, 18 Aug 2009 23:04:31 +0800
Subject: [Freeswitch-users] memory leak with switch_core_hash_insert
Message-ID: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>

Hi,

I am getting some strange vg malloc error message in switch_core_hash_insert.
Does anyone know what is wrong with these few lines?  Am I missing
something?


switch_core_hash_init(&hash,pool);
param_name =switch_core_sprintf(pool,"%s", key);
param_value =switch_core_sprintf(pool,"%s", value);
switch_core_hash_insert(hash,param_name,param_value);


==5459==
==5459== 8,973 (2,344 direct, 6,629 indirect) bytes in 10 blocks are
definitely lost in loss record 62 of 89
==5459==    at 0x4A05809: malloc (vg_replace_malloc.c:149)
==5459==    by 0x4CEBB68: sqlite3MallocX (sqliteInt.h:278)
==5459==    by 0x4CE1C7E: rehash (hash.c:227)
==5459==    by 0x4CE206F: sqlite3HashInsert (hash.c:386)
==5459==    by 0x4C70EA4: switch_core_hash_insert (switch_core_hash.c:67)
==5459==    by 0x7CEAD07: ??? (mod_specialivr.c:343)
==5459==    by 0x7CEBF87: ??? (mod_specialivr.c:784)
==5459==    by 0x7CEDBF0: ??? (mod_specialivr.c:1778)
==5459==    by 0x4C790AC: switch_core_session_run
(switch_core_state_machine.c:109)
==5459==    by 0x4C751FF: switch_core_session_thread
(switch_core_session.c:1066)
==5459==    by 0x4FBC306: start_thread (in /lib64/libpthread-2.5.so)
==5459==    by 0x35018D1DEC: clone (in /lib64/libc-2.5.so)
==5459==
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From mrene_lists at avgs.ca  Tue Aug 18 08:09:02 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 18 Aug 2009 11:09:02 -0400
Subject: [Freeswitch-users] memory leak with switch_core_hash_insert
In-Reply-To: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>
References: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>
Message-ID: <5BB3EE22-66F9-4D9B-9CB9-23A7680757AB@avgs.ca>

Hi,

You're missing:
switch_core_hash_destroy(&hash)

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 18-Aug-09, at 11:04 AM, Juan Backson wrote:

> Hi,
>
> I am getting some strange vg malloc error message in  
> switch_core_hash_insert.  Does anyone know what is wrong with these  
> few lines?  Am I missing something?
>
>
> switch_core_hash_init(&hash,pool);
> param_name =switch_core_sprintf(pool,"%s", key);
> param_value =switch_core_sprintf(pool,"%s", value);
> switch_core_hash_insert(hash,param_name,param_value);
>
>
> ==5459==
> ==5459== 8,973 (2,344 direct, 6,629 indirect) bytes in 10 blocks are  
> definitely lost in loss record 62 of 89
> ==5459==    at 0x4A05809: malloc (vg_replace_malloc.c:149)
> ==5459==    by 0x4CEBB68: sqlite3MallocX (sqliteInt.h:278)
> ==5459==    by 0x4CE1C7E: rehash (hash.c:227)
> ==5459==    by 0x4CE206F: sqlite3HashInsert (hash.c:386)
> ==5459==    by 0x4C70EA4: switch_core_hash_insert  
> (switch_core_hash.c:67)
> ==5459==    by 0x7CEAD07: ??? (mod_specialivr.c:343)
> ==5459==    by 0x7CEBF87: ??? (mod_specialivr.c:784)
> ==5459==    by 0x7CEDBF0: ??? (mod_specialivr.c:1778)
> ==5459==    by 0x4C790AC: switch_core_session_run  
> (switch_core_state_machine.c:109)
> ==5459==    by 0x4C751FF: switch_core_session_thread  
> (switch_core_session.c:1066)
> ==5459==    by 0x4FBC306: start_thread (in /lib64/libpthread-2.5.so)
> ==5459==    by 0x35018D1DEC: clone (in /lib64/libc-2.5.so)
> ==5459==
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From markmorreny at gmail.com  Tue Aug 18 08:18:33 2009
From: markmorreny at gmail.com (mark morreny)
Date: Tue, 18 Aug 2009 23:18:33 +0800
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <FD54C936-D80E-4F96-8502-740ADDD1225C@avgs.ca>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
	<20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
	<EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>
	<20ad6b920908141919u1b2aa1c4jacc75cd4896d72b0@mail.gmail.com>
	<EF4B4642-E586-42E0-8D40-D9FCB7A93DD2@jerris.com>
	<FD54C936-D80E-4F96-8502-740ADDD1225C@avgs.ca>
Message-ID: <20ad6b920908180818q7c5a7214vc20aa8ed864953e9@mail.gmail.com>

Hi,

Thank you very much.  I am able to get my sched api to work now.

I still have one problem.  I am getting core dump if freeswitch is shut down
by typing in "shutdown" in the CLI.  How can I control the shutdown process
so that it will wait until all the existing sched jobs are done before
starting to free the memory?

Mark

On Sat, Aug 15, 2009 at 11:26 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Because switch_time_now() is in microseconds.
>    Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
>   On 14-Aug-09, at 11:16 PM, Michael Jerris wrote:
>
>    task->runtime = switch_epoch_time_now(NULL) + 10;
>  On Aug 14, 2009, at 10:19 PM, mark morreny wrote:
>
>  Hi Michael,
>
> The following code was executed once, but not after the next 10 s.
>
> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to flush
> data  buffer...\n");
>
>     task->runtime = switch_time_now() + 10;
> }
>
> Any suggestion why?
>
>
> Thanks,
> Mark
> On Sat, Aug 15, 2009 at 2:13 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> thats in seconds.
>> Mike
>>
>>  On Aug 14, 2009, at 8:32 AM, mark morreny wrote:
>>
>>  Hi,
>>
>> Thank you for your help.
>>
>> I get that too, but the callback does not execute the second time.
>>
>> When I do task->runtime = switch_time_now() + 10;, what does +10 mean?
>> Does it mean 10 s or 10 mins?
>>
>> Thanks,
>> Mark
>>
>> On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene <mrene_lists at avgs.ca>wrote:
>>
>>> Hi,
>>> I did the same thing on my side....
>>>  API CALL [load(mod_skel)] output:
>>> +OK
>>>
>>> 2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task 2
>>> data_flush (core) to run at 1250089698
>>> 2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889
>>> Successfully Loaded [mod_skel]
>>> 2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270 Adding
>>> API Function 'skel'
>>> freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]
>>> mod_skel.c:120 starting to flush data  buffer...
>>>
>>> Note that you don't need to start the thread manually, the core already
>>> has threads running for the scheduler.
>>>
>>>   Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> mrene at avgs.ca
>>>
>>>
>>>
>>>
>>>  On 12-Aug-09, at 7:26 AM, mark morreny wrote:
>>>
>>> Hi,
>>>
>>> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out some
>>> data every 10 s.  The following lines of code does not show any effect at
>>> all.
>>>
>>>     switch_scheduler_task_thread_start();
>>>     switch_scheduler_add_task(switch_epoch_time_now(NULL),
>>> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|SSHF_NO_DEL);
>>>
>>>
>>> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>>>
>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to
>>> flush data  buffer...\n");
>>>
>>>
>>>     task->runtime = switch_time_now() + 10;
>>>
>>> }
>>>
>>> Does anyone know how to get it to work?
>>>
>>> Thanks,
>>> Mark
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Brian West <brian at freeswitch.org>
>>> Date: Mon, Aug 10, 2009 at 8:53 PM
>>> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
>>> To: freeswitch-users at lists.freeswitch.org
>>>
>>>
>>> switch_rtp.c has a simple one for the zrtp cache storing.
>>>
>>> /b
>>>
>>> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>>>
>>> > Re schedule is done in your callback, take a look at places that use
>>> > these apis in the code for details.
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Aug 18 08:21:21 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 18 Aug 2009 11:21:21 -0400
Subject: [Freeswitch-users] memory leak with switch_core_hash_insert
In-Reply-To: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>
References: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>
Message-ID: <BB05B7CA-F29D-48E0-A102-7518866AFAB9@jerris.com>

try running freeswitch with -vg command line arg.

Mike

On Aug 18, 2009, at 11:04 AM, Juan Backson wrote:

> Hi,
>
> I am getting some strange vg malloc error message in  
> switch_core_hash_insert.  Does anyone know what is wrong with these  
> few lines?  Am I missing something?
>
>
> switch_core_hash_init(&hash,pool);
> param_name =switch_core_sprintf(pool,"%s", key);
> param_value =switch_core_sprintf(pool,"%s", value);
> switch_core_hash_insert(hash,param_name,param_value);
>
>
> ==5459==
> ==5459== 8,973 (2,344 direct, 6,629 indirect) bytes in 10 blocks are  
> definitely lost in loss record 62 of 89
> ==5459==    at 0x4A05809: malloc (vg_replace_malloc.c:149)
> ==5459==    by 0x4CEBB68: sqlite3MallocX (sqliteInt.h:278)
> ==5459==    by 0x4CE1C7E: rehash (hash.c:227)
> ==5459==    by 0x4CE206F: sqlite3HashInsert (hash.c:386)
> ==5459==    by 0x4C70EA4: switch_core_hash_insert  
> (switch_core_hash.c:67)
> ==5459==    by 0x7CEAD07: ??? (mod_specialivr.c:343)
> ==5459==    by 0x7CEBF87: ??? (mod_specialivr.c:784)
> ==5459==    by 0x7CEDBF0: ??? (mod_specialivr.c:1778)
> ==5459==    by 0x4C790AC: switch_core_session_run  
> (switch_core_state_machine.c:109)
> ==5459==    by 0x4C751FF: switch_core_session_thread  
> (switch_core_session.c:1066)
> ==5459==    by 0x4FBC306: start_thread (in /lib64/libpthread-2.5.so)
> ==5459==    by 0x35018D1DEC: clone (in /lib64/libc-2.5.so)
> ==5459==
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tzury.by at reguluslabs.com  Tue Aug 18 08:22:19 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Tue, 18 Aug 2009 18:22:19 +0300
Subject: [Freeswitch-users] how to use freeswitch as a sip client
Message-ID: <10128ef10908180822p373b2415rb558bbaa8df5020a@mail.gmail.com>

Hi,

Can someone instruct me (or point to a page where I can found this
information) how can freeswitch be used as a sip client.
I would like to test a server transporting SIP packets over TCP and was
reported on this mailing list that it is possible to do so with FS.

thanks,
Tzury
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From dujinfang at gmail.com  Tue Aug 18 08:26:37 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 18 Aug 2009 23:26:37 +0800
Subject: [Freeswitch-users] how to use freeswitch as a sip client
In-Reply-To: <10128ef10908180822p373b2415rb558bbaa8df5020a@mail.gmail.com>
References: <10128ef10908180822p373b2415rb558bbaa8df5020a@mail.gmail.com>
Message-ID: <7BC38AD3-79BB-4533-A4B7-501539DE678C@gmail.com>

http://wiki.freeswitch.org/wiki/Mod_portaudio
http://wiki.freeswitch.org/wiki/FsAir


On Aug 18, 2009, at 11:22 PM, Tzury Bar Yochay wrote:
> Hi,
>
> Can someone instruct me (or point to a page where I can found this  
> information) how can freeswitch be used as a sip client.
> I would like to test a server transporting SIP packets over TCP and  
> was reported on this mailing list that it is possible to do so with  
> FS.
>
> thanks,
> Tzury
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dist.lists at gmail.com  Tue Aug 18 08:24:48 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Tue, 18 Aug 2009 18:24:48 +0300
Subject: [Freeswitch-users] failover based on initial INVITE timeout
Message-ID: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>

I am trying to implement failover dialing plan as described in:
http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout

I figured out that originate_timeout must be passed as
{originate_timeout=<timeout>} in front of the dial string to have any
effect (setting it as channel variable as described in the example
above has no effect).

I have set the timeout to 1 second, so expected behavior is to try the
second gateway if no response is received from the first one in 1
second. The problem is that FS cancels the first request with
[NO_ANSWER] and tries to route the call via the second gateway even
though it receives response from the first during that 1 second.

The response received is "100 Trying" provisional response (checked
with sofia siptrace). I'm guessing that either 100 provisional
responses don't cancel the originate_timeout timer (bug?) or I am
doing it the wrong way.

I was also thinking of using the timer-T1 or timer-T1X64 parameter in
the sip profile, but I need this to be set per dial string, not per
profile, besides, it seems that these timers (T1, T1X64) affect both
invite and non-invite requests, so this is not really an option.
Also, I tried leg_timeout, but it doesn't really do what I need it to.

Anyone has any idea how to implement this?


From brian at freeswitch.org  Tue Aug 18 08:26:57 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Aug 2009 10:26:57 -0500
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <20ad6b920908180818q7c5a7214vc20aa8ed864953e9@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
	<20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
	<EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>
	<20ad6b920908141919u1b2aa1c4jacc75cd4896d72b0@mail.gmail.com>
	<EF4B4642-E586-42E0-8D40-D9FCB7A93DD2@jerris.com>
	<FD54C936-D80E-4F96-8502-740ADDD1225C@avgs.ca>
	<20ad6b920908180818q7c5a7214vc20aa8ed864953e9@mail.gmail.com>
Message-ID: <28DCCC86-C909-4AD7-9AD5-3D5AC9D39EB1@freeswitch.org>

Sounds like you need to put a mutex on your scheduler routine.

/b

On Aug 18, 2009, at 10:18 AM, mark morreny wrote:

> Hi,
>
> Thank you very much.  I am able to get my sched api to work now.
>
> I still have one problem.  I am getting core dump if freeswitch is  
> shut down by typing in "shutdown" in the CLI.  How can I control the  
> shutdown process so that it will wait until all the existing sched  
> jobs are done before starting to free the memory?
>
> Mark



From brian at freeswitch.org  Tue Aug 18 08:33:01 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Aug 2009 10:33:01 -0500
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
Message-ID: <45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>

I think you wanna use progress_timeout

http://wiki.freeswitch.org/wiki/Channel_Variables#progress_timeout

/b

On Aug 18, 2009, at 10:24 AM, Hristo Trendev wrote:

> I am trying to implement failover dialing plan as described in:
> http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout
>
> I figured out that originate_timeout must be passed as
> {originate_timeout=<timeout>} in front of the dial string to have any
> effect (setting it as channel variable as described in the example
> above has no effect).
>
> I have set the timeout to 1 second, so expected behavior is to try the
> second gateway if no response is received from the first one in 1
> second. The problem is that FS cancels the first request with
> [NO_ANSWER] and tries to route the call via the second gateway even
> though it receives response from the first during that 1 second.
>
> The response received is "100 Trying" provisional response (checked
> with sofia siptrace). I'm guessing that either 100 provisional
> responses don't cancel the originate_timeout timer (bug?) or I am
> doing it the wrong way.
>
> I was also thinking of using the timer-T1 or timer-T1X64 parameter in
> the sip profile, but I need this to be set per dial string, not per
> profile, besides, it seems that these timers (T1, T1X64) affect both
> invite and non-invite requests, so this is not really an option.
> Also, I tried leg_timeout, but it doesn't really do what I need it to.
>
> Anyone has any idea how to implement this?

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From msc at freeswitch.org  Tue Aug 18 09:02:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Aug 2009 09:02:33 -0700
Subject: [Freeswitch-users] memory leak with switch_core_hash_insert
In-Reply-To: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>
References: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>
Message-ID: <87f2f3b90908180902i4ddb6c75mbc4c6b353ce39821@mail.gmail.com>

Just a heads up: this is a rather technical subject. I highly recommend that
you use freeswitch-dev list for this kind of thing.
-MC

On Tue, Aug 18, 2009 at 8:04 AM, Juan Backson <juanbackson at gmail.com> wrote:

> Hi,
>
> I am getting some strange vg malloc error message in switch_core_hash_insert.
> Does anyone know what is wrong with these few lines?  Am I missing
> something?
>
>
> switch_core_hash_init(&hash,pool);
> param_name =switch_core_sprintf(pool,"%s", key);
> param_value =switch_core_sprintf(pool,"%s", value);
> switch_core_hash_insert(hash,param_name,param_value);
>
>
> ==5459==
> ==5459== 8,973 (2,344 direct, 6,629 indirect) bytes in 10 blocks are
> definitely lost in loss record 62 of 89
> ==5459==    at 0x4A05809: malloc (vg_replace_malloc.c:149)
> ==5459==    by 0x4CEBB68: sqlite3MallocX (sqliteInt.h:278)
> ==5459==    by 0x4CE1C7E: rehash (hash.c:227)
> ==5459==    by 0x4CE206F: sqlite3HashInsert (hash.c:386)
> ==5459==    by 0x4C70EA4: switch_core_hash_insert (switch_core_hash.c:67)
> ==5459==    by 0x7CEAD07: ??? (mod_specialivr.c:343)
> ==5459==    by 0x7CEBF87: ??? (mod_specialivr.c:784)
> ==5459==    by 0x7CEDBF0: ??? (mod_specialivr.c:1778)
> ==5459==    by 0x4C790AC: switch_core_session_run
> (switch_core_state_machine.c:109)
> ==5459==    by 0x4C751FF: switch_core_session_thread
> (switch_core_session.c:1066)
> ==5459==    by 0x4FBC306: start_thread (in /lib64/libpthread-2.5.so)
> ==5459==    by 0x35018D1DEC: clone (in /lib64/libc-2.5.so)
> ==5459==
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From woodydickson at gmail.com  Tue Aug 18 09:23:02 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 19 Aug 2009 00:23:02 +0800
Subject: [Freeswitch-users] how to set different action for different
	cause code
In-Reply-To: <87f2f3b90908170859p52259449q913b2ddab224709f@mail.gmail.com>
References: <a44688a0908160424n3569cef1vad8fcdc68cbfb96e@mail.gmail.com>
	<87f2f3b90908170859p52259449q913b2ddab224709f@mail.gmail.com>
Message-ID: <a44688a0908180923q4b4b8e48od86d676872c97c3a@mail.gmail.com>

Hi,

I have my dialplan to do some simple routing. What I need to do is when
certain hangup code is received, route advance to the next or the route
after next based on the hangup code received.

So, I have:

<condition>
   <action application="set" data="hangup_after_bridge=true"/>
   <action application="set" data="continue_on_fail="false"/>
   <action application="bridge" data="sofia/internal/aaa at 123.3.3./>
   <action application="bridge" data="sofia/internal/bbb at 123.3.3./>
   <action application="bridge" data="sofia/internal/ccc at 123.3.3./>
   <action application="bridge" data="sofia/internal/ddd at 123.3.3./>

</condition>

What I want to do is that based on the error code received from ( bridge to
aaa), it can route to either bbb or ccc depends on the hangup code received
from aaa.

So there anyway to do it in the dialplan without any scripting?

thannks,
woody




On Mon, Aug 17, 2009 at 11:59 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Sun, Aug 16, 2009 at 4:24 AM, Woody Dickson <woodydickson at gmail.com>wrote:
>
>> Hello,
>>
>> I find hangup_hook, but I would like to define different actions for
>> different hangup codes.  Is there anyway to do that?
>>
>>
>
> I can think of at least two ways you could do this: one that uses only the
> dialplan and one that uses a script. If you don't mind using a scripting
> language then you can make it very clean:
>
> <action application="set" data="api_hangup_hook=luarun chancleanup.lua
> ${hangup_cause}"/>
>
> Then have your Lua script handle all the if-then-else or case stuff.
>
> Question: are you trying to transfer the a-leg to some other destination if
> the b-leg hangup is a specific cause, or are you just doing some external
> cleanup stuff? Just curious...
>
> -MC
>
>
>> Woody
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dist.lists at gmail.com  Tue Aug 18 10:48:19 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Tue, 18 Aug 2009 20:48:19 +0300
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
Message-ID: <2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>

progress_timeout will wait for media (tried it, but I don't need
that). I want to detect the case when the destination gateway is down
and there is no response (not even 100, 407, etc) to the initial
INVITE sent by FS.

According to the wiki, this is exactly what originate_timeout is used
for. Actually the wiki gives as example for originate_timeout exactly
what I'm trying to accomplish.

It seems to me that FS ignores 100 and alike messages, which are sent
as response to initial INVITE and doesn't cancel originate_timeout
timer if such message is received.

The more I look into this, the more I start to think that it's a bug.

On Tue, Aug 18, 2009 at 6:33 PM, Brian West<brian at freeswitch.org> wrote:
> I think you wanna use progress_timeout
> http://wiki.freeswitch.org/wiki/Channel_Variables#progress_timeout
> /b
> On Aug 18, 2009, at 10:24 AM, Hristo Trendev wrote:
>
> I am trying to implement failover dialing plan as described in:
> http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout
>
> I figured out that originate_timeout must be passed as
> {originate_timeout=<timeout>} in front of the dial string to have any
> effect (setting it as channel variable as described in the example
> above has no effect).
>
> I have set the timeout to 1 second, so expected behavior is to try the
> second gateway if no response is received from the first one in 1
> second. The problem is that FS cancels the first request with
> [NO_ANSWER] and tries to route the call via the second gateway even
> though it receives response from the first during that 1 second.
>
> The response received is "100 Trying" provisional response (checked
> with sofia siptrace). I'm guessing that either 100 provisional
> responses don't cancel the originate_timeout timer (bug?) or I am
> doing it the wrong way.
>
> I was also thinking of using the timer-T1 or timer-T1X64 parameter in
> the sip profile, but I need this to be set per dial string, not per
> profile, besides, it seems that these timers (T1, T1X64) affect both
> invite and non-invite requests, so this is not really an option.
> Also, I tried leg_timeout, but it doesn't really do what I need it to.
>
> Anyone has any idea how to implement this?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Tue Aug 18 11:00:39 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Aug 2009 13:00:39 -0500
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
Message-ID: <F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>

use TCP then if you get an ICMP unreachable it'll move on instantly.

/b

On Aug 18, 2009, at 12:48 PM, Hristo Trendev wrote:

> progress_timeout will wait for media (tried it, but I don't need
> that). I want to detect the case when the destination gateway is down
> and there is no response (not even 100, 407, etc) to the initial
> INVITE sent by FS.
>
> According to the wiki, this is exactly what originate_timeout is used
> for. Actually the wiki gives as example for originate_timeout exactly
> what I'm trying to accomplish.
>
> It seems to me that FS ignores 100 and alike messages, which are sent
> as response to initial INVITE and doesn't cancel originate_timeout
> timer if such message is received.
>
> The more I look into this, the more I start to think that it's a bug.



From dist.lists at gmail.com  Tue Aug 18 11:24:57 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Tue, 18 Aug 2009 21:24:57 +0300
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
Message-ID: <2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>

TCP is not really an option for me. I have tried using different sofia
profile for that and set:

<param name="timer-T1X64" value="1000" />

This works the way I want and INVITEs to dead gateways are
disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
1. It affects all calls sent via this "custom timer" profile
2. I need to use one more SIP port to bind to (I need keep the
"default timers" profile as well)
3. The timer is used by transactions other than the initial invite
message and that may cause unexpected problems.

Obviously originate_timeout doesn't work the way it's supposed to
(according to the wiki) so I will report it as bug.

The profile trick above actually solves my problem and may happen to
be a better solution after all, but I will need to test it for some
time before I know.

On Tue, Aug 18, 2009 at 9:00 PM, Brian West<brian at freeswitch.org> wrote:
> use TCP then if you get an ICMP unreachable it'll move on instantly.
>
> /b
>
> On Aug 18, 2009, at 12:48 PM, Hristo Trendev wrote:
>
>> progress_timeout will wait for media (tried it, but I don't need
>> that). I want to detect the case when the destination gateway is down
>> and there is no response (not even 100, 407, etc) to the initial
>> INVITE sent by FS.
>>
>> According to the wiki, this is exactly what originate_timeout is used
>> for. Actually the wiki gives as example for originate_timeout exactly
>> what I'm trying to accomplish.
>>
>> It seems to me that FS ignores 100 and alike messages, which are sent
>> as response to initial INVITE and doesn't cancel originate_timeout
>> timer if such message is received.
>>
>> The more I look into this, the more I start to think that it's a bug.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Tue Aug 18 11:30:56 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Aug 2009 13:30:56 -0500
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
Message-ID: <A7B2812C-DA99-4284-AC44-9BCD5C78E967@freeswitch.org>

setup gateways and turn on the ping option.
/b

On Aug 18, 2009, at 1:24 PM, Hristo Trendev wrote:

> TCP is not really an option for me. I have tried using different sofia
> profile for that and set:
>
> <param name="timer-T1X64" value="1000" />
>
> This works the way I want and INVITEs to dead gateways are
> disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
> 1. It affects all calls sent via this "custom timer" profile
> 2. I need to use one more SIP port to bind to (I need keep the
> "default timers" profile as well)
> 3. The timer is used by transactions other than the initial invite
> message and that may cause unexpected problems.
>
> Obviously originate_timeout doesn't work the way it's supposed to
> (according to the wiki) so I will report it as bug.
>
> The profile trick above actually solves my problem and may happen to
> be a better solution after all, but I will need to test it for some
> time before I know.



From mrene_lists at avgs.ca  Tue Aug 18 11:32:59 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 18 Aug 2009 14:32:59 -0400
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
Message-ID: <C01FD4EC-8F34-4C64-8EE7-C4E18276B2C3@avgs.ca>

Hi,

originate_timeout does exactly what it is supposed to do, cancel the  
new call if originate doesnt return within X seconds.
In my opinion, I would keep state wether the gateway is alive or not,  
and have a reasonable t1x64 timeout value (5000 should be enough).
This way it'll only timeout the first time the call is attempted.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 18-Aug-09, at 2:24 PM, Hristo Trendev wrote:

> TCP is not really an option for me. I have tried using different sofia
> profile for that and set:
>
> <param name="timer-T1X64" value="1000" />
>
> This works the way I want and INVITEs to dead gateways are
> disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
> 1. It affects all calls sent via this "custom timer" profile
> 2. I need to use one more SIP port to bind to (I need keep the
> "default timers" profile as well)
> 3. The timer is used by transactions other than the initial invite
> message and that may cause unexpected problems.
>
> Obviously originate_timeout doesn't work the way it's supposed to
> (according to the wiki) so I will report it as bug.
>
> The profile trick above actually solves my problem and may happen to
> be a better solution after all, but I will need to test it for some
> time before I know.
>
> On Tue, Aug 18, 2009 at 9:00 PM, Brian West<brian at freeswitch.org>  
> wrote:
>> use TCP then if you get an ICMP unreachable it'll move on instantly.
>>
>> /b
>>
>> On Aug 18, 2009, at 12:48 PM, Hristo Trendev wrote:
>>
>>> progress_timeout will wait for media (tried it, but I don't need
>>> that). I want to detect the case when the destination gateway is  
>>> down
>>> and there is no response (not even 100, 407, etc) to the initial
>>> INVITE sent by FS.
>>>
>>> According to the wiki, this is exactly what originate_timeout is  
>>> used
>>> for. Actually the wiki gives as example for originate_timeout  
>>> exactly
>>> what I'm trying to accomplish.
>>>
>>> It seems to me that FS ignores 100 and alike messages, which are  
>>> sent
>>> as response to initial INVITE and doesn't cancel originate_timeout
>>> timer if such message is received.
>>>
>>> The more I look into this, the more I start to think that it's a  
>>> bug.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dist.lists at gmail.com  Tue Aug 18 11:40:56 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Tue, 18 Aug 2009 21:40:56 +0300
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <A7B2812C-DA99-4284-AC44-9BCD5C78E967@freeswitch.org>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
	<A7B2812C-DA99-4284-AC44-9BCD5C78E967@freeswitch.org>
Message-ID: <2a73afe0908181140j5371cea8p79e51f2139646806@mail.gmail.com>

I've seen this parameter when I started testing, but the name suggest
that it will do icmp ping to detect failed gateways (haven't really
tried it though). I need to detect sip service failure as well - the
case where the destination server is up (ping replies are received),
but the sip process is down and there is no response to sip requests.

If the ping parameter uses sip-based ping (like OPTIONS ping) this may
be a solution for me as well.

On Tue, Aug 18, 2009 at 9:30 PM, Brian West<brian at freeswitch.org> wrote:
> setup gateways and turn on the ping option.
> /b
>
> On Aug 18, 2009, at 1:24 PM, Hristo Trendev wrote:
>
>> TCP is not really an option for me. I have tried using different sofia
>> profile for that and set:
>>
>> <param name="timer-T1X64" value="1000" />
>>
>> This works the way I want and INVITEs to dead gateways are
>> disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
>> 1. It affects all calls sent via this "custom timer" profile
>> 2. I need to use one more SIP port to bind to (I need keep the
>> "default timers" profile as well)
>> 3. The timer is used by transactions other than the initial invite
>> message and that may cause unexpected problems.
>>
>> Obviously originate_timeout doesn't work the way it's supposed to
>> (according to the wiki) so I will report it as bug.
>>
>> The profile trick above actually solves my problem and may happen to
>> be a better solution after all, but I will need to test it for some
>> time before I know.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mrene_lists at avgs.ca  Tue Aug 18 11:42:36 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 18 Aug 2009 14:42:36 -0400
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <2a73afe0908181140j5371cea8p79e51f2139646806@mail.gmail.com>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
	<A7B2812C-DA99-4284-AC44-9BCD5C78E967@freeswitch.org>
	<2a73afe0908181140j5371cea8p79e51f2139646806@mail.gmail.com>
Message-ID: <E4D7F0B9-8036-4187-99F5-171A507230F6@avgs.ca>

It does use SIP OPTIONS.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 18-Aug-09, at 2:40 PM, Hristo Trendev wrote:

> I've seen this parameter when I started testing, but the name suggest
> that it will do icmp ping to detect failed gateways (haven't really
> tried it though). I need to detect sip service failure as well - the
> case where the destination server is up (ping replies are received),
> but the sip process is down and there is no response to sip requests.
>
> If the ping parameter uses sip-based ping (like OPTIONS ping) this may
> be a solution for me as well.
>
> On Tue, Aug 18, 2009 at 9:30 PM, Brian West<brian at freeswitch.org>  
> wrote:
>> setup gateways and turn on the ping option.
>> /b
>>
>> On Aug 18, 2009, at 1:24 PM, Hristo Trendev wrote:
>>
>>> TCP is not really an option for me. I have tried using different  
>>> sofia
>>> profile for that and set:
>>>
>>> <param name="timer-T1X64" value="1000" />
>>>
>>> This works the way I want and INVITEs to dead gateways are
>>> disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
>>> 1. It affects all calls sent via this "custom timer" profile
>>> 2. I need to use one more SIP port to bind to (I need keep the
>>> "default timers" profile as well)
>>> 3. The timer is used by transactions other than the initial invite
>>> message and that may cause unexpected problems.
>>>
>>> Obviously originate_timeout doesn't work the way it's supposed to
>>> (according to the wiki) so I will report it as bug.
>>>
>>> The profile trick above actually solves my problem and may happen to
>>> be a better solution after all, but I will need to test it for some
>>> time before I know.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Tue Aug 18 11:45:11 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Aug 2009 13:45:11 -0500
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <2a73afe0908181140j5371cea8p79e51f2139646806@mail.gmail.com>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
	<A7B2812C-DA99-4284-AC44-9BCD5C78E967@freeswitch.org>
	<2a73afe0908181140j5371cea8p79e51f2139646806@mail.gmail.com>
Message-ID: <11870622-2039-412A-ADB8-D58F04F4F737@freeswitch.org>

Why would it make you think its ICMP?  Weird.  Its a sip based options  
ping.

/b

On Aug 18, 2009, at 1:40 PM, Hristo Trendev wrote:

> I've seen this parameter when I started testing, but the name suggest
> that it will do icmp ping to detect failed gateways (haven't really
> tried it though). I need to detect sip service failure as well - the
> case where the destination server is up (ping replies are received),
> but the sip process is down and there is no response to sip requests.
>
> If the ping parameter uses sip-based ping (like OPTIONS ping) this may
> be a solution for me as well.



From dist.lists at gmail.com  Tue Aug 18 11:50:01 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Tue, 18 Aug 2009 21:50:01 +0300
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <C01FD4EC-8F34-4C64-8EE7-C4E18276B2C3@avgs.ca>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
	<C01FD4EC-8F34-4C64-8EE7-C4E18276B2C3@avgs.ca>
Message-ID: <2a73afe0908181150o3d868f0fgc18f1129c9cee002@mail.gmail.com>

That's exactly what I'm trying to do. This used to be implemented with
openser and on timeout it will mark destination as inactive so
subsequent calls don't get sent to dead gateways.

If originate_timeout is intended for something else, then maybe the
example given in the wiki can be changed, because may be a bit
misleading for someone trying to do failover.

On Tue, Aug 18, 2009 at 9:32 PM, Mathieu Rene<mrene_lists at avgs.ca> wrote:
> Hi,
>
> originate_timeout does exactly what it is supposed to do, cancel the
> new call if originate doesnt return within X seconds.
> In my opinion, I would keep state wether the gateway is alive or not,
> and have a reasonable t1x64 timeout value (5000 should be enough).
> This way it'll only timeout the first time the call is attempted.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 18-Aug-09, at 2:24 PM, Hristo Trendev wrote:
>
>> TCP is not really an option for me. I have tried using different sofia
>> profile for that and set:
>>
>> <param name="timer-T1X64" value="1000" />
>>
>> This works the way I want and INVITEs to dead gateways are
>> disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
>> 1. It affects all calls sent via this "custom timer" profile
>> 2. I need to use one more SIP port to bind to (I need keep the
>> "default timers" profile as well)
>> 3. The timer is used by transactions other than the initial invite
>> message and that may cause unexpected problems.
>>
>> Obviously originate_timeout doesn't work the way it's supposed to
>> (according to the wiki) so I will report it as bug.
>>
>> The profile trick above actually solves my problem and may happen to
>> be a better solution after all, but I will need to test it for some
>> time before I know.
>>
>> On Tue, Aug 18, 2009 at 9:00 PM, Brian West<brian at freeswitch.org>
>> wrote:
>>> use TCP then if you get an ICMP unreachable it'll move on instantly.
>>>
>>> /b
>>>
>>> On Aug 18, 2009, at 12:48 PM, Hristo Trendev wrote:
>>>
>>>> progress_timeout will wait for media (tried it, but I don't need
>>>> that). I want to detect the case when the destination gateway is
>>>> down
>>>> and there is no response (not even 100, 407, etc) to the initial
>>>> INVITE sent by FS.
>>>>
>>>> According to the wiki, this is exactly what originate_timeout is
>>>> used
>>>> for. Actually the wiki gives as example for originate_timeout
>>>> exactly
>>>> what I'm trying to accomplish.
>>>>
>>>> It seems to me that FS ignores 100 and alike messages, which are
>>>> sent
>>>> as response to initial INVITE and doesn't cancel originate_timeout
>>>> timer if such message is received.
>>>>
>>>> The more I look into this, the more I start to think that it's a
>>>> bug.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From dist.lists at gmail.com  Tue Aug 18 11:51:42 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Tue, 18 Aug 2009 21:51:42 +0300
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <E4D7F0B9-8036-4187-99F5-171A507230F6@avgs.ca>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
	<A7B2812C-DA99-4284-AC44-9BCD5C78E967@freeswitch.org>
	<2a73afe0908181140j5371cea8p79e51f2139646806@mail.gmail.com>
	<E4D7F0B9-8036-4187-99F5-171A507230F6@avgs.ca>
Message-ID: <2a73afe0908181151v6bb638bx70b86ffd30d3276b@mail.gmail.com>

Thanks. I guess I will try that as well and see which one works better
for me then.

On Tue, Aug 18, 2009 at 9:42 PM, Mathieu Rene<mrene_lists at avgs.ca> wrote:
> It does use SIP OPTIONS.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 18-Aug-09, at 2:40 PM, Hristo Trendev wrote:
>
>> I've seen this parameter when I started testing, but the name suggest
>> that it will do icmp ping to detect failed gateways (haven't really
>> tried it though). I need to detect sip service failure as well - the
>> case where the destination server is up (ping replies are received),
>> but the sip process is down and there is no response to sip requests.
>>
>> If the ping parameter uses sip-based ping (like OPTIONS ping) this may
>> be a solution for me as well.
>>
>> On Tue, Aug 18, 2009 at 9:30 PM, Brian West<brian at freeswitch.org>
>> wrote:
>>> setup gateways and turn on the ping option.
>>> /b
>>>
>>> On Aug 18, 2009, at 1:24 PM, Hristo Trendev wrote:
>>>
>>>> TCP is not really an option for me. I have tried using different
>>>> sofia
>>>> profile for that and set:
>>>>
>>>> <param name="timer-T1X64" value="1000" />
>>>>
>>>> This works the way I want and INVITEs to dead gateways are
>>>> disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
>>>> 1. It affects all calls sent via this "custom timer" profile
>>>> 2. I need to use one more SIP port to bind to (I need keep the
>>>> "default timers" profile as well)
>>>> 3. The timer is used by transactions other than the initial invite
>>>> message and that may cause unexpected problems.
>>>>
>>>> Obviously originate_timeout doesn't work the way it's supposed to
>>>> (according to the wiki) so I will report it as bug.
>>>>
>>>> The profile trick above actually solves my problem and may happen to
>>>> be a better solution after all, but I will need to test it for some
>>>> time before I know.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From alan at chandlerfamily.org.uk  Tue Aug 18 11:59:29 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Tue, 18 Aug 2009 19:59:29 +0100
Subject: [Freeswitch-users] Stuck on my first attempt at dialplanning
Message-ID: <4A8AFA11.2080603@chandlerfamily.org.uk>

This is my first attempt at setting up the dial plan to do anything more 
than the basic default stuff.  What I am trying to do is set up some 
conferencing as follows

1. A calls B
2. B puts both A and B into a conference
3. A or B calls out from the conference (via the * set in caller 
controls) to C
4a If C hangs up or rejects the call, A/B should go back to the conference
4b If A/B types *1 before the call is answered he should go back to the 
conference
4c If A/B types *1 after the call is answered both A/B and C should go 
back into the conference.




It nearly all works except

i) A seemingly can't use the * caller control, where as B can.  The 
other caller control I set up (mute) seems to work

ii) If C rejects the call before answering, B gets hung_up (because of 
i) I can only test with B)

iii) When B gets back to the conference, the * caller control doesn't 
work (although the mute does)

Can someone tell me where I am going wrong.


----------------------

Here is the dialplan snippet that I use to handle the dialout from the 
conference (pressing the * in step 3)

     <!-- Conference Callout To Another Extension -->
     <extension name="cco">
       <condition field="destination_number" expression="^cco$">
         <action application="read" data="3 3 
'tone_stream://%(10000,0,350,440)' digits 5000 #"/>
         <action application="bind_meta_app" data="2 a s 
execute_extension::conf_transfer XML features" />
         <action application="set" data="hangup_after_bridge=false" />
         <action application="bridge" 
data="user/${digits}@${domain_name}" />
         <action application="info" />
          <action application="conference" 
data="${conference_name}@default" />
       </condition>
     </extension>

     <extension name="conf_transfer">
       <condition field="destination_number" expression="^conf_transfer$" >
         <action application="answer" />
         <action application="transfer" data="-both ${conference_name} 
XML default" />
       </condition>
     </extension>


And here is the bit in my default XML where I define the conference

     <extension name="nb_conferences">
       <condition field="destination_number" 
expression="^(2(0[1-9]|[1-9][0-9]))$">
	<action application="answer"/>
	<action application="conference" data="$1 at default"/>
       </condition>
     </extension>

The call out to the conference from the original call is done via 
changing the bind meta app in the default dialplan for local extension

	<action application="bind_meta_app" data="2 ab s execute_extension::cf 
XML features"/>

and this extension in features

     <!-- Used to transfer both legs into a conference -->
     <extension name="cf">
       <condition field="destination_number" expression="^cf$">
	<action application="answer"/>
	<action application="transfer" data="-both 2${dialed_extension:1} XML 
default"/>
       </condition>
     </extension>








I can get as far as talking to C after dialing him, but if he doesn't 
answer, or rejects the call I get lost in space somewhere.  If I type *1 
during the call with C I get back to the conference, and C appears to be 
  in the conference (hear him speak) but he gets music on hold.

What am I doing wrong?



-- 
Alan Chandler
http://www.chandlerfamily.org.uk



From alan at chandlerfamily.org.uk  Tue Aug 18 12:53:03 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Tue, 18 Aug 2009 20:53:03 +0100
Subject: [Freeswitch-users] Stuck on my first attempt at dialplanning
In-Reply-To: <4A8AFA11.2080603@chandlerfamily.org.uk>
References: <4A8AFA11.2080603@chandlerfamily.org.uk>
Message-ID: <4A8B069F.9030404@chandlerfamily.org.uk>

Ignore the very last but one paragraph of this e-mail - that's the text 
from an old attempt before I fixed some things before sending this e-mail



Alan Chandler wrote:
> This is my first attempt at setting up the dial plan to do anything more 
> than the basic default stuff.  What I am trying to do is set up some 
> conferencing as follows
> 
> 1. A calls B
> 2. B puts both A and B into a conference
> 3. A or B calls out from the conference (via the * set in caller 
> controls) to C
> 4a If C hangs up or rejects the call, A/B should go back to the conference
> 4b If A/B types *1 before the call is answered he should go back to the 
> conference
> 4c If A/B types *1 after the call is answered both A/B and C should go 
> back into the conference.
> 
> 
> 
> 
> It nearly all works except
> 
> i) A seemingly can't use the * caller control, where as B can.  The 
> other caller control I set up (mute) seems to work
> 
> ii) If C rejects the call before answering, B gets hung_up (because of 
> i) I can only test with B)
> 
> iii) When B gets back to the conference, the * caller control doesn't 
> work (although the mute does)
> 
> Can someone tell me where I am going wrong.
> 
> 
> ----------------------
> 
> Here is the dialplan snippet that I use to handle the dialout from the 
> conference (pressing the * in step 3)
> 
>      <!-- Conference Callout To Another Extension -->
>      <extension name="cco">
>        <condition field="destination_number" expression="^cco$">
>          <action application="read" data="3 3 
> 'tone_stream://%(10000,0,350,440)' digits 5000 #"/>
>          <action application="bind_meta_app" data="2 a s 
> execute_extension::conf_transfer XML features" />
>          <action application="set" data="hangup_after_bridge=false" />
>          <action application="bridge" 
> data="user/${digits}@${domain_name}" />
>          <action application="info" />
>           <action application="conference" 
> data="${conference_name}@default" />
>        </condition>
>      </extension>
> 
>      <extension name="conf_transfer">
>        <condition field="destination_number" expression="^conf_transfer$" >
>          <action application="answer" />
>          <action application="transfer" data="-both ${conference_name} 
> XML default" />
>        </condition>
>      </extension>
> 
> 
> And here is the bit in my default XML where I define the conference
> 
>      <extension name="nb_conferences">
>        <condition field="destination_number" 
> expression="^(2(0[1-9]|[1-9][0-9]))$">
> 	<action application="answer"/>
> 	<action application="conference" data="$1 at default"/>
>        </condition>
>      </extension>
> 
> The call out to the conference from the original call is done via 
> changing the bind meta app in the default dialplan for local extension
> 
> 	<action application="bind_meta_app" data="2 ab s execute_extension::cf 
> XML features"/>
> 
> and this extension in features
> 
>      <!-- Used to transfer both legs into a conference -->
>      <extension name="cf">
>        <condition field="destination_number" expression="^cf$">
> 	<action application="answer"/>
> 	<action application="transfer" data="-both 2${dialed_extension:1} XML 
> default"/>
>        </condition>
>      </extension>
> 
> 
> 
> 
> 
> 
> 
> 
> I can get as far as talking to C after dialing him, but if he doesn't 
> answer, or rejects the call I get lost in space somewhere.  If I type *1 
> during the call with C I get back to the conference, and C appears to be 
>   in the conference (hear him speak) but he gets music on hold.
> 
> What am I doing wrong?
> 
> 
> 


-- 
Alan Chandler
http://www.chandlerfamily.org.uk



From christian.loeschenkohl at xpirio.com  Tue Aug 18 14:41:41 2009
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Tue, 18 Aug 2009 23:41:41 +0200
Subject: [Freeswitch-users] send sip options message
Message-ID: <4A8B2015.2060605@xpirio.com>

hi

does anybody know how to send a sip options message to a registered user, using the event socket
or something else build in freeswitch
i think the ping parameter does something like this for gateways.

what i want/need is the same thing that is provided in asterisk with the qualifying option,
to see how "reachable" a certain client is.

br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com


From alan at chandlerfamily.org.uk  Tue Aug 18 16:30:40 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Wed, 19 Aug 2009 00:30:40 +0100
Subject: [Freeswitch-users] Conference Status
Message-ID: <4A8B39A0.4010206@chandlerfamily.org.uk>

In the command to lists a conference it talks about member status, shown 
as hear|speak|floor

What does the floor mean as a status - I can't find any reference to it 
anywhere.


-- 
Alan Chandler
http://www.chandlerfamily.org.uk



From msc at freeswitch.org  Tue Aug 18 17:10:32 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Aug 2009 17:10:32 -0700
Subject: [Freeswitch-users] Conference Status
In-Reply-To: <4A8B39A0.4010206@chandlerfamily.org.uk>
References: <4A8B39A0.4010206@chandlerfamily.org.uk>
Message-ID: <87f2f3b90908181710i4c645f06gedc00689da979677@mail.gmail.com>

On Tue, Aug 18, 2009 at 4:30 PM, Alan Chandler
<alan at chandlerfamily.org.uk>wrote:

> In the command to lists a conference it talks about member status, shown
> as hear|speak|floor
>
> What does the floor mean as a status - I can't find any reference to it
> anywhere.
>

I just spoke with Mike J about this. The floor is basically "who has the
floor" in the conference. In most cases you'll see that it is the person who
is talking or who spoke last. He said there's some algorithmic magic in
there but we don't actually use floor for much of anything...

-MC
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From mattdfong at gmail.com  Wed Aug 19 01:43:29 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Wed, 19 Aug 2009 01:43:29 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
Message-ID: <4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>

Hi Eric,
Thanks for these recommendations.

for vmd-not-panasonic-home-ans.wav changing MAX_FREQ to 1450 WORKED!

but I'm still having problems picking out the ~750Hz beep of sprint,
tmobile, and verizon. I tried first cutting POINTS and VALID in half, then
in half again, while also reducing MIN_AMPL in half but still no luck. I
assumed from the descriptions of each, that reducing the numbers would make
the algorithm less picky at finding a beep. Is this correct?

Any other recommendations on picking up these ~750Hz beeps? Thanks again for
the help.

--matt

On Mon, Aug 17, 2009 at 2:15 PM, Eric des Courtis <
eric.des.courtis at gmail.com> wrote:

> Matt,
>
> Okay the good news is vmd should be able to handle these cases. The
> bad news is for whatever reason they are not getting detected at the
> moment.
>
> vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
> MAX_FREQ to 1450 and play with MIN_AMPL  if that still doesn't help.


> The following seem to use the same beep:
>
> vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>
> You can try to play with these values:
>
> POINTS 32
> VALID 22
> MAX_CHIRP 22
>
> If that doesn't work let me know I will try to improve the algorithm
> to detect the providers.
>
> Cheers!
>
> Eric des Courtis
>
>
>
> On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong at gmail.com> wrote:
> > Hi Eric,
> > Thanks for the response. I had tried emailing you @brenbria.com and the
> > email had bounced, thanks for responding to my mail.
> > If you'd be interested I .zipped up my sample voicemail beeps
> > at http://bandcon.hellohunter.com/vmd_wav.zip
> > I'm relatively new to telephony, but can you point me in the right
> direction
> > for figuring out if the beeps are sinewaves. About as far as I've come
> with
> > audio is being able to open the .wav files in audacity. Any website
> >  recommendations I can read? Thanks so much.
> > --matt
> >
> > On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
> > <eric.des.courtis at gmail.com> wrote:
> >>
> >> Matt,
> >>
> >> You must first capture the audio beeps and verify that they are sine
> >> waves. If not, simply tweaking the algorithm will not give you better
> >> results.
> >>
> >> It might be possible to use FFT and I would be happy to help you
> >> implement such a solution but keep in mind FFT is very very demanding
> >> on the hardware. Ideally what you want to find out is what functions
> >> was use to generate the beep in the first place so that it can be
> >> detected. Is it two sines waves like in DTMF? Or something more
> >> complex?
> >>
> >> Anyway my email is eric.des.courtis at benbria.ca.
> >>
> >> Cheers.
> >>
> >> Eric des Courtis
> >>
> >> On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong at gmail.com>
> wrote:
> >> > I tried emailed Eric, seeking advice on this, but his email (the one
> in
> >> > the
> >> > source code) is bouncing email (invalid user), so thought I would ask
> >> > here
> >> > instead. If anyone has eric's new email address, I'd be interesting in
> >> > it.
> >> >
> >> > I did some tests with mod_vmd this afternoon, but I'm only finding
> about
> >> > 33%
> >> > of the voice mail beeps and did have 1 false-positive in my test of
> >> > 7 voice
> >> > mail machines. I've recorded the audio of the session in .wav files
> that
> >> > were both successful and not, as a comparison. I can upload the .wav
> >> > files
> >> > if they would be useful.
> >> > mod_vmd works great for voicemails of Skype Users, and kall8.com, but
> >> > has
> >> > issues dealing with mobile phone carriers.
> >> > sprint - not successful
> >> > tmobile - not successful
> >> > verizon - not successful
> >> > panasonic home answering machine system - not successful
> >> > kall8 - SUCCESS
> >> > skype - SUCCESS
> >> > I'm wondering if you can recommend a simple fix, like changing some of
> >> > the
> >> > constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c
> source
> >> > file, or if better success requires more complex analysis.  Do you
> have
> >> > any
> >> > recommendations on how this might be done? Listening to the .wav's
> >> > its apparent the beeps are not as loud for the mobile phone carriers
> as
> >> > they
> >> > are with skype and kall8. Any guidance would be greatly appreciated.
> >> > --matt
> >> > hello hunter
> >> > http://www.hellohunter.com
> >> > voice broadcasting & hosted dialer
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >
> >
>
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From rdenert at tng.de  Wed Aug 19 02:17:26 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Wed, 19 Aug 2009 11:17:26 +0200 (CEST)
Subject: [Freeswitch-users] "mute" channel programmatically with
 mod_event_socket
In-Reply-To: <28651154.332401250673051662.JavaMail.root@zimbra.tng.de>
Message-ID: <17888468.332601250673446046.JavaMail.root@zimbra.tng.de>

You can use the caller controlls in the conference.conf.xml to implement your own features something like mute or kick. Or do you want to mute mute other conference members like a moderator can do this.

BR

----- Urspr?ngliche Mail -----
Von: "Artem Shiyanov" <shiyanov at gmail.com>
An: freeswitch-users at lists.freeswitch.org
Gesendet: Dienstag, 18. August 2009 14:54:04 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: [Freeswitch-users] "mute" channel programmatically with mod_event_socket


Hello all! 

I'm trying to implement "mute" feature with mod_event_socket: I want programmatically mute/unmute some channel in a call.. And I don't see any other ways except to use conference room with special rule "mute". 
Anybody knows the better way? 


Thanks, 
Artem 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


From shiyanov at gmail.com  Wed Aug 19 04:13:15 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Wed, 19 Aug 2009 15:13:15 +0400
Subject: [Freeswitch-users] "mute" channel programmatically with
	mod_event_socket
In-Reply-To: <17888468.332601250673446046.JavaMail.root@zimbra.tng.de>
References: <28651154.332401250673051662.JavaMail.root@zimbra.tng.de>
	<17888468.332601250673446046.JavaMail.root@zimbra.tng.de>
Message-ID: <d211f54a0908190413p17311b15u42f57dd3417ec634@mail.gmail.com>

The point is - a simple call flow is desired- if I have a ordinary 1-to-1
call and one of the participators decides to mute call - I don't want to put
both channels into a conference room but it looks like I have no other
choices.

BUT:
I found brilliant app - eavesdrop! If I do this for one-to-one call - mute
works!
SendMsg <uuid_channel_making_mute>
call-command: execute
execute-app-name: eavesdrop
execute-app-arg: <uuid_participator>

But the problem appears when I want to unmute.. the call! I've tried to
re-bridge channels, intercept them- nothing happens- one channel (muted one)
doesn't hear the participator. And CLI command 'show channels' shows that
channel with uui=<uuid_channel_making_mute> still process eavesdrop app.
Maybe someone know how to switch off eavesdrop app?


Artem


On Wed, Aug 19, 2009 at 1:17 PM, Rudolf Denert <rdenert at tng.de> wrote:

> You can use the caller controlls in the conference.conf.xml to implement
> your own features something like mute or kick. Or do you want to mute mute
> other conference members like a moderator can do this.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Artem Shiyanov" <shiyanov at gmail.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Dienstag, 18. August 2009 14:54:04 GMT +01:00
> Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
> Betreff: [Freeswitch-users] "mute" channel programmatically with
> mod_event_socket
>
>
> Hello all!
>
> I'm trying to implement "mute" feature with mod_event_socket: I want
> programmatically mute/unmute some channel in a call.. And I don't see any
> other ways except to use conference room with special rule "mute".
> Anybody knows the better way?
>
>
> Thanks,
> Artem
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bruce.mcalister at blueface.ie  Wed Aug 19 05:53:05 2009
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Wed, 19 Aug 2009 13:53:05 +0100
Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 &
 v1.0.4)
In-Reply-To: <4A891472.5060302@blueface.ie>
References: <4A8280B8.6050308@blueface.ie> <4A891472.5060302@blueface.ie>
Message-ID: <4A8BF5B1.2090801@blueface.ie>

Hi All,

JIRA FSBUILD-186 BugID has been logged for this issue.

Thanks
Bruce

Bruce McAlister wrote:
> Hi All,
> 
> Shall I log a JIRA for this issue?
> 
> Thanks
> Bruce
> 
> Bruce McAlister wrote:
>> Hi All,
>>
>> I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and 
>> 1.0.4.
>>
>> I am running on Solaris 10 Update 5 on x86 hardware (32-bit).
>>
>> The build fails with:
>>
>> --- snip ---
>> make: Fatal error: Command failed for target `all-recursive'
>> Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4
>> *** Error code 1
>> make: Fatal error: Command failed for target `all'
>> ---
>>
>> Looking back through the build I can see the following error:
>>
>> --- snip ---
>> creating libfreeswitch.la
>> (cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la 
>> libfreeswitch.la)
>> /usr/bin/cc 
>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
>> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
>> -DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch 
>> freeswitch-switch.o  ./.libs/libfreeswitch.so 
>> -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib 
>> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib/.libs/libexpat.a 
>> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libapr-1.a 
>> -lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp 
>> -L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt 
>> -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl 
>> -lcurses -lsocket  -R/opt/freeswitch/lib -R/usr/sfw/lib
>> Undefined                       first referenced
>>   symbol                             in file
>> herror                              ./.libs/libfreeswitch.so
>> ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch
>> *** Error code 1
>> The following command caused the error:
>> `if test -z "" ; then echo /bin/bash 
>> /export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else 
>> echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` 
>> --tag=CC --mode=link /usr/bin/cc 
>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
>> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
>> -DPATH_MAX=2048 -g -v -Xc -xc99=all  -lm -R/opt/freeswitch/lib  -o 
>> freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib 
>> freeswitch-switch.o   libfreeswitch.la libs/apr/libapr-1.la 
>> libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl 
>> -lcrypto -lsocket -lnsl -ldl   -lcurses -lsocket
>> --- snip ---
>>
>> Then a little above this error, there is the following warning that is 
>> displayed (I'm not sure if it is related):
>>
>> --- snip ---
>> *** Warning: Linking the shared library libfreeswitch.la against the
>> *** static library libs/libedit/src/.libs/libedit.a is not portable!
>> --- snip ---
>>
>> My configure line is as follows:
>>
>> ---
>> ./configure --prefix=/opt/freeswitch
>> ---
>>
>> I have the complete configure and make output if anyone needs them.
>>
>> Any help/pointers would be greatly appreciated.
>>
>> Thanks
>> Bruce
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 


From mrene_lists at avgs.ca  Wed Aug 19 06:02:19 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 19 Aug 2009 09:02:19 -0400
Subject: [Freeswitch-users] "mute" channel programmatically with
	mod_event_socket
In-Reply-To: <d211f54a0908190413p17311b15u42f57dd3417ec634@mail.gmail.com>
References: <28651154.332401250673051662.JavaMail.root@zimbra.tng.de>
	<17888468.332601250673446046.JavaMail.root@zimbra.tng.de>
	<d211f54a0908190413p17311b15u42f57dd3417ec634@mail.gmail.com>
Message-ID: <36B2A0E3-6139-4724-A7F3-CAD89FB98688@avgs.ca>

Hi,

Eavesdrop kind of works yeah, you can use the intercept application to  
re-bridge the channels together, like:

SendMsg <uuid_channel_making_mute>
call-command: execute
execute-app-name: intercept
execute-app-arg: <uuid_participator>


The same can be done with the uuid_bridge api.

api uuid_bridge <uuid1> <uuid2>

If you want the cleaner way you could implement a media bug that  
replaces all the audio of the channel by silence, but that'd require  
some C coding.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 19-Aug-09, at 7:13 AM, Artem Shiyanov wrote:

> The point is - a simple call flow is desired- if I have a ordinary 1- 
> to-1 call and one of the participators decides to mute call - I  
> don't want to put both channels into a conference room but it looks  
> like I have no other choices.
>
> BUT:
> I found brilliant app - eavesdrop! If I do this for one-to-one call  
> - mute works!
> SendMsg <uuid_channel_making_mute>
> call-command: execute
> execute-app-name: eavesdrop
> execute-app-arg: <uuid_participator>
>
> But the problem appears when I want to unmute.. the call! I've tried  
> to re-bridge channels, intercept them- nothing happens- one channel  
> (muted one) doesn't hear the participator. And CLI command 'show  
> channels' shows that channel with uui=<uuid_channel_making_mute>  
> still process eavesdrop app. Maybe someone know how to switch off  
> eavesdrop app?
>
>
> Artem
>
>
> On Wed, Aug 19, 2009 at 1:17 PM, Rudolf Denert <rdenert at tng.de> wrote:
> You can use the caller controlls in the conference.conf.xml to  
> implement your own features something like mute or kick. Or do you  
> want to mute mute other conference members like a moderator can do  
> this.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Artem Shiyanov" <shiyanov at gmail.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Dienstag, 18. August 2009 14:54:04 GMT +01:00 Amsterdam/ 
> Berlin/Bern/Rom/Stockholm/Wien
> Betreff: [Freeswitch-users] "mute" channel programmatically with  
> mod_event_socket
>
>
> Hello all!
>
> I'm trying to implement "mute" feature with mod_event_socket: I want  
> programmatically mute/unmute some channel in a call.. And I don't  
> see any other ways except to use conference room with special rule  
> "mute".
> Anybody knows the better way?
>
>
> Thanks,
> Artem
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ivanov.maxim at gmail.com  Wed Aug 19 02:31:30 2009
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Wed, 19 Aug 2009 13:31:30 +0400
Subject: [Freeswitch-users] playback to legB before bridge?
Message-ID: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>

Hi all!
Is it possible to execute  playback  application to legB before
bridge? I mean sequence of actions similar to this:

1. originate legB
2. playback
3. bridge legA with legB using existing session from step 1 (no new
incoming call)


From t.mahe at telemaque.fr  Wed Aug 19 06:23:42 2009
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Wed, 19 Aug 2009 15:23:42 +0200
Subject: [Freeswitch-users] playback to legB before bridge?
In-Reply-To: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
References: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
Message-ID: <4A8BFCDE.4020700@telemaque.fr>

Hi Max,

Read the wiki, you'll find your answer

( here's a tip: http://wiki.freeswitch.org/wiki/Channel_Variables )

Have fun :)

Gled

Max Ivanov a ?crit :
> Hi all!
> Is it possible to execute  playback  application to legB before
> bridge? I mean sequence of actions similar to this:
>
> 1. originate legB
> 2. playback
> 3. bridge legA with legB using existing session from step 1 (no new
> incoming call)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From asannucci at gmail.com  Wed Aug 19 07:22:23 2009
From: asannucci at gmail.com (bakko)
Date: Wed, 19 Aug 2009 16:22:23 +0200
Subject: [Freeswitch-users] playback to legB before bridge?
In-Reply-To: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
References: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
Message-ID: <F0EB2D01BAF54938B619700CD0295D20@voztovoice>

http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_file

Look this variable.

I'd like know if exist similar variable for a-leg for use in the dialplan.

A call a number
A lissen a announce
A digit one or more DTMF 
The A channel is bridge with the B channel

Thank you in advance.

BR


From mrene_lists at avgs.ca  Wed Aug 19 07:29:04 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 19 Aug 2009 10:29:04 -0400
Subject: [Freeswitch-users] playback to legB before bridge?
In-Reply-To: <F0EB2D01BAF54938B619700CD0295D20@voztovoice>
References: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
	<F0EB2D01BAF54938B619700CD0295D20@voztovoice>
Message-ID: <34D37CC9-FE37-4B91-B82E-93E2312B5A1F@avgs.ca>

<action application="playback" data="file_here" />
<action application="bridge" data="dial string for b chan" />

or if you're talking about another B channel (I don't quite understand  
what you're trying to do here)
<action application="playback" data="file_here" />
<action application="intercept" data="<uuid here>" />

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 19-Aug-09, at 10:22 AM, bakko wrote:

> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_file
>
> Look this variable.
>
> I'd like know if exist similar variable for a-leg for use in the  
> dialplan.
>
> A call a number
> A lissen a announce
> A digit one or more DTMF
> The A channel is bridge with the B channel
>
> Thank you in advance.
>
> BR
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From asannucci at gmail.com  Wed Aug 19 07:48:12 2009
From: asannucci at gmail.com (bakko)
Date: Wed, 19 Aug 2009 16:48:12 +0200
Subject: [Freeswitch-users] playback to legB before bridge?
In-Reply-To: <34D37CC9-FE37-4B91-B82E-93E2312B5A1F@avgs.ca>
References: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com><F0EB2D01BAF54938B619700CD0295D20@voztovoice>
	<34D37CC9-FE37-4B91-B82E-93E2312B5A1F@avgs.ca>
Message-ID: <4EDE5C8143E7459AB02759B1AFFB7DC8@voztovoice>

before to bridge the call the a-leg (the caller) have to lissen a announce 
and then have to digit one or more dtmf.
If the dtmf are rights then bridge the caller with the callee, if not hangup 
the call.
I hope now my question is more clear :)
Sorry for my english.
BR 



From mrene_lists at avgs.ca  Wed Aug 19 07:51:05 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 19 Aug 2009 10:51:05 -0400
Subject: [Freeswitch-users] playback to legB before bridge?
In-Reply-To: <4EDE5C8143E7459AB02759B1AFFB7DC8@voztovoice>
References: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com><F0EB2D01BAF54938B619700CD0295D20@voztovoice>
	<34D37CC9-FE37-4B91-B82E-93E2312B5A1F@avgs.ca>
	<4EDE5C8143E7459AB02759B1AFFB7DC8@voztovoice>
Message-ID: <45F6928B-8DAE-4E6B-B801-D3909E0ABD58@avgs.ca>

Oh,

You can use embedded languages (like Lua or Javascript) to take  
control of the call and playback files / get dtmf digits, make another  
call, etc. Check on the wiki for samples.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 19-Aug-09, at 10:48 AM, bakko wrote:

> before to bridge the call the a-leg (the caller) have to lissen a  
> announce
> and then have to digit one or more dtmf.
> If the dtmf are rights then bridge the caller with the callee, if  
> not hangup
> the call.
> I hope now my question is more clear :)
> Sorry for my english.
> BR
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jmesquita at gmail.com  Wed Aug 19 09:27:17 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 19 Aug 2009 13:27:17 -0300
Subject: [Freeswitch-users] FreeSWITCH Console Released For iPhone/iPod
	Touch Users
In-Reply-To: <4A85A4E2.2030702@maxpowersoft.com>
References: <4A85A4E2.2030702@maxpowersoft.com>
Message-ID: <5a8712120908190927l15f83c6dm98d16952ad9daa37@mail.gmail.com>

No one said a thing but I really feel like initiatives like this
should be cheered.
Thank you for this app and thank you for making it free as well.

On 8/14/09, Chris Danielson <chris at maxpowersoft.com> wrote:
> Announcing the release of FreeSWITCH Console in the Apple Application
> Store.  The application is FREE and allows you to connect to a
> FreeSWITCH event socket layer module that is bound to an external
> interface.  Great for development purposes and general remote debugging.
>
> Blog announcement:
> http://www.chrisdanielson.com/2009/08/14/release-iphoneipod-touch-freeswitch-console/
>
> iTunes Store Link:
> http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221&mt=8
> <http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221&mt=8>
>
> Kind Regards,
> Chris Danielson
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From msc at freeswitch.org  Wed Aug 19 09:36:01 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Aug 2009 09:36:01 -0700
Subject: [Freeswitch-users] how to set different action for different
	cause code
In-Reply-To: <a44688a0908180923q4b4b8e48od86d676872c97c3a@mail.gmail.com>
References: <a44688a0908160424n3569cef1vad8fcdc68cbfb96e@mail.gmail.com>
	<87f2f3b90908170859p52259449q913b2ddab224709f@mail.gmail.com>
	<a44688a0908180923q4b4b8e48od86d676872c97c3a@mail.gmail.com>
Message-ID: <87f2f3b90908190936i50c12104vc252599fe2caf051@mail.gmail.com>

On Tue, Aug 18, 2009 at 9:23 AM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> I have my dialplan to do some simple routing. What I need to do is when
> certain hangup code is received, route advance to the next or the route
> after next based on the hangup code received.
>
> So, I have:
>
> <condition>
>    <action application="set" data="hangup_after_bridge=true"/>
>    <action application="set" data="continue_on_fail="false"/>
>    <action application="bridge" data="sofia/internal/aaa at 123.3.3./>
>    <action application="bridge" data="sofia/internal/bbb at 123.3.3./>
>    <action application="bridge" data="sofia/internal/ccc at 123.3.3./>
>    <action application="bridge" data="sofia/internal/ddd at 123.3.3./>
>
> </condition>
>
> What I want to do is that based on the error code received from ( bridge to
> aaa), it can route to either bbb or ccc depends on the hangup code received
> from aaa.
>
> So there anyway to do it in the dialplan without any scripting?


Yes, you can use the result of the previous bridge to execute a transfer.
You'd need to use the hangup cause as the transfer destination, or part of
the destination. For example, you could try something like this:

   <action application="set" data="hangup_after_bridge=true"/>
   <action application="set" data="continue_on_fail="false"/>
   <action application="bridge" data="sofia/internal/aaa at 123.3.3./>
   <action application="transfer" data="${hangup_cause}_AAA XML default"/>
...

And then have an extension for each of the various hangup_causes
<extension name="normal clearing">
  <condition field="destination_number"
expression="^(normal_clearing_AAA)$">
    <!-- Do whatever you want for "normal clearing" bridged calls -->
  </condition>
</extension>

<extension name="user busy">
  <condition field="destination_number" expression="^(user_busy_AAA)$">
    <!-- Do whatever you want for "user busy" bridged calls -->
  </condition>
</extension>

<extension name="no answer">
  <condition field="destination_number" expression="^(no_answer_AAA)$">
    <!-- Do whatever you want for "no answer" bridged calls -->
  </condition>
</extension>

Etc., etc.

I don't claim it's pretty, but it would let you avoid the use of scripting
languages. However, I would caution you to examine just why you want/need to
avoid scripting langs. The dp-only method may indeed be best for you, or
perhaps using mod_xml_curl would be better. It does depend on your specific
needs and your environment.

-MC


>
> thannks,
> woody
>
>
>
>
>
> On Mon, Aug 17, 2009 at 11:59 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Sun, Aug 16, 2009 at 4:24 AM, Woody Dickson <woodydickson at gmail.com>wrote:
>>
>>> Hello,
>>>
>>> I find hangup_hook, but I would like to define different actions for
>>> different hangup codes.  Is there anyway to do that?
>>>
>>>
>>
>> I can think of at least two ways you could do this: one that uses only the
>> dialplan and one that uses a script. If you don't mind using a scripting
>> language then you can make it very clean:
>>
>> <action application="set" data="api_hangup_hook=luarun chancleanup.lua
>> ${hangup_cause}"/>
>>
>> Then have your Lua script handle all the if-then-else or case stuff.
>>
>> Question: are you trying to transfer the a-leg to some other destination
>> if the b-leg hangup is a specific cause, or are you just doing some external
>> cleanup stuff? Just curious...
>>
>> -MC
>>
>>
>>> Woody
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From klaus.teller at gmx.net  Wed Aug 19 09:39:41 2009
From: klaus.teller at gmx.net (Klaus Teller)
Date: Wed, 19 Aug 2009 18:39:41 +0200
Subject: [Freeswitch-users] Passing Caller ID in Junction Network
Message-ID: <20090819163941.156100@gmx.net>

Hi,

Anybody with experience in passing caller ID number in junction network? I need some help.

I have the following origination string:

{origination_caller_id_number=1514xxxxxx,ignore_early_media=true}sofia/gateway/sip.jnctn.net/1514xxxxxx

Unfortunately, with this, junctionetwork won't pass the caller ID. Any idea?

Thanks,
Klaus.
-- 
Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 -
sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser


From msc at freeswitch.org  Wed Aug 19 09:41:16 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Aug 2009 09:41:16 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>
Message-ID: <87f2f3b90908190941m49c030e4w747e053e76f9b818@mail.gmail.com>

On Wed, Aug 19, 2009 at 1:43 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> Hi Eric,
> Thanks for these recommendations.
>
> for vmd-not-panasonic-home-ans.wav changing MAX_FREQ to 1450 WORKED!
>
> but I'm still having problems picking out the ~750Hz beep of sprint,
> tmobile, and verizon. I tried first cutting POINTS and VALID in half, then
> in half again, while also reducing MIN_AMPL in half but still no luck. I
> assumed from the descriptions of each, that reducing the numbers would make
> the algorithm less picky at finding a beep. Is this correct?
>
> Any other recommendations on picking up these ~750Hz beeps? Thanks again
> for the help.
>

How close are they to 750Hz? If they're not more than say +/- 16Hz then the
tone_detect app *should* be able to detect them. At the very least I would
try it. See if tone_detect can detect those beeps. While it may not be the
most elegant solution, having mod_vmd looking for one set of tones and
tone_detect looking for the 750Hz tones might actually get the job done, at
least until you and Eric can get together to see what's happening on the
750's.

-MC


>
> --matt
>
> On Mon, Aug 17, 2009 at 2:15 PM, Eric des Courtis <
> eric.des.courtis at gmail.com> wrote:
>
>> Matt,
>>
>> Okay the good news is vmd should be able to handle these cases. The
>> bad news is for whatever reason they are not getting detected at the
>> moment.
>>
>> vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
>> MAX_FREQ to 1450 and play with MIN_AMPL  if that still doesn't help.
>
>
>> The following seem to use the same beep:
>>
>> vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
>> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>>
>> You can try to play with these values:
>>
>> POINTS 32
>> VALID 22
>> MAX_CHIRP 22
>>
>> If that doesn't work let me know I will try to improve the algorithm
>> to detect the providers.
>>
>> Cheers!
>>
>> Eric des Courtis
>>
>>
>>
>> On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong at gmail.com> wrote:
>> > Hi Eric,
>> > Thanks for the response. I had tried emailing you @brenbria.com and the
>> > email had bounced, thanks for responding to my mail.
>> > If you'd be interested I .zipped up my sample voicemail beeps
>> > at http://bandcon.hellohunter.com/vmd_wav.zip
>> > I'm relatively new to telephony, but can you point me in the right
>> direction
>> > for figuring out if the beeps are sinewaves. About as far as I've come
>> with
>> > audio is being able to open the .wav files in audacity. Any website
>> >  recommendations I can read? Thanks so much.
>> > --matt
>> >
>> > On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
>> > <eric.des.courtis at gmail.com> wrote:
>> >>
>> >> Matt,
>> >>
>> >> You must first capture the audio beeps and verify that they are sine
>> >> waves. If not, simply tweaking the algorithm will not give you better
>> >> results.
>> >>
>> >> It might be possible to use FFT and I would be happy to help you
>> >> implement such a solution but keep in mind FFT is very very demanding
>> >> on the hardware. Ideally what you want to find out is what functions
>> >> was use to generate the beep in the first place so that it can be
>> >> detected. Is it two sines waves like in DTMF? Or something more
>> >> complex?
>> >>
>> >> Anyway my email is eric.des.courtis at benbria.ca.
>> >>
>> >> Cheers.
>> >>
>> >> Eric des Courtis
>> >>
>> >> On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong at gmail.com>
>> wrote:
>> >> > I tried emailed Eric, seeking advice on this, but his email (the one
>> in
>> >> > the
>> >> > source code) is bouncing email (invalid user), so thought I would ask
>> >> > here
>> >> > instead. If anyone has eric's new email address, I'd be interesting
>> in
>> >> > it.
>> >> >
>> >> > I did some tests with mod_vmd this afternoon, but I'm only finding
>> about
>> >> > 33%
>> >> > of the voice mail beeps and did have 1 false-positive in my test of
>> >> > 7 voice
>> >> > mail machines. I've recorded the audio of the session in .wav files
>> that
>> >> > were both successful and not, as a comparison. I can upload the .wav
>> >> > files
>> >> > if they would be useful.
>> >> > mod_vmd works great for voicemails of Skype Users, and kall8.com,
>> but
>> >> > has
>> >> > issues dealing with mobile phone carriers.
>> >> > sprint - not successful
>> >> > tmobile - not successful
>> >> > verizon - not successful
>> >> > panasonic home answering machine system - not successful
>> >> > kall8 - SUCCESS
>> >> > skype - SUCCESS
>> >> > I'm wondering if you can recommend a simple fix, like changing some
>> of
>> >> > the
>> >> > constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c
>> source
>> >> > file, or if better success requires more complex analysis.  Do you
>> have
>> >> > any
>> >> > recommendations on how this might be done? Listening to the .wav's
>> >> > its apparent the beeps are not as loud for the mobile phone carriers
>> as
>> >> > they
>> >> > are with skype and kall8. Any guidance would be greatly appreciated.
>> >> > --matt
>> >> > hello hunter
>> >> > http://www.hellohunter.com
>> >> > voice broadcasting & hosted dialer
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >
>> >
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Aug 19 09:49:24 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Aug 2009 09:49:24 -0700
Subject: [Freeswitch-users] Passing Caller ID in Junction Network
In-Reply-To: <20090819163941.156100@gmx.net>
References: <20090819163941.156100@gmx.net>
Message-ID: <87f2f3b90908190949q33751864l495e463769585b7d@mail.gmail.com>

On Wed, Aug 19, 2009 at 9:39 AM, Klaus Teller <klaus.teller at gmx.net> wrote:

> Hi,
>
> Anybody with experience in passing caller ID number in junction network? I
> need some help.
>
> I have the following origination string:
>
>
> {origination_caller_id_number=1514xxxxxx,ignore_early_media=true}sofia/gateway/
> sip.jnctn.net/1514xxxxxx
>
> Unfortunately, with this, junctionetwork won't pass the caller ID. Any
> idea?


Does Junction Networks pass any caller ID info at all?  Also, are you sure
that they support customized caller ID sending?
-MC


>
> Thanks,
> Klaus.
> --
> Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 -
> sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Aug 19 09:50:54 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 Aug 2009 11:50:54 -0500
Subject: [Freeswitch-users] Passing Caller ID in Junction Network
In-Reply-To: <20090819163941.156100@gmx.net>
References: <20090819163941.156100@gmx.net>
Message-ID: <89755F1E-97E4-4ECA-8A6D-9D1B92B062F1@freeswitch.org>

Can you show me a sip invite of an inbound call from them?  and an  
invite you're sending to them?

/b

On Aug 19, 2009, at 11:39 AM, Klaus Teller wrote:

> Hi,
>
> Anybody with experience in passing caller ID number in junction  
> network? I need some help.
>
> I have the following origination string:
>
> {origination_caller_id_number 
> =1514xxxxxx,ignore_early_media=true}sofia/gateway/sip.jnctn.net/ 
> 1514xxxxxx
>
> Unfortunately, with this, junctionetwork won't pass the caller ID.  
> Any idea?
>
> Thanks,
> Klaus.



From mrene_lists at avgs.ca  Wed Aug 19 09:46:59 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 19 Aug 2009 12:46:59 -0400
Subject: [Freeswitch-users] Passing Caller ID in Junction Network
In-Reply-To: <20090819163941.156100@gmx.net>
References: <20090819163941.156100@gmx.net>
Message-ID: <8B74E348-B4C4-4AA0-B3A5-B0AA395DCDFF@avgs.ca>

Hi,

I would send a sip trace to your carrier and/or ask them how they like  
to receive caller id information.

PS: Are you in Montreal?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 19-Aug-09, at 12:39 PM, Klaus Teller wrote:

> Hi,
>
> Anybody with experience in passing caller ID number in junction  
> network? I need some help.
>
> I have the following origination string:
>
> {origination_caller_id_number 
> =1514xxxxxx,ignore_early_media=true}sofia/gateway/sip.jnctn.net/ 
> 1514xxxxxx
>
> Unfortunately, with this, junctionetwork won't pass the caller ID.  
> Any idea?
>
> Thanks,
> Klaus.
> -- 
> Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla  
> Firefox 3 -
> sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From chris at maxpowersoft.com  Wed Aug 19 10:04:26 2009
From: chris at maxpowersoft.com (Chris Danielson)
Date: Wed, 19 Aug 2009 10:04:26 -0700
Subject: [Freeswitch-users] FreeSWITCH Console Released For iPhone/iPod
 Touch Users
In-Reply-To: <5a8712120908190927l15f83c6dm98d16952ad9daa37@mail.gmail.com>
References: <4A85A4E2.2030702@maxpowersoft.com>
	<5a8712120908190927l15f83c6dm98d16952ad9daa37@mail.gmail.com>
Message-ID: <4A8C309A.2060405@maxpowersoft.com>

Thanks!

Jo?o Mesquita wrote:
> No one said a thing but I really feel like initiatives like this
> should be cheered.
> Thank you for this app and thank you for making it free as well.
>
> On 8/14/09, Chris Danielson <chris at maxpowersoft.com> wrote:
>   
>> Announcing the release of FreeSWITCH Console in the Apple Application
>> Store.  The application is FREE and allows you to connect to a
>> FreeSWITCH event socket layer module that is bound to an external
>> interface.  Great for development purposes and general remote debugging.
>>
>> Blog announcement:
>> http://www.chrisdanielson.com/2009/08/14/release-iphoneipod-touch-freeswitch-console/
>>
>> iTunes Store Link:
>> http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221&mt=8
>> <http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221&mt=8>
>>
>> Kind Regards,
>> Chris Danielson
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>   
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From mike at jerris.com  Wed Aug 19 10:26:19 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 19 Aug 2009 13:26:19 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <87f2f3b90908190941m49c030e4w747e053e76f9b818@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>
	<87f2f3b90908190941m49c030e4w747e053e76f9b818@mail.gmail.com>
Message-ID: <8E7379D0-FB06-4C93-95B4-C493F8355A88@jerris.com>

my bet is if mod_vmd is not getting them that they are not going to  
work with tone detect either.  Someone needs to look at the tone and  
see what frequencies are really involved and if they change throughout  
the beep.

Mike

On Aug 19, 2009, at 12:41 PM, Michael Collins wrote:

>
>
> On Wed, Aug 19, 2009 at 1:43 AM, Matthew Fong <mattdfong at gmail.com>  
> wrote:
> Hi Eric,
>
> Thanks for these recommendations.
>
> for vmd-not-panasonic-home-ans.wav changing MAX_FREQ to 1450 WORKED!
>
> but I'm still having problems picking out the ~750Hz beep of sprint,  
> tmobile, and verizon. I tried first cutting POINTS and VALID in  
> half, then in half again, while also reducing MIN_AMPL in half but  
> still no luck. I assumed from the descriptions of each, that  
> reducing the numbers would make the algorithm less picky at finding  
> a beep. Is this correct?
>
> Any other recommendations on picking up these ~750Hz beeps? Thanks  
> again for the help.
>
> How close are they to 750Hz? If they're not more than say +/- 16Hz  
> then the tone_detect app *should* be able to detect them. At the  
> very least I would try it. See if tone_detect can detect those  
> beeps. While it may not be the most elegant solution, having mod_vmd  
> looking for one set of tones and tone_detect looking for the 750Hz  
> tones might actually get the job done, at least until you and Eric  
> can get together to see what's happening on the 750's.
>
> -MC
>
>
> --matt
>
> On Mon, Aug 17, 2009 at 2:15 PM, Eric des Courtis <eric.des.courtis at gmail.com 
> > wrote:
> Matt,
>
> Okay the good news is vmd should be able to handle these cases. The
> bad news is for whatever reason they are not getting detected at the
> moment.
>
> vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
> MAX_FREQ to 1450 and play with MIN_AMPL  if that still doesn't help.
>
> The following seem to use the same beep:
>
> vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>
> You can try to play with these values:
>
> POINTS 32
> VALID 22
> MAX_CHIRP 22
>
> If that doesn't work let me know I will try to improve the algorithm
> to detect the providers.
>
> Cheers!
>
> Eric des Courtis
>
>
>
> On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong at gmail.com>  
> wrote:
> > Hi Eric,
> > Thanks for the response. I had tried emailing you @brenbria.com  
> and the
> > email had bounced, thanks for responding to my mail.
> > If you'd be interested I .zipped up my sample voicemail beeps
> > at http://bandcon.hellohunter.com/vmd_wav.zip
> > I'm relatively new to telephony, but can you point me in the right  
> direction
> > for figuring out if the beeps are sinewaves. About as far as I've  
> come with
> > audio is being able to open the .wav files in audacity. Any website
> >  recommendations I can read? Thanks so much.
> > --matt
> >
> > On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
> > <eric.des.courtis at gmail.com> wrote:
> >>
> >> Matt,
> >>
> >> You must first capture the audio beeps and verify that they are  
> sine
> >> waves. If not, simply tweaking the algorithm will not give you  
> better
> >> results.
> >>
> >> It might be possible to use FFT and I would be happy to help you
> >> implement such a solution but keep in mind FFT is very very  
> demanding
> >> on the hardware. Ideally what you want to find out is what  
> functions
> >> was use to generate the beep in the first place so that it can be
> >> detected. Is it two sines waves like in DTMF? Or something more
> >> complex?
> >>
> >> Anyway my email is eric.des.courtis at benbria.ca.
> >>
> >> Cheers.
> >>
> >> Eric des Courtis
> >>
> >> On Mon, Aug 17, 2009 at 12:26 AM, Matthew  
> Fong<mattdfong at gmail.com> wrote:
> >> > I tried emailed Eric, seeking advice on this, but his email  
> (the one in
> >> > the
> >> > source code) is bouncing email (invalid user), so thought I  
> would ask
> >> > here
> >> > instead. If anyone has eric's new email address, I'd be  
> interesting in
> >> > it.
> >> >
> >> > I did some tests with mod_vmd this afternoon, but I'm only  
> finding about
> >> > 33%
> >> > of the voice mail beeps and did have 1 false-positive in my  
> test of
> >> > 7 voice
> >> > mail machines. I've recorded the audio of the session in .wav  
> files that
> >> > were both successful and not, as a comparison. I can upload  
> the .wav
> >> > files
> >> > if they would be useful.
> >> > mod_vmd works great for voicemails of Skype Users, and  
> kall8.com, but
> >> > has
> >> > issues dealing with mobile phone carriers.
> >> > sprint - not successful
> >> > tmobile - not successful
> >> > verizon - not successful
> >> > panasonic home answering machine system - not successful
> >> > kall8 - SUCCESS
> >> > skype - SUCCESS
> >> > I'm wondering if you can recommend a simple fix, like changing  
> some of
> >> > the
> >> > constants like MAX_FREQ, or MIN_TIME at the top of the  
> mod_vmd.c source
> >> > file, or if better success requires more complex analysis.  Do  
> you have
> >> > any
> >> > recommendations on how this might be done? Listening to  
> the .wav's
> >> > its apparent the beeps are not as loud for the mobile phone  
> carriers as
> >> > they
> >> > are with skype and kall8. Any guidance would be greatly  
> appreciated.
> >> > --matt
> >> > hello hunter
> >> > http://www.hellohunter.com
> >> > voice broadcasting & hosted dialer
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Wed Aug 19 10:30:04 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 19 Aug 2009 13:30:04 -0400
Subject: [Freeswitch-users] playback to legB before bridge?
In-Reply-To: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
References: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
Message-ID: <78ACC57E-6AD6-4026-A715-ABB8D0F4D162@jerris.com>

you can export nolocal the execute_on_answer var to do this.  Set it  
to playback the file.

Mike

On Aug 19, 2009, at 5:31 AM, Max Ivanov wrote:

> Hi all!
> Is it possible to execute  playback  application to legB before
> bridge? I mean sequence of actions similar to this:
>
> 1. originate legB
> 2. playback
> 3. bridge legA with legB using existing session from step 1 (no new
> incoming call)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dist.lists at gmail.com  Wed Aug 19 11:11:05 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Wed, 19 Aug 2009 21:11:05 +0300
Subject: [Freeswitch-users] custom avp in radius
Message-ID: <2a73afe0908191111h528d0a6dod257df4384318c90@mail.gmail.com>

Hi,

Does anyone know of a way to add custom information in AVP in radius
accounting messages? I can see in the radius dictionary there are many
AVPs defined, but I'd like to add some additional custom text
information. "Freeswitch-AVPair" seems like a generic AVP, but I
cannot figure out how to use it (if it's intended for something like
that at all).

Maybe there is a way to set some variable in the dial plan and to use
its contents to populate a general purpose AVP?

If there is no such functionality what will be the best way to do it?
I found similar feature added in the past
(http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=7529) and it
doesn't seem too complicated.

BR


From max.bridgewater at gmail.com  Wed Aug 19 11:14:36 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Wed, 19 Aug 2009 14:14:36 -0400
Subject: [Freeswitch-users] No Dialplan on answered channel,
	changing state to HANGUP
Message-ID: <c55d46950908191114q67f90333m4ccd53320989f965@mail.gmail.com>

Hi,

I'm trying to use les.net for termination. When i place a call from the
command line, i get the following error message:  No Dialplan on answered
channel, changing state to HANGUP. What does it mean?

My dialing string is as simple as it could be: sofia/gateway/
did.voip.les.net/13028838864 & park()
My les.net profile is located under
/usr/loca/freeswitch/conf/sip_profiles/external


<include>
  <gateway name="did.voip.les.net">
    <param name="username" value="1690xxxxxx"/>
    <param name="from-user" value="1690xxxxxx"/>
    <param name="realm" value="did.voip.les.net"/>
    <param name="password" value="ma34n2th./an4"/>
    <param name="caller-id-in-from" value="true"/>
  </gateway>
</include>

Regards,
Max.
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From brian at freeswitch.org  Wed Aug 19 11:19:01 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 Aug 2009 13:19:01 -0500
Subject: [Freeswitch-users] No Dialplan on answered channel,
	changing state to HANGUP
In-Reply-To: <c55d46950908191114q67f90333m4ccd53320989f965@mail.gmail.com>
References: <c55d46950908191114q67f90333m4ccd53320989f965@mail.gmail.com>
Message-ID: <17EBEEA2-A63B-44F2-84BC-CC946AC69C31@freeswitch.org>

you don't put a space after the &

/b

On Aug 19, 2009, at 1:14 PM, Max Bridgewater wrote:

> My dialing string is as simple as it could be: sofia/gateway/ 
> did.voip.les.net/13028838864 & park()

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From klaus.teller at gmx.net  Wed Aug 19 11:38:18 2009
From: klaus.teller at gmx.net (Klaus Teller)
Date: Wed, 19 Aug 2009 20:38:18 +0200
Subject: [Freeswitch-users] Passing Caller ID in Junction Network
In-Reply-To: <89755F1E-97E4-4ECA-8A6D-9D1B92B062F1@freeswitch.org>
References: <20090819163941.156100@gmx.net>
	<89755F1E-97E4-4ECA-8A6D-9D1B92B062F1@freeswitch.org>
Message-ID: <20090819183818.156130@gmx.net>

Hi Brian,

Please find my SIP trace attached. I called the Junction Networks guys and they suggest that they pass the caller id number only.

Could you confirm that everything is fine on my side so i can send the same to them?

Thanks so much,
Klaus.


-------- Original-Nachricht --------
> Datum: Wed, 19 Aug 2009 11:50:54 -0500
> Von: Brian West <brian at freeswitch.org>
> An: freeswitch-users at lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] Passing Caller ID in Junction Network

> Can you show me a sip invite of an inbound call from them?  and an  
> invite you're sending to them?
> 
> /b
> 
> On Aug 19, 2009, at 11:39 AM, Klaus Teller wrote:
> 
> > Hi,
> >
> > Anybody with experience in passing caller ID number in junction  
> > network? I need some help.
> >
> > I have the following origination string:
> >
> > {origination_caller_id_number 
> > =1514xxxxxx,ignore_early_media=true}sofia/gateway/sip.jnctn.net/ 
> > 1514xxxxxx
> >
> > Unfortunately, with this, junctionetwork won't pass the caller ID.  
> > Any idea?
> >
> > Thanks,
> > Klaus.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT!
Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01
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From matt at hellohunter.com  Wed Aug 19 11:50:02 2009
From: matt at hellohunter.com (Matt Hunter)
Date: Wed, 19 Aug 2009 11:50:02 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <8E7379D0-FB06-4C93-95B4-C493F8355A88@jerris.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>
	<87f2f3b90908190941m49c030e4w747e053e76f9b818@mail.gmail.com>
	<8E7379D0-FB06-4C93-95B4-C493F8355A88@jerris.com>
Message-ID: <4256bf830908191150v5c01a829s7b5138d902f7e9e2@mail.gmail.com>

Can anyone recommend a tool to analyze the wave files to see what's causing
the sine wav not to be detected? I have them zipped at
http://bandcon.hellohunter.com/vmd_wav.zip
<http://bandcon.hellohunter.com/vmd_wav.zip>I was trying to use audacity,
but not sure how to tell the exact frequency.

--matt

On Wed, Aug 19, 2009 at 10:26 AM, Michael Jerris <mike at jerris.com> wrote:

> my bet is if mod_vmd is not getting them that they are not going to work
> with tone detect either.  Someone needs to look at the tone and see what
> frequencies are really involved and if they change throughout the beep.
> Mike
>
> On Aug 19, 2009, at 12:41 PM, Michael Collins wrote:
>
>
>
> On Wed, Aug 19, 2009 at 1:43 AM, Matthew Fong <mattdfong at gmail.com> wrote:
>
>> Hi Eric,
>> Thanks for these recommendations.
>>
>> for vmd-not-panasonic-home-ans.wav changing MAX_FREQ to 1450 WORKED!
>>
>> but I'm still having problems picking out the ~750Hz beep of sprint,
>> tmobile, and verizon. I tried first cutting POINTS and VALID in half, then
>> in half again, while also reducing MIN_AMPL in half but still no luck. I
>> assumed from the descriptions of each, that reducing the numbers would make
>> the algorithm less picky at finding a beep. Is this correct?
>>
>> Any other recommendations on picking up these ~750Hz beeps? Thanks again
>> for the help.
>>
>
> How close are they to 750Hz? If they're not more than say +/- 16Hz then the
> tone_detect app *should* be able to detect them. At the very least I would
> try it. See if tone_detect can detect those beeps. While it may not be the
> most elegant solution, having mod_vmd looking for one set of tones and
> tone_detect looking for the 750Hz tones might actually get the job done, at
> least until you and Eric can get together to see what's happening on the
> 750's.
>
> -MC
>
>
>>
>> --matt
>>
>> On Mon, Aug 17, 2009 at 2:15 PM, Eric des Courtis <
>> eric.des.courtis at gmail.com> wrote:
>>
>>> Matt,
>>>
>>> Okay the good news is vmd should be able to handle these cases. The
>>> bad news is for whatever reason they are not getting detected at the
>>> moment.
>>>
>>> vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
>>> MAX_FREQ to 1450 and play with MIN_AMPL  if that still doesn't help.
>>
>>
>>> The following seem to use the same beep:
>>>
>>> vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
>>> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>>> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>>>
>>> You can try to play with these values:
>>>
>>> POINTS 32
>>> VALID 22
>>> MAX_CHIRP 22
>>>
>>> If that doesn't work let me know I will try to improve the algorithm
>>> to detect the providers.
>>>
>>> Cheers!
>>>
>>> Eric des Courtis
>>>
>>>
>>>
>>> On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong at gmail.com>
>>> wrote:
>>> > Hi Eric,
>>> > Thanks for the response. I had tried emailing you @brenbria.com and
>>> the
>>> > email had bounced, thanks for responding to my mail.
>>> > If you'd be interested I .zipped up my sample voicemail beeps
>>> > at http://bandcon.hellohunter.com/vmd_wav.zip
>>> > I'm relatively new to telephony, but can you point me in the right
>>> direction
>>> > for figuring out if the beeps are sinewaves. About as far as I've come
>>> with
>>> > audio is being able to open the .wav files in audacity. Any website
>>> >  recommendations I can read? Thanks so much.
>>> > --matt
>>> >
>>> > On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
>>> > <eric.des.courtis at gmail.com> wrote:
>>> >>
>>> >> Matt,
>>> >>
>>> >> You must first capture the audio beeps and verify that they are sine
>>> >> waves. If not, simply tweaking the algorithm will not give you better
>>> >> results.
>>> >>
>>> >> It might be possible to use FFT and I would be happy to help you
>>> >> implement such a solution but keep in mind FFT is very very demanding
>>> >> on the hardware. Ideally what you want to find out is what functions
>>> >> was use to generate the beep in the first place so that it can be
>>> >> detected. Is it two sines waves like in DTMF? Or something more
>>> >> complex?
>>> >>
>>> >> Anyway my email is eric.des.courtis at benbria.ca.
>>> >>
>>> >> Cheers.
>>> >>
>>> >> Eric des Courtis
>>> >>
>>> >> On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong at gmail.com>
>>> wrote:
>>> >> > I tried emailed Eric, seeking advice on this, but his email (the one
>>> in
>>> >> > the
>>> >> > source code) is bouncing email (invalid user), so thought I would
>>> ask
>>> >> > here
>>> >> > instead. If anyone has eric's new email address, I'd be interesting
>>> in
>>> >> > it.
>>> >> >
>>> >> > I did some tests with mod_vmd this afternoon, but I'm only finding
>>> about
>>> >> > 33%
>>> >> > of the voice mail beeps and did have 1 false-positive in my test of
>>> >> > 7 voice
>>> >> > mail machines. I've recorded the audio of the session in .wav files
>>> that
>>> >> > were both successful and not, as a comparison. I can upload the .wav
>>> >> > files
>>> >> > if they would be useful.
>>> >> > mod_vmd works great for voicemails of Skype Users, and kall8.com,
>>> but
>>> >> > has
>>> >> > issues dealing with mobile phone carriers.
>>> >> > sprint - not successful
>>> >> > tmobile - not successful
>>> >> > verizon - not successful
>>> >> > panasonic home answering machine system - not successful
>>> >> > kall8 - SUCCESS
>>> >> > skype - SUCCESS
>>> >> > I'm wondering if you can recommend a simple fix, like changing some
>>> of
>>> >> > the
>>> >> > constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c
>>> source
>>> >> > file, or if better success requires more complex analysis.  Do you
>>> have
>>> >> > any
>>> >> > recommendations on how this might be done? Listening to the .wav's
>>> >> > its apparent the beeps are not as loud for the mobile phone carriers
>>> as
>>> >> > they
>>> >> > are with skype and kall8. Any guidance would be greatly appreciated.
>>> >> > --matt
>>> >> > hello hunter
>>> >> > http://www.hellohunter.com
>>> >> > voice broadcasting & hosted dialer
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >
>>> >
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mattdfong at gmail.com  Wed Aug 19 11:50:34 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Wed, 19 Aug 2009 11:50:34 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <8E7379D0-FB06-4C93-95B4-C493F8355A88@jerris.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>
	<87f2f3b90908190941m49c030e4w747e053e76f9b818@mail.gmail.com>
	<8E7379D0-FB06-4C93-95B4-C493F8355A88@jerris.com>
Message-ID: <4256bf830908191150x479ba3b7i35b78053c8b6123c@mail.gmail.com>

Can anyone recommend a tool to analyze the wave files to see what's causing
the sine wav not to be detected? I have them zipped at
http://bandcon.hellohunter.com/vmd_wav.zip
<http://bandcon.hellohunter.com/vmd_wav.zip>I was trying to use audacity,
but not sure how to tell the exact frequency.

sorry if this was a double post...

On Wed, Aug 19, 2009 at 10:26 AM, Michael Jerris <mike at jerris.com> wrote:

> my bet is if mod_vmd is not getting them that they are not going to work
> with tone detect either.  Someone needs to look at the tone and see what
> frequencies are really involved and if they change throughout the beep.
> Mike
>
> On Aug 19, 2009, at 12:41 PM, Michael Collins wrote:
>
>
>
> On Wed, Aug 19, 2009 at 1:43 AM, Matthew Fong <mattdfong at gmail.com> wrote:
>
>> Hi Eric,
>> Thanks for these recommendations.
>>
>> for vmd-not-panasonic-home-ans.wav changing MAX_FREQ to 1450 WORKED!
>>
>> but I'm still having problems picking out the ~750Hz beep of sprint,
>> tmobile, and verizon. I tried first cutting POINTS and VALID in half, then
>> in half again, while also reducing MIN_AMPL in half but still no luck. I
>> assumed from the descriptions of each, that reducing the numbers would make
>> the algorithm less picky at finding a beep. Is this correct?
>>
>> Any other recommendations on picking up these ~750Hz beeps? Thanks again
>> for the help.
>>
>
> How close are they to 750Hz? If they're not more than say +/- 16Hz then the
> tone_detect app *should* be able to detect them. At the very least I would
> try it. See if tone_detect can detect those beeps. While it may not be the
> most elegant solution, having mod_vmd looking for one set of tones and
> tone_detect looking for the 750Hz tones might actually get the job done, at
> least until you and Eric can get together to see what's happening on the
> 750's.
>
> -MC
>
>
>>
>> --matt
>>
>> On Mon, Aug 17, 2009 at 2:15 PM, Eric des Courtis <
>> eric.des.courtis at gmail.com> wrote:
>>
>>> Matt,
>>>
>>> Okay the good news is vmd should be able to handle these cases. The
>>> bad news is for whatever reason they are not getting detected at the
>>> moment.
>>>
>>> vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
>>> MAX_FREQ to 1450 and play with MIN_AMPL  if that still doesn't help.
>>
>>
>>> The following seem to use the same beep:
>>>
>>> vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
>>> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>>> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>>>
>>> You can try to play with these values:
>>>
>>> POINTS 32
>>> VALID 22
>>> MAX_CHIRP 22
>>>
>>> If that doesn't work let me know I will try to improve the algorithm
>>> to detect the providers.
>>>
>>> Cheers!
>>>
>>> Eric des Courtis
>>>
>>>
>>>
>>> On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong at gmail.com>
>>> wrote:
>>> > Hi Eric,
>>> > Thanks for the response. I had tried emailing you @brenbria.com and
>>> the
>>> > email had bounced, thanks for responding to my mail.
>>> > If you'd be interested I .zipped up my sample voicemail beeps
>>> > at http://bandcon.hellohunter.com/vmd_wav.zip
>>> > I'm relatively new to telephony, but can you point me in the right
>>> direction
>>> > for figuring out if the beeps are sinewaves. About as far as I've come
>>> with
>>> > audio is being able to open the .wav files in audacity. Any website
>>> >  recommendations I can read? Thanks so much.
>>> > --matt
>>> >
>>> > On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
>>> > <eric.des.courtis at gmail.com> wrote:
>>> >>
>>> >> Matt,
>>> >>
>>> >> You must first capture the audio beeps and verify that they are sine
>>> >> waves. If not, simply tweaking the algorithm will not give you better
>>> >> results.
>>> >>
>>> >> It might be possible to use FFT and I would be happy to help you
>>> >> implement such a solution but keep in mind FFT is very very demanding
>>> >> on the hardware. Ideally what you want to find out is what functions
>>> >> was use to generate the beep in the first place so that it can be
>>> >> detected. Is it two sines waves like in DTMF? Or something more
>>> >> complex?
>>> >>
>>> >> Anyway my email is eric.des.courtis at benbria.ca.
>>> >>
>>> >> Cheers.
>>> >>
>>> >> Eric des Courtis
>>> >>
>>> >> On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong at gmail.com>
>>> wrote:
>>> >> > I tried emailed Eric, seeking advice on this, but his email (the one
>>> in
>>> >> > the
>>> >> > source code) is bouncing email (invalid user), so thought I would
>>> ask
>>> >> > here
>>> >> > instead. If anyone has eric's new email address, I'd be interesting
>>> in
>>> >> > it.
>>> >> >
>>> >> > I did some tests with mod_vmd this afternoon, but I'm only finding
>>> about
>>> >> > 33%
>>> >> > of the voice mail beeps and did have 1 false-positive in my test of
>>> >> > 7 voice
>>> >> > mail machines. I've recorded the audio of the session in .wav files
>>> that
>>> >> > were both successful and not, as a comparison. I can upload the .wav
>>> >> > files
>>> >> > if they would be useful.
>>> >> > mod_vmd works great for voicemails of Skype Users, and kall8.com,
>>> but
>>> >> > has
>>> >> > issues dealing with mobile phone carriers.
>>> >> > sprint - not successful
>>> >> > tmobile - not successful
>>> >> > verizon - not successful
>>> >> > panasonic home answering machine system - not successful
>>> >> > kall8 - SUCCESS
>>> >> > skype - SUCCESS
>>> >> > I'm wondering if you can recommend a simple fix, like changing some
>>> of
>>> >> > the
>>> >> > constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c
>>> source
>>> >> > file, or if better success requires more complex analysis.  Do you
>>> have
>>> >> > any
>>> >> > recommendations on how this might be done? Listening to the .wav's
>>> >> > its apparent the beeps are not as loud for the mobile phone carriers
>>> as
>>> >> > they
>>> >> > are with skype and kall8. Any guidance would be greatly appreciated.
>>> >> > --matt
>>> >> > hello hunter
>>> >> > http://www.hellohunter.com
>>> >> > voice broadcasting & hosted dialer
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >
>>> >
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From woof at iwoof.org  Wed Aug 19 12:38:57 2009
From: woof at iwoof.org (Andy Spitzer)
Date: Wed, 19 Aug 2009 15:38:57 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4256bf830908191150x479ba3b7i35b78053c8b6123c@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>
	<87f2f3b90908190941m49c030e4w747e053e76f9b818@mail.gmail.com>
	<8E7379D0-FB06-4C93-95B4-C493F8355A88@jerris.com>
	<4256bf830908191150x479ba3b7i35b78053c8b6123c@mail.gmail.com>
Message-ID: <op.uyxcu7rdg0dksq@woof3.fuzzy>

On Wed, 19 Aug 2009 14:50:34 -0400, Matthew Fong <mattdfong at gmail.com> wrote:

> I was trying to use audacity, but not sure how to tell the exact  
> frequency.

Audacity can do it.  Highlight the "beep" (and nothing but the beep) with the selection cursor, then click "Analyze->Plot Spectrum..."  In the "Frequency Analysis" window that opens, place the cursor on the center of the largest peak.  Near the base of that window it will show the Cursor and Peak frequencies.  For example, in the "vmd-not-panasonic-home-ans.wav" file, the "beep" is 1400 Hz (for a very short 1 second).  Then later it switches to FAX CED tone (2100 Hz) and FAX preamble.

--Woof!




From steveu at coppice.org  Wed Aug 19 13:22:57 2009
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 20 Aug 2009 04:22:57 +0800
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
Message-ID: <4A8C5F21.8080401@coppice.org>

On 08/18/2009 05:15 AM, Eric des Courtis wrote:
> Matt,
>
> Okay the good news is vmd should be able to handle these cases. The
> bad news is for whatever reason they are not getting detected at the
> moment.
>
> vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
> MAX_FREQ to 1450 and play with MIN_AMPL if that still doesn't help.
>
> The following seem to use the same beep:
>
> vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>
> You can try to play with these values:
>
> POINTS 32
> VALID 22
> MAX_CHIRP 22
>
> If that doesn't work let me know I will try to improve the algorithm
> to detect the providers.
>    
There is no noise on those 3 beeps. In fact, for something that's been 
through ulaw/alaw compression those beeps are very clean. They are quite 
short, though.

Steve



From msc at freeswitch.org  Wed Aug 19 14:22:37 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Aug 2009 14:22:37 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4A8C5F21.8080401@coppice.org>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
Message-ID: <87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>

> There is no noise on those 3 beeps. In fact, for something that's been
> through ulaw/alaw compression those beeps are very clean. They are quite
> short, though.
>

Heck yeah they're short! Steve, in your experience is there a practical way
to detect a beep that short without chewing up system resources or having
lots of false positives?
-MC
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From intralanman at freeswitch.org  Wed Aug 19 19:27:05 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Wed, 19 Aug 2009 22:27:05 -0400
Subject: [Freeswitch-users] Passing Caller ID in Junction Network
In-Reply-To: <20090819183818.156130@gmx.net>
References: <20090819163941.156100@gmx.net>
	<89755F1E-97E4-4ECA-8A6D-9D1B92B062F1@freeswitch.org>
	<20090819183818.156130@gmx.net>
Message-ID: <CE062C17-16E6-4474-87FC-29A0CFB5A7CF@freeswitch.org>

I think i remember needing to send the callerid in the From: when  
using them, try throwing
<param name="caller-id-in-from" value="true"/>
on the gateway and restarting the profile to see if it works after that

Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     

On Aug 19, 2009, at 2:38 PM, Klaus Teller wrote:

> Hi Brian,
>
> Please find my SIP trace attached. I called the Junction Networks  
> guys and they suggest that they pass the caller id number only.
>
> Could you confirm that everything is fine on my side so i can send  
> the same to them?
>
> Thanks so much,
> Klaus.
>
>
> -------- Original-Nachricht --------
>> Datum: Wed, 19 Aug 2009 11:50:54 -0500
>> Von: Brian West <brian at freeswitch.org>
>> An: freeswitch-users at lists.freeswitch.org
>> Betreff: Re: [Freeswitch-users] Passing Caller ID in Junction Network
>
>> Can you show me a sip invite of an inbound call from them?  and an
>> invite you're sending to them?
>>
>> /b
>>
>> On Aug 19, 2009, at 11:39 AM, Klaus Teller wrote:
>>
>>> Hi,
>>>
>>> Anybody with experience in passing caller ID number in junction
>>> network? I need some help.
>>>
>>> I have the following origination string:
>>>
>>> {origination_caller_id_number
>>> =1514xxxxxx,ignore_early_media=true}sofia/gateway/sip.jnctn.net/
>>> 1514xxxxxx
>>>
>>> Unfortunately, with this, junctionetwork won't pass the caller ID.
>>> Any idea?
>>>
>>> Thanks,
>>> Klaus.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> -- 
> GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT!
> Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01
> <etherXXXXFyLDxJ.cap>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dist.lists at gmail.com  Wed Aug 19 23:43:13 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Thu, 20 Aug 2009 09:43:13 +0300
Subject: [Freeswitch-users] custom avp in radius
In-Reply-To: <2a73afe0908191111h528d0a6dod257df4384318c90@mail.gmail.com>
References: <2a73afe0908191111h528d0a6dod257df4384318c90@mail.gmail.com>
Message-ID: <2a73afe0908192343n7c0313d6g87e650c5ecd1a457@mail.gmail.com>

I believe, I'm looking for a radius equivalent of the "userfield"
(http://wiki.freeswitch.org/wiki/Mod_cdr_csv#userfield)

br

On Wed, Aug 19, 2009 at 9:11 PM, Hristo Trendev<dist.lists at gmail.com> wrote:
> Hi,
>
> Does anyone know of a way to add custom information in AVP in radius
> accounting messages? I can see in the radius dictionary there are many
> AVPs defined, but I'd like to add some additional custom text
> information. "Freeswitch-AVPair" seems like a generic AVP, but I
> cannot figure out how to use it (if it's intended for something like
> that at all).
>
> Maybe there is a way to set some variable in the dial plan and to use
> its contents to populate a general purpose AVP?
>
> If there is no such functionality what will be the best way to do it?
> I found similar feature added in the past
> (http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=7529) and it
> doesn't seem too complicated.
>
> BR
>


From ken at expitrans.com  Wed Aug 19 23:51:53 2009
From: ken at expitrans.com (Kenneth Shaw)
Date: Wed, 19 Aug 2009 23:51:53 -0700
Subject: [Freeswitch-users] Sharing Presence Information Across Separate
	Offices
Message-ID: <20090820065153.ac3a26cf@kms.expitrans.com>

I apologize in advance if this message duplicates a previous post or the topic has been covered elsewhere. If that's the case, a pointer would be appreciated!

I have multiple offices each running Freeswitch and a number of Polycom phones and softphones.

I would like to share the presence information between the offices, such that if a person in Office B can see if a person in Office A is on the phone. I personally don't know of any way to do this through Freeswitch as it stands. I have thought about using a VPN for the offices and just setting up the phones to all register to a single server, however I don't want calls between phones in the same office (intra-office) to have to be routed all the way through a different office, which may be on the other side of the world and costs bandwidth.

Any suggestions as to the best way to accomplish this would be greatly appreciated. Thanks!

--
Kenneth Shaw
ExpiTrans, Inc.
129 W. Wilson St., Suite 204
Costa Mesa, CA 92627
tel: 949.650.4600
fax: 949.642.6044
ken at expitrans.com
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From mattdfong at gmail.com  Thu Aug 20 00:29:08 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Thu, 20 Aug 2009 00:29:08 -0700
Subject: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with same
	Gateway
Message-ID: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>

I'm trying to get FreeSWITCH to bridge two channels together through the
same external gateway, but I'm having issues hearing audio. Both legs if
setup independently and forwarded to 5000 (test ivr) work fine for both
inbound and outbound media, but when I try to bridge them together,
everything seems fine in FreeSWITCH, but neither party can hear the other
speak. I'm thinking the external gateway might be having some issues because
I've been able to bridge 2 channels together through the same gateway on
different providers, but thought I'd also try to seek some help here.
FreeSWITCH should be handling the media for both channels, so I can't figure
out why if Leg A and Leg B work independently, but not if they are bridged
together. Is there a setting somewhere in FS that I'm missing?
Below is a ngrep of the SIP interactions if it's useful. Thanks for the
help.

--matt

interface: eth0 (172.24.200.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )

U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060
INVITE sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>SIP/2.0.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Max-Forwards: 70.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>>.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 293.
Remote-Party-ID: "FreeSWITCH"
<sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
s=FreeSWITCH.
c=IN IP4 216.81.56.198.
t=0 0.
m=audio 24700 RTP/AVP 0 8 3 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 100 Trying.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Contact: <sip:914159927717 at 38.98.58.148:5060>.
Content-Length: 0.
.


U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 183 Session Progress.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Contact: <sip:914159927717 at 38.98.58.148:5060>.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Content-Type: application/sdp.
Content-Length: 227.
.
v=0.
o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
s=M6 Call.
c=IN IP4 38.98.58.148.
t=0 0.
m=audio 42554 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=rtcp:6461 IN IP4 10.10.24.50.


U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 200 OK.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Contact: <sip:914159927717 at 38.98.58.148:5060>.
Session-Expires: 1800;refresher=uas.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Supported: timer.
Content-Type: application/sdp.
Content-Length: 227.
.
v=0.
o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
s=M6 Call.
c=IN IP4 38.98.58.148.
t=0 0.
m=audio 42554 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=rtcp:6461 IN IP4 10.10.24.50.


U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060
ACK sip:914159927717 at 38.98.58.148:5060 SIP/2.0.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc.
Max-Forwards: 70.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 ACK.
Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
Content-Length: 0.
.


U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060
INVITE sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>SIP/2.0.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
Max-Forwards: 70.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>>.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257815 INVITE.
Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 293.
Remote-Party-ID: "FreeSWITCH"
<sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198.
s=FreeSWITCH.
c=IN IP4 216.81.56.198.
t=0 0.
m=audio 24798 RTP/AVP 0 8 3 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 100 Trying.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>;tag=F72E.2D4E.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257815 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
Contact: <sip:914154650027 at 38.98.58.148:5060>.
Content-Length: 0.
.


U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 183 Session Progress.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>;tag=F72E.2D4E.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257815 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
Contact: <sip:914154650027 at 38.98.58.148:5060>.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Content-Type: application/sdp.
Content-Length: 225.
.
v=0.
o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
s=M6 Call.
c=IN IP4 38.98.58.148.
t=0 0.
m=audio 46520 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=rtcp:6451 IN IP4 10.10.24.50.


U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 200 OK.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>;tag=F72E.2D4E.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257815 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
Contact: <sip:914154650027 at 38.98.58.148:5060>.
Session-Expires: 1800;refresher=uas.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Supported: timer.
Content-Type: application/sdp.
Content-Length: 225.
.
v=0.
o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
s=M6 Call.
c=IN IP4 38.98.58.148.
t=0 0.
m=audio 46520 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=rtcp:6451 IN IP4 10.10.24.50.


U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060
ACK sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK.
Max-Forwards: 70.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>;tag=F72E.2D4E.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257815 ACK.
Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
Content-Length: 0.
.


U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080
BYE sip:gw+epik.com at 216.81.56.198:5080 SIP/2.0.
From: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>;tag=F725.2C49.
To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 4817 BYE.
Max-Forwards: 70.
Via: SIP/2.0/UDP 38.98.58.148:5060
;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
Contact: <sip:914159927717 at 38.98.58.148:5060>.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Content-Length: 0.
.


U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 38.98.58.148:5060
;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
From: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>;tag=F725.2C49.
To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 4817 BYE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Content-Length: 0.
.


U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060
BYE sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
Max-Forwards: 70.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>;tag=F72E.2D4E.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257816 BYE.
Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
.


U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 200 OK.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>;tag=F72E.2D4E.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257816 BYE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
Contact: <sip:914154650027 at 38.98.58.148:5060>.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Content-Length: 0.
.
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From a.afzali2003 at gmail.com  Thu Aug 20 01:29:19 2009
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Thu, 20 Aug 2009 11:59:19 +0330
Subject: [Freeswitch-users] Sharing Presence Information Across Separate
	Offices
In-Reply-To: <20090820065153.ac3a26cf@kms.expitrans.com>
References: <20090820065153.ac3a26cf@kms.expitrans.com>
Message-ID: <debff5a80908200129q60d91a76v395781da1aabd661@mail.gmail.com>

Hi Kenneth,

I'm not going to answer your question! Instead I would like to emphasize on
the thing you are going to achieve because some times ago I've post this
question in some other title but unfortunately did not get any answer. As
the SIP protocol's point of view you should be able to subscribe every
FreeSWITCH machines for external events in another one (or probably in a
central PRESENCE server such as SER) to get NOTIFY messages which indicate
the information you wish to have. Although the sofia SIP endpoint can do
this (as you can find in its features) I could not find any guidelines for
it in FreeSWITCH.

Regards,
-- Afshin Afzali

On Thu, Aug 20, 2009 at 10:21 AM, Kenneth Shaw <ken at expitrans.com> wrote:

>  I apologize in advance if this message duplicates a previous post or the
> topic has been covered elsewhere. If that's the case, a pointer would be
> appreciated!
>
> I have multiple offices each running Freeswitch and a number of Polycom
> phones and softphones.
>
> I would like to share the presence information between the offices, such
> that if a person in Office B can see if a person in Office A is on the
> phone. I personally don't know of any way to do this through Freeswitch as
> it stands. I have thought about using a VPN for the offices and just setting
> up the phones to all register to a single server, however I don't want calls
> between phones in the same office (intra-office) to have to be routed all
> the way through a different office, which may be on the other side of the
> world and costs bandwidth.
>
> Any suggestions as to the best way to accomplish this would be greatly
> appreciated. Thanks!
>
> --
> Kenneth Shaw
> ExpiTrans, Inc.
> 129 W. Wilson St., Suite 204
> Costa Mesa, CA 92627
> tel: 949.650.4600
> fax: 949.642.6044
> ken at expitrans.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Thu Aug 20 03:42:37 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 20 Aug 2009 17:42:37 +0700
Subject: [Freeswitch-users] FS and SS7
Message-ID: <8ccbff060908200342p53474f58m645159a191db0364@mail.gmail.com>

Dear All,
            my telco can provider  SS7 (Now i use ISDN PRI).   I have
question about SS7.
        1. Can i setup SS7 with out ss7box ?
        2. I found some tutorial about libss7 in Asterisk. Is FS
support all feature like that.


Best Regards.

Dome C.


From mrene_lists at avgs.ca  Thu Aug 20 06:00:52 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 20 Aug 2009 09:00:52 -0400
Subject: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with
	same Gateway
In-Reply-To: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>
References: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>
Message-ID: <6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>

Hi

How are you bridging the calls in FS (which api call or C function are  
you using)?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:

> I'm trying to get FreeSWITCH to bridge two channels together through  
> the same external gateway, but I'm having issues hearing audio. Both  
> legs if setup independently and forwarded to 5000 (test ivr) work  
> fine for both inbound and outbound media, but when I try to bridge  
> them together, everything seems fine in FreeSWITCH, but neither  
> party can hear the other speak. I'm thinking the external gateway  
> might be having some issues because I've been able to bridge 2  
> channels together through the same gateway on different providers,  
> but thought I'd also try to seek some help here. FreeSWITCH should  
> be handling the media for both channels, so I can't figure out why  
> if Leg A and Leg B work independently, but not if they are bridged  
> together. Is there a setting somewhere in FS that I'm missing?
>
> Below is a ngrep of the SIP interactions if it's useful. Thanks for  
> the help.
>
> --matt
>
> interface: eth0 (172.24.200.0/255.255.255.0)
> filter: (ip or ip6) and ( port 5060 )
>
> U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060
> INVITE sip:914159927717 at 38.98.58.148 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148>.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 293.
> Remote-Party-ID: "FreeSWITCH" <sip: 
> 0000000000 at 216.81.56.198>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
> s=FreeSWITCH.
> c=IN IP4 216.81.56.198.
> t=0 0.
> m=audio 24700 RTP/AVP 0 8 3 101 13.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
>
>
> U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 100 Trying.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 183 Session Progress.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Allow:  
> INVITE 
> ,BYE 
> ,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Type: application/sdp.
> Content-Length: 227.
> .
> v=0.
> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 42554 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6461 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 200 OK.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Session-Expires: 1800;refresher=uas.
> Allow:  
> INVITE 
> ,BYE 
> ,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Supported: timer.
> Content-Type: application/sdp.
> Content-Length: 227.
> .
> v=0.
> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 42554 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6461 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060
> ACK sip:914159927717 at 38.98.58.148:5060 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 ACK.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060
> INVITE sip:914154650027 at 38.98.58.148 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 293.
> Remote-Party-ID: "FreeSWITCH" <sip: 
> 0000000000 at 216.81.56.198>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198.
> s=FreeSWITCH.
> c=IN IP4 216.81.56.198.
> t=0 0.
> m=audio 24798 RTP/AVP 0 8 3 101 13.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
>
>
> U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 100 Trying.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 183 Session Progress.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Allow:  
> INVITE 
> ,BYE 
> ,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Type: application/sdp.
> Content-Length: 225.
> .
> v=0.
> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 46520 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6451 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 200 OK.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Session-Expires: 1800;refresher=uas.
> Allow:  
> INVITE 
> ,BYE 
> ,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Supported: timer.
> Content-Type: application/sdp.
> Content-Length: 225.
> .
> v=0.
> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 46520 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6451 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060
> ACK sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 ACK.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080
> BYE sip:gw+epik.com at 216.81.56.198:5080 SIP/2.0.
> From: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 4817 BYE.
> Max-Forwards: 70.
> Via: SIP/2.0/UDP  
> 38.98.58.148:5060;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Allow:  
> INVITE 
> ,BYE 
> ,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP  
> 38.98.58.148:5060;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
> From: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 4817 BYE.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060
> BYE sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257816 BYE.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 200 OK.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257816 BYE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Allow:  
> INVITE 
> ,BYE 
> ,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Length: 0.
> .
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Thu Aug 20 06:19:50 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 08:19:50 -0500
Subject: [Freeswitch-users] FS and SS7
In-Reply-To: <8ccbff060908200342p53474f58m645159a191db0364@mail.gmail.com>
References: <8ccbff060908200342p53474f58m645159a191db0364@mail.gmail.com>
Message-ID: <D7C4EE10-6998-4FDC-BCC5-E3180F3157A8@freeswitch.org>


On Aug 20, 2009, at 5:42 AM, Dome Charoenyost wrote:

>        1. Can i setup SS7 with out ss7box ?

Sangoma's ss7box is the only solution right now thats turn key.

>        2. I found some tutorial about libss7 in Asterisk. Is FS
> support all feature like that.

You can't really use libss7 in FreeSWITCH as its not license  
compatible... neither is OpenSS7.



From matt at venturevoip.com  Thu Aug 20 02:35:31 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Thu, 20 Aug 2009 21:35:31 +1200
Subject: [Freeswitch-users] Recommendations for an interface between
	FreeSwitch and ANSI C code
Message-ID: <4A8D18E3.9090302@venturevoip.com>

Hi,

We have an application written in ANSI C which currently talks to the 
Asterisk Manager to make phone calls.

We're possibly looking at converting this to FreeSwitch.

At the moment it has an abstraction layer from Asterisk and speaks to 
between 1 and 80 Asterisk machines using round robin for distribution.

Would you recommend that I use the mod_event socket and basically work 
with FreeSwitch in the same way as I work with Asterisk or am I 
overlooking a possibly different way of doing things?

Cheers,

Matt Riddell


From mrene_lists at avgs.ca  Thu Aug 20 06:26:34 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 20 Aug 2009 09:26:34 -0400
Subject: [Freeswitch-users] Recommendations for an interface between
	FreeSwitch and ANSI C code
In-Reply-To: <4A8D18E3.9090302@venturevoip.com>
References: <4A8D18E3.9090302@venturevoip.com>
Message-ID: <85BE2921-4C4F-4FCA-B4BE-2D7C6A88B3FD@avgs.ca>

Hi,

I would recommand mod_event_socket, also look at libs/esl (event  
socket library), a C library that takes care of all the parsing to  
talk to mod_event_socket.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 20-Aug-09, at 5:35 AM, Matt Riddell wrote:

> Hi,
>
> We have an application written in ANSI C which currently talks to the
> Asterisk Manager to make phone calls.
>
> We're possibly looking at converting this to FreeSwitch.
>
> At the moment it has an abstraction layer from Asterisk and speaks to
> between 1 and 80 Asterisk machines using round robin for distribution.
>
> Would you recommend that I use the mod_event socket and basically work
> with FreeSwitch in the same way as I work with Asterisk or am I
> overlooking a possibly different way of doing things?
>
> Cheers,
>
> Matt Riddell
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Thu Aug 20 06:26:37 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 08:26:37 -0500
Subject: [Freeswitch-users] Recommendations for an interface between
	FreeSwitch and ANSI C code
In-Reply-To: <4A8D18E3.9090302@venturevoip.com>
References: <4A8D18E3.9090302@venturevoip.com>
Message-ID: <03AAD9B3-EF35-4615-892F-2A3354409A3E@freeswitch.org>

Matt,
	Event Socket would be perfect for your needs.  If you run into any  
troubles don't hesitate to join #freeswitch or ask on the list to help  
you get on the right track.

Thanks,
/b

On Aug 20, 2009, at 4:35 AM, Matt Riddell wrote:

> Would you recommend that I use the mod_event socket and basically work
> with FreeSwitch in the same way as I work with Asterisk or am I
> overlooking a possibly different way of doing things?



From matt at venturevoip.com  Thu Aug 20 06:32:05 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Fri, 21 Aug 2009 01:32:05 +1200
Subject: [Freeswitch-users] Recommendations for an interface
 between	FreeSwitch and ANSI C code
In-Reply-To: <85BE2921-4C4F-4FCA-B4BE-2D7C6A88B3FD@avgs.ca>
References: <4A8D18E3.9090302@venturevoip.com>
	<85BE2921-4C4F-4FCA-B4BE-2D7C6A88B3FD@avgs.ca>
Message-ID: <4A8D5055.9050801@venturevoip.com>

On 21/08/09 1:26 AM, Mathieu Rene wrote:
> Hi,
>
> I would recommand mod_event_socket, also look at libs/esl (event
> socket library), a C library that takes care of all the parsing to
> talk to mod_event_socket.

Awesome thanks man :)

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)


From matt at venturevoip.com  Thu Aug 20 06:32:20 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Fri, 21 Aug 2009 01:32:20 +1200
Subject: [Freeswitch-users] Recommendations for an interface
 between	FreeSwitch and ANSI C code
In-Reply-To: <03AAD9B3-EF35-4615-892F-2A3354409A3E@freeswitch.org>
References: <4A8D18E3.9090302@venturevoip.com>
	<03AAD9B3-EF35-4615-892F-2A3354409A3E@freeswitch.org>
Message-ID: <4A8D5064.60507@venturevoip.com>

On 21/08/09 1:26 AM, Brian West wrote:
> Matt,
> 	Event Socket would be perfect for your needs.  If you run into any
> troubles don't hesitate to join #freeswitch or ask on the list to help
> you get on the right track.

Cheers, will do!

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)


From Prometheus001 at gmx.net  Thu Aug 20 06:35:46 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 20 Aug 2009 15:35:46 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
Message-ID: <4A8D5132.7010807@gmx.net>

Hello,

when calling from Fritzbox to a Snom Phone , sound is fine. But when
calling an internal Freeswitch number (conference, mailbox) i hear a
very choppy voice coming from the fritzbox side. I think it may have to
do with the ptime 20msec/30msec.

Example: When calling from the fritzbox to a voicemail then the
annoucement from Freeswitch is choppy (too slow with interrups), but the
recorded message is fine.

Did anybody experience the same problem?

Best regards
Peter

Here are some SIP messages:
U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060
INVITE sip:0123456 at my.domain SIP/2.0.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 55 INVITE.
Contact:
<sip:02xxxxxxxxx at 112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3>.
Max-Forwards: 70.
Expires: 120.
User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009).
Supported: 100rel,replaces,timer.
Allow-Events: telephone-event,refer.
Allow:
INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH.
Content-Type: application/sdp.
Accept: application/sdp, multipart/mixed.
Accept-Encoding: identity.
Content-Length:   359.
.
v=0.
o=user 16423733 16423733 IN IP4 112.xxx.xx.xxx.
s=call.
c=IN IP4 112.xxx.xx.xxx.
t=0 0.
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101.
a=sendrecv.
a=rtpmap:2 G726-32/8000.
a=rtpmap:102 G726-32/8000.
a=rtpmap:100 G726-40/8000.
a=rtpmap:99 G726-24/8000.
a=rtpmap:97 iLBC/8000.
a=fmtp:97.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
a=rtcp:7079.

#
U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 55 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552.
Content-Length: 0.
.

#
U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>;tag=7t1e8BQg5B7yK.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 55 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Proxy-Authenticate: Digest realm="my.domain",
nonce="900b46a0-8d88-11de-a6a1-098738f35adb", algorithm=MD5, qop="auth".
Content-Length: 0.
.

#
U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060
ACK sip:0123456 at my.domain SIP/2.0.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>;tag=7t1e8BQg5B7yK.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 55 ACK.
User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009).
Content-Length: 0.
.

#
U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060
INVITE sip:0123456 at my.domain SIP/2.0.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 56 INVITE.
Contact:
<sip:02xxxxxxxxx at 112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3>.
Proxy-Authorization: Digest username="02xxxxxxxxx", realm="my.domain",
nonce="900b46a0-8d88-11de-a6a1-098738f35adb", uri="sip:0123456 at my.domain",
 response="276b44e261c13bd17218adff1150f414", algorithm=MD5,
cnonce="CADBE5D624516E8A", qop=auth, nc=00000001.
Max-Forwards: 70.
Expires: 120.
User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009).
Supported: 100rel,replaces,timer.
Allow-Events: telephone-event,refer.
Allow:
INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH.
Content-Type: application/sdp.
Accept: application/sdp, multipart/mixed.
Accept-Encoding: identity.
Content-Length:   359.
.
v=0.
o=user 16423733 16423733 IN IP4 112.xxx.xx.xxx.
s=call.
c=IN IP4 112.xxx.xx.xxx.
t=0 0.
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101.
a=sendrecv.
a=rtpmap:2 G726-32/8000.
a=rtpmap:102 G726-32/8000.
a=rtpmap:100 G726-40/8000.
a=rtpmap:99 G726-24/8000.
a=rtpmap:97 iLBC/8000.
a=fmtp:97 mode=30.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
a=rtcp:7079.

#
U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 56 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552.
Content-Length: 0.
.

#
U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>;tag=83t7967K2mXHF.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 56 INVITE.
Contact: <sip:0123456 at 182.xxx.xx.xxx:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 249.
.
v=0.
o=FreeSWITCH 1250746184 1250746185 IN IP4 182.xxx.xx.xxx.
s=FreeSWITCH.
c=IN IP4 182.xxx.xx.xxx.
t=0 0.
m=audio 26670 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.

#
U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060
ACK sip:0123456 at 182.xxx.xx.xxx:5060;transport=udp SIP/2.0.
Via: SIP/2.0/udp 112.xxx.xx.xxx:5060;branch=z9hG4bK39E0B6E237E408FC.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>;tag=83t7967K2mXHF.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 56 ACK.
Contact:
<sip:02xxxxxxxxx at 112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3>.
Max-Forwards: 70.
User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009).
Content-Length: 0.
.

#
U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060
INVITE sip:02xxxxxxxxx at 112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3
SIP/2.0.
Via: SIP/2.0/UDP 182.xxx.xx.xxx;rport;branch=z9hG4bK7rpNeaN67y9ta.
Max-Forwards: 70.
From: <sip:0123456 at my.domain>;tag=83t7967K2mXHF.
To: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 119268091 INVITE.
Contact: <sip:0123456 at 182.xxx.xx.xxx:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 249.
.
v=0.
o=FreeSWITCH 1250746184 1250746186 IN IP4 182.xxx.xx.xxx.
s=FreeSWITCH.
c=IN IP4 182.xxx.xx.xxx.
t=0 0.
m=audio 26670 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:30.







From brian at freeswitch.org  Thu Aug 20 06:38:34 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 08:38:34 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4A8D5132.7010807@gmx.net>
References: <4A8D5132.7010807@gmx.net>
Message-ID: <04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>

This is a bug in the fritzbox... you have to set your codec neg. to  
greedy on the sofia profile and that should fix it.

/b

On Aug 20, 2009, at 8:35 AM, Peter P GMX wrote:

> Hello,
>
> when calling from Fritzbox to a Snom Phone , sound is fine. But when
> calling an internal Freeswitch number (conference, mailbox) i hear a
> very choppy voice coming from the fritzbox side. I think it may have  
> to
> do with the ptime 20msec/30msec.
>
> Example: When calling from the fritzbox to a voicemail then the
> annoucement from Freeswitch is choppy (too slow with interrups), but  
> the
> recorded message is fine.
>
> Did anybody experience the same problem?
>
> Best regards
> Peter



From Prometheus001 at gmx.net  Thu Aug 20 07:27:15 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 20 Aug 2009 16:27:15 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>
References: <4A8D5132.7010807@gmx.net>
	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>
Message-ID: <4A8D5D43.3030300@gmx.net>

Hello Brian,

I have added
<param name="inbound-codec-negotiation" value="greedy"/>
to internal and external conf.
We have only allowed one codec now (G711A)

Conferencing MOH is fine. However hearing the other party (Fritzbox is
almost impossible).
Hearing voicemail announcements is also very choppy with seconds of delay.

So, in one case it works fine, in other s not.

Any more hints?

Best regards
Peter



Brian West schrieb:
> This is a bug in the fritzbox... you have to set your codec neg. to  
> greedy on the sofia profile and that should fix it.
>
> /b
>
> On Aug 20, 2009, at 8:35 AM, Peter P GMX wrote:
>
>   
>> Hello,
>>
>> when calling from Fritzbox to a Snom Phone , sound is fine. But when
>> calling an internal Freeswitch number (conference, mailbox) i hear a
>> very choppy voice coming from the fritzbox side. I think it may have  
>> to
>> do with the ptime 20msec/30msec.
>>
>> Example: When calling from the fritzbox to a voicemail then the
>> annoucement from Freeswitch is choppy (too slow with interrups), but  
>> the
>> recorded message is fine.
>>
>> Did anybody experience the same problem?
>>
>> Best regards
>> Peter
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From dave at 3c.co.uk  Thu Aug 20 07:29:09 2009
From: dave at 3c.co.uk (David Knell)
Date: Thu, 20 Aug 2009 17:29:09 +0300
Subject: [Freeswitch-users] Recommendations for an interface between
 FreeSwitch and ANSI C code
In-Reply-To: <4A8D18E3.9090302@venturevoip.com>
References: <4A8D18E3.9090302@venturevoip.com>
Message-ID: <1250778549.6309.28.camel@dk-d820>

On Thu, 2009-08-20 at 21:35 +1200, Matt Riddell wrote:
> We have an application written in ANSI C which currently talks to the 
> Asterisk Manager to make phone calls.
> 
> We're possibly looking at converting this to FreeSwitch.
> 
> At the moment it has an abstraction layer from Asterisk and speaks to 
> between 1 and 80 Asterisk machines using round robin for distribution.
> 
> Would you recommend that I use the mod_event socket and basically work 
> with FreeSwitch in the same way as I work with Asterisk or am I 
> overlooking a possibly different way of doing things?

Just to echo the sound advice from others, the event socket is almost
certainly the right place to start.  We've used it exclusively for our
stuff for getting on for two years, and I've no regrets.

--Dave

-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From shiyanov at gmail.com  Thu Aug 20 07:30:52 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Thu, 20 Aug 2009 18:30:52 +0400
Subject: [Freeswitch-users] "mute" channel programmatically with
	mod_event_socket
In-Reply-To: <36B2A0E3-6139-4724-A7F3-CAD89FB98688@avgs.ca>
References: <28651154.332401250673051662.JavaMail.root@zimbra.tng.de>
	<17888468.332601250673446046.JavaMail.root@zimbra.tng.de>
	<d211f54a0908190413p17311b15u42f57dd3417ec634@mail.gmail.com>
	<36B2A0E3-6139-4724-A7F3-CAD89FB98688@avgs.ca>
Message-ID: <d211f54a0908200730i6fe7d7ffm998cbdad601dac9b@mail.gmail.com>

Mathieu,

have to confess- you are right! uuid_bridge works as expected. Usual saying
- is didn't work last time I tried!
Anyway, thank you much!

Artem


On Wed, Aug 19, 2009 at 5:02 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> Eavesdrop kind of works yeah, you can use the intercept application to
> re-bridge the channels together, like:
>
> SendMsg <uuid_channel_making_mute>
> call-command: execute
> execute-app-name: intercept
> execute-app-arg: <uuid_participator>
>
> The same can be done with the uuid_bridge api.
>
> api uuid_bridge <uuid1> <uuid2>
>
> If you want the
> cleaner way you could implement a media bug that replaces all the audio of the channel by silence, but that'd require some C coding.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 19-Aug-09, at 7:13 AM, Artem Shiyanov wrote:
>
> The point is - a simple call flow is desired- if I have a ordinary 1-to-1
> call and one of the participators decides to mute call - I don't want to put
> both channels into a conference room but it looks like I have no other
> choices.
>
> BUT:
> I found brilliant app - eavesdrop! If I do this for one-to-one call - mute
> works!
> SendMsg <uuid_channel_making_mute>
> call-command: execute
> execute-app-name: eavesdrop
> execute-app-arg: <uuid_participator>
>
> But the problem appears when I want to unmute.. the call! I've tried to
> re-bridge channels, intercept them- nothing happens- one channel (muted one)
> doesn't hear the participator. And CLI command 'show channels' shows that
> channel with uui=<uuid_channel_making_mute> still process eavesdrop app.
> Maybe someone know how to switch off eavesdrop app?
>
>
> Artem
>
>
> On Wed, Aug 19, 2009 at 1:17 PM, Rudolf Denert <rdenert at tng.de> wrote:
>
>> You can use the caller controlls in the conference.conf.xml to implement
>> your own features something like mute or kick. Or do you want to mute mute
>> other conference members like a moderator can do this.
>>
>> BR
>>
>> ----- Urspr?ngliche Mail -----
>> Von: "Artem Shiyanov" <shiyanov at gmail.com>
>> An: freeswitch-users at lists.freeswitch.org
>> Gesendet: Dienstag, 18. August 2009 14:54:04 GMT +01:00
>> Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
>> Betreff: [Freeswitch-users] "mute" channel programmatically with
>> mod_event_socket
>>
>>
>> Hello all!
>>
>> I'm trying to implement "mute" feature with mod_event_socket: I want
>> programmatically mute/unmute some channel in a call.. And I don't see any
>> other ways except to use conference room with special rule "mute".
>> Anybody knows the better way?
>>
>>
>> Thanks,
>> Artem
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Thu Aug 20 07:39:33 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 09:39:33 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4A8D5D43.3030300@gmx.net>
References: <4A8D5132.7010807@gmx.net>
	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>
	<4A8D5D43.3030300@gmx.net>
Message-ID: <C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>

Besides taking a hammer to it?  Have you tried to make sure you have  
the latest firmware?  Try setting the ptime on the fritz to 20ms?

I really can't trust a product that has fritz in its name... it might  
go on the fritz  :P  pun intended.

/b

On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:

> Any more hints?



From msc at freeswitch.org  Thu Aug 20 08:16:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Aug 2009 08:16:54 -0700
Subject: [Freeswitch-users] send sip options message
In-Reply-To: <4A8B2015.2060605@xpirio.com>
References: <4A8B2015.2060605@xpirio.com>
Message-ID: <87f2f3b90908200816x65d296dckc5052fcebad89e4a@mail.gmail.com>

2009/8/18 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hi
>
> does anybody know how to send a sip options message to a registered user,
> using the event socket
> or something else build in freeswitch
> i think the ping parameter does something like this for gateways.
>
> what i want/need is the same thing that is provided in asterisk with the
> qualifying option,
> to see how "reachable" a certain client is.


Nothing exists at present but Mathieu Rene was looking at possibly adding an
API to do this. Stay tuned for more information...

-MC


> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From boonedox at gmail.com  Thu Aug 20 08:20:17 2009
From: boonedox at gmail.com (Jeremiah Johnson)
Date: Thu, 20 Aug 2009 09:20:17 -0600
Subject: [Freeswitch-users] Media Not Heard When Bridging 2 Calls
	withsame Gateway
In-Reply-To: <6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>
References: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>
	<6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>
Message-ID: <15FBC1BB717E4A9CB3E15E01D299B672@johnson>

For my gateway, I had to set ignore_early_media=true.


From: Mathieu Rene 
Sent: Thursday, August 20, 2009 7:00 AM
To: freeswitch-users at lists.freeswitch.org 
Subject: Re: [Freeswitch-users] Media Not Heard When Bridging 2 Calls withsame Gateway


Hi


How are you bridging the calls in FS (which api call or C function are you using)? 


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca








On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:


  I'm trying to get FreeSWITCH to bridge two channels together through the same external gateway, but I'm having issues hearing audio. Both legs if setup independently and forwarded to 5000 (test ivr) work fine for both inbound and outbound media, but when I try to bridge them together, everything seems fine in FreeSWITCH, but neither party can hear the other speak. I'm thinking the external gateway might be having some issues because I've been able to bridge 2 channels together through the same gateway on different providers, but thought I'd also try to seek some help here. FreeSWITCH should be handling the media for both channels, so I can't figure out why if Leg A and Leg B work independently, but not if they are bridged together. Is there a setting somewhere in FS that I'm missing? 


  Below is a ngrep of the SIP interactions if it's useful. Thanks for the help.


  --matt



  interface: eth0 (172.24.200.0/255.255.255.0)
  filter: (ip or ip6) and ( port 5060 )


  U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060
  INVITE sip:914159927717 at 38.98.58.148 SIP/2.0.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
  Max-Forwards: 70.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  To: <sip:914159927717 at 38.98.58.148>.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257811 INVITE.
  Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
  Supported: timer, precondition, path, replaces.
  Allow-Events: talk, refer.
  Content-Type: application/sdp.
  Content-Disposition: session.
  Content-Length: 293.
  Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;party=calling;screen=yes;privacy=off.
  .
  v=0.
  o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
  s=FreeSWITCH.
  c=IN IP4 216.81.56.198.
  t=0 0.
  m=audio 24700 RTP/AVP 0 8 3 101 13.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:3 GSM/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=rtpmap:13 CN/8000.
  a=ptime:20.




  U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 100 Trying.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257811 INVITE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
  Contact: <sip:914159927717 at 38.98.58.148:5060>.
  Content-Length: 0.
  .




  U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 183 Session Progress.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257811 INVITE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
  Contact: <sip:914159927717 at 38.98.58.148:5060>.
  Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
  Content-Type: application/sdp.
  Content-Length: 227.
  .
  v=0.
  o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
  s=M6 Call.
  c=IN IP4 38.98.58.148.
  t=0 0.
  m=audio 42554 RTP/AVP 0 101.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-15.
  a=ptime:20.
  a=sendrecv.
  a=rtcp:6461 IN IP4 10.10.24.50.




  U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 200 OK.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257811 INVITE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
  Contact: <sip:914159927717 at 38.98.58.148:5060>.
  Session-Expires: 1800;refresher=uas.
  Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
  Supported: timer.
  Content-Type: application/sdp.
  Content-Length: 227.
  .
  v=0.
  o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
  s=M6 Call.
  c=IN IP4 38.98.58.148.
  t=0 0.
  m=audio 42554 RTP/AVP 0 101.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-15.
  a=ptime:20.
  a=sendrecv.
  a=rtcp:6461 IN IP4 10.10.24.50.




  U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060
  ACK sip:914159927717 at 38.98.58.148:5060 SIP/2.0.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc.
  Max-Forwards: 70.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257811 ACK.
  Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
  Content-Length: 0.
  .




  U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060
  INVITE sip:914154650027 at 38.98.58.148 SIP/2.0.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
  Max-Forwards: 70.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257815 INVITE.
  Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
  Supported: timer, precondition, path, replaces.
  Allow-Events: talk, refer.
  Content-Type: application/sdp.
  Content-Disposition: session.
  Content-Length: 293.
  Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;party=calling;screen=yes;privacy=off.
  .
  v=0.
  o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198.
  s=FreeSWITCH.
  c=IN IP4 216.81.56.198.
  t=0 0.
  m=audio 24798 RTP/AVP 0 8 3 101 13.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:3 GSM/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=rtpmap:13 CN/8000.
  a=ptime:20.




  U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 100 Trying.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257815 INVITE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
  Contact: <sip:914154650027 at 38.98.58.148:5060>.
  Content-Length: 0.
  .




  U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 183 Session Progress.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257815 INVITE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
  Contact: <sip:914154650027 at 38.98.58.148:5060>.
  Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
  Content-Type: application/sdp.
  Content-Length: 225.
  .
  v=0.
  o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
  s=M6 Call.
  c=IN IP4 38.98.58.148.
  t=0 0.
  m=audio 46520 RTP/AVP 0 101.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-15.
  a=ptime:20.
  a=sendrecv.
  a=rtcp:6451 IN IP4 10.10.24.50.




  U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 200 OK.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257815 INVITE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
  Contact: <sip:914154650027 at 38.98.58.148:5060>.
  Session-Expires: 1800;refresher=uas.
  Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
  Supported: timer.
  Content-Type: application/sdp.
  Content-Length: 225.
  .
  v=0.
  o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
  s=M6 Call.
  c=IN IP4 38.98.58.148.
  t=0 0.
  m=audio 46520 RTP/AVP 0 101.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-15.
  a=ptime:20.
  a=sendrecv.
  a=rtcp:6451 IN IP4 10.10.24.50.




  U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060
  ACK sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK.
  Max-Forwards: 70.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257815 ACK.
  Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
  Content-Length: 0.
  .




  U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080
  BYE sip:gw+epik.com at 216.81.56.198:5080 SIP/2.0.
  From: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
  To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 4817 BYE.
  Max-Forwards: 70.
  Via: SIP/2.0/UDP 38.98.58.148:5060;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
  Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
  Contact: <sip:914159927717 at 38.98.58.148:5060>.
  Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
  Content-Length: 0.
  .




  U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 38.98.58.148:5060;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
  Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
  From: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
  To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 4817 BYE.
  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
  Supported: timer, precondition, path, replaces.
  Content-Length: 0.
  .




  U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060
  BYE sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
  Max-Forwards: 70.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257816 BYE.
  Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
  Supported: timer, precondition, path, replaces.
  Reason: Q.850;cause=16;text="NORMAL_CLEARING".
  Content-Length: 0.
  .




  U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 200 OK.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257816 BYE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
  Contact: <sip:914154650027 at 38.98.58.148:5060>.
  Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
  Content-Length: 0.
  .
  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org





--------------------------------------------------------------------------------


_______________________________________________
FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jan.kubr at gmail.com  Thu Aug 20 08:43:57 2009
From: jan.kubr at gmail.com (Jan Kubr)
Date: Thu, 20 Aug 2009 17:43:57 +0200
Subject: [Freeswitch-users] How to delay IVR answer during an outbound
	call
In-Reply-To: <c43084ae0908112323m1c8d5856kc05acc6664b635cd@mail.gmail.com>
References: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com> 
	<c43084ae0908112323m1c8d5856kc05acc6664b635cd@mail.gmail.com>
Message-ID: <698401620908200843p414f426cudcda3a40ac2d91ca@mail.gmail.com>

I have been experiencing this as well. It happens randomly and I
haven't been able to find out what the issue is. I think there is some
delay when the RTP ports are being negotiated/allocated. Or something.
What helped me a bit: I start with playing a file containing 1 second
of silence and only then do whatever I want to do.

Jan

On Wed, Aug 12, 2009 at 8:23 AM, Paul Li<plite2012 at gmail.com> wrote:
> I am actually doing a lua script for IVR as follows
>
> -- answer the call
> session:answer();
>
> while session:ready() == true do
> ? ? ? ?-- sleep a second
> ? ? ? ?session:sleep(1000);
>
> ? ? ? ?-- play a file
> ? ? ? ?session:streamFile("/path/to/blah.wav");
>
> ? ? ? ?-- hangup
> ? ? ? ?session:hangup();
> end
>
> The problem lies in: when I picked up my phone, blah.wav was already
> played for a while, instead of from the beginning.
>
> I shall greatly appreciate any input.
>
> On Wed, Aug 12, 2009 at 12:58 AM, Paul Li<plite2012 at gmail.com> wrote:
>> I have a dummy question. Say, you have an outbound call to the demo
>> IVR as below:
>>
>> originate sofia/gateway/myvoip/19876543210 5000
>>
>> How do I delay the IVR response until the recipient at 19876543210
>> picks up the call? I tried "ignore_early_media=true", which had no
>> effect.
>>
>> Many thanks in advance.
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From brian at freeswitch.org  Thu Aug 20 09:04:33 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 11:04:33 -0500
Subject: [Freeswitch-users] How to delay IVR answer during an outbound
	call
In-Reply-To: <698401620908200843p414f426cudcda3a40ac2d91ca@mail.gmail.com>
References: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com>
	<c43084ae0908112323m1c8d5856kc05acc6664b635cd@mail.gmail.com>
	<698401620908200843p414f426cudcda3a40ac2d91ca@mail.gmail.com>
Message-ID: <EFF7C574-B430-4DC9-BE1B-01241FDBA232@freeswitch.org>

Nope.  The issue is the far end isn't done in time.  So we send  
packets and they ignore them.  The only way to prevent this is to  
answer, silence_stream://1000, then playback.

/b

On Aug 20, 2009, at 10:43 AM, Jan Kubr wrote:

> I have been experiencing this as well. It happens randomly and I
> haven't been able to find out what the issue is. I think there is some
> delay when the RTP ports are being negotiated/allocated. Or something.
> What helped me a bit: I start with playing a file containing 1 second
> of silence and only then do whatever I want to do.
>
> Jan



From dome at tel.co.th  Thu Aug 20 09:11:08 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 20 Aug 2009 23:11:08 +0700
Subject: [Freeswitch-users] FS and SS7
In-Reply-To: <D7C4EE10-6998-4FDC-BCC5-E3180F3157A8@freeswitch.org>
References: <8ccbff060908200342p53474f58m645159a191db0364@mail.gmail.com>
	<D7C4EE10-6998-4FDC-BCC5-E3180F3157A8@freeswitch.org>
Message-ID: <8ccbff060908200911u30c7f169nb9016733aec2460f@mail.gmail.com>

2009/8/20 Brian West <brian at freeswitch.org>:
>
> On Aug 20, 2009, at 5:42 AM, Dome Charoenyost wrote:
>
>> ? ? ? ?1. Can i setup SS7 with out ss7box ?
>
> Sangoma's ss7box is the only solution right now thats turn key.

If i seting up asterisk with libss7. should be same ss7box right ?
>
>> ? ? ? ?2. I found some tutorial about libss7 in Asterisk. Is FS
>> support all feature like that.
>
> You can't really use libss7 in FreeSWITCH as its not license
> compatible... neither is OpenSS7.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mattdfong at gmail.com  Thu Aug 20 10:21:40 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Thu, 20 Aug 2009 10:21:40 -0700
Subject: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with
	same Gateway
In-Reply-To: <6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>
References: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>
	<6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>
Message-ID: <4256bf830908201021s7da0bff0wed8e060dbac0db2c@mail.gmail.com>

originate {ignore_early_media=true}sofia/gateway/epik.com/914159927717&bridge(sofia/gateway/
epik.com/914154650027)
is the string I was using from the console.

On Thu, Aug 20, 2009 at 6:00 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi
>
> How are you bridging the calls in FS (which api call or C function are you
> using)?
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:
>
> I'm trying to get FreeSWITCH to bridge two channels together through the
> same external gateway, but I'm having issues hearing audio. Both legs if
> setup independently and forwarded to 5000 (test ivr) work fine for both
> inbound and outbound media, but when I try to bridge them together,
> everything seems fine in FreeSWITCH, but neither party can hear the other
> speak. I'm thinking the external gateway might be having some issues because
> I've been able to bridge 2 channels together through the same gateway on
> different providers, but thought I'd also try to seek some help here.
> FreeSWITCH should be handling the media for both channels, so I can't figure
> out why if Leg A and Leg B work independently, but not if they are bridged
> together. Is there a setting somewhere in FS that I'm missing?
> Below is a ngrep of the SIP interactions if it's useful. Thanks for the
> help.
>
> --matt
>
> interface: eth0 (172.24.200.0/255.255.255.0)
> filter: (ip or ip6) and ( port 5060 )
>
> U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060
> INVITE sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>>.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 293.
> Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
> s=FreeSWITCH.
> c=IN IP4 216.81.56.198.
> t=0 0.
> m=audio 24700 RTP/AVP 0 8 3 101 13.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
>
>
> U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 100 Trying.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
> >;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 183 Session Progress.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
> >;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Allow:
> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Type: application/sdp.
> Content-Length: 227.
> .
> v=0.
> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 42554 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6461 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 200 OK.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
> >;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Session-Expires: 1800;refresher=uas.
> Allow:
> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Supported: timer.
> Content-Type: application/sdp.
> Content-Length: 227.
> .
> v=0.
> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 42554 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6461 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060
> ACK sip:914159927717 at 38.98.58.148:5060 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
> >;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 ACK.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060
> INVITE sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>>.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 293.
> Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198.
> s=FreeSWITCH.
> c=IN IP4 216.81.56.198.
> t=0 0.
> m=audio 24798 RTP/AVP 0 8 3 101 13.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
>
>
> U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 100 Trying.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
> >;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 183 Session Progress.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
> >;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Allow:
> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Type: application/sdp.
> Content-Length: 225.
> .
> v=0.
> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 46520 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6451 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 200 OK.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
> >;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Session-Expires: 1800;refresher=uas.
> Allow:
> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Supported: timer.
> Content-Type: application/sdp.
> Content-Length: 225.
> .
> v=0.
> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 46520 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6451 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060
> ACK sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
> >;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 ACK.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080
> BYE sip:gw+epik.com at 216.81.56.198:5080 SIP/2.0.
> From: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
> >;tag=F725.2C49.
> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 4817 BYE.
> Max-Forwards: 70.
> Via: SIP/2.0/UDP 38.98.58.148:5060
> ;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Allow:
> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 38.98.58.148:5060
> ;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
> From: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
> >;tag=F725.2C49.
> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 4817 BYE.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060
> BYE sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
> >;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257816 BYE.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 200 OK.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
> >;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257816 BYE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Allow:
> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Length: 0.
> .
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From csa at nowthor.com  Thu Aug 20 10:28:56 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 20 Aug 2009 13:28:56 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
Message-ID: <4A8D87D8.8080001@nowthor.com>

Hello!

I am using Callcentric for my tests and have observed what appears to me 
a possible bug in the way Freeswitch handles DNS SRV records.

Callcentric uses DNS SRV records as a way to direct traffic to their SIP 
server. A 'srv' 'dig' of '_sip._udp.callcentric.com' returns:

_sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080 
alpha6.callcentric.com.
_sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080 
alpha7.callcentric.com.
_sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080 
alpha1.callcentric.com.
_sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080 
alpha3.callcentric.com.

Based on this information, Freeswitch appears to correctly round robin 
all available IP addresses except in a particular situation: in the 
middle of authorizations.

For example, in a registration, Freeswitch send a packet to 
alpha1.callcentric.com. Callcentric then challenges Freeswitch with a 
Proxy Authorization request. Freeswitch then sends the packet with the 
requested credentials but not necessarily to alpha1.callcentric.com! In 
many cases, instead of sticking to the 'challenging' server, Freeswitch 
round robins and sends the second packet to, one of the other servers. 
This continues for a little while and eventually, simply by luck, the 
second packet is sent to the 'challenging' Callcentric server.

Shouldn't Freeswitch stick to the same server when challenged for 
credentials? Is this a bug? Is there a way to make Freeswitch behave 
differently?

Thanks!

Carlos Antunes
Nowthor Corporation





From intralanman at freeswitch.org  Thu Aug 20 10:54:58 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 20 Aug 2009 13:54:58 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8D87D8.8080001@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>
Message-ID: <C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>

On Aug 20, 2009, at 1:28 PM, Carlos S. Antunes wrote:

> Hello!
>
> I am using Callcentric for my tests and have observed what appears  
> to me
> a possible bug in the way Freeswitch handles DNS SRV records.
>
> Callcentric uses DNS SRV records as a way to direct traffic to their  
> SIP
> server. A 'srv' 'dig' of '_sip._udp.callcentric.com' returns:
>
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha6.callcentric.com.
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha7.callcentric.com.
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha1.callcentric.com.
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha3.callcentric.com.
>
> Based on this information, Freeswitch appears to correctly round robin
> all available IP addresses except in a particular situation: in the
> middle of authorizations.
>

very true, but i've been reading over the RFCs on this, and it seems  
that FreeSWITCH isn't doing anything incorrectly.

in RFC3263 (section 4), when talking about client usage of SRV:
The procedures here MUST be done exactly once per transaction, where  
transaction is as defined in [1].
[1] being RFC3261

in RFC3261 (section 8.1.3.5), when talking about 4xx responses:
In all of the above cases, the request is retried by creating a new
    request with the appropriate modifications.  This new request
    constitutes a new transaction and SHOULD have the same value of the
    Call-ID, To, and From of the previous request, but the CSeq should
    contain a new sequence number that is one higher than the previous.

> For example, in a registration, Freeswitch send a packet to
> alpha1.callcentric.com. Callcentric then challenges Freeswitch with a
> Proxy Authorization request. Freeswitch then sends the packet with the
> requested credentials but not necessarily to alpha1.callcentric.com!  
> In
> many cases, instead of sticking to the 'challenging' server,  
> Freeswitch
> round robins and sends the second packet to, one of the other servers.
> This continues for a little while and eventually, simply by luck, the
> second packet is sent to the 'challenging' Callcentric server.
>
> Shouldn't Freeswitch stick to the same server when challenged for
> credentials?

can you show anything in the RFCs that says so?



> Is this a bug?

see above



> Is there a way to make Freeswitch behave
> differently?
disable-srv on the profile



All that said, if it has to be a "bug", then it seems to me that it's  
more of a "bug" in callcentric's service. They'd probably be better of  
actually prioritizing their SRV records. If they want load balancing  
and want to do checking for stale nonces, then they should be sharing  
nonces across all of their proxies or using a proper load balancer.


Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     

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From steveu at coppice.org  Thu Aug 20 11:06:54 2009
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 21 Aug 2009 02:06:54 +0800
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
Message-ID: <4A8D90BE.2090007@coppice.org>

On 08/20/2009 05:22 AM, Michael Collins wrote:
>
>     There is no noise on those 3 beeps. In fact, for something that's been
>     through ulaw/alaw compression those beeps are very clean. They are
>     quite
>     short, though.
>
>
> Heck yeah they're short! Steve, in your experience is there a 
> practical way to detect a beep that short without chewing up system 
> resources or having lots of false positives?
> -MC
>
The tone samples I just looked at are about 130ms long. The problem is 
the detector is trying to be a very open ended detector of anything 
narrowband enough to be a single tone, and of any duration beyond some 
small minimum. Its difficult to make such a thing voice immune unless 
you can also count on a very large signal to noise ratio. With a digital 
trunk you can probably rely on a large SNR, but what happens when people 
use analogue lines? There is a reason why DTMF detectors try hard to 
work down to about 10dB SNR. :-)

Steve



From msc at freeswitch.org  Thu Aug 20 11:20:17 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Aug 2009 11:20:17 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4A8D90BE.2090007@coppice.org>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
Message-ID: <87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>

On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood <steveu at coppice.org>wrote:

> On 08/20/2009 05:22 AM, Michael Collins wrote:
> >
> >     There is no noise on those 3 beeps. In fact, for something that's
> been
> >     through ulaw/alaw compression those beeps are very clean. They are
> >     quite
> >     short, though.
> >
> >
> > Heck yeah they're short! Steve, in your experience is there a
> > practical way to detect a beep that short without chewing up system
> > resources or having lots of false positives?
> > -MC
> >
> The tone samples I just looked at are about 130ms long. The problem is
> the detector is trying to be a very open ended detector of anything
> narrowband enough to be a single tone, and of any duration beyond some
> small minimum. Its difficult to make such a thing voice immune unless
> you can also count on a very large signal to noise ratio. With a digital
> trunk you can probably rely on a large SNR, but what happens when people
> use analogue lines? There is a reason why DTMF detectors try hard to
> work down to about 10dB SNR. :-)
>
> Steve
>

Thanks for the lesson uncle Steve! I'm guessing that the OP will need a new
strategy. Possibly waiting for silence? Not sure what's the best way to go
but I'm interested in hearing if someone has a solution.

-MC
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From mike at jerris.com  Thu Aug 20 11:25:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 20 Aug 2009 14:25:51 -0400
Subject: [Freeswitch-users] Sharing Presence Information Across Separate
	Offices
In-Reply-To: <debff5a80908200129q60d91a76v395781da1aabd661@mail.gmail.com>
References: <20090820065153.ac3a26cf@kms.expitrans.com>
	<debff5a80908200129q60d91a76v395781da1aabd661@mail.gmail.com>
Message-ID: <4C4DB254-9F49-4181-AD9E-592571240844@jerris.com>

Check out mod_event_multicast.  I think you should be able to share  
using that to share events between boxes (although I have never tried  
it).

Mike

On Aug 20, 2009, at 4:29 AM, afshin afzali wrote:

> Hi Kenneth,
>
> I'm not going to answer your question! Instead I would like to  
> emphasize on the thing you are going to achieve because some times  
> ago I've post this question in some other title but unfortunately  
> did not get any answer. As the SIP protocol's point of view you  
> should be able to subscribe every FreeSWITCH machines for external  
> events in another one (or probably in a central PRESENCE server such  
> as SER) to get NOTIFY messages which indicate the information you  
> wish to have. Although the sofia SIP endpoint can do this (as you  
> can find in its features) I could not find any guidelines for it in  
> FreeSWITCH.
>
> Regards,
> -- Afshin Afzali   
>
> On Thu, Aug 20, 2009 at 10:21 AM, Kenneth Shaw <ken at expitrans.com>  
> wrote:
> I apologize in advance if this message duplicates a previous post or  
> the topic has been covered elsewhere. If that's the case, a pointer  
> would be appreciated!
>
> I have multiple offices each running Freeswitch and a number of  
> Polycom phones and softphones.
>
> I would like to share the presence information between the offices,  
> such that if a person in Office B can see if a person in Office A is  
> on the phone. I personally don't know of any way to do this through  
> Freeswitch as it stands. I have thought about using a VPN for the  
> offices and just setting up the phones to all register to a single  
> server, however I don't want calls between phones in the same office  
> (intra-office) to have to be routed all the way through a different  
> office, which may be on the other side of the world and costs  
> bandwidth.
>
> Any suggestions as to the best way to accomplish this would be  
> greatly appreciated. Thanks!
>
> --
> Kenneth Shaw
> ExpiTrans, Inc.
> 129 W. Wilson St., Suite 204
> Costa Mesa, CA 92627
> tel: 949.650.4600
> fax: 949.642.6044
> ken at expitrans.com

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From mike at jerris.com  Thu Aug 20 11:27:43 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 20 Aug 2009 14:27:43 -0400
Subject: [Freeswitch-users] send sip options message
In-Reply-To: <87f2f3b90908200816x65d296dckc5052fcebad89e4a@mail.gmail.com>
References: <4A8B2015.2060605@xpirio.com>
	<87f2f3b90908200816x65d296dckc5052fcebad89e4a@mail.gmail.com>
Message-ID: <B769F00C-C305-45BA-87B2-C050C5CD136B@jerris.com>

There may be a magic event you can send to do this.  I can't recall if  
we did that for options or just info/notify.  A grep for nua_options  
in mod_sofia dir should answer the question.

Mike

On Aug 20, 2009, at 11:16 AM, Michael Collins wrote:

>
>
> 2009/8/18 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
> hi
>
> does anybody know how to send a sip options message to a registered  
> user, using the event socket
> or something else build in freeswitch
> i think the ping parameter does something like this for gateways.
>
> what i want/need is the same thing that is provided in asterisk with  
> the qualifying option,
> to see how "reachable" a certain client is.
>
> Nothing exists at present but Mathieu Rene was looking at possibly  
> adding an API to do this. Stay tuned for more information...
>
> -MC
>
>
> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP

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From mrene_lists at avgs.ca  Thu Aug 20 11:31:55 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 20 Aug 2009 14:31:55 -0400
Subject: [Freeswitch-users] send sip options message
In-Reply-To: <B769F00C-C305-45BA-87B2-C050C5CD136B@jerris.com>
References: <4A8B2015.2060605@xpirio.com>
	<87f2f3b90908200816x65d296dckc5052fcebad89e4a@mail.gmail.com>
	<B769F00C-C305-45BA-87B2-C050C5CD136B@jerris.com>
Message-ID: <09DDA10A-47F5-4DAE-8B8B-82A613BE2A53@avgs.ca>

Oh, I missed that one.

<param name="nat-options-ping" value="true" /> (in the sip profile)

It does it only when NAT is detected though.

Tschuess,

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 20-Aug-09, at 2:27 PM, Michael Jerris wrote:

> There may be a magic event you can send to do this.  I can't recall  
> if we did that for options or just info/notify.  A grep for  
> nua_options in mod_sofia dir should answer the question.
>
> Mike
>
> On Aug 20, 2009, at 11:16 AM, Michael Collins wrote:
>
>>
>>
>> 2009/8/18 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
>> hi
>>
>> does anybody know how to send a sip options message to a registered  
>> user, using the event socket
>> or something else build in freeswitch
>> i think the ping parameter does something like this for gateways.
>>
>> what i want/need is the same thing that is provided in asterisk  
>> with the qualifying option,
>> to see how "reachable" a certain client is.
>>
>> Nothing exists at present but Mathieu Rene was looking at possibly  
>> adding an API to do this. Stay tuned for more information...
>>
>> -MC
>>
>>
>> br
>>
>> --
>> Ing. Christian L?schenkohl
>> Technische Leitung, Forschung & Entwicklung VoIP
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Thu Aug 20 11:36:49 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 20 Aug 2009 14:36:49 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8D87D8.8080001@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>
Message-ID: <700500D9-4508-4848-A198-76692574AF15@jerris.com>

You can bypass the srv records if you like by passing a :port with the  
hostname where you use it in freeswitch.


On Aug 20, 2009, at 1:28 PM, Carlos S. Antunes wrote:

> Hello!
>
> I am using Callcentric for my tests and have observed what appears  
> to me
> a possible bug in the way Freeswitch handles DNS SRV records.
>
> Callcentric uses DNS SRV records as a way to direct traffic to their  
> SIP
> server. A 'srv' 'dig' of '_sip._udp.callcentric.com' returns:
>
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha6.callcentric.com.
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha7.callcentric.com.
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha1.callcentric.com.
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha3.callcentric.com.
>
> Based on this information, Freeswitch appears to correctly round robin
> all available IP addresses except in a particular situation: in the
> middle of authorizations.
>
> For example, in a registration, Freeswitch send a packet to
> alpha1.callcentric.com. Callcentric then challenges Freeswitch with a
> Proxy Authorization request. Freeswitch then sends the packet with the
> requested credentials but not necessarily to alpha1.callcentric.com!  
> In
> many cases, instead of sticking to the 'challenging' server,  
> Freeswitch
> round robins and sends the second packet to, one of the other servers.
> This continues for a little while and eventually, simply by luck, the
> second packet is sent to the 'challenging' Callcentric server.
>
> Shouldn't Freeswitch stick to the same server when challenged for
> credentials? Is this a bug? Is there a way to make Freeswitch behave
> differently?
>
> Thanks!
>
> Carlos Antunes
> Nowthor Corporation



From brian at freeswitch.org  Thu Aug 20 11:39:44 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 13:39:44 -0500
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <700500D9-4508-4848-A198-76692574AF15@jerris.com>
References: <4A8D87D8.8080001@nowthor.com>
	<700500D9-4508-4848-A198-76692574AF15@jerris.com>
Message-ID: <DEBF02AF-E36C-47FA-88C7-363BCC91282F@freeswitch.org>

Or as I have argued today they should fix their SRV records to be zero  
weighted.

/b

On Aug 20, 2009, at 1:36 PM, Michael Jerris wrote:

> You can bypass the srv records if you like by passing a :port with the
> hostname where you use it in freeswitch.



From csa at nowthor.com  Thu Aug 20 12:56:43 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 20 Aug 2009 15:56:43 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>
References: <4A8D87D8.8080001@nowthor.com>
	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>
Message-ID: <4A8DAA7B.8090601@nowthor.com>

Raymond Chandler wrote:
> very true, but i've been reading over the RFCs on this, and it seems 
> that FreeSWITCH isn't doing anything incorrectly.
>
> in RFC3263 (section 4), when talking about client usage of SRV:
> The procedures here MUST be done exactly once per transaction, where transaction is as defined in [1]. 
> [1] being RFC3261
>
> in RFC3261 (section 8.1.3.5), when talking about 4xx responses:
> In all of the above cases, the request is retried by creating a new
>    request with the appropriate modifications.  This new request
>    constitutes a new transaction and SHOULD have the same value of the
>    Call-ID, To, and From of the previous request, but the CSeq should
>    contain a new sequence number that is one higher than the previous.
>

I'd say that based on my own and brief perusal of the relevant RFC's, 
you are right that Freeswitch is operating without violating the specs. 
Therefore, there is no bug in Freeswitch.


>
>> Is there a way to make Freeswitch behave
>> differently?
> disable-srv on the profile
>

In Callcentric's case, it doesn't help either. A request for 'A' recods 
for 'callcentric.com' returns a bunch. The only way appears to select 
and stick with only one.

> All that said, if it has to be a "bug", then it seems to me that it's 
> more of a "bug" in callcentric's service. They'd probably be better of 
> actually prioritizing their SRV records. If they want load balancing 
> and want to do checking for stale nonces, then they should be sharing 
> nonces across all of their proxies or using a proper load balancer.
>

Agreed. That being said, having a way to force Freeswitch to stick to 
the same IP address in the middle of authorization/authentication 
wouldn't violate any specs but would certainly make things easier when 
dealing with not so well implemented round robin scenarios. Do you think 
a new option could be added to Freeswitch to achieve round robin 
avoidance in these cases?

Thanks!

Carlos Antunes
Nowthor Corporation

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From alan at chandlerfamily.org.uk  Thu Aug 20 13:09:21 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Thu, 20 Aug 2009 21:09:21 +0100
Subject: [Freeswitch-users] Is it possible to unbind the * key?
Message-ID: <4A8DAD71.5000404@chandlerfamily.org.uk>

In a previous e-mail to this list I explained some problems I was having 
with the re-entering a conference and the caller keys not working. 
Nobody responded (See "Stuck on my first attempt at dialplanning" above).

I have made some good progress in getting things to work.  What appears 
to be the case is that as I callout from the conference the first time, 
I call bind_meta_app in order to enable me to get back to the conference 
from the callout call.

When I get back to the conference the * key no longer works as a 
conference control, presumably because bind_meta_app still has control 
of it.

Is there a way to unbind it before re-entering the conference?
-- 
Alan Chandler
http://www.chandlerfamily.org.uk



From intralanman at freeswitch.org  Thu Aug 20 13:14:36 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 20 Aug 2009 16:14:36 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8DAA7B.8090601@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>
	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>
	<4A8DAA7B.8090601@nowthor.com>
Message-ID: <CD7267C3-5FE4-407E-AA8B-D4D54D8B2152@freeswitch.org>

>
> Agreed. That being said, having a way to force Freeswitch to stick  
> to the same IP address in the middle of authorization/authentication  
> wouldn't violate any specs but would certainly make things easier  
> when dealing with not so well implemented round robin scenarios. Do  
> you think a new option could be added to Freeswitch to achieve round  
> robin avoidance in these cases?


patches gladly accepted ;-)

The code that handles the dns srv stuff is apparently buried in the  
nta code of sofia, so it's not as easy as it sounds to "just add an  
option".  I personally don't have the skillz needed to take on such an  
endeavor, but would love to see that as an option.  Callcentric isn't  
the only provider that has equally weighted SRVs and doing checks for  
stale nonces, so i'm sure we'll here more of this issue when we get  
more residential users using more various providers.


Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     

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From intralanman at freeswitch.org  Thu Aug 20 13:54:12 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 20 Aug 2009 16:54:12 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8DAA7B.8090601@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>
	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>
	<4A8DAA7B.8090601@nowthor.com>
Message-ID: <8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>

Actually, disregard my previous mail... this patch probably wouldn't  
be gladly accepted... I seem to have forgotten about the part where  
the DNS SRV lookup "MUST be done once per transaction"....

So if we don't do that, then we would be breaking spec. Just disabling  
SRV on the profile to which the gateway to the faulty carrier is  
attached should fix the problem.  You shouldn't actually need to hard  
code any IPs into your hosts file.

Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     



From Prometheus001 at gmx.net  Thu Aug 20 13:54:32 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 20 Aug 2009 22:54:32 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>
	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>
Message-ID: <4A8DB808.1030903@gmx.net>

Hello Brian,

yes we have updated to the latest Fritzbox Firmware. These FritzBoxes
are widely spread here in Germany. I know of a SIP provider who has > 5
Mio FritzBoxes out there. So overall I expect about 10Mio FritzBoxes in
Germany, and they are covering a big stake of in the market. So they
generally they work. I tested mine against my Asterisk without problems.

But in my Freeswitch environment this is not working, and we have manage
to couple of these Boxes. So any help is appreciated.

Best regards
Peter


Brian West schrieb:
> Besides taking a hammer to it?  Have you tried to make sure you have  
> the latest firmware?  Try setting the ptime on the fritz to 20ms?
>
> I really can't trust a product that has fritz in its name... it might  
> go on the fritz  :P  pun intended.
>
> /b
>
> On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:
>
>   
>> Any more hints?
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From csa at nowthor.com  Thu Aug 20 14:03:32 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 20 Aug 2009 17:03:32 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>
References: <4A8D87D8.8080001@nowthor.com>	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>	<4A8DAA7B.8090601@nowthor.com>
	<8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>
Message-ID: <4A8DBA24.6010203@nowthor.com>

Raymond Chandler wrote:
> Actually, disregard my previous mail... this patch probably wouldn't  
> be gladly accepted... I seem to have forgotten about the part where  
> the DNS SRV lookup "MUST be done once per transaction"....
>
>   

Hmm, where does it say that, after the lookup, one cannot use the same 
IP address as before? :)

Carlos Antunes
Nowthor Corporation



From intralanman at freeswitch.org  Thu Aug 20 14:13:07 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 20 Aug 2009 17:13:07 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8DBA24.6010203@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>	<4A8DAA7B.8090601@nowthor.com>
	<8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>
	<4A8DBA24.6010203@nowthor.com>
Message-ID: <4D73B669-478E-44EF-83D7-0AA745286ACB@freeswitch.org>

On Aug 20, 2009, at 5:03 PM, Carlos S. Antunes wrote:
> Hmm, where does it say that, after the lookup, one cannot use the same
> IP address as before? :)

Section 4 of RFC3263 as quoted in my first email....

"The procedures here MUST be done exactly once per transaction, where  
transaction is as defined in [1].
"

Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     

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From csa at nowthor.com  Thu Aug 20 14:25:22 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 20 Aug 2009 17:25:22 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4D73B669-478E-44EF-83D7-0AA745286ACB@freeswitch.org>
References: <4A8D87D8.8080001@nowthor.com>	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>	<4A8DAA7B.8090601@nowthor.com>	<8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>	<4A8DBA24.6010203@nowthor.com>
	<4D73B669-478E-44EF-83D7-0AA745286ACB@freeswitch.org>
Message-ID: <4A8DBF42.9050503@nowthor.com>

Raymond Chandler wrote:
> On Aug 20, 2009, at 5:03 PM, Carlos S. Antunes wrote:
>> Hmm, where does it say that, after the lookup, one cannot use the same
>> IP address as before? :)
>
> Section 4 of RFC3263 as quoted in my first email....
>
> "The procedures here MUST be done exactly once per transaction, where 
> transaction is as defined in [1]. 
> "
>

Raymond, sure. But do the "procedures here" preclude one from choosing 
the same host given that both the priorities and weights are the same 
for all the hosts?
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From msc at freeswitch.org  Thu Aug 20 14:35:08 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Aug 2009 14:35:08 -0700
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4A8DB808.1030903@gmx.net>
References: <4A8D5132.7010807@gmx.net>
	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>
	<4A8D5D43.3030300@gmx.net>
	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>
	<4A8DB808.1030903@gmx.net>
Message-ID: <87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>

On Thu, Aug 20, 2009 at 1:54 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello Brian,
>
> yes we have updated to the latest Fritzbox Firmware. These FritzBoxes
> are widely spread here in Germany. I know of a SIP provider who has > 5
> Mio FritzBoxes out there. So overall I expect about 10Mio FritzBoxes in
> Germany, and they are covering a big stake of in the market. So they
> generally they work. I tested mine against my Asterisk without problems.
>
> But in my Freeswitch environment this is not working, and we have manage
> to couple of these Boxes. So any help is appreciated.
>

Just curious - if it seems to be working with Asterisk but not FreeSWITCH
then could you do some tcpdumps of working vs. non-working calls and then
analyze them with Wireshark? I think Jason Garland's ClueCon presentation(s)
might be applicable here.

Thoughts?

-MC


>
> Best regards
> Peter
>
>
> Brian West schrieb:
> > Besides taking a hammer to it?  Have you tried to make sure you have
> > the latest firmware?  Try setting the ptime on the fritz to 20ms?
> >
> > I really can't trust a product that has fritz in its name... it might
> > go on the fritz  :P  pun intended.
> >
> > /b
> >
> > On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:
> >
> >
> >> Any more hints?
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From intralanman at freeswitch.org  Thu Aug 20 14:55:40 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 20 Aug 2009 17:55:40 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8DBF42.9050503@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>	<4A8DAA7B.8090601@nowthor.com>	<8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>	<4A8DBA24.6010203@nowthor.com>
	<4D73B669-478E-44EF-83D7-0AA745286ACB@freeswitch.org>
	<4A8DBF42.9050503@nowthor.com>
Message-ID: <773E8CAB-F146-452D-A3FD-AE202157A2B2@freeswitch.org>


On Aug 20, 2009, at 5:25 PM, Carlos S. Antunes wrote:

> Raymond Chandler wrote:
>>
>> On Aug 20, 2009, at 5:03 PM, Carlos S. Antunes wrote:
>>> Hmm, where does it say that, after the lookup, one cannot use the  
>>> same
>>> IP address as before? :)
>>
>> Section 4 of RFC3263 as quoted in my first email....
>>
>> "The procedures here MUST be done exactly once per transaction,  
>> where transaction is as defined in [1].
>> "
>>
>
> Raymond, sure. But do the "procedures here" preclude one from  
> choosing the same host given that both the priorities and weights  
> are the same for all the hosts?


well, not exactly, in fact... every so often, you will end up choosing  
the same 1 out of 4 hosts twice in a row at random, but the procedures  
basically say to choose one at random if they're evenly prioritized  
and evenly weighted.... so saying "i'm gonna keep this one for later  
user" kind of goes against "random"

That said, I'm really just about done with this thread since I don't  
personally agree with the spec in this case anyway since stale nonce  
checking makes sense to avoid replay attacks, etc.  Although, I  
haven't looked at the specs to see if checking for stale nonces breaks  
any specs... that might be an interesting search ;-)

Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     
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From csa at nowthor.com  Thu Aug 20 15:18:22 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 20 Aug 2009 18:18:22 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <773E8CAB-F146-452D-A3FD-AE202157A2B2@freeswitch.org>
References: <4A8D87D8.8080001@nowthor.com>	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>	<4A8DAA7B.8090601@nowthor.com>	<8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>	<4A8DBA24.6010203@nowthor.com>	<4D73B669-478E-44EF-83D7-0AA745286ACB@freeswitch.org>	<4A8DBF42.9050503@nowthor.com>
	<773E8CAB-F146-452D-A3FD-AE202157A2B2@freeswitch.org>
Message-ID: <4A8DCBAE.60107@nowthor.com>

Raymond Chandler wrote:
>
> On Aug 20, 2009, at 5:25 PM, Carlos S. Antunes wrote:
>
>> Raymond Chandler wrote:
>>> On Aug 20, 2009, at 5:03 PM, Carlos S. Antunes wrote:
>>>> Hmm, where does it say that, after the lookup, one cannot use the same
>>>> IP address as before? :)
>>>
>>> Section 4 of RFC3263 as quoted in my first email....
>>>
>>> "The procedures here MUST be done exactly once per transaction, 
>>> where transaction is as defined in [1]. 
>>> "
>>>
>>
>> Raymond, sure. But do the "procedures here" preclude one from 
>> choosing the same host given that both the priorities and weights are 
>> the same for all the hosts?
>
>
> well, not exactly, in fact... every so often, you will end up choosing 
> the same 1 out of 4 hosts twice in a row at random

Exactly! :)

> but the procedures basically say to choose one at random

Right, but isn't the generator pseudo-random, instead? (If one takes 
"random" literally, pseudo-random would violate the spec!) What prevents 
one from using the "right" pseudo-random generator?


>
> That said, I'm really just about done with this thread since I don't 
> personally agree with the spec in this case anyway since stale nonce 
> checking makes sense to avoid replay attacks, etc.  Although, I 
> haven't looked at the specs to see if checking for stale nonces breaks 
> any specs... that might be an interesting search ;-)
>

I am going to try to find an appropriate IETF mailing list to ask some 
questions about this random SRV stuff and will repost here once I have 
some additional info.

Thanks for trying to keep me honest, though! :)

Carlos Antunes
Nowthor Corporation

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From brian at freeswitch.org  Thu Aug 20 15:24:50 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 17:24:50 -0500
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8DCBAE.60107@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>	<4A8DAA7B.8090601@nowthor.com>	<8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>	<4A8DBA24.6010203@nowthor.com>	<4D73B669-478E-44EF-83D7-0AA745286ACB@freeswitch.org>	<4A8DBF42.9050503@nowthor.com>
	<773E8CAB-F146-452D-A3FD-AE202157A2B2@freeswitch.org>
	<4A8DCBAE.60107@nowthor.com>
Message-ID: <3CB13AD6-A2A0-448B-821A-5878961E110C@freeswitch.org>

Read RFC 2782, About the significance of a 0 weight vs weighted.  Ray  
proved that if you have your records weighted at 0 it behaves correctly.

/b

On Aug 20, 2009, at 5:18 PM, Carlos S. Antunes wrote:

> I am going to try to find an appropriate IETF mailing list to ask  
> some questions about this random SRV stuff and will repost here once  
> I have some additional info.
>
> Thanks for trying to keep me honest, though! :)



From eric.des.courtis at gmail.com  Thu Aug 20 16:51:01 2009
From: eric.des.courtis at gmail.com (Eric des Courtis)
Date: Thu, 20 Aug 2009 19:51:01 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
	<87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
Message-ID: <539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>

Matt,

For your information the tones you gave me are exactly 738Hz. If you
want to try that tone detection thing.

Cheers.

Eric des Courtis

On Thu, Aug 20, 2009 at 2:20 PM, Michael Collins<msc at freeswitch.org> wrote:
>
>
> On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood <steveu at coppice.org>
> wrote:
>>
>> On 08/20/2009 05:22 AM, Michael Collins wrote:
>> >
>> > ? ? There is no noise on those 3 beeps. In fact, for something that's
>> > been
>> > ? ? through ulaw/alaw compression those beeps are very clean. They are
>> > ? ? quite
>> > ? ? short, though.
>> >
>> >
>> > Heck yeah they're short! Steve, in your experience is there a
>> > practical way to detect a beep that short without chewing up system
>> > resources or having lots of false positives?
>> > -MC
>> >
>> The tone samples I just looked at are about 130ms long. The problem is
>> the detector is trying to be a very open ended detector of anything
>> narrowband enough to be a single tone, and of any duration beyond some
>> small minimum. Its difficult to make such a thing voice immune unless
>> you can also count on a very large signal to noise ratio. With a digital
>> trunk you can probably rely on a large SNR, but what happens when people
>> use analogue lines? There is a reason why DTMF detectors try hard to
>> work down to about 10dB SNR. :-)
>>
>> Steve
>
> Thanks for the lesson uncle Steve! I'm guessing that the OP will need a new
> strategy. Possibly waiting for silence? Not sure what's the best way to go
> but I'm interested in hearing if someone has a solution.
>
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From eric.des.courtis at gmail.com  Thu Aug 20 16:54:52 2009
From: eric.des.courtis at gmail.com (Eric des Courtis)
Date: Thu, 20 Aug 2009 19:54:52 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
	<87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
	<539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>
Message-ID: <539da8a30908201654u3f660a43pf32100223fa1863a@mail.gmail.com>

Matt,

As is mod_vmd will not detect tones shorter then 138ms. However I
could get that value down to ~30ms at best by making a few
modifications to the algorithm.

Cheers.

Eric des Courtis


On Thu, Aug 20, 2009 at 7:51 PM, Eric des
Courtis<eric.des.courtis at gmail.com> wrote:
> Matt,
>
> For your information the tones you gave me are exactly 738Hz. If you
> want to try that tone detection thing.
>
> Cheers.
>
> Eric des Courtis
>
> On Thu, Aug 20, 2009 at 2:20 PM, Michael Collins<msc at freeswitch.org> wrote:
>>
>>
>> On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood <steveu at coppice.org>
>> wrote:
>>>
>>> On 08/20/2009 05:22 AM, Michael Collins wrote:
>>> >
>>> > ? ? There is no noise on those 3 beeps. In fact, for something that's
>>> > been
>>> > ? ? through ulaw/alaw compression those beeps are very clean. They are
>>> > ? ? quite
>>> > ? ? short, though.
>>> >
>>> >
>>> > Heck yeah they're short! Steve, in your experience is there a
>>> > practical way to detect a beep that short without chewing up system
>>> > resources or having lots of false positives?
>>> > -MC
>>> >
>>> The tone samples I just looked at are about 130ms long. The problem is
>>> the detector is trying to be a very open ended detector of anything
>>> narrowband enough to be a single tone, and of any duration beyond some
>>> small minimum. Its difficult to make such a thing voice immune unless
>>> you can also count on a very large signal to noise ratio. With a digital
>>> trunk you can probably rely on a large SNR, but what happens when people
>>> use analogue lines? There is a reason why DTMF detectors try hard to
>>> work down to about 10dB SNR. :-)
>>>
>>> Steve
>>
>> Thanks for the lesson uncle Steve! I'm guessing that the OP will need a new
>> strategy. Possibly waiting for silence? Not sure what's the best way to go
>> but I'm interested in hearing if someone has a solution.
>>
>> -MC
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>


From mattdfong at gmail.com  Thu Aug 20 17:36:43 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Thu, 20 Aug 2009 17:36:43 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <539da8a30908201654u3f660a43pf32100223fa1863a@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
	<87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
	<539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>
	<539da8a30908201654u3f660a43pf32100223fa1863a@mail.gmail.com>
Message-ID: <4256bf830908201736v63eb9188if1d82842a885b9d8@mail.gmail.com>

I changed

/*! Minimum time for a beep. */
#define MIN_TIME 8000
to 6500 and it seemed to work, but I'm not sure how many false positives I
will get in a real-world environment. at 4000 it fired the event like 5
times in a session, but 6500 only once. Do you think I should expect a lot
of false positives after changing this value?

--matt
http://www.hellohunter.com

On Thu, Aug 20, 2009 at 4:54 PM, Eric des Courtis <
eric.des.courtis at gmail.com> wrote:

> Matt,
>
> As is mod_vmd will not detect tones shorter then 138ms. However I
> could get that value down to ~30ms at best by making a few
> modifications to the algorithm.
>
> Cheers.
>
> Eric des Courtis
>
>
> On Thu, Aug 20, 2009 at 7:51 PM, Eric des
> Courtis<eric.des.courtis at gmail.com> wrote:
>  > Matt,
> >
> > For your information the tones you gave me are exactly 738Hz. If you
> > want to try that tone detection thing.
> >
> > Cheers.
> >
> > Eric des Courtis
> >
> > On Thu, Aug 20, 2009 at 2:20 PM, Michael Collins<msc at freeswitch.org>
> wrote:
> >>
> >>
> >> On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood <steveu at coppice.org>
> >> wrote:
> >>>
> >>> On 08/20/2009 05:22 AM, Michael Collins wrote:
> >>> >
> >>> >     There is no noise on those 3 beeps. In fact, for something that's
> >>> > been
> >>> >     through ulaw/alaw compression those beeps are very clean. They
> are
> >>> >     quite
> >>> >     short, though.
> >>> >
> >>> >
> >>> > Heck yeah they're short! Steve, in your experience is there a
> >>> > practical way to detect a beep that short without chewing up system
> >>> > resources or having lots of false positives?
> >>> > -MC
> >>> >
> >>> The tone samples I just looked at are about 130ms long. The problem is
> >>> the detector is trying to be a very open ended detector of anything
> >>> narrowband enough to be a single tone, and of any duration beyond some
> >>> small minimum. Its difficult to make such a thing voice immune unless
> >>> you can also count on a very large signal to noise ratio. With a
> digital
> >>> trunk you can probably rely on a large SNR, but what happens when
> people
> >>> use analogue lines? There is a reason why DTMF detectors try hard to
> >>> work down to about 10dB SNR. :-)
> >>>
> >>> Steve
> >>
> >> Thanks for the lesson uncle Steve! I'm guessing that the OP will need a
> new
> >> strategy. Possibly waiting for silence? Not sure what's the best way to
> go
> >> but I'm interested in hearing if someone has a solution.
> >>
> >> -MC
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From eric.des.courtis at gmail.com  Thu Aug 20 18:53:25 2009
From: eric.des.courtis at gmail.com (Eric des Courtis)
Date: Thu, 20 Aug 2009 21:53:25 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4256bf830908201736v63eb9188if1d82842a885b9d8@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
	<87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
	<539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>
	<539da8a30908201654u3f660a43pf32100223fa1863a@mail.gmail.com>
	<4256bf830908201736v63eb9188if1d82842a885b9d8@mail.gmail.com>
Message-ID: <539da8a30908201853s6d4307e7j996849a4388aa3d3@mail.gmail.com>

Matt,

I think the only way to know for sure is to try it. I would try to get
the value as high as possible while still detecting that 738Hz sine
(with a small margin of error). Lowering the value increases false
positives rapidly.

Eric des Courtis

On Thu, Aug 20, 2009 at 8:36 PM, Matthew Fong<mattdfong at gmail.com> wrote:
> I changed
>
> /*! Minimum time for a beep. */
> #define MIN_TIME 8000
> to 6500 and it seemed to work, but I'm not sure how many false positives I
> will get in a real-world environment. at 4000 it fired the event like 5
> times in a session, but 6500 only once. Do you think I should expect a lot
> of false positives after changing this value?
>
> --matt
> http://www.hellohunter.com
>
> On Thu, Aug 20, 2009 at 4:54 PM, Eric des Courtis
> <eric.des.courtis at gmail.com> wrote:
>>
>> Matt,
>>
>> As is mod_vmd will not detect tones shorter then 138ms. However I
>> could get that value down to ~30ms at best by making a few
>> modifications to the algorithm.
>>
>> Cheers.
>>
>> Eric des Courtis
>>
>>
>> On Thu, Aug 20, 2009 at 7:51 PM, Eric des
>> Courtis<eric.des.courtis at gmail.com> wrote:
>> > Matt,
>> >
>> > For your information the tones you gave me are exactly 738Hz. If you
>> > want to try that tone detection thing.
>> >
>> > Cheers.
>> >
>> > Eric des Courtis
>> >
>> > On Thu, Aug 20, 2009 at 2:20 PM, Michael Collins<msc at freeswitch.org>
>> > wrote:
>> >>
>> >>
>> >> On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood <steveu at coppice.org>
>> >> wrote:
>> >>>
>> >>> On 08/20/2009 05:22 AM, Michael Collins wrote:
>> >>> >
>> >>> > ? ? There is no noise on those 3 beeps. In fact, for something
>> >>> > that's
>> >>> > been
>> >>> > ? ? through ulaw/alaw compression those beeps are very clean. They
>> >>> > are
>> >>> > ? ? quite
>> >>> > ? ? short, though.
>> >>> >
>> >>> >
>> >>> > Heck yeah they're short! Steve, in your experience is there a
>> >>> > practical way to detect a beep that short without chewing up system
>> >>> > resources or having lots of false positives?
>> >>> > -MC
>> >>> >
>> >>> The tone samples I just looked at are about 130ms long. The problem is
>> >>> the detector is trying to be a very open ended detector of anything
>> >>> narrowband enough to be a single tone, and of any duration beyond some
>> >>> small minimum. Its difficult to make such a thing voice immune unless
>> >>> you can also count on a very large signal to noise ratio. With a
>> >>> digital
>> >>> trunk you can probably rely on a large SNR, but what happens when
>> >>> people
>> >>> use analogue lines? There is a reason why DTMF detectors try hard to
>> >>> work down to about 10dB SNR. :-)
>> >>>
>> >>> Steve
>> >>
>> >> Thanks for the lesson uncle Steve! I'm guessing that the OP will need a
>> >> new
>> >> strategy. Possibly waiting for silence? Not sure what's the best way to
>> >> go
>> >> but I'm interested in hearing if someone has a solution.
>> >>
>> >> -MC
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From gabe at gundy.org  Thu Aug 20 20:16:42 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 20 Aug 2009 21:16:42 -0600
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4256bf830908201736v63eb9188if1d82842a885b9d8@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
	<87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
	<539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>
	<539da8a30908201654u3f660a43pf32100223fa1863a@mail.gmail.com>
	<4256bf830908201736v63eb9188if1d82842a885b9d8@mail.gmail.com>
Message-ID: <903da5680908202016w2e146c79q408146001faeecfe@mail.gmail.com>

On Thu, Aug 20, 2009 at 6:36 PM, Matthew Fong<mattdfong at gmail.com> wrote:
> /*! Minimum time for a beep. */
> #define MIN_TIME 8000
> to 6500 and it seemed to work, but I'm not sure how many false positives I
> will get in a real-world environment. at 4000 it fired the event like 5
> times in a session, but 6500 only once. Do you think I should expect a lot
> of false positives after changing this value?

YES.


From gabe at gundy.org  Thu Aug 20 20:17:02 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 20 Aug 2009 21:17:02 -0600
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <903da5680908202016w2e146c79q408146001faeecfe@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
	<87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
	<539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>
	<539da8a30908201654u3f660a43pf32100223fa1863a@mail.gmail.com>
	<4256bf830908201736v63eb9188if1d82842a885b9d8@mail.gmail.com>
	<903da5680908202016w2e146c79q408146001faeecfe@mail.gmail.com>
Message-ID: <903da5680908202017w5babe925uea912a36b63d9c2c@mail.gmail.com>

On Thu, Aug 20, 2009 at 9:16 PM, Gabriel Gunderson<gabe at gundy.org> wrote:
> On Thu, Aug 20, 2009 at 6:36 PM, Matthew Fong<mattdfong at gmail.com> wrote:
>> /*! Minimum time for a beep. */
>> #define MIN_TIME 8000
>> to 6500 and it seemed to work, but I'm not sure how many false positives I
>> will get in a real-world environment. at 4000 it fired the event like 5
>> times in a session, but 6500 only once. Do you think I should expect a lot
>> of false positives after changing this value?
>
> YES.

Err, NO.


From ahmedmunir007 at gmail.com  Thu Aug 20 21:54:44 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Fri, 21 Aug 2009 10:54:44 +0600
Subject: [Freeswitch-users] Application Variable list needed
Message-ID: <a60fed5e0908202154pd0bb1f6ga4a060be8b8e5c34@mail.gmail.com>

Hi,
I'm newbie in FreeSwitch, currently I'm replacing asterisk with FreeSwitch.
I need list of application variables so I can easily translate my asterisk
configuration in FreeSwitch. Like I don't know how I translate my dial plan
in FreeSwitch as I like to forward it to another context as sample listed
below;

exten => _X.,1,Goto(origination_incoming,${EXTEN},1)
exten => _X.,n,Hangup()

My another request is how can I change extensions.ael in asterisk to
FreeSwitch? Is there alternative way of doing this? Kindly do let me know.

-- 
Regards,

Ahmed Munir
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From diego.viola at gmail.com  Thu Aug 20 22:35:13 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 21 Aug 2009 01:35:13 -0400
Subject: [Freeswitch-users] Application Variable list needed
In-Reply-To: <a60fed5e0908202154pd0bb1f6ga4a060be8b8e5c34@mail.gmail.com>
References: <a60fed5e0908202154pd0bb1f6ga4a060be8b8e5c34@mail.gmail.com>
Message-ID: <86a32abc0908202235x6ca9cbdap8a0574429da46da@mail.gmail.com>

For dialplan apps take a look at.

http://wiki.freeswitch.org/wiki/Mod_dptools

For channel variables:

http://wiki.freeswitch.org/wiki/Channel_Variables

For FS commands:

http://wiki.freeswitch.org/wiki/Mod_commands

You could use transfer in order to send a call to another extension/context.


On Fri, Aug 21, 2009 at 12:54 AM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:

> Hi,
> I'm newbie in FreeSwitch, currently I'm replacing asterisk with FreeSwitch.
> I need list of application variables so I can easily translate my asterisk
> configuration in FreeSwitch. Like I don't know how I translate my dial plan
> in FreeSwitch as I like to forward it to another context as sample listed
> below;
>
> exten => _X.,1,Goto(origination_incoming,${EXTEN},1)
> exten => _X.,n,Hangup()
>
> My another request is how can I change extensions.ael in asterisk to
> FreeSwitch? Is there alternative way of doing this? Kindly do let me know.
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Thu Aug 20 22:51:05 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 21 Aug 2009 01:51:05 -0400
Subject: [Freeswitch-users] Application Variable list needed
In-Reply-To: <86a32abc0908202235x6ca9cbdap8a0574429da46da@mail.gmail.com>
References: <a60fed5e0908202154pd0bb1f6ga4a060be8b8e5c34@mail.gmail.com>
	<86a32abc0908202235x6ca9cbdap8a0574429da46da@mail.gmail.com>
Message-ID: <86a32abc0908202251y68ee91fhb94f5b5718ba4597@mail.gmail.com>

Let us know if you have more questions or need more help :)

On Fri, Aug 21, 2009 at 1:35 AM, Diego Viola <diego.viola at gmail.com> wrote:

> For dialplan apps take a look at.
>
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> For channel variables:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables
>
> For FS commands:
>
> http://wiki.freeswitch.org/wiki/Mod_commands
>
> You could use transfer in order to send a call to another
> extension/context.
>
>
> On Fri, Aug 21, 2009 at 12:54 AM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:
>
>> Hi,
>> I'm newbie in FreeSwitch, currently I'm replacing asterisk with
>> FreeSwitch. I need list of application variables so I can easily translate
>> my asterisk configuration in FreeSwitch. Like I don't know how I translate
>> my dial plan in FreeSwitch as I like to forward it to another context as
>> sample listed below;
>>
>> exten => _X.,1,Goto(origination_incoming,${EXTEN},1)
>> exten => _X.,n,Hangup()
>>
>> My another request is how can I change extensions.ael in asterisk to
>> FreeSwitch? Is there alternative way of doing this? Kindly do let me know.
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From b_ball_henry at hotmail.com  Fri Aug 21 01:06:24 2009
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Fri, 21 Aug 2009 16:06:24 +0800
Subject: [Freeswitch-users] Dialing SIP URL issue
Message-ID: <59ad9ca10908210106g2541ac79x4e511be8820ffa20@mail.gmail.com>

Hi:

I try to dial sip url from my softphone but seems like the sip address is
being processed by sofia before it pass to the dialplan. The example here is
:

*X-lite(softphone) dials -> 1009 at 4.2.2.2 (it's fake sip address, the purpose
was just to test what's being passed to dialplan)
sofia receives the invite and return with trying
sofia pass the destination number to dailplan with "1009" (without the
"sip:" in front and without the "@4.2.2.2" after it)
*
Please see pastebin for full log. http://pastebin.freeswitch.org/10089
ignore anything after line 80, because it's not my point, and the
destination is a fake address.

I would like to know how do you actually pass a full sip url to the dialplan
to do the regex match. Because from the default.xml dialplan, it comes with
an example sip url dialing extension that match's *^sip:(.*)$ *. So I assume
there must be a way of passing full sip url to the dialplan. Here is the
example dialplan expecting sofia to pass it a full sip url:

 <!-- dial via SIP uri -->
    <extension name="sip_uri">
      <condition field="destination_number" expression="^sip:(.*)$">
        <action application="bridge" data="sofia/${use_profile}/$1"/>
      </condition>
    </extension>

Thanks
-- 
Henry Huang
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From woodydickson at gmail.com  Fri Aug 21 02:54:09 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Fri, 21 Aug 2009 17:54:09 +0800
Subject: [Freeswitch-users] zombie channels
Message-ID: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>

Hi,

I am running 1.0.4 right now using latest trunk.

After a high traffic session, I do "show channels", I would find a bunch of
"CS_HIBERNATE" channels that don't get removed after all the traffic is
gone.

Does anyone know what is the case of thoes CS_HIBERNATE'd channels?  How can
I set  a timeout for those channels to be removed?

Thanks,
Woody
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From dave at 3c.co.uk  Fri Aug 21 03:07:04 2009
From: dave at 3c.co.uk (David Knell)
Date: Fri, 21 Aug 2009 13:07:04 +0300
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
References: <4A8D5132.7010807@gmx.net>
	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>
	<4A8D5D43.3030300@gmx.net>
	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>
	<4A8DB808.1030903@gmx.net>
	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
Message-ID: <1250849224.4663.68.camel@dk-d820>

On Thu, 2009-08-20 at 14:35 -0700, Michael Collins wrote:


> Just curious - if it seems to be working with Asterisk but not
> FreeSWITCH then could you do some tcpdumps of working vs. non-working
> calls and then analyze them with Wireshark? I think Jason Garland's
> ClueCon presentation(s) might be applicable here. 

Just to deepen the mystery a little, we have a FRITZ!Box here in Greece,
and it works like a little champ for us.  Firmware's 06.04.49, it's
talking to a FreeSWITCH box in London.  It's set to pick its own codec
(but the other end only supports G.711), VAD's off.

Cheers --

Dave





From fraunhofer.lists.freeswitch-001 at traced.net  Fri Aug 21 04:03:07 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Fri, 21 Aug 2009 13:03:07 +0200
Subject: [Freeswitch-users] zombie channels
In-Reply-To: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
References: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
Message-ID: <a37510170908210403o66aebaffw7cf913fec6e3615@mail.gmail.com>

Hi Woody,

2009/8/21 Woody Dickson <woodydickson at gmail.com>:

> After a high traffic session, I do "show channels", I would find a bunch of
> "CS_HIBERNATE" channels that don't get removed after all the traffic is
> gone.
>
> Does anyone know what is the case of thoes CS_HIBERNATE'd channels?? How can
> I set? a timeout for those channels to be removed?

Is this simiilar to "my" bugreport?

http://jira.freeswitch.org/browse/FSCORE-415

what do you do in your "high traffic"-scenario? We've got tons of
CS_HIBERNATE channels and even the sip stack is somehow affected
(ignoring messages) but it looks like it's the scheduler to blame.
In your example this would mean that the "garbage collection" for
finished sessions is not run.
Can you get rid of the channels using "uuid_kill" or a sweeping swipe
"fsctl hupall"?

Greetinx
  Beni.


From tparikh at gmail.com  Thu Aug 20 22:30:33 2009
From: tparikh at gmail.com (Tapan Parikh)
Date: Thu, 20 Aug 2009 22:30:33 -0700
Subject: [Freeswitch-users] Accepting google talk friend requests
Message-ID: <1ecdcb6a0908202230s7324790bvf0c5adae7397b4a2@mail.gmail.com>

Hi Folks -

Im a relative newbie freeswitch and first of all wanted to thank you
for all the great work u have done here.

My question is about mod dingaling, and specifically being able to get
incoming calls from Google Talk.  Ive got the client set up on
Freeswitch, and am able to receive calls, IMs, etc. once I accept the
friend invite.

However, new accounts cannot talk to FS until they are already
friends.  Is there a way to get FS / mod dingaling to automatically
accept incoming friend requests?

I saw something in the archives about this, but no obvious resolution.
 People seem to say this is already true, but I dont even see the
incoming invite requests coming in w/ DL in debug mode.

Thanks in advance,
Tapan


From panayotov.vd at gmail.com  Thu Aug 20 23:45:58 2009
From: panayotov.vd at gmail.com (Vassil Panayotov)
Date: Fri, 21 Aug 2009 09:45:58 +0300
Subject: [Freeswitch-users] Sangoma A500 and FreeSWITCH
Message-ID: <8a9b664c0908202345g6dd56154v2abba5d76a9e663e@mail.gmail.com>

Hi,

Is it already possible to use FreeSWITCH with A500 BRI card? I found a
thread (
http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg09633.html)
stating that the support is not quite ready.

Can you give me some pointers about A500 configuration with FreeSWITCH? I
need nothing fancy - just inbound/outbound calls to telco.

Best regards,
V. Panayotov
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From odermann at googlemail.com  Fri Aug 21 05:46:56 2009
From: odermann at googlemail.com (Dennis)
Date: Fri, 21 Aug 2009 14:46:56 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup cause
Message-ID: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>

hi,

we are using fs for different services, but we never used it, to
connect sip-phones directly with fs.

now we want to do so, but we encounter big problems. everything works
fine, but after 120 seconds fs hangs up with the hangup cause
RECOVERY_ON_TIMER_EXPIRE.

it seems that this has something to do with nat problems (i read about
it in nabble), but we can not figure out, how to fix the problem.

we are not using stun, because all ip-addresses are static. we opened
our firewall for our ip-addresses and opened the following ports:
5060 - 5091 TCP / UDP
10000 - 32767 UDP

we are using a quite actual version of fs - the revision ist 13783.

we set up a profile in the internal.xml and played a lot with
different settings. we can talk to each other, make outbound calls and
receive calls - but we can not figure out, how to avoid the hangup
after 120 seconds.

thanks a lot for your help
dennis


From chris.chen2004 at gmail.com  Fri Aug 21 06:16:50 2009
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Fri, 21 Aug 2009 09:16:50 -0400
Subject: [Freeswitch-users] Accepting google talk friend requests
In-Reply-To: <1ecdcb6a0908202230s7324790bvf0c5adae7397b4a2@mail.gmail.com>
References: <1ecdcb6a0908202230s7324790bvf0c5adae7397b4a2@mail.gmail.com>
Message-ID: <507898380908210616p4a0f9f29v98287286fa108d6a@mail.gmail.com>

Hi Tapan, if your google talk is loaded as client mode in FreeSWITCH, you
cannot automatically accept new incoming
invite requests. But if you have google talk (mod_dingaling) loaded as
component mode (assuming you have your
own jabber server setup properly with google federation  etc, and loaded
with server.xml), for the invite requests to some
FreeSWITCH built-in accounts such as "user+bla blah at your jabber server"
"ext+blah blah@ your jabber server",
they will be automatically accepted.

Hope this helps.

Chris


On Fri, Aug 21, 2009 at 1:30 AM, Tapan Parikh <tparikh at gmail.com> wrote:

> Hi Folks -
>
> Im a relative newbie freeswitch and first of all wanted to thank you
> for all the great work u have done here.
>
> My question is about mod dingaling, and specifically being able to get
> incoming calls from Google Talk.  Ive got the client set up on
> Freeswitch, and am able to receive calls, IMs, etc. once I accept the
> friend invite.
>
> However, new accounts cannot talk to FS until they are already
> friends.  Is there a way to get FS / mod dingaling to automatically
> accept incoming friend requests?
>
> I saw something in the archives about this, but no obvious resolution.
>  People seem to say this is already true, but I dont even see the
> incoming invite requests coming in w/ DL in debug mode.
>
> Thanks in advance,
> Tapan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Aug 21 06:20:42 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 21 Aug 2009 08:20:42 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
Message-ID: <B30F06D6-3D0E-4405-B63B-907246EEEEC8@freeswitch.org>

You have a NAT issue.

/b

On Aug 21, 2009, at 7:46 AM, Dennis wrote:

> hi,
>
> we are using fs for different services, but we never used it, to
> connect sip-phones directly with fs.
>
> now we want to do so, but we encounter big problems. everything works
> fine, but after 120 seconds fs hangs up with the hangup cause
> RECOVERY_ON_TIMER_EXPIRE.
>
> it seems that this has something to do with nat problems (i read about
> it in nabble), but we can not figure out, how to fix the problem.
>
> we are not using stun, because all ip-addresses are static. we opened
> our firewall for our ip-addresses and opened the following ports:
> 5060 - 5091 TCP / UDP
> 10000 - 32767 UDP
>
> we are using a quite actual version of fs - the revision ist 13783.
>
> we set up a profile in the internal.xml and played a lot with
> different settings. we can talk to each other, make outbound calls and
> receive calls - but we can not figure out, how to avoid the hangup
> after 120 seconds.
>
> thanks a lot for your help
> dennis
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mrene_lists at avgs.ca  Fri Aug 21 06:31:30 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 21 Aug 2009 09:31:30 -0400
Subject: [Freeswitch-users] zombie channels
In-Reply-To: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
References: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
Message-ID: <87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>

Hi,

CS_REPORTING is the state in which cdrs are written, if the channel  
gets stuck in that state, the cdr module you are using is probably  
hanging somewhere.

Use the "freeswitch-gcore" script in your source tree's scripts  
directory to generate a bug report for hanging channels.

should be like..

cd /usr/src/freeswitch # or whatever your source tree is
bash ./scripts/freeswitch-gcore > bugreport.txt

then submit it on http://jira.freeswitch.org/ so we can look at it.

As you wish, you can also hop on #freeswitch / irc.freenode.net and  
have someone look into it.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 21-Aug-09, at 5:54 AM, Woody Dickson wrote:

> Hi,
>
> I am running 1.0.4 right now using latest trunk.
>
> After a high traffic session, I do "show channels", I would find a  
> bunch of "CS_HIBERNATE" channels that don't get removed after all  
> the traffic is gone.
>
> Does anyone know what is the case of thoes CS_HIBERNATE'd channels?   
> How can I set  a timeout for those channels to be removed?
>
> Thanks,
> Woody
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From asannucci at gmail.com  Fri Aug 21 06:35:26 2009
From: asannucci at gmail.com (bakko)
Date: Fri, 21 Aug 2009 15:35:26 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
Message-ID: <E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>

Do you have those lines in switch.conf file?

    <!--RTP port range -->
    <param name="rtp-start-port" value="10000"/>
    <param name="rtp-end-port" value="32767"/>


BR


From moises.silva at gmail.com  Fri Aug 21 07:26:56 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 21 Aug 2009 10:26:56 -0400
Subject: [Freeswitch-users] Sangoma A500 and FreeSWITCH
In-Reply-To: <8a9b664c0908202345g6dd56154v2abba5d76a9e663e@mail.gmail.com>
References: <8a9b664c0908202345g6dd56154v2abba5d76a9e663e@mail.gmail.com>
Message-ID: <c4d05cbe0908210726j3b643fcapd3901fa5d92f28f6@mail.gmail.com>

>
> Can you give me some pointers about A500 configuration with FreeSWITCH? I
> need nothing fancy - just inbound/outbound calls to telco.
>

There is some progress and you should be able to get it working with this:
http://wiki.sangoma.com/boostbri

If you find any problem please contact techdesk at sangoma.com and they will
take care of helping you to get you up and running.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From msc at freeswitch.org  Fri Aug 21 08:06:56 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 21 Aug 2009 08:06:56 -0700
Subject: [Freeswitch-users] Application Variable list needed
In-Reply-To: <a60fed5e0908202154pd0bb1f6ga4a060be8b8e5c34@mail.gmail.com>
References: <a60fed5e0908202154pd0bb1f6ga4a060be8b8e5c34@mail.gmail.com>
Message-ID: <5C49E69C-5C6E-4B26-9845-CAB0CE765705@freeswitch.org>

Be sure to visit wiki.freeswitch.org and search for "rosetta stone"  
which will take you to a page that helps translate Asterisk concepts  
into FreeSWITCH concepts.

It will seem strange at first but once you get over the hump you will  
really appreciate the power of FreeSWITCH.  Be sure to visit the IRC  
channel for a friendly place to ask questions.

-MC

Sent from my iPhone

On Aug 20, 2009, at 9:54 PM, Ahmed Munir <ahmedmunir007 at gmail.com>  
wrote:

> Hi,
>
> I'm newbie in FreeSwitch, currently I'm replacing asterisk with  
> FreeSwitch. I need list of application variables so I can easily  
> translate my asterisk configuration in FreeSwitch. Like I don't know  
> how I translate my dial plan in FreeSwitch as I like to forward it  
> to another context as sample listed below;
>
> exten => _X.,1,Goto(origination_incoming,${EXTEN},1)
> exten => _X.,n,Hangup()
>
> My another request is how can I change extensions.ael in asterisk to  
> FreeSwitch? Is there alternative way of doing this? Kindly do let me  
> know.
>
> -- 
> Regards,
>
> Ahmed Munir
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From Prometheus001 at gmx.net  Fri Aug 21 08:38:31 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 21 Aug 2009 17:38:31 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>	<4A8DB808.1030903@gmx.net>
	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
Message-ID: <4A8EBF77.1080204@gmx.net>

Hello Michael,

I made some tests with Freeswitch and Fritzbox and found by Wireshark that:
within one call

    * Freeswitch starts sending 20msec packets, then after ~0,2 second
      sends 30msec packets
    * FritzBox always sends 30msec packets.

The average jitter is below 2 msec in both directions.
 
The below logs shows that Freeswitch considers the FritzBox to be broken
and starts using 30msec packets. But there is no SIP message from FS to
Fritzbox telling him that FB will use 30msec packets. SDP from FS to
Fritzbox always shows ptime:20

BTW: We can ship you a FritzBox if you need one for testing.

Best regards
Peter


Log:

2009-08-21 17:15:50.271555 [DEBUG] switch_rtp.c:1138 Starting timer
[soft] 160 bytes per 20ms
2009-08-21 17:15:50.281563 [INFO] mod_sofia.c:1499 Ring SDP:
v=0
o=FreeSWITCH 1250837460 1250837461 IN IP4 182.xxx.xx.xxx
s=FreeSWITCH
c=IN IP4 182.xxx.xx.xxx
t=0 0
m=audio 30290 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-08-21 17:15:50.281563 [NOTICE] mod_sofia.c:1502 Pre-Answer
sofia/internal/02xxxxxxxxx at fs1.my.domain!
2009-08-21 17:15:50.281563 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/02xxxxxxxxx at fs1.my.domain [BREAK]
EXECUTE sofia/internal/02xxxxxxxxx at fs1.my.domain
playback(voicemail/8000/vm-that_was_an_invalid_ext.wav)
2009-08-21 17:15:50.281563 [DEBUG] switch_ivr_play_say.c:1097 Codec
Activated L16 at 8000hz 1 channels 20ms
2009-08-21 17:15:50.281563 [DEBUG] sofia.c:3302 Channel
sofia/internal/02xxxxxxxxx at fs1.my.domain entering state [early][183]
2009-08-21 17:15:50.281563 [DEBUG] switch_core_io.c:649
sofia/internal/02xxxxxxxxx at fs1.my.domain receive message
[TRANSCODING_NECESSARY]
2009-08-21 17:15:50.531551 [WARNING] mod_sofia.c:799 We were told to use
ptime 20 but what they meant to say was 30
This issue has so far been identified to happen on the following broken
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken
who knows what will happen..







From markmorreny at gmail.com  Fri Aug 21 08:41:13 2009
From: markmorreny at gmail.com (mark morreny)
Date: Fri, 21 Aug 2009 23:41:13 +0800
Subject: [Freeswitch-users] hanging problem with switch_scheduler_add_task
Message-ID: <20ad6b920908210841l453b2833v9ac12612c174820b@mail.gmail.com>

Hi,

I am experiencing some hanging when fs is executing
switch_scheduler_add_task.

switch_scheduler_add_task(switch_epoch_time_now(NULL) , data_flush_callback,
"data_flush","core",0,NULL,SSHF_NONE|SSHF_NO_DEL);

I place switch_scheduler_add_task in my SWITCH_MODULE_LOAD_FUNCTION and
sometimes, hanging occurs on that particular line.  My data_flush_callback
is as follows.  I debug the module and it does not even enter
data_flush_callback.  It is hanging at switch_scheduler_add_task().

SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {

    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "starting to
flush cdr record...\n");

    int last_sequence = flush_data_to_csv();
    if (last_sequence > -1 ) {

        update_last_seq(last_sequence);
    }
    task->runtime = switch_epoch_time_now(NULL) + globals.cycle_time;

}

Does anyone know why?

Thanks,
Mark
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From panayotov.vd at gmail.com  Fri Aug 21 08:56:56 2009
From: panayotov.vd at gmail.com (Vassil Panayotov)
Date: Fri, 21 Aug 2009 18:56:56 +0300
Subject: [Freeswitch-users] Sangoma A500 and FreeSWITCH
Message-ID: <8a9b664c0908210856m3b90f9e1n233b7fe7eb9467a9@mail.gmail.com>

I already tried the boostbri config, but with no success...
I will contact the support.

Thank you!

Best regards,
V. Panayotov
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From mrene_lists at avgs.ca  Fri Aug 21 08:59:04 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 21 Aug 2009 11:59:04 -0400
Subject: [Freeswitch-users] hanging problem with
	switch_scheduler_add_task
In-Reply-To: <20ad6b920908210841l453b2833v9ac12612c174820b@mail.gmail.com>
References: <20ad6b920908210841l453b2833v9ac12612c174820b@mail.gmail.com>
Message-ID: <B4BB7488-5C31-4447-BAD7-EAA062C467CC@avgs.ca>

Means another task is hanging, do a "thread apply all bt" in gdb and  
look for scheduler tasks.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 21-Aug-09, at 11:41 AM, mark morreny wrote:

> Hi,
>
> I am experiencing some hanging when fs is executing  
> switch_scheduler_add_task.
>
> switch_scheduler_add_task(switch_epoch_time_now(NULL) ,  
> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE| 
> SSHF_NO_DEL);
>
> I place switch_scheduler_add_task in my SWITCH_MODULE_LOAD_FUNCTION  
> and sometimes, hanging occurs on that particular line.  My  
> data_flush_callback is as follows.  I debug the module and it does  
> not even enter data_flush_callback.  It is hanging at  
> switch_scheduler_add_task().
>
> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>
>     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,  
> "starting to flush cdr record...\n");
>
>     int last_sequence = flush_data_to_csv();
>     if (last_sequence > -1 ) {
>
>         update_last_seq(last_sequence);
>     }
>     task->runtime = switch_epoch_time_now(NULL) + globals.cycle_time;
>
> }
>
> Does anyone know why?
>
> Thanks,
> Mark
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mrene_lists at avgs.ca  Fri Aug 21 09:06:23 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 21 Aug 2009 12:06:23 -0400
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4A8EBF77.1080204@gmx.net>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>	<4A8DB808.1030903@gmx.net>
	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
	<4A8EBF77.1080204@gmx.net>
Message-ID: <50DA88E6-832E-4596-851D-CB8FFFC54EAA@avgs.ca>

Try setting that in your sip profile:

<param name="rtp-autofix-timing" value="false" />

Thats a feature to work around with devices lying about their ptime in  
their sdp payload.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 21-Aug-09, at 11:38 AM, Peter P GMX wrote:

> Hello Michael,
>
> I made some tests with Freeswitch and Fritzbox and found by  
> Wireshark that:
> within one call
>
>    * Freeswitch starts sending 20msec packets, then after ~0,2 second
>      sends 30msec packets
>    * FritzBox always sends 30msec packets.
>
> The average jitter is below 2 msec in both directions.
>
> The below logs shows that Freeswitch considers the FritzBox to be  
> broken
> and starts using 30msec packets. But there is no SIP message from FS  
> to
> Fritzbox telling him that FB will use 30msec packets. SDP from FS to
> Fritzbox always shows ptime:20
>
> BTW: We can ship you a FritzBox if you need one for testing.
>
> Best regards
> Peter
>
>
> Log:
>
> 2009-08-21 17:15:50.271555 [DEBUG] switch_rtp.c:1138 Starting timer
> [soft] 160 bytes per 20ms
> 2009-08-21 17:15:50.281563 [INFO] mod_sofia.c:1499 Ring SDP:
> v=0
> o=FreeSWITCH 1250837460 1250837461 IN IP4 182.xxx.xx.xxx
> s=FreeSWITCH
> c=IN IP4 182.xxx.xx.xxx
> t=0 0
> m=audio 30290 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 2009-08-21 17:15:50.281563 [NOTICE] mod_sofia.c:1502 Pre-Answer
> sofia/internal/02xxxxxxxxx at fs1.my.domain!
> 2009-08-21 17:15:50.281563 [DEBUG] switch_core_session.c:630 Send  
> signal
> sofia/internal/02xxxxxxxxx at fs1.my.domain [BREAK]
> EXECUTE sofia/internal/02xxxxxxxxx at fs1.my.domain
> playback(voicemail/8000/vm-that_was_an_invalid_ext.wav)
> 2009-08-21 17:15:50.281563 [DEBUG] switch_ivr_play_say.c:1097 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2009-08-21 17:15:50.281563 [DEBUG] sofia.c:3302 Channel
> sofia/internal/02xxxxxxxxx at fs1.my.domain entering state [early][183]
> 2009-08-21 17:15:50.281563 [DEBUG] switch_core_io.c:649
> sofia/internal/02xxxxxxxxx at fs1.my.domain receive message
> [TRANSCODING_NECESSARY]
> 2009-08-21 17:15:50.531551 [WARNING] mod_sofia.c:799 We were told to  
> use
> ptime 20 but what they meant to say was 30
> This issue has so far been identified to happen on the following  
> broken
> platforms/devices:
> Linksys/Sipura aka Cisco
> ShoreTel
> Sonus/L3
> We will try to fix it but some of the devices on this list are so  
> broken
> who knows what will happen..
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From markmorreny at gmail.com  Fri Aug 21 09:24:08 2009
From: markmorreny at gmail.com (mark morreny)
Date: Sat, 22 Aug 2009 00:24:08 +0800
Subject: [Freeswitch-users] hanging problem with
	switch_scheduler_add_task
In-Reply-To: <B4BB7488-5C31-4447-BAD7-EAA062C467CC@avgs.ca>
References: <20ad6b920908210841l453b2833v9ac12612c174820b@mail.gmail.com>
	<B4BB7488-5C31-4447-BAD7-EAA062C467CC@avgs.ca>
Message-ID: <20ad6b920908210924y7e11c91di9613ce74fa18ef35@mail.gmail.com>

Hi Mathieu,

Thanks for your help.

How can I intentionally crash a hanging freeswitch and obtain the core file
to run gdb?

Thanks,
Mark

On Fri, Aug 21, 2009 at 11:59 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Means another task is hanging, do a "thread apply all bt" in gdb and
> look for scheduler tasks.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 21-Aug-09, at 11:41 AM, mark morreny wrote:
>
> > Hi,
> >
> > I am experiencing some hanging when fs is executing
> > switch_scheduler_add_task.
> >
> > switch_scheduler_add_task(switch_epoch_time_now(NULL) ,
> > data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|
> > SSHF_NO_DEL);
> >
> > I place switch_scheduler_add_task in my SWITCH_MODULE_LOAD_FUNCTION
> > and sometimes, hanging occurs on that particular line.  My
> > data_flush_callback is as follows.  I debug the module and it does
> > not even enter data_flush_callback.  It is hanging at
> > switch_scheduler_add_task().
> >
> > SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
> >
> >     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,
> > "starting to flush cdr record...\n");
> >
> >     int last_sequence = flush_data_to_csv();
> >     if (last_sequence > -1 ) {
> >
> >         update_last_seq(last_sequence);
> >     }
> >     task->runtime = switch_epoch_time_now(NULL) + globals.cycle_time;
> >
> > }
> >
> > Does anyone know why?
> >
> > Thanks,
> > Mark
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From woodydickson at gmail.com  Fri Aug 21 09:31:34 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 22 Aug 2009 00:31:34 +0800
Subject: [Freeswitch-users] zombie channels
In-Reply-To: <87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>
References: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
	<87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>
Message-ID: <a44688a0908210931ocafd373qc0465f8f24d35335@mail.gmail.com>

Hello,

Yes, I am using cdr, so I guess CS_REPORTING could be a problem.

I tried running the core, but I am getting some errors:
 ./freeswitch-gcore
/usr/local/freeswitch/log/freeswitch.gcore.fm5478:1: Error in sourced
command file:
ptrace: No such process.
gcore: failed to create /usr/local/freeswitch/log/freeswitch.gcore.16240

What is the proper way of using freeswitch-gcore?

Thanks,
Woody


On Fri, Aug 21, 2009 at 9:31 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> CS_REPORTING is the state in which cdrs are written, if the channel
> gets stuck in that state, the cdr module you are using is probably
> hanging somewhere.
>
> Use the "freeswitch-gcore" script in your source tree's scripts
> directory to generate a bug report for hanging channels.
>
> should be like..
>
> cd /usr/src/freeswitch # or whatever your source tree is
> bash ./scripts/freeswitch-gcore > bugreport.txt
>
> then submit it on http://jira.freeswitch.org/ so we can look at it.
>
> As you wish, you can also hop on #freeswitch / irc.freenode.net and
> have someone look into it.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 21-Aug-09, at 5:54 AM, Woody Dickson wrote:
>
> > Hi,
> >
> > I am running 1.0.4 right now using latest trunk.
> >
> > After a high traffic session, I do "show channels", I would find a
> > bunch of "CS_HIBERNATE" channels that don't get removed after all
> > the traffic is gone.
> >
> > Does anyone know what is the case of thoes CS_HIBERNATE'd channels?
> > How can I set  a timeout for those channels to be removed?
> >
> > Thanks,
> > Woody
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fraunhofer.lists.freeswitch-001 at traced.net  Fri Aug 21 10:02:31 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Fri, 21 Aug 2009 19:02:31 +0200
Subject: [Freeswitch-users] zombie channels
In-Reply-To: <a44688a0908210931ocafd373qc0465f8f24d35335@mail.gmail.com>
References: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
	<87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>
	<a44688a0908210931ocafd373qc0465f8f24d35335@mail.gmail.com>
Message-ID: <a37510170908211002h4ec4b4fek44518118d85e2594@mail.gmail.com>

Hello,

> I tried running the core, but I am getting some errors:
> ?./freeswitch-gcore
> /usr/local/freeswitch/log/freeswitch.gcore.fm5478:1: Error in sourced
> command file:
> ptrace: No such process.
> gcore: failed to create /usr/local/freeswitch/log/freeswitch.gcore.16240
>
> What is the proper way of using freeswitch-gcore?

I had the same problem... look at the source, it expects the pid file to be in
/usr/local/freeswitch/log/freeswitch.pid
and freeswitch to be installed at
/usr/local/freeswitch/bin/freeswitch

you can do that manualle, tough.

get the pid of freeswitch (something like "ps waux | grep freeswitch")
then run
gcore -o freeswitch_coredump1.core [pid found above]
gdb [location of freeswitch binary] -c freeswitch_coredump1.core \
        --eval-command='set pagination off' \
        --eval-command='bt' \
        --eval-command='bt full' \
        --eval-command='thread apply all bt' \
        --eval-command='thread apply all bt full' \
        --eval-command='quit' \
       >  freeswitch_report.txt

Beni.


From rupa at rupa.com  Fri Aug 21 10:15:26 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 21 Aug 2009 12:15:26 -0500
Subject: [Freeswitch-users] hanging problem with
	switch_scheduler_add_task
In-Reply-To: <20ad6b920908210924y7e11c91di9613ce74fa18ef35@mail.gmail.com>
References: <20ad6b920908210841l453b2833v9ac12612c174820b@mail.gmail.com>
	<B4BB7488-5C31-4447-BAD7-EAA062C467CC@avgs.ca>
	<20ad6b920908210924y7e11c91di9613ce74fa18ef35@mail.gmail.com>
Message-ID: <b39273530908211015u117dc01ep982f720334e3cdf9@mail.gmail.com>

Use gcore to get a core dump.  It will pause the process for the
duration of the dump, but will not kill the process.

On Fri, Aug 21, 2009 at 11:24 AM, mark morreny<markmorreny at gmail.com> wrote:
> Hi Mathieu,
>
> Thanks for your help.
>
> How can I intentionally crash a hanging freeswitch and obtain the core file
> to run gdb?
>
> Thanks,
> Mark
>
> On Fri, Aug 21, 2009 at 11:59 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>>
>> Means another task is hanging, do a "thread apply all bt" in gdb and
>> look for scheduler tasks.
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 21-Aug-09, at 11:41 AM, mark morreny wrote:
>>
>> > Hi,
>> >
>> > I am experiencing some hanging when fs is executing
>> > switch_scheduler_add_task.
>> >
>> > switch_scheduler_add_task(switch_epoch_time_now(NULL) ,
>> > data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|
>> > SSHF_NO_DEL);
>> >
>> > I place switch_scheduler_add_task in my SWITCH_MODULE_LOAD_FUNCTION
>> > and sometimes, hanging occurs on that particular line. ?My
>> > data_flush_callback is as follows. ?I debug the module and it does
>> > not even enter data_flush_callback. ?It is hanging at
>> > switch_scheduler_add_task().
>> >
>> > SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>> >
>> > ? ? switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,
>> > "starting to flush cdr record...\n");
>> >
>> > ? ? int last_sequence = flush_data_to_csv();
>> > ? ? if (last_sequence > -1 ) {
>> >
>> > ? ? ? ? update_last_seq(last_sequence);
>> > ? ? }
>> > ? ? task->runtime = switch_epoch_time_now(NULL) + globals.cycle_time;
>> >
>> > }
>> >
>> > Does anyone know why?
>> >
>> > Thanks,
>> > Mark
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From Prometheus001 at gmx.net  Fri Aug 21 11:28:12 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 21 Aug 2009 20:28:12 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <50DA88E6-832E-4596-851D-CB8FFFC54EAA@avgs.ca>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>	<4A8DB808.1030903@gmx.net>	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>	<4A8EBF77.1080204@gmx.net>
	<50DA88E6-832E-4596-851D-CB8FFFC54EAA@avgs.ca>
Message-ID: <4A8EE73C.2070006@gmx.net>

Hello Mathieu,

thank for your help. But this however didn't change the behaviour.

I've read of a patch in mod_sofia.c which partly corrects the problem
temporarily:

When I change Line 784 to

if (switch_rtp_ready(tech_pvt->rtp_session) && codec_ms != tech_pvt->codec_ms) {

to

if (switch_rtp_ready(tech_pvt->rtp_session) && codec_ms != tech_pvt->codec_ms && 0) {

(add a "&& 0") to deactivate this expression)

the announcements are played correctly to the Fritzbox. Connections to
other SIP phones (Snom) are also fine.

However the person at the Fritzbox still sounds very choppy in a
conference, but this is another module where I do not have a patch
available.

Best regards
Peter

Mathieu Rene schrieb:
> Try setting that in your sip profile:
>
> <param name="rtp-autofix-timing" value="false" />
>
> Thats a feature to work around with devices lying about their ptime in  
> their sdp payload.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 21-Aug-09, at 11:38 AM, Peter P GMX wrote:
>
>   
>> Hello Michael,
>>
>> I made some tests with Freeswitch and Fritzbox and found by  
>> Wireshark that:
>> within one call
>>
>>    * Freeswitch starts sending 20msec packets, then after ~0,2 second
>>      sends 30msec packets
>>    * FritzBox always sends 30msec packets.
>>
>> The average jitter is below 2 msec in both directions.
>>
>> The below logs shows that Freeswitch considers the FritzBox to be  
>> broken
>> and starts using 30msec packets. But there is no SIP message from FS  
>> to
>> Fritzbox telling him that FB will use 30msec packets. SDP from FS to
>> Fritzbox always shows ptime:20
>>
>> BTW: We can ship you a FritzBox if you need one for testing.
>>
>> Best regards
>> Peter
>>
>>
>> Log:
>>
>> 2009-08-21 17:15:50.271555 [DEBUG] switch_rtp.c:1138 Starting timer
>> [soft] 160 bytes per 20ms
>> 2009-08-21 17:15:50.281563 [INFO] mod_sofia.c:1499 Ring SDP:
>> v=0
>> o=FreeSWITCH 1250837460 1250837461 IN IP4 182.xxx.xx.xxx
>> s=FreeSWITCH
>> c=IN IP4 182.xxx.xx.xxx
>> t=0 0
>> m=audio 30290 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> 2009-08-21 17:15:50.281563 [NOTICE] mod_sofia.c:1502 Pre-Answer
>> sofia/internal/02xxxxxxxxx at fs1.my.domain!
>> 2009-08-21 17:15:50.281563 [DEBUG] switch_core_session.c:630 Send  
>> signal
>> sofia/internal/02xxxxxxxxx at fs1.my.domain [BREAK]
>> EXECUTE sofia/internal/02xxxxxxxxx at fs1.my.domain
>> playback(voicemail/8000/vm-that_was_an_invalid_ext.wav)
>> 2009-08-21 17:15:50.281563 [DEBUG] switch_ivr_play_say.c:1097 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>> 2009-08-21 17:15:50.281563 [DEBUG] sofia.c:3302 Channel
>> sofia/internal/02xxxxxxxxx at fs1.my.domain entering state [early][183]
>> 2009-08-21 17:15:50.281563 [DEBUG] switch_core_io.c:649
>> sofia/internal/02xxxxxxxxx at fs1.my.domain receive message
>> [TRANSCODING_NECESSARY]
>> 2009-08-21 17:15:50.531551 [WARNING] mod_sofia.c:799 We were told to  
>> use
>> ptime 20 but what they meant to say was 30
>> This issue has so far been identified to happen on the following  
>> broken
>> platforms/devices:
>> Linksys/Sipura aka Cisco
>> ShoreTel
>> Sonus/L3
>> We will try to fix it but some of the devices on this list are so  
>> broken
>> who knows what will happen..
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From brian at freeswitch.org  Fri Aug 21 11:38:50 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 21 Aug 2009 13:38:50 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4A8EBF77.1080204@gmx.net>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>	<4A8DB808.1030903@gmx.net>
	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
	<4A8EBF77.1080204@gmx.net>
Message-ID: <BD3DE8E0-3363-465C-985D-EC593AE45694@freeswitch.org>

You can ship me one whois bkw.org, I can add it to my lab.

/b

On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:

>
> BTW: We can ship you a FritzBox if you need one for testing.



From pjintheusa at gmail.com  Fri Aug 21 12:53:30 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 21 Aug 2009 15:53:30 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908141217g2facd41av25f65d6321967d74@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
	<b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
	<367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>
	<FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>
	<b39273530908132333y52279d64qe926eca47e15b5eb@mail.gmail.com>
	<367751820908141217g2facd41av25f65d6321967d74@mail.gmail.com>
Message-ID: <367751820908211253j33776506y8264f6f2c6bad74a@mail.gmail.com>

Hi there,

I created a feature request to cover this issue:

http://jira.freeswitch.org/browse/FSCORE-422 - The ability to support
Call FollowMe (or Call Blast) and multiple termination carriers -
without Loopback

If anybody wants comment on its merits and/or make the request clearer
- that would be great.

Thanks for every bodies help on this.

Phillip Jones

On Fri, Aug 14, 2009 at 3:17 PM, Phillip Jones<pjintheusa at gmail.com> wrote:
> Hi Rupa,
>
> What about my suggestion above introduce a "api_after_bridge" event
> that fires when the switch_ivr_uuid_bridge() bridges to the two sofia
> channels that Mathieu mentioned?
>
> Is that suggestion just way off the mark? If possible that would allow
> me to move forward - although I agree that supporting groupings of
> carriers is would be the most elegant solution.
>
> Let me know if I am just talking rubbish re the api_after_bridge" event.
>
> Thanks!
>
>
> Phillip Jones
>
>
> On Fri, Aug 14, 2009 at 2:33 AM, Rupa Schomaker<rupa at rupa.com> wrote:
>> On Thu, Aug 13, 2009 at 6:54 PM, Mathieu Rene<mrene_lists at avgs.ca> wrote:
>>> Hi All,
>>>
>>> The reason it works when you wait 3 seconds is that mod_loopback bails
>>
>> [snip]
>>
>> Thanks for that explanation. ?It umm.. explains a lot. :)
>>
>>> On another note, mod_sofia will behave differently when it detects its
>>> being bridge with another sofia channel, providing optimizations when
>>> both call legs are SIP.
>>>
>>> My personal opinion is not to use mod_loopback unless absolutely
>>> necessary, FreeSWITCH's core is very flexible and there's often a
>>> (better) way than using mod_loopback.
>>
>> So, I think the temp solution is to use loopback+delayed no media.
>>
>> but the real "solution" is to either drive the forked dialing logic
>> externally (event socket) or consider supporting groupings in the
>> bridge which.. umm... ?is gonna be a pain and will need buy in from
>> from Tony and other core devs since that is a core (no pun intended)
>> piece of code that nearly everything uses.
>>
>> I'm not sure I want to take a wack at it.
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From anthony.minessale at gmail.com  Fri Aug 21 13:09:25 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Aug 2009 15:09:25 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <BD3DE8E0-3363-465C-985D-EC593AE45694@freeswitch.org>
References: <4A8D5132.7010807@gmx.net>
	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>
	<4A8D5D43.3030300@gmx.net>
	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>
	<4A8DB808.1030903@gmx.net>
	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
	<4A8EBF77.1080204@gmx.net>
	<BD3DE8E0-3363-465C-985D-EC593AE45694@freeswitch.org>
Message-ID: <191c3a030908211309j60c16b26h1e6445891136ae7f@mail.gmail.com>

try setting FS to 30ms too

edit vars.xml and add @30i to everywhere you see  PCMU or PCMA so it looks
like PCMU at 30i

from:

  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>

to:

  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
,G7221 at 16000h,G722,PCMU at 30i,PCMA at 30i,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i
,GSM"/>


On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian at freeswitch.org> wrote:

> You can ship me one whois bkw.org, I can add it to my lab.
>
> /b
>
> On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
>
> >
> > BTW: We can ship you a FritzBox if you need one for testing.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From rogelio.perez at gmail.com  Fri Aug 21 12:15:13 2009
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Fri, 21 Aug 2009 16:15:13 -0300
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
Message-ID: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>

Hi Everyone,

I'm working on a PBX project for the Sheevaplug ARM based computer,  
with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
So far I've found a big difference between Freeswitch and Asterisk  
performance times.
This is a comparison of the time it takes them to perform different  
actions:

startup Freeswitch:			3 min.
startup Asterisk:			2 sec.

call extension Freeswitch:	6 sec.
call extension Asterisk:		0 sec.

shutdown Freeswitch:		6.5 sec
shutdown Asterisk:			0 sec.

reload config Freeswitch: 	1 sec.
reload config Asterisk:		1 sec.

Both were built from sources natively (no cross-compiling), and they  
use the default startup configurations.
I have managed to lower the Freeswitch times by disabling most of the  
modules and recompiling, but it is still far away from Asterisk (i.e.  
FS startup time 2.5 min).

1. Is there any way to further improve Freeswitch performance for the  
ARM architecture?
2. Can this be related to the lack of a FPU (the Sheevalug emulates  
the floating point operations).
3. On the startup I see this error repeated many times: [ERR]  
switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be  
related?

Thanks,
Rogelio Perez

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From andrew at hijacked.us  Fri Aug 21 13:20:06 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Fri, 21 Aug 2009 16:20:06 -0400
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
Message-ID: <20090821202005.GB28302@hijacked.us>

On Fri, Aug 21, 2009 at 04:15:13PM -0300, Rogelio Perez wrote:
> Hi Everyone,
> 
> I'm working on a PBX project for the Sheevaplug ARM based computer,  
> with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
> So far I've found a big difference between Freeswitch and Asterisk  
> performance times.
> This is a comparison of the time it takes them to perform different  
> actions:
> 
> startup Freeswitch:			3 min.
> startup Asterisk:			2 sec.
> 
> call extension Freeswitch:	6 sec.
> call extension Asterisk:		0 sec.
> 
> shutdown Freeswitch:		6.5 sec
> shutdown Asterisk:			0 sec.
> 
> reload config Freeswitch: 	1 sec.
> reload config Asterisk:		1 sec.
> 
> Both were built from sources natively (no cross-compiling), and they  
> use the default startup configurations.
> I have managed to lower the Freeswitch times by disabling most of the  
> modules and recompiling, but it is still far away from Asterisk (i.e.  
> FS startup time 2.5 min).
> 
> 1. Is there any way to further improve Freeswitch performance for the  
> ARM architecture?
> 2. Can this be related to the lack of a FPU (the Sheevalug emulates  
> the floating point operations).
> 3. On the startup I see this error repeated many times: [ERR]  
> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be  
> related?
>

Try making where freeswitch stores it's sqlite databases
/usr/local/freeswitch/db (by default) a ramdisk. I've had this vastly
improve FS performance on embedded devices.

Andrew


From anthony.minessale at gmail.com  Fri Aug 21 13:20:57 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Aug 2009 15:20:57 -0500
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
Message-ID: <191c3a030908211320s6feb5f10sf8670575069ec95a@mail.gmail.com>

probably disk i/o.

Is it some kind of flash drive?

make a ramdisk and simlink in /usr/local/freeswitch/db and
/usr/local/freeswitch/log to it
the default configuration uses a lot of high level features that use the
sqlite db on the disk.

We also offer commercial support where we could dig deeper into the problem
if you can't figure it out
consulting at freeswitch.org



On Fri, Aug 21, 2009 at 2:15 PM, Rogelio Perez <rogelio.perez at gmail.com>wrote:

> Hi Everyone,
>
> I'm working on a PBX project for the Sheevaplug<http://www.google.com/url?sa=t&source=web&ct=res&cd=1&url=http%253A%252F%252Fwww.marvell.com%252Fproducts%252Fembedded_processors%252Fdeveloper%252Fkirkwood%252Fsheevaplug.jsp&ei=EOOOSo6ILcyOtgeNruTOBA&usg=AFQjCNFREhfy_erj5irBWk8XFUjkQOP-aw&sig2=pAHYlI15IbZ5Kcw0n-nIvA> ARM
> based computer, with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
> So far I've found a big difference between Freeswitch and Asterisk
> performance times.
> This is a comparison of the time it takes them to perform different
> actions:
>
> startup Freeswitch: 3 min.
> startup Asterisk: 2 sec.
>
> call extension Freeswitch: 6 sec.
> call extension Asterisk: 0 sec.
>
> shutdown Freeswitch: 6.5 sec
>
> shutdown Asterisk: 0 sec.
>
>
> reload config Freeswitch: 1 sec.
> reload config Asterisk: 1 sec.
>
>
> Both were built from sources natively (no cross-compiling), and they use
> the default startup configurations.
> I have managed to lower the Freeswitch times by disabling most of the
> modules and recompiling, but it is still far away from Asterisk (i.e. FS
> startup time 2.5 min).
>
> 1. Is there any way to further improve Freeswitch performance for the ARM
> architecture?
> 2. Can this be related to the lack of a FPU (the Sheevalug emulates the
> floating point operations).
> 3. On the startup I see this error repeated many times: [ERR]
> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be related?
>
> Thanks,
> Rogelio Perez
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From moises.silva at gmail.com  Fri Aug 21 15:29:26 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 21 Aug 2009 18:29:26 -0400
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
Message-ID: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>

So, I finally took some days to put up OpenR2 working with OpenZAP, which
means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
support for. Including Mexico, Brazil, Argentina and others. The stack has
been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most
countries that users may be interested in, support for new variants will be
added on-demand only (in any case users can always tweak the advanced
configuration file to create their own variants as a last resort).
I created a wiki page to illustrate the basic setup:
http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2

<http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2>Now is time for testing. I
just did minimal testing on my development environment, no serious testing,
and I know that some stuff is not working at this point (I had some issues
with variable length DNIS and ANI) which should be fixed soon.

If anyone around happens to have an R2 link and wants to test R2 support in
OpenZAP, I can give them a hand with the configuration and any issues you
may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
#openzap as moy.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From msc at freeswitch.org  Fri Aug 21 15:59:03 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Aug 2009 15:59:03 -0700
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
In-Reply-To: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
References: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
Message-ID: <87f2f3b90908211559n163aa631j9b74a9e66cf7d95a@mail.gmail.com>

On Fri, Aug 21, 2009 at 3:29 PM, Moises Silva <moises.silva at gmail.com>wrote:

> So, I finally took some days to put up OpenR2 working with OpenZAP, which
> means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
> support for. Including Mexico, Brazil, Argentina and others. The stack has
> been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most
> countries that users may be interested in, support for new variants will be
> added on-demand only (in any case users can always tweak the advanced
> configuration file to create their own variants as a last resort).
> I created a wiki page to illustrate the basic setup:
> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
>
> <http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2>Now is time for testing. I
> just did minimal testing on my development environment, no serious testing,
> and I know that some stuff is not working at this point (I had some issues
> with variable length DNIS and ANI) which should be fixed soon.
>
> If anyone around happens to have an R2 link and wants to test R2 support in
> OpenZAP, I can give them a hand with the configuration and any issues you
> may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
> #openzap as moy.
>

You rock, dude!
-MC


>
> --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at gmail.com  Fri Aug 21 18:20:12 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Aug 2009 22:20:12 -0300
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
In-Reply-To: <87f2f3b90908211559n163aa631j9b74a9e66cf7d95a@mail.gmail.com>
References: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
	<87f2f3b90908211559n163aa631j9b74a9e66cf7d95a@mail.gmail.com>
Message-ID: <5a8712120908211820y4a674e81wba7af5cc532c3f69@mail.gmail.com>

Way to go moy!

On Fri, Aug 21, 2009 at 7:59 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Aug 21, 2009 at 3:29 PM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> So, I finally took some days to put up OpenR2 working with OpenZAP, which
>> means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
>> support for. Including Mexico, Brazil, Argentina and others. The stack has
>> been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most
>> countries that users may be interested in, support for new variants will be
>> added on-demand only (in any case users can always tweak the advanced
>> configuration file to create their own variants as a last resort).
>> I created a wiki page to illustrate the basic setup:
>> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
>>
>> <http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2>Now is time for testing.
>> I just did minimal testing on my development environment, no serious
>> testing, and I know that some stuff is not working at this point (I had some
>> issues with variable length DNIS and ANI) which should be fixed soon.
>>
>> If anyone around happens to have an R2 link and wants to test R2 support
>> in OpenZAP, I can give them a hand with the configuration and any issues you
>> may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
>> #openzap as moy.
>>
>
> You rock, dude!
> -MC
>
>
>>
>> --
>> Moises Silva
>> Software Developer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>> 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From b_ball_henry at hotmail.com  Fri Aug 21 23:40:40 2009
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sat, 22 Aug 2009 14:40:40 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
Message-ID: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>

Hi:

I try to dial sip url from my softphone but seems like the sip address is
being processed by sofia before it pass to the dialplan. The example here is
:

*X-lite(softphone) dials -> 1009 at 4.2.2.2 (it's fake sip address, the purpose
was just to test what's being passed to dialplan)
sofia receives the invite and return with trying
sofia pass the destination number to dailplan with "1009" (without the
"sip:" in front and without the "@4.2.2.2" after it)
*
Please see pastebin for full log. http://pastebin.freeswitch.org/10089
ignore anything after line 80, because it's not my point, and the
destination is a fake address.

I would like to know how do you actually pass a full sip url to the dialplan
to do the regex match. Because from the default.xml dialplan, it comes with
an example sip url dialing extension that match's *^sip:(.*)$ *. So I assume
there must be a way of passing full sip url to the dialplan. Here is the
example dialplan expecting sofia to pass it a full sip url:

 <!-- dial via SIP uri -->
    <extension name="sip_uri">
      <condition field="destination_number" expression="^sip:(.*)$">
        <action application="bridge" data="sofia/${use_profile}/$1"/>
      </condition>
    </extension>

Thanks

-- 
Henry Huang
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From mattdfong at gmail.com  Fri Aug 21 23:41:09 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 21 Aug 2009 23:41:09 -0700
Subject: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with
	same Gateway
In-Reply-To: <4256bf830908201021s7da0bff0wed8e060dbac0db2c@mail.gmail.com>
References: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>
	<6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>
	<4256bf830908201021s7da0bff0wed8e060dbac0db2c@mail.gmail.com>
Message-ID: <4256bf830908212341k1c0a9c9x7c8a9c70460965f7@mail.gmail.com>

So there seems to be some sort of error when bridging directly like
originate {ignore_early_media=true}sofia/gateway/XXXX.com/91415992<http://epik.com/914159927717>
XXXX &bridge(sofia/gateway/XXXX.com/91415465 <http://epik.com/914154650027>
XXXX)

BUT

if I get FS to send media to leg A, and then bridge to leg B by using a lua
script like

session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/hh/hh-welcome.wav");
session:execute("bridge", "sofia/gateway/epik.com/91415XXXXXXX");

then the legs bridge together OK. This happens when trying to bridge two
channels via the same Broadsoft SBC. Does this sound like a bug that should
be submitted to JIRA?

--matt
http://www.hellohunter.com


On Thu, Aug 20, 2009 at 10:21 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> originate {ignore_early_media=true}sofia/gateway/epik.com/914159927717
>  &bridge(sofia/gateway/epik.com/914154650027)
>
> is the string I was using from the console.
>
>
> On Thu, Aug 20, 2009 at 6:00 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
>> Hi
>>
>> How are you bridging the calls in FS (which api call or C function are you
>> using)?
>>  Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:
>>
>> I'm trying to get FreeSWITCH to bridge two channels together through the
>> same external gateway, but I'm having issues hearing audio. Both legs if
>> setup independently and forwarded to 5000 (test ivr) work fine for both
>> inbound and outbound media, but when I try to bridge them together,
>> everything seems fine in FreeSWITCH, but neither party can hear the other
>> speak. I'm thinking the external gateway might be having some issues because
>> I've been able to bridge 2 channels together through the same gateway on
>> different providers, but thought I'd also try to seek some help here.
>> FreeSWITCH should be handling the media for both channels, so I can't figure
>> out why if Leg A and Leg B work independently, but not if they are bridged
>> together. Is there a setting somewhere in FS that I'm missing?
>> Below is a ngrep of the SIP interactions if it's useful. Thanks for the
>> help.
>>
>> --matt
>>
>> interface: eth0 (172.24.200.0/255.255.255.0)
>> filter: (ip or ip6) and ( port 5060 )
>>
>> U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060
>> INVITE sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>SIP/2.0.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>> Max-Forwards: 70.
>>  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>>.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257811 INVITE.
>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>> Supported: timer, precondition, path, replaces.
>> Allow-Events: talk, refer.
>> Content-Type: application/sdp.
>> Content-Disposition: session.
>> Content-Length: 293.
>> Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;party=calling;screen=yes;privacy=off.
>> .
>> v=0.
>> o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
>> s=FreeSWITCH.
>> c=IN IP4 216.81.56.198.
>> t=0 0.
>> m=audio 24700 RTP/AVP 0 8 3 101 13.
>> a=rtpmap:0 PCMU/8000.
>> a=rtpmap:8 PCMA/8000.
>> a=rtpmap:3 GSM/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=rtpmap:13 CN/8000.
>> a=ptime:20.
>>
>>
>> U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 100 Trying.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>> >;tag=F725.2C49.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257811 INVITE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 183 Session Progress.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>> >;tag=F725.2C49.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257811 INVITE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>> Allow:
>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>> Content-Type: application/sdp.
>> Content-Length: 227.
>> .
>> v=0.
>> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
>> s=M6 Call.
>> c=IN IP4 38.98.58.148.
>> t=0 0.
>> m=audio 42554 RTP/AVP 0 101.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>> a=ptime:20.
>> a=sendrecv.
>> a=rtcp:6461 IN IP4 10.10.24.50.
>>
>>
>> U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 200 OK.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>> >;tag=F725.2C49.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257811 INVITE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>> Session-Expires: 1800;refresher=uas.
>> Allow:
>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>> Supported: timer.
>> Content-Type: application/sdp.
>> Content-Length: 227.
>> .
>> v=0.
>> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
>> s=M6 Call.
>> c=IN IP4 38.98.58.148.
>> t=0 0.
>> m=audio 42554 RTP/AVP 0 101.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>> a=ptime:20.
>> a=sendrecv.
>> a=rtcp:6461 IN IP4 10.10.24.50.
>>
>>
>> U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060
>> ACK sip:914159927717 at 38.98.58.148:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc.
>> Max-Forwards: 70.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>> >;tag=F725.2C49.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257811 ACK.
>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060
>> INVITE sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>SIP/2.0.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>> Max-Forwards: 70.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>>.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257815 INVITE.
>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>> Supported: timer, precondition, path, replaces.
>> Allow-Events: talk, refer.
>> Content-Type: application/sdp.
>> Content-Disposition: session.
>> Content-Length: 293.
>> Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;party=calling;screen=yes;privacy=off.
>> .
>> v=0.
>> o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198.
>> s=FreeSWITCH.
>> c=IN IP4 216.81.56.198.
>> t=0 0.
>> m=audio 24798 RTP/AVP 0 8 3 101 13.
>> a=rtpmap:0 PCMU/8000.
>> a=rtpmap:8 PCMA/8000.
>> a=rtpmap:3 GSM/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=rtpmap:13 CN/8000.
>> a=ptime:20.
>>
>>
>> U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 100 Trying.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>> >;tag=F72E.2D4E.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257815 INVITE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 183 Session Progress.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>> >;tag=F72E.2D4E.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257815 INVITE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>> Allow:
>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>> Content-Type: application/sdp.
>> Content-Length: 225.
>> .
>> v=0.
>> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
>> s=M6 Call.
>> c=IN IP4 38.98.58.148.
>> t=0 0.
>> m=audio 46520 RTP/AVP 0 101.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>> a=ptime:20.
>> a=sendrecv.
>> a=rtcp:6451 IN IP4 10.10.24.50.
>>
>>
>> U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 200 OK.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>> >;tag=F72E.2D4E.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257815 INVITE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>> Session-Expires: 1800;refresher=uas.
>> Allow:
>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>> Supported: timer.
>> Content-Type: application/sdp.
>> Content-Length: 225.
>> .
>> v=0.
>> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
>> s=M6 Call.
>> c=IN IP4 38.98.58.148.
>> t=0 0.
>> m=audio 46520 RTP/AVP 0 101.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>> a=ptime:20.
>> a=sendrecv.
>> a=rtcp:6451 IN IP4 10.10.24.50.
>>
>>
>> U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060
>> ACK sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK.
>> Max-Forwards: 70.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>> >;tag=F72E.2D4E.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257815 ACK.
>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080
>> BYE sip:gw+epik.com at 216.81.56.198:5080 SIP/2.0.
>> From: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>> >;tag=F725.2C49.
>> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 4817 BYE.
>> Max-Forwards: 70.
>> Via: SIP/2.0/UDP 38.98.58.148:5060
>> ;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
>> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>> Allow:
>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>  Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 38.98.58.148:5060
>> ;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
>> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
>> From: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>> >;tag=F725.2C49.
>> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 4817 BYE.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>> Supported: timer, precondition, path, replaces.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060
>> BYE sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
>> Max-Forwards: 70.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>> >;tag=F72E.2D4E.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257816 BYE.
>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>> Supported: timer, precondition, path, replaces.
>> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 200 OK.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>> >;tag=F72E.2D4E.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257816 BYE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>> Allow:
>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>> Content-Length: 0.
>> .
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From woodydickson at gmail.com  Sat Aug 22 01:08:12 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 22 Aug 2009 16:08:12 +0800
Subject: [Freeswitch-users] zombie channels
In-Reply-To: <87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>
References: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
	<87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>
Message-ID: <a44688a0908220108i4d9a63a1lbe4c36f4ed7c3f31@mail.gmail.com>

Hi

I checked and there is no looping in cdr.

Also, only a very small percentage of the channels become zombie.

What could cause fs to not releasing the channels?

Also, it seems to happen on under high traffic.  Could the fact that FS does
not receive BYE or BYE timing out on the uac side may cause this problem?

Thanks,
woody

On Fri, Aug 21, 2009 at 9:31 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> CS_REPORTING is the state in which cdrs are written, if the channel
> gets stuck in that state, the cdr module you are using is probably
> hanging somewhere.
>
> Use the "freeswitch-gcore" script in your source tree's scripts
> directory to generate a bug report for hanging channels.
>
> should be like..
>
> cd /usr/src/freeswitch # or whatever your source tree is
> bash ./scripts/freeswitch-gcore > bugreport.txt
>
> then submit it on http://jira.freeswitch.org/ so we can look at it.
>
> As you wish, you can also hop on #freeswitch / irc.freenode.net and
> have someone look into it.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 21-Aug-09, at 5:54 AM, Woody Dickson wrote:
>
> > Hi,
> >
> > I am running 1.0.4 right now using latest trunk.
> >
> > After a high traffic session, I do "show channels", I would find a
> > bunch of "CS_HIBERNATE" channels that don't get removed after all
> > the traffic is gone.
> >
> > Does anyone know what is the case of thoes CS_HIBERNATE'd channels?
> > How can I set  a timeout for those channels to be removed?
> >
> > Thanks,
> > Woody
>  > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Sat Aug 22 01:35:07 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 22 Aug 2009 04:35:07 -0400
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
Message-ID: <EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>

No, you don't get the full sip uri in the dialplan like that.   You do  
have a whole bunch of variables of the parsed sip header you can use.   
Use the "info" application to see all the vars so you can see what you  
have to route the call on.

Mike

On Aug 22, 2009, at 2:40 AM, Henry Huang wrote:

> Hi:
>
> I try to dial sip url from my softphone but seems like the sip  
> address is being processed by sofia before it pass to the dialplan.  
> The example here is :
>
> X-lite(softphone) dials -> 1009 at 4.2.2.2 (it's fake sip address, the  
> purpose was just to test what's being passed to dialplan)
> sofia receives the invite and return with trying
> sofia pass the destination number to dailplan with "1009" (without  
> the "sip:" in front and without the "@4.2.2.2" after it)
>
> Please see pastebin for full log. http://pastebin.freeswitch.org/10089
> ignore anything after line 80, because it's not my point, and the  
> destination is a fake address.
>
> I would like to know how do you actually pass a full sip url to the  
> dialplan to do the regex match. Because from the default.xml  
> dialplan, it comes with an example sip url dialing extension that  
> match's ^sip:(.*)$ . So I assume there must be a way of passing full  
> sip url to the dialplan. Here is the example dialplan expecting  
> sofia to pass it a full sip url:
>
>  <!-- dial via SIP uri -->
>     <extension name="sip_uri">
>       <condition field="destination_number" expression="^sip:(.*)$">
>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>       </condition>
>     </extension>
>

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From red.rain.seven at gmail.com  Sat Aug 22 01:41:15 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 22 Aug 2009 16:41:15 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
Message-ID: <59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>

It that case, the example of dialing sip_uri in the dialplan/default.xml
should be removed to prevent confusion. Because according to what you said,
one can never be able to hit this extension:

<!-- dial via SIP uri -->
    <extension name="sip_uri">
      <condition field="destination_number" expression="^sip:(.*)$">
        <action application="bridge" data="sofia/${use_profile}/$1"/>
      </condition>
    </extension>

And thanks for the tip, I will use variable instead.


On Sat, Aug 22, 2009 at 4:35 PM, Michael Jerris <mike at jerris.com> wrote:

> No, you don't get the full sip uri in the dialplan like that.   You do have
> a whole bunch of variables of the parsed sip header you can use.  Use the
> "info" application to see all the vars so you can see what you have to route
> the call on.
> Mike
>
> On Aug 22, 2009, at 2:40 AM, Henry Huang wrote:
>
> Hi:
>
> I try to dial sip url from my softphone but seems like the sip address is
> being processed by sofia before it pass to the dialplan. The example here is
> :
>
> *X-lite(softphone) dials -> 1009 at 4.2.2.2 (it's fake sip address, the
> purpose was just to test what's being passed to dialplan)
> sofia receives the invite and return with trying
> sofia pass the destination number to dailplan with "1009" (without the
> "sip:" in front and without the "@4.2.2.2" after it)
> *
> Please see pastebin for full log. http://pastebin.freeswitch.org/10089
> ignore anything after line 80, because it's not my point, and the
> destination is a fake address.
>
> I would like to know how do you actually pass a full sip url to the
> dialplan to do the regex match. Because from the default.xml dialplan, it
> comes with an example sip url dialing extension that match's *^sip:(.*)$ *.
> So I assume there must be a way of passing full sip url to the dialplan.
> Here is the example dialplan expecting sofia to pass it a full sip url:
>
>  <!-- dial via SIP uri -->
>     <extension name="sip_uri">
>       <condition field="destination_number" expression="^sip:(.*)$">
>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>       </condition>
>     </extension>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From jason at jasonjgw.net  Sat Aug 22 03:30:05 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 22 Aug 2009 20:30:05 +1000
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
Message-ID: <20090822103005.GA4299@jdc.jasonjgw.net>

Henry Huang <red.rain.seven at gmail.com> wrote:
> It that case, the example of dialing sip_uri in the dialplan/default.xml
> should be removed to prevent confusion. Because according to what you said,
> one can never be able to hit this extension:

It is entirely possible to reach this extension, but notice that the "sip:"
prefix is removed before the rest of the URI is used in calling the bridge
application.

If you don't understand why this dial-plan entry works, go back and read about
regular expressions and the format of destinations used with the bridge
application. There are examples and explanations on the wiki.



From red.rain.seven at gmail.com  Sat Aug 22 07:09:26 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 22 Aug 2009 22:09:26 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <20090822103005.GA4299@jdc.jasonjgw.net>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
Message-ID: <59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>

Jason:

I fully understand how the regex works in the dialplan. If you look closely
in my original email and check out the pastebin. You will see that sofia
does not pass the "sip:" to dialplan. I can do any combination of letters
that dials from my softphone, and it will pass them to the dialplan. but if
I put "sip:" in the front of my dial string. The "sip:" gets trunkated by
sofia module so does the "@xx.xx.xx.xx" gets trunkated before it reaches
dialplan to for regex matching. Therefore I say you can never reach the
example sip uri extension because sofia will trunkate "sip:" .

Here is the excert from pastebin:
3.    INVITE sip:1009 at 4.2.2.2 SIP/2.0
(line 3 , the freeswitch has successfuly received my dialying to sip:
1009 at 4.2.2.2)73.  2009-08-20 16:37:28.982772 [INFO]
mod_dialplan_xml.c:315Processing
1001->1009 in context Global
74.  Dialplan: sofia/trunkgroup_1/1001 at 192.168.1.67 parsing [
Global->number_1] continue=false
75.  Dialplan: sofia/trunkgroup_1/1001 at 192.168.1.67 Regex (FAIL)
[number_1]destination_number
(1009) =~ /^sip(.*)$/ break=on-false
(line 73~75, you can see that on line 73, sofia has trunkated the "sip: " &
"@4.2.2.2" and only leave "1009" as the destination to pass to dialplan for
regex match.)



On Sat, Aug 22, 2009 at 6:30 PM, Jason White <jason at jasonjgw.net> wrote:

> Henry Huang <red.rain.seven at gmail.com> wrote:
> > It that case, the example of dialing sip_uri in the dialplan/default.xml
> > should be removed to prevent confusion. Because according to what you
> said,
> > one can never be able to hit this extension:
>
> It is entirely possible to reach this extension, but notice that the "sip:"
> prefix is removed before the rest of the URI is used in calling the bridge
> application.
>
> If you don't understand why this dial-plan entry works, go back and read
> about
> regular expressions and the format of destinations used with the bridge
> application. There are examples and explanations on the wiki.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From mike at jerris.com  Sat Aug 22 07:27:26 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 22 Aug 2009 10:27:26 -0400
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
Message-ID: <EBDCE6C6-CB7C-4652-8A4D-850B364D0BB8@jerris.com>

a call coming from sofia would never hit that in the dialplan.  That  
extension is useful for dialing a sip url from mod_portaudio.

Mike

On Aug 22, 2009, at 10:09 AM, Henry Huang wrote:

> Jason:
>
> I fully understand how the regex works in the dialplan. If you look  
> closely in my original email and check out the pastebin. You will  
> see that sofia does not pass the "sip:" to dialplan. I can do any  
> combination of letters that dials from my softphone, and it will  
> pass them to the dialplan. but if I put "sip:" in the front of my  
> dial string. The "sip:" gets trunkated by sofia module so does the  
> "@xx.xx.xx.xx" gets trunkated before it reaches dialplan to for  
> regex matching. Therefore I say you can never reach the example sip  
> uri extension because sofia will trunkate "sip:" .
>
> Here is the excert from pastebin:
> 3.    INVITE sip:1009 at 4.2.2.2 SIP/2.0
> (line 3 , the freeswitch has successfuly received my dialying to  
> sip: 1009 at 4.2.2.2)
> 73.  2009-08-20 16:37:28.982772 [INFO] mod_dialplan_xml.c:315  
> Processing 1001->1009 in context Global
> 74.  Dialplan: sofia/trunkgroup_1/1001 at 192.168.1.67 parsing [Global- 
> >number_1] continue=false
> 75.  Dialplan: sofia/trunkgroup_1/1001 at 192.168.1.67 Regex (FAIL)  
> [number_1] destination_number(1009) =~ /^sip(.*)$/ break=on-false
> (line 73~75, you can see that on line 73, sofia has trunkated the  
> "sip: " & "@4.2.2.2" and only leave "1009" as the destination to  
> pass to dialplan for regex match.)
>
>
>
> On Sat, Aug 22, 2009 at 6:30 PM, Jason White <jason at jasonjgw.net>  
> wrote:
> Henry Huang <red.rain.seven at gmail.com> wrote:
> > It that case, the example of dialing sip_uri in the dialplan/ 
> default.xml
> > should be removed to prevent confusion. Because according to what  
> you said,
> > one can never be able to hit this extension:
>
> It is entirely possible to reach this extension, but notice that the  
> "sip:"
> prefix is removed before the rest of the URI is used in calling the  
> bridge
> application.
>
> If you don't understand why this dial-plan entry works, go back and  
> read about
> regular expressions and the format of destinations used with the  
> bridge
> application. There are examples and explanations on the wiki.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -- 
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sat Aug 22 07:30:14 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 22 Aug 2009 09:30:14 -0500
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
Message-ID: <51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>

Remember the dialplan is agnostic... it has no clue about SIP, IAX,  
Jingle, H323... it routes... you have various other variables you can  
condition on also... route on destination_number and you'll be fine.

/b

On Aug 22, 2009, at 9:09 AM, Henry Huang wrote:

> I fully understand how the regex works in the dialplan. If you look  
> closely in my original email and check out the pastebin. You will  
> see that sofia does not pass the "sip:" to dialplan. I can do any  
> combination of letters that dials from my softphone, and it will  
> pass them to the dialplan. but if I put "sip:" in the front of my  
> dial string. The "sip:" gets trunkated by sofia module so does the  
> "@xx.xx.xx.xx" gets trunkated before it reaches dialplan to for  
> regex matching. Therefore I say you can never reach the example sip  
> uri extension because sofia will trunkate "sip:" .



From red.rain.seven at gmail.com  Sat Aug 22 07:46:14 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 22 Aug 2009 22:46:14 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
Message-ID: <59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>

Brian:

but why can't I pass "sip:" to dialplan? seems like it's being truncated by
sofia..
Can you confirm that?

On Sat, Aug 22, 2009 at 10:30 PM, Brian West <brian at freeswitch.org> wrote:

> Remember the dialplan is agnostic... it has no clue about SIP, IAX,
> Jingle, H323... it routes... you have various other variables you can
> condition on also... route on destination_number and you'll be fine.
>
> /b
>
> On Aug 22, 2009, at 9:09 AM, Henry Huang wrote:
>
> > I fully understand how the regex works in the dialplan. If you look
> > closely in my original email and check out the pastebin. You will
> > see that sofia does not pass the "sip:" to dialplan. I can do any
> > combination of letters that dials from my softphone, and it will
> > pass them to the dialplan. but if I put "sip:" in the front of my
> > dial string. The "sip:" gets trunkated by sofia module so does the
> > "@xx.xx.xx.xx" gets trunkated before it reaches dialplan to for
> > regex matching. Therefore I say you can never reach the example sip
> > uri extension because sofia will trunkate "sip:" .
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From brian at freeswitch.org  Sat Aug 22 07:59:00 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 22 Aug 2009 09:59:00 -0500
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
Message-ID: <66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>

Because the dial plan is technology agnostic... you have been told  
more than once it won't pass it to the dialplan from mod_sofia...

/b

On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:

> Brian:
>
> but why can't I pass "sip:" to dialplan? seems like it's being  
> truncated by sofia..
> Can you confirm that?



From red.rain.seven at gmail.com  Sat Aug 22 08:02:22 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 22 Aug 2009 23:02:22 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
	<66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
Message-ID: <59ad9ca10908220802n5285e36qbbdca2c3d50b5311@mail.gmail.com>

Brian:

Sorry, it's my English. I didn't understand what you meant by "agnostic"
back there. Now I know.

Thank you.

On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian at freeswitch.org> wrote:

> Because the dial plan is technology agnostic... you have been told
> more than once it won't pass it to the dialplan from mod_sofia...
>
> /b
>
> On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
>
> > Brian:
> >
> > but why can't I pass "sip:" to dialplan? seems like it's being
> > truncated by sofia..
> > Can you confirm that?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From red.rain.seven at gmail.com  Sat Aug 22 08:07:09 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 22 Aug 2009 23:07:09 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
	<66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
Message-ID: <59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com>

Brian:

Oh, and again, if it's not passing it to the dialplan. I had suggested to
remove the sample "sip uri" extension in the default.xml dialplan. because
no one can reach the dialplan with prefix "sip:" because sofia is going to
remove that prefix.

<!-- dial via SIP uri -->
    <extension name="sip_uri">
      <condition field="destination_number" expression="^sip:(.*)$">
        <action application="bridge" data="sofia/${use_profile}/$1"/>
      </condition>
    </extension>

On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian at freeswitch.org> wrote:

> Because the dial plan is technology agnostic... you have been told
> more than once it won't pass it to the dialplan from mod_sofia...
>
> /b
>
> On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
>
> > Brian:
> >
> > but why can't I pass "sip:" to dialplan? seems like it's being
> > truncated by sofia..
> > Can you confirm that?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From brian at freeswitch.org  Sat Aug 22 08:15:21 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 22 Aug 2009 10:15:21 -0500
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
	<66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
	<59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com>
Message-ID: <366174E9-8213-4F20-81FC-FA26D0E4ED3F@freeswitch.org>

You were told already this was used by mod_portaudio.  So that you can  
pa call sip:blah at domain.com which portaudio passes the exact string  
you dial with pa call to the dialplan.

/b

On Aug 22, 2009, at 10:07 AM, Henry Huang wrote:

> Brian:
>
> Oh, and again, if it's not passing it to the dialplan. I had  
> suggested to remove the sample "sip uri" extension in the  
> default.xml dialplan. because no one can reach the dialplan with  
> prefix "sip:" because sofia is going to remove that prefix.
>
> <!-- dial via SIP uri -->
>     <extension name="sip_uri">
>       <condition field="destination_number" expression="^sip:(.*)$">
>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>       </condition>
>     </extension>

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From msc at freeswitch.org  Sat Aug 22 08:35:29 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 22 Aug 2009 08:35:29 -0700
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
	<66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
	<59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com>
Message-ID: <87f2f3b90908220835w6e5c98ecpeecdf81e6ac46f4c@mail.gmail.com>

On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> Brian:
>
> Oh, and again, if it's not passing it to the dialplan. I had suggested to
> remove the sample "sip uri" extension in the default.xml dialplan. because
> no one can reach the dialplan with prefix "sip:" because sofia is going to
> remove that prefix.


Well, this isn't entirely accurate. Like Mike J said, if you dialed
something like this at the CLI:

pa call sip:user at domain.com <sip%3Auser at domain.com>

Then you'd need the dialplan entry that handles the SIP URI.

Going back to the original question...
X-Lite dials 1009 at 4.2.2.2 correct?
But you're saying that the dialplan simply sees "1009" as the destination
number? I'm looking at the pastebin (10089) and trying to figure out exactly
what is happening. All I can see is that you have a context named "Global"
so I'm assuming you've made at least some modifications to the default
dialplan. Can you pastebin that whole context?

The other thing that you should probably do is create an extension in this
global context that routes a call to the info application. You could do
something like this so that "9992" would do an info dump:
<extension name="info">
  <condition field="destination_number" expression="^(9992)$">
    <action application="info"/>
  </condition>
</extension>

Then reloadxml and make a call to 9992 from your X-Lite client. The CLI will
have a dump and you'll see all sorts of variables listed. Many of those are
available for you to use for condition matches and routing in the dialplan.

Let us know how the info application does in giving you information about
the A leg of the call.
-MC

>
>
> <!-- dial via SIP uri -->
>     <extension name="sip_uri">
>       <condition field="destination_number" expression="^sip:(.*)$">
>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>       </condition>
>     </extension>
>
> On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Because the dial plan is technology agnostic... you have been told
>> more than once it won't pass it to the dialplan from mod_sofia...
>>
>> /b
>>
>> On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
>>
>> > Brian:
>> >
>> > but why can't I pass "sip:" to dialplan? seems like it's being
>> > truncated by sofia..
>> > Can you confirm that?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From larclap at yahoo.com  Sat Aug 22 09:42:47 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Sat, 22 Aug 2009 09:42:47 -0700
Subject: [Freeswitch-users] Problem with cnam.js?
Message-ID: <00b901ca2347$950feee0$bf2fcca0$@com>

I think there's something wrong with the script at
http://wiki.freeswitch.org/wiki/Examples_cnam.js.

 

If you use it as is, it displays "Content-type: text/html" for the
effective_caller_id_name. In cnam.pl, the first two output lines are
generated by:

 

if (!$debug) {print "Content-type: text/html\n\n";}

 

with the actual name in the third line.

 

So I changed:

 

fd.open("read");

buff = fd.readln();

 

if(buff) {

   logger(buff, "info");

   session.setVariable("effective_caller_id_name", buff);

}

 

To:

 

fd.open("read");

buff = fd.readAll();

 

if(buff[2]) {

   logger(buff, "info");

   session.setVariable("effective_caller_id_name", buff[2]);

}

 

Or remove the print statement from cnam.pl.

 

Sorry for the code, but the page was not editable.

 

Lars

 

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From red.rain.seven at gmail.com  Sat Aug 22 10:35:01 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sun, 23 Aug 2009 01:35:01 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <87f2f3b90908220835w6e5c98ecpeecdf81e6ac46f4c@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
	<66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
	<59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com>
	<87f2f3b90908220835w6e5c98ecpeecdf81e6ac46f4c@mail.gmail.com>
Message-ID: <59ad9ca10908221035l448a9f91ncf26f978506487ad@mail.gmail.com>

Michael:

Thank you for making it in "for dummies" format. :P
These are really nice tips I can use. thanks.

On Sat, Aug 22, 2009 at 11:35 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>
>> Brian:
>>
>> Oh, and again, if it's not passing it to the dialplan. I had suggested to
>> remove the sample "sip uri" extension in the default.xml dialplan. because
>> no one can reach the dialplan with prefix "sip:" because sofia is going to
>> remove that prefix.
>
>
> Well, this isn't entirely accurate. Like Mike J said, if you dialed
> something like this at the CLI:
>
> pa call sip:user at domain.com <sip%3Auser at domain.com>
>
> Then you'd need the dialplan entry that handles the SIP URI.
>
> Going back to the original question...
> X-Lite dials 1009 at 4.2.2.2 correct?
> But you're saying that the dialplan simply sees "1009" as the destination
> number? I'm looking at the pastebin (10089) and trying to figure out exactly
> what is happening. All I can see is that you have a context named "Global"
> so I'm assuming you've made at least some modifications to the default
> dialplan. Can you pastebin that whole context?
>
> The other thing that you should probably do is create an extension in this
> global context that routes a call to the info application. You could do
> something like this so that "9992" would do an info dump:
> <extension name="info">
>   <condition field="destination_number" expression="^(9992)$">
>     <action application="info"/>
>   </condition>
> </extension>
>
> Then reloadxml and make a call to 9992 from your X-Lite client. The CLI
> will have a dump and you'll see all sorts of variables listed. Many of those
> are available for you to use for condition matches and routing in the
> dialplan.
>
> Let us know how the info application does in giving you information about
> the A leg of the call.
> -MC
>
>>
>>
>> <!-- dial via SIP uri -->
>>     <extension name="sip_uri">
>>       <condition field="destination_number" expression="^sip:(.*)$">
>>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>>       </condition>
>>     </extension>
>>
>> On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian at freeswitch.org>wrote:
>>
>>> Because the dial plan is technology agnostic... you have been told
>>> more than once it won't pass it to the dialplan from mod_sofia...
>>>
>>> /b
>>>
>>> On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
>>>
>>> > Brian:
>>> >
>>> > but why can't I pass "sip:" to dialplan? seems like it's being
>>> > truncated by sofia..
>>> > Can you confirm that?
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Henry Huang
>> UniC Solution - Communication Unified
>> VoIP & Open Source software Consultant
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From anthony.minessale at gmail.com  Sat Aug 22 11:11:55 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 22 Aug 2009 13:11:55 -0500
Subject: [Freeswitch-users] zombie channels
In-Reply-To: <a44688a0908220108i4d9a63a1lbe4c36f4ed7c3f31@mail.gmail.com>
References: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
	<87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>
	<a44688a0908220108i4d9a63a1lbe4c36f4ed7c3f31@mail.gmail.com>
Message-ID: <191c3a030908221111m2181977dn782da42afee84726@mail.gmail.com>

Yes try enabling session timers

On Aug 22, 2009 3:10 AM, "Woody Dickson" <woodydickson at gmail.com> wrote:

Hi

I checked and there is no looping in cdr.

Also, only a very small percentage of the channels become zombie.

What could cause fs to not releasing the channels?

Also, it seems to happen on under high traffic.  Could the fact that FS does
not receive BYE or BYE timing out on the uac side may cause this problem?

Thanks,
woody

On Fri, Aug 21, 2009 at 9:31 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> > Hi, > > CS_REPORTING is the state in which cdrs are written, if the
channel > gets stuck in that...


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From shaheryarkh at googlemail.com  Sat Aug 22 13:53:20 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Sun, 23 Aug 2009 01:53:20 +0500
Subject: [Freeswitch-users] SIP codec preference order
Message-ID: <b16156850908221353m3e219f7eyefd29870584e121c@mail.gmail.com>

Hi,

I have a FS gateway (SVN revision 14537) that is is receiving SIP calls from
different source gateways and sending it to one single destination gateway.
Now each source gateway can talk in one specific codec and FS itself is not
doing any transcoding. So i enabled all possible codecs that this FS may
receive from source gateways.

The problem is that the source gateways who are talking in codec that are in
first three preferred codecs list in sip profile are working fine, while the
codecs that are at 4th or greater preference order number do not work. The
call is received and accepted by destination gateway then it gets terminated
almost immediately.

Do note that destination gateway is not FS, its some CISCO device that is
also accepting all possible codecs.

Can you guys suggest why it is happening and what are the possible
solutions, other then transcode of course.

Thank you.


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From brian at freeswitch.org  Sat Aug 22 14:07:09 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 22 Aug 2009 16:07:09 -0500
Subject: [Freeswitch-users] SIP codec preference order
In-Reply-To: <b16156850908221353m3e219f7eyefd29870584e121c@mail.gmail.com>
References: <b16156850908221353m3e219f7eyefd29870584e121c@mail.gmail.com>
Message-ID: <50B93B44-B096-44CD-A44F-7A229D0A7B6A@freeswitch.org>

Can you provide a little bit of log detail?

/b

On Aug 22, 2009, at 3:53 PM, Muhammad Shahzad wrote:

> Can you guys suggest why it is happening and what are the possible  
> solutions, other then transcode of course.



From shaheryarkh at googlemail.com  Sat Aug 22 15:16:01 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Sun, 23 Aug 2009 03:16:01 +0500
Subject: [Freeswitch-users] SIP codec preference order
In-Reply-To: <50B93B44-B096-44CD-A44F-7A229D0A7B6A@freeswitch.org>
References: <b16156850908221353m3e219f7eyefd29870584e121c@mail.gmail.com>
	<50B93B44-B096-44CD-A44F-7A229D0A7B6A@freeswitch.org>
Message-ID: <b16156850908221516h535ea11fh354c7d31ce316e87@mail.gmail.com>

i just upgraded it to 14599 and its working fine now.

Thank you.


On Sun, Aug 23, 2009 at 2:07 AM, Brian West <brian at freeswitch.org> wrote:

> Can you provide a little bit of log detail?
>
> /b
>
> On Aug 22, 2009, at 3:53 PM, Muhammad Shahzad wrote:
>
> > Can you guys suggest why it is happening and what are the possible
> > solutions, other then transcode of course.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From kevin at johnnyvoip.com  Sat Aug 22 16:24:26 2009
From: kevin at johnnyvoip.com (Kevin Green)
Date: Sat, 22 Aug 2009 19:24:26 -0400
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908221035l448a9f91ncf26f978506487ad@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com> 
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com> 
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com> 
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com> 
	<66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
	<59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com> 
	<87f2f3b90908220835w6e5c98ecpeecdf81e6ac46f4c@mail.gmail.com> 
	<59ad9ca10908221035l448a9f91ncf26f978506487ad@mail.gmail.com>
Message-ID: <ab57a3e80908221624j7e32d5b3me9a8571f6e66dd59@mail.gmail.com>

X-lite I believe handles the sip: by itself sometime and therefore will try
and place a call to the sip address directly from x-lite without touching
FreeSWITCH. Be aware of this while testing and watch for this behavior
because it might throw off your expectations.

Regards,
   Kevin Green


On Sat, Aug 22, 2009 at 1:35 PM, Henry Huang <red.rain.seven at gmail.com>wrote:

> Michael:
>
> Thank you for making it in "for dummies" format. :P
> These are really nice tips I can use. thanks.
>
>
> On Sat, Aug 22, 2009 at 11:35 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>>
>>> Brian:
>>>
>>> Oh, and again, if it's not passing it to the dialplan. I had suggested to
>>> remove the sample "sip uri" extension in the default.xml dialplan. because
>>> no one can reach the dialplan with prefix "sip:" because sofia is going to
>>> remove that prefix.
>>
>>
>> Well, this isn't entirely accurate. Like Mike J said, if you dialed
>> something like this at the CLI:
>>
>> pa call sip:user at domain.com <sip%3Auser at domain.com>
>>
>> Then you'd need the dialplan entry that handles the SIP URI.
>>
>> Going back to the original question...
>> X-Lite dials 1009 at 4.2.2.2 correct?
>> But you're saying that the dialplan simply sees "1009" as the destination
>> number? I'm looking at the pastebin (10089) and trying to figure out exactly
>> what is happening. All I can see is that you have a context named "Global"
>> so I'm assuming you've made at least some modifications to the default
>> dialplan. Can you pastebin that whole context?
>>
>> The other thing that you should probably do is create an extension in this
>> global context that routes a call to the info application. You could do
>> something like this so that "9992" would do an info dump:
>> <extension name="info">
>>   <condition field="destination_number" expression="^(9992)$">
>>     <action application="info"/>
>>   </condition>
>> </extension>
>>
>> Then reloadxml and make a call to 9992 from your X-Lite client. The CLI
>> will have a dump and you'll see all sorts of variables listed. Many of those
>> are available for you to use for condition matches and routing in the
>> dialplan.
>>
>> Let us know how the info application does in giving you information about
>> the A leg of the call.
>> -MC
>>
>>>
>>>
>>> <!-- dial via SIP uri -->
>>>     <extension name="sip_uri">
>>>       <condition field="destination_number" expression="^sip:(.*)$">
>>>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>>>       </condition>
>>>     </extension>
>>>
>>> On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian at freeswitch.org>wrote:
>>>
>>>> Because the dial plan is technology agnostic... you have been told
>>>> more than once it won't pass it to the dialplan from mod_sofia...
>>>>
>>>> /b
>>>>
>>>> On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
>>>>
>>>> > Brian:
>>>> >
>>>> > but why can't I pass "sip:" to dialplan? seems like it's being
>>>> > truncated by sofia..
>>>> > Can you confirm that?
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Henry Huang
>>> UniC Solution - Communication Unified
>>> VoIP & Open Source software Consultant
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Sat Aug 22 22:40:25 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 23 Aug 2009 01:40:25 -0400
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
In-Reply-To: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
References: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
Message-ID: <86a32abc0908222240r2543ebbbhea758cbae17e49cc@mail.gmail.com>

Nice work, keep up the great work :).

On Fri, Aug 21, 2009 at 6:29 PM, Moises Silva<moises.silva at gmail.com> wrote:
> So, I finally took some days to put up OpenR2 working with OpenZAP, which
> means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
> support for. Including Mexico, Brazil, Argentina and others. The stack has
> been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most
> countries that users may be interested in, support for new variants will be
> added on-demand only (in any case users can always tweak the advanced
> configuration file to create their own variants as a last resort).
> I created a wiki page to illustrate the basic
> setup:?http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
> Now is time for testing. I just did minimal testing on my development
> environment, no serious testing, and I know that some stuff is not working
> at this point (I had some issues with variable length DNIS and ANI) which
> should be fixed soon.
> If anyone around happens to have an R2 link and wants to test R2 support in
> OpenZAP, I can give them a hand with the configuration and any issues you
> may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
> #openzap as moy.
>
> --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
> Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From panayotov.vd at gmail.com  Sat Aug 22 23:21:10 2009
From: panayotov.vd at gmail.com (Vassil Panayotov)
Date: Sun, 23 Aug 2009 09:21:10 +0300
Subject: [Freeswitch-users] Yet another question about A500 + FS
Message-ID: <8a9b664c0908222321r20c5c4d8wd08721dc8955f62c@mail.gmail.com>

Hi,

I managed to get our A500 running with FreeSWITCH 1.0.4 stable using wanpipe
3.4.4 drivers. But now I have another problem...
I want to originate calls through event socket, and I only want to receive
ANSWERED(+OK) reply when the user actually answers.

Now the situation is:

====================================
originate openzap/1/a/123456 023
2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT:
CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456]
Ci=[0000000000]
2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4
2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel
OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082]
2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer
OpenZAP/1:1/123456!
API CALL [originate(openzap/1/a/123456 023)] output:
+OK f8fca2be-8fa7-11de-9076-511e29dfc082

2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer
OpenZAP/1:1/123456 to XML[023 at default]
freeswitch at emo-voip> 2009-08-23 08:44:06.743475 [INFO]
mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default
2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
[OpenZAP/1:1/123456] has been answered
2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5
2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6
2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup
OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT (N):
CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3
2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086 Session 2
(OpenZAP/1:1/123456) Ended
2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close Channel
OpenZAP/1:1/123456 [CS_DESTROY]
====================================

Extension 023 is an IVR. As you can see FreeSWITCH answers the call
(2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
[OpenZAP/1:1/123456] has been answered) 20 seconds before user actually pick
up the phone (2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX
EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5).

So Sangoma drivers/daemons report the events correctly.
How can I set FreeSWITCH to answer after receiving RX EVENT (N):
CALL_ANSWERED from the driver?

Thank you,
V. Panayotov
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From diego.viola at gmail.com  Sat Aug 22 23:38:05 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 23 Aug 2009 02:38:05 -0400
Subject: [Freeswitch-users] Inboud Call Queue
In-Reply-To: <7F5652780AE84DFFB371EF62042540A8@saeedlaptop>
References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop>
	<87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com>
	<77308CE88F604444863741D590835B10@saeedlaptop>
	<63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com>
	<EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop>
	<18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com>
	<191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com>
	<62BE31C473E04988BB3AA9553A56C4F8@saeedlaptop>
	<2FED90CE-0CBB-43DF-80E7-7593B236281E@gmail.com>
	<7F5652780AE84DFFB371EF62042540A8@saeedlaptop>
Message-ID: <86a32abc0908222338hc0ee765y8802f406d74f6460@mail.gmail.com>

Hi guys,

I was wondering if some of you run FreeSWITCH on a call center
environment, I ask this because I plan to do that soon and I was
wondering how well mod_fifo works for queues, etc.

Thanks,

Diego

On Thu, May 7, 2009 at 6:08 AM, Saeed Ahmed<saeedahmad1981 at gmail.com> wrote:
> Thanks Seven I?ll try it very soon.
>
>
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of seven
> Sent: Thursday, May 07, 2009 5:42 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Inboud Call Queue
>
>
>
> See this:
>
>
>
> http://wiki.freeswitch.org/wiki/Simple_call_center_using_mod_fifo
>
>
>
>
>
> On May 6, 2009, at 10:15 PM, Saeed Ahmed wrote:
>
> Thanks Guys
>
>
>
> ________________________________
>
> From:?freeswitch-users-bounces at lists.freeswitch.org?[mailto:freeswitch-users-bounces at lists.freeswitch.org]?On
> Behalf Of?Anthony Minessale
> Sent:?Wednesday, May 06, 2009 3:57 PM
> To:?freeswitch-users at lists.freeswitch.org
> Subject:?Re: [Freeswitch-users] Inboud Call Queue
>
>
>
> I worked on the patch and added it to trunk rev 13240
>
>
> On Wed, May 6, 2009 at 7:53 AM, dujinfang <dujinfang at gmail.com> wrote:
>
> The patch haven't been merged into trunk. It should be as easy as execute
> the following command in the FS source code root dir:
>
>
>
> patch < /tmp/the_patch_file_name.diff
>
>
>
> I will post an example on the wiki when I finished, hope be soon.
>
>
>
> On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote:
>
>
> Hi Seven,
>
> I am exactly looking for this functionality.
>
> Please let me know when you are finished with new queue manager app. I?ll
> try it in my call center.
>
> Regarding Patch: is it already part of SVN trunk? If not then could you help
> me how to install it, I have no programming background.
>
> Many Thanks.
>
>
>
> ________________________________
>
> From:?freeswitch-users-bounces at lists.freeswitch.org?[mailto:freeswitch-users-bounces at lists.freeswitch.org]?On
> Behalf Of?seven
> Sent:?Wednesday, May 06, 2009 4:17 AM
> To:?freeswitch-users at lists.freeswitch.org
> Subject:?Re: [Freeswitch-users] Inboud Call Queue
>
>
>
>
>
> On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote:
>
>
>
> Hi Michael,
>
> Thanks for a quick reply.
>
> I would definitely create a test environment, but my question is that will
> it work in required way?
>
> I read that in Mod_fifo agent has to call in queue but I need that all
> incoming calls are automatically distributed between available agents or if
> all are busy then should go to voicemail.
>
> I'm working on a call center like queue scenario right now, I'm pretty sure
> it call automatically distributed to available agents, but the customer will
> stay in the queue if all agents are busy by default. You can bind a key to
> the channel and play a message repeatedly to guide the customer to voicemail
> by press a key.
>
>
>
> Also maybe you need this patch to make the fifo works as desired.
>
>
>
> http://jira.freeswitch.org/browse/MODAPP-272
>
>
>
> I would join IRC for further assistance.
>
>
>
> Thanks.
>
>
>
> ________________________________
>
> From:?freeswitch-users-bounces at lists.freeswitch.org?[mailto:freeswitch-users-bounces at lists.freeswitch.org]?On
> Behalf Of?Michael Collins
> Sent:?Tuesday, May 05, 2009 7:19 PM
> To:?freeswitch-users at lists.freeswitch.org
> Subject:?Re: [Freeswitch-users] Inboud Call Queue
>
>
>
>
>
> On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed <saeedahmad1981 at gmail.com>
> wrote:
>
> Hi All,
>
> In an inbound call center scenario is it possible that customers calls in
> and calls are distributed between online (who are registered on FS and in
> idle state) agents. I saw some on going discussion on list where it looks
> that currently it?s not possible but I am newbie so maybe I didn?t
> understand it well. If it?s possible then please give me a start point that
> how can I implement it.
>
> I would start here:
> http://wiki.freeswitch.org/wiki/Mod_fifo
>
> I strongly recommend that you set up a FreeSWITCH server and play around
> with it so that you can learn the pros and cons of using the FIFO queues. It
> would be best if you could set up a few phones and set them as FIFO agents
> and then have someone help you make test calls so that you can emulate your
> CC environment.
>
> Also, you might want to join us on IRC: #freeswitch on?irc.freenode.net?-
> there are several users who've had real world experience with mod_fifo and
> they might be in a good position to answer your questions real-time.
>
> -MC
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH?http://www.freeswitch.org/
> ClueCon?http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC:?irc.freenode.net?#freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jason at jasonjgw.net  Sun Aug 23 00:08:48 2009
From: jason at jasonjgw.net (Jason White)
Date: Sun, 23 Aug 2009 17:08:48 +1000
Subject: [Freeswitch-users] Inboud Call Queue
In-Reply-To: <86a32abc0908222338hc0ee765y8802f406d74f6460@mail.gmail.com>
References: <87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com>
	<77308CE88F604444863741D590835B10@saeedlaptop>
	<63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com>
	<EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop>
	<18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com>
	<191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com>
	<62BE31C473E04988BB3AA9553A56C4F8@saeedlaptop>
	<2FED90CE-0CBB-43DF-80E7-7593B236281E@gmail.com>
	<7F5652780AE84DFFB371EF62042540A8@saeedlaptop>
	<86a32abc0908222338hc0ee765y8802f406d74f6460@mail.gmail.com>
Message-ID: <20090823070848.GA16072@jdc.jasonjgw.net>

Diego Viola <diego.viola at gmail.com> wrote:
> I was wondering if some of you run FreeSWITCH on a call center
> environment, I ask this because I plan to do that soon and I was
> wondering how well mod_fifo works for queues, etc.

This was mentioned on the list once before, and it might be what you want:
http://wiki.opencsm.org/wiki/index.php/Spice_Telephony
(Spice Telephony).



From darklion11 at yahoo.com  Sun Aug 23 00:26:44 2009
From: darklion11 at yahoo.com (Edmar Cruz)
Date: Sun, 23 Aug 2009 00:26:44 -0700 (PDT)
Subject: [Freeswitch-users]  Starting Freeswitch using Intercom
Message-ID: <25100943.post@talk.nabble.com>


Hi,

  I dont know how can I start Freeswitch using Intercom device, can you help
me on this? Is there an alternative software like X-Lite but only when I
press call on the Intercom device? 

Thanks, 
Edmar
-- 
View this message in context: http://www.nabble.com/Starting-Freeswitch-using-Intercom-tp25100943p25100943.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From diego.viola at gmail.com  Sun Aug 23 01:02:59 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 23 Aug 2009 04:02:59 -0400
Subject: [Freeswitch-users] Inboud Call Queue
In-Reply-To: <20090823070848.GA16072@jdc.jasonjgw.net>
References: <87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com>
	<63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com>
	<EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop>
	<18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com>
	<191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com>
	<62BE31C473E04988BB3AA9553A56C4F8@saeedlaptop>
	<2FED90CE-0CBB-43DF-80E7-7593B236281E@gmail.com>
	<7F5652780AE84DFFB371EF62042540A8@saeedlaptop>
	<86a32abc0908222338hc0ee765y8802f406d74f6460@mail.gmail.com>
	<20090823070848.GA16072@jdc.jasonjgw.net>
Message-ID: <86a32abc0908230102i79bbbe7eoee7313f329f55607@mail.gmail.com>

Looks nice, is anyone running that in production?

On Sun, Aug 23, 2009 at 3:08 AM, Jason White <jason at jasonjgw.net> wrote:

> Diego Viola <diego.viola at gmail.com> wrote:
> > I was wondering if some of you run FreeSWITCH on a call center
> > environment, I ask this because I plan to do that soon and I was
> > wondering how well mod_fifo works for queues, etc.
>
> This was mentioned on the list once before, and it might be what you want:
> http://wiki.opencsm.org/wiki/index.php/Spice_Telephony
> (Spice Telephony).
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From egghunt at gmail.com  Sun Aug 23 04:25:20 2009
From: egghunt at gmail.com (Arnaldo de Moraes Pereira)
Date: Sun, 23 Aug 2009 08:25:20 -0300
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
In-Reply-To: <86a32abc0908222240r2543ebbbhea758cbae17e49cc@mail.gmail.com>
References: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com> 
	<86a32abc0908222240r2543ebbbhea758cbae17e49cc@mail.gmail.com>
Message-ID: <ff2539090908230425i345b9450gfdc0cb4d015fc93c@mail.gmail.com>

Thanks a lot, moy, this is great. I'll check to see if there's somewhere I
can test it.

On Sun, Aug 23, 2009 at 2:40 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Nice work, keep up the great work :).
>
> On Fri, Aug 21, 2009 at 6:29 PM, Moises Silva<moises.silva at gmail.com>
> wrote:
> > So, I finally took some days to put up OpenR2 working with OpenZAP, which
> > means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
> > support for. Including Mexico, Brazil, Argentina and others. The stack
> has
> > been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers
> most
> > countries that users may be interested in, support for new variants will
> be
> > added on-demand only (in any case users can always tweak the advanced
> > configuration file to create their own variants as a last resort).
> > I created a wiki page to illustrate the basic
> > setup: http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
> > Now is time for testing. I just did minimal testing on my development
> > environment, no serious testing, and I know that some stuff is not
> working
> > at this point (I had some issues with variable length DNIS and ANI) which
> > should be fixed soon.
> > If anyone around happens to have an R2 link and wants to test R2 support
> in
> > OpenZAP, I can give them a hand with the configuration and any issues you
> > may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
> > #openzap as moy.
> >
> > --
> > Moises Silva
> > Software Developer
> > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3
> > Canada
> > t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Arnaldo M Pereira
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From merul at mac.com  Sun Aug 23 10:27:45 2009
From: merul at mac.com (Merul Patel)
Date: Sun, 23 Aug 2009 18:27:45 +0100
Subject: [Freeswitch-users] FXO and analogue phones
Message-ID: <C281504B-3D79-465E-ADA9-25C140FCF50F@mac.com>

I have a Freeswitch setup working on an Alix embedded platform in  
conjunction with a USB FXO device from Sangoma. My goal is to be able  
to either answer incoming calls on a softphone or on a POTS handset  
elsewhere in the building, and to also be able to make outgoing calls  
from either. For clarity, the analogue line has two physical  
extensions, one connected to the POTS and the other to the FXO.

I can make and receive calls fine, but have problems when the call is  
answered on the POTS handset.

Here is the dialplan I initially used  in /opt/freeswitch/conf/ 
dialplans/public/01_incoming.xml:

<include>
   <extension name="public_did">
     <condition field="${strftime(%w)}" expression="^(\d)$">
       <!-- There seems to be a delay of 7 seconds from when FS starts  
dealing with the call and from when it starts ringing -->
       <action application="sleep" data="23000"/>
       <action application="set" data="domain_name=$${domain}"/>
       <action application="transfer" data="1001 XML default"/>
     </condition>
   </extension>
</include>

It's pretty basic, and if the softphone is not registered or does not  
answer then the call goes to voicemail. However the call will always  
go to voicemail, and the voicemail application will begin to execute  
after the call has been answered on the POTS handset.

I've been trying to make the dialplan more useful, by having it ring  
the softphone immediately, and only transfer the call to the voicemail  
application if the line is still ringing. I'm in the UK, hence my  
choice of frequencies in the tone_detect application:

<include>
   <extension name="public_did">
     <condition field="${strftime(%w)}" expression="^(\d)$">
       <!-- There seems to be a delay of 7 seconds from when FS starts  
dealing with the call and from when it starts ringing -->
       <action application="set" data="call_timeout=23"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="bridge" data="sofia/internal/1001%$$ 
{domain}"/>
       <action application="sleep" data="23000"/>
       <action application="tone_detect" data="ring 400,450 r +5000  
set RING=true"/>
       <action application="transfer" data="public_answer_and_email"/>
     </condition>
   </extension>

<extension name="public_answer_and_email">
     <condition field="RING" expression="true">
        <action application="answer"/>
        <action application="voicemail" data="default $${domain} 1001"/>
     </condition>
   </extension>
</include>

Unfortunately, this is not working, and the logs are not yielding  
anything the is helpful to me. Is my use of the tone_detect  
application and the basic dialplan correct?

Merul
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From help at pdscc.com  Sun Aug 23 11:51:53 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 11:51:53 -0700
Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3
Message-ID: <20090823185152.D17845FE@sinclaire.sibble.net>

Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server, 
then went to install FreePBX v3, I've gotten all the prerequisities in the 
wizard fixed except for ESL

As per

http://wiki.freeswitch.org/wiki/Event_Socket_Library
http://wiki.freeswitch.org/wiki/Event_Socket

I go into my FS source dir

/home/sibbleh/freeswitch-1.0.4/libs/esl

Run make and then "sudo make phpmod-install" 

and I get 


$ sudo make phpmod-install
make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-
I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb 
-I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-
variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-
I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb 
-I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" 
CXX_CFLAGS="" -C php
make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
g++  -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g 
-ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -
I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -
I/usr/include/php5/Zend -I/usr/include/php5/ext -
I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -
Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
cc1plus: warnings being treated as errors
esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1047: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1073: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int, 
zval*, zval**, zval*, int)':
esl_wrap.cpp:1111: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int, 
zval*, zval**, zval*, int)':
esl_wrap.cpp:1141: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1172: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1198: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1234: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1269: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1294: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**, zval*, 
int)':
esl_wrap.cpp:1346: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1403: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1441: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1478: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1508: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1538: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1571: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_addHeader(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1611: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_delHeader(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1644: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_firstHeader(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1674: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_nextHeader(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1704: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_0(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1744: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_1(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1770: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLconnection(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1803: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_socketDescriptor(int, 
zval*, zval**, zval*, int)':
esl_wrap.cpp:1846: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_connected(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1872: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_getInfo(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1898: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_send(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1931: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendRecv(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1964: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_api(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:2007: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_bgapi(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:2050: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendEvent(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2082: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEvent(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2108: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEventTimed(int, 
zval*, zval**, zval*, int)':
esl_wrap.cpp:2141: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_filter(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2181: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_events(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2221: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_execute(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2272: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_executeAsync(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2323: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_setAsyncExecute(int, 
zval*, zval**, zval*, int)':
esl_wrap.cpp:2356: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_setEventLock(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2389: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_disconnect(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2415: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_eslSetLogLevel(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:2438: error: format not a string literal and no format arguments
make[1]: *** [esl_wrap.o] Error 1
make[1]: Leaving directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
make: *** [phpmod] Error 2

Same thing happens if I try sudo make everymod

Checking the list archives I found this thread

http://www.nabble.com/ESL-Wrapper-td22209991.html#a22222338

I've made sure that the php-dev packages are installed. Any suggestions on 
what to do next?
-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From Prometheus001 at gmx.net  Sun Aug 23 13:20:16 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Sun, 23 Aug 2009 22:20:16 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <191c3a030908211309j60c16b26h1e6445891136ae7f@mail.gmail.com>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>	<4A8DB808.1030903@gmx.net>	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>	<4A8EBF77.1080204@gmx.net>	<BD3DE8E0-3363-465C-985D-EC593AE45694@freeswitch.org>
	<191c3a030908211309j60c16b26h1e6445891136ae7f@mail.gmail.com>
Message-ID: <4A91A480.7080307@gmx.net>

Hello Anthony,

I set PCMA at 30i,PCMU at 30i and I can see in the logs that PCMA is used.
However ptime is set to 20 msec as shown in the Logs:

2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP:
v=0
o=user 2075230 2075230 IN IP4 217.xx.xx.xxx
s=call
c=IN IP4 217.xx.xx.xxx
t=0 0
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtcp:7079

2009-08-23 22:11:29.530301 [DEBUG] switch_core_state_machine.c:404
(sofia/internal/02xxxxxxxxx at fs1.my.domain) State NEW
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare
[PCMA:8:8000:0]/[G722:9:8000:20]
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare
[PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare
[PCMA:8:8000:0]/[PCMA:8:8000:20]
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:2090 Set Codec
sofia/internal/02xxxxxxxxx at fs1.my.domain PCMA/8000 20 ms 160 samples
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
payload to 101

Sound from FS to Fritzbox is fine. Sound from Fritzbox to FS is horrible.

Best regards
Peter

Anthony Minessale schrieb:
> try setting FS to 30ms too
>
> edit vars.xml and add @30i to everywhere you see  PCMU or PCMA so it
> looks like PCMU at 30i
>
> from:
>
>   <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
> to:
>
>   <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU at 30i,PCMA at 30i,GSM"/>
>   <X-PRE-PROCESS cmd="set"
> data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i,GSM"/>
>
>
> On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian at freeswitch.org
> <mailto:brian at freeswitch.org>> wrote:
>
>     You can ship me one whois bkw.org <http://bkw.org>, I can add it
>     to my lab.
>
>     /b
>
>     On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
>
>     >
>     > BTW: We can ship you a FritzBox if you need one for testing.
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From help at pdscc.com  Sun Aug 23 14:37:47 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 14:37:47 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
Message-ID: <20090823213745.D81819D5@sinclaire.sibble.net>

I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp capable 
endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone client, 
2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i (symbian 
s60) and an O2 Xda Flame (windows mobile 5).

All 3 endpoints are registered with FS using the default extensions of 1000-
1003

With global_setvar zrtp_secure_media=true the zrtp negotiation between end 
points happens but the SAS never matches,below is console output for a call 
between 2 of the endpoints 


2009-08-23 14:10:17.643073 [NOTICE] mod_sofia.c:1509 Pre-Answer 
sofia/internal/1003 at 10.12.13.45!
2009-08-23 14:10:21.257568 [NOTICE] sofia.c:3794 Channel 
[sofia/internal/sip:1000 at 10.12.13.166:5062] has been answered
2009-08-23 14:10:21.275521 [NOTICE] switch_ivr_originate.c:2015 Channel 
[sofia/internal/1003 at 10.12.13.45] has been answered
2009-08-23 14:10:22.232053 [WARNING] mod_sofia.c:810 We were told to use 
ptime 20 but what they meant to say was 80
This issue has so far been identified to happen on the following broken 
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who 
knows what will happen..
2009-08-23 14:11:34.496118 [NOTICE] sofia.c:322 Hangup 
sofia/internal/sip:1000 at 10.12.13.166:5062 [CS_EXCHANGE_MEDIA] 
[NORMAL_CLEARING]
2009-08-23 14:11:34.512100 [NOTICE] switch_ivr_bridge.c:1016 Hangup 
sofia/internal/1003 at 10.12.13.45 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1086 Session 16 
(sofia/internal/sip:1000 at 10.12.13.166:5062) Ended
2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/sip:1000 at 10.12.13.166:5062 [CS_DESTROY]
2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1086 Session 15 
(sofia/internal/1003 at 10.12.13.45) Ended
2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/1003 at 10.12.13.45 [CS_DESTROY]

Of note, with the endpoints registered through the Ekiga sip server, the sas 
DOES match on both ends.

With global_setvar zrtp_secure_media=false, the endpoints can't detect a zrtp 
peer.  

Reading the list archives hasn't enlightened me.

I see this comment from 2008

http://www.nabble.com/Freeswitch-and-Twinkle-and-ZRTP-
td18518140.html#a18518343

    On Jul 17, 2008, at 4:23 PM, Michael Jerris wrote:

    > it should in bypass_media or proxy_media modes.  in the other modes we
    > are in the media path and would not know how to handle the encrypted
    > packets.
    >
    > Mike

Is this still relevant? Or is there some other setting not covered here

http://wiki.freeswitch.org/wiki/ZRTP

to make this work properly? I ask firstly about  this in the context of a 
peer 2 peer zrtp communication between the endpoints, then secondly in the 
case of FS acting as a trusted middleman as in section 2 here

http://www.zfoneproject.com/docs/asterisk/man/html/u_guide.html#passthrough

Lastly how does one implement the security enrollment as noted above with FS
-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From brian at freeswitch.org  Sun Aug 23 15:09:56 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 17:09:56 -0500
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090823213745.D81819D5@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>
Message-ID: <FC5818F6-C173-4D47-9C2A-4CCC24A46D6B@freeswitch.org>

This is because you didn't install the zrtpagent.lua script and dial  
zrtp on your keypad to enroll the FS box as a trusted man in the  
middle... which btw will only work with the unreleased zfone3.

/b

On Aug 23, 2009, at 4:37 PM, Harondel J. Sibble wrote:

> I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp  
> capable
> endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone  
> client,
> 2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i  
> (symbian
> s60) and an O2 Xda Flame (windows mobile 5).
>
> All 3 endpoints are registered with FS using the default extensions  
> of 1000-
> 1003
>
> With global_setvar zrtp_secure_media=true the zrtp negotiation  
> between end
> points happens but the SAS never matches,below is console output for  
> a call
> between 2 of the endpoints



From help at pdscc.com  Sun Aug 23 15:39:38 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 15:39:38 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <FC5818F6-C173-4D47-9C2A-4CCC24A46D6B@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<FC5818F6-C173-4D47-9C2A-4CCC24A46D6B@freeswitch.org>
Message-ID: <20090823223936.84B1373E@sinclaire.sibble.net>

Brian, okay, that answers the case with FS acting as a trusted man in the 
middle, but what about in the peer to peer case? Shouldn't FS just be passing 
the ztrp traffic through to the endpoints? Or am I misunderstanding how it's 
supposed to work?

Secondly where would I find info about zrtpagent.lua? Doing a search for that 
term on the wiki returns no results, ditto for a search of the nabble list 
archives (other than your response today of course). Also ditto for a search 
of the box running FS.

On 23 Aug 2009 at 17:09, Brian West wrote:

> This is because you didn't install the zrtpagent.lua script and dial  
> zrtp on your keypad to enroll the FS box as a trusted man in the  
> middle... which btw will only work with the unreleased zfone3.
> 
> /b

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From brian at freeswitch.org  Sun Aug 23 15:48:27 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 17:48:27 -0500
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090823223936.84B1373E@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<FC5818F6-C173-4D47-9C2A-4CCC24A46D6B@freeswitch.org>
	<20090823223936.84B1373E@sinclaire.sibble.net>
Message-ID: <B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>


On Aug 23, 2009, at 5:39 PM, Harondel J. Sibble wrote:

> Brian, okay, that answers the case with FS acting as a trusted man  
> in the
> middle, but what about in the peer to peer case? Shouldn't FS just  
> be passing
> the ztrp traffic through to the endpoints? Or am I misunderstanding  
> how it's
> supposed to work?

Nope.  FreeSWITCH is a b2bua so the traffic is decrypted... and  
relayed and encrypted again to the far side.  Which is why you have to  
use the trusted man in the middle stuff but you can't use it fully  
unless you have zfone3 beta or release.

> Secondly where would I find info about zrtpagent.lua? Doing a search  
> for that
> term on the wiki returns no results, ditto for a search of the  
> nabble list
> archives (other than your response today of course). Also ditto for  
> a search
> of the box running FS.

scripts/lua/ (in src tree)

/b



From help at pdscc.com  Sun Aug 23 16:16:11 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 16:16:11 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090823223936.84B1373E@sinclaire.sibble.net>,
	<B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>
Message-ID: <20090823231610.3398273E@sinclaire.sibble.net>



On 23 Aug 2009 at 17:48, Brian West wrote:

> Nope.  FreeSWITCH is a b2bua so the traffic is decrypted... and  

Hadn't heard that term before

http://en.wikipedia.org/wiki/Back-to-back_user_agent

that clears it up.  Any plans to offer straight proxy/passthru?

> relayed and encrypted again to the far side.  Which is why you have to  
> use the trusted man in the middle stuff but you can't use it fully  
> unless you have zfone3 beta or release.

Which explains the mismatched sas on each end. Okay got it.
 
> scripts/lua/ (in src tree)

ahh, it's zrtp_agent.lua which is why I didn't find it. I'll add that into 
the mix and cross my fingers that zfone3 gets released soon along with it's 
inclusion into the softphones I have on my smartphone devices.
-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From intralanman at freeswitch.org  Sun Aug 23 16:23:23 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Sun, 23 Aug 2009 19:23:23 -0400
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
Message-ID: <3BF035D5-ED50-406E-8D19-FAAD314BB045@freeswitch.org>

Could you load freeswitch with a couple hundred calls then run the  
test again.. and do the same to asterisk and see how the numbers stack  
up then? I'm just curious to see what happens at that point.

-Ray

On Aug 21, 2009, at 3:15 PM, Rogelio Perez wrote:

> Hi Everyone,
>
> I'm working on a PBX project for the Sheevaplug ARM based computer,  
> with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
> So far I've found a big difference between Freeswitch and Asterisk  
> performance times.
> This is a comparison of the time it takes them to perform different  
> actions:
>
> startup Freeswitch:			3 min.
> startup Asterisk:			2 sec.
>
> call extension Freeswitch:	6 sec.
> call extension Asterisk:		0 sec.
>
> shutdown Freeswitch:		6.5 sec
> shutdown Asterisk:			0 sec.
>
> reload config Freeswitch: 	1 sec.
> reload config Asterisk:		1 sec.
>
> Both were built from sources natively (no cross-compiling), and they  
> use the default startup configurations.
> I have managed to lower the Freeswitch times by disabling most of  
> the modules and recompiling, but it is still far away from Asterisk  
> (i.e. FS startup time 2.5 min).
>
> 1. Is there any way to further improve Freeswitch performance for  
> the ARM architecture?
> 2. Can this be related to the lack of a FPU (the Sheevalug emulates  
> the floating point operations).
> 3. On the startup I see this error repeated many times: [ERR]  
> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be  
> related?
>
> Thanks,
> Rogelio Perez
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From matt at venturevoip.com  Sun Aug 23 16:27:56 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Mon, 24 Aug 2009 11:27:56 +1200
Subject: [Freeswitch-users] Couple of questions
Message-ID: <4A91D07C.1080805@venturevoip.com>

Hi,

I don't see how I can read some responses to command using esl.

I.E. esl_send_recv(&handle, "api show calls count\n\n");

and

printf("Header Test %s\n", esl_event_get_header(event, "API-Command"));
printf("Body Test %s\n", esl_event_get_body(event));

the header details are returned.

The body is null.

Also, I can originate a call and set the account code for it, but how do 
I get a list of calls with their account codes?

Do I get a list of calls then go through them one by one and get the 
variables for those calls by uuid?

Does anyone have any documentation for the esl api?

Even if I could read some comments from a usage of it would be useful.

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)


From sprice at gmail.com  Sun Aug 23 16:31:16 2009
From: sprice at gmail.com (SP)
Date: Sun, 23 Aug 2009 18:31:16 -0500
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <3BF035D5-ED50-406E-8D19-FAAD314BB045@freeswitch.org>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
	<3BF035D5-ED50-406E-8D19-FAAD314BB045@freeswitch.org>
Message-ID: <7e2ac3270908231631y5cc4d0bbmc0ceedcb070a04fa@mail.gmail.com>

Don't forget to press tab at the asterisk console!  :)

On Sun, Aug 23, 2009 at 18:23, Raymond Chandler
<intralanman at freeswitch.org>wrote:

> Could you load freeswitch with a couple hundred calls then run the test
> again.. and do the same to asterisk and see how the numbers stack up then?
> I'm just curious to see what happens at that point.
> -Ray
>
> On Aug 21, 2009, at 3:15 PM, Rogelio Perez wrote:
>
> Hi Everyone,
>
> I'm working on a PBX project for the Sheevaplug<http://www.google.com/url?sa=t&source=web&ct=res&cd=1&url=http%253A%252F%252Fwww.marvell.com%252Fproducts%252Fembedded_processors%252Fdeveloper%252Fkirkwood%252Fsheevaplug.jsp&ei=EOOOSo6ILcyOtgeNruTOBA&usg=AFQjCNFREhfy_erj5irBWk8XFUjkQOP-aw&sig2=pAHYlI15IbZ5Kcw0n-nIvA> ARM
> based computer, with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
> So far I've found a big difference between Freeswitch and Asterisk
> performance times.
> This is a comparison of the time it takes them to perform different
> actions:
>
> startup Freeswitch: 3 min.
> startup Asterisk: 2 sec.
>
> call extension Freeswitch: 6 sec.
> call extension Asterisk: 0 sec.
>
> shutdown Freeswitch: 6.5 sec
>
> shutdown Asterisk: 0 sec.
>
>
> reload config Freeswitch: 1 sec.
> reload config Asterisk: 1 sec.
>
>
> Both were built from sources natively (no cross-compiling), and they use
> the default startup configurations.
> I have managed to lower the Freeswitch times by disabling most of the
> modules and recompiling, but it is still far away from Asterisk (i.e. FS
> startup time 2.5 min).
>
> 1. Is there any way to further improve Freeswitch performance for the ARM
> architecture?
> 2. Can this be related to the lack of a FPU (the Sheevalug emulates the
> floating point operations).
> 3. On the startup I see this error repeated many times: [ERR]
> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be related?
>
> Thanks,
> Rogelio Perez
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Shannon
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From brian at freeswitch.org  Sun Aug 23 16:43:31 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 18:43:31 -0500
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090823231610.3398273E@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090823223936.84B1373E@sinclaire.sibble.net>,
	<B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>
	<20090823231610.3398273E@sinclaire.sibble.net>
Message-ID: <A4BF66BF-AE4F-4D1B-9102-569EBF16A08C@freeswitch.org>

I can confirm that it works 100% correct passing the SAS across the  
bridge correctly once you trust the switch in the middle.

/b

On Aug 23, 2009, at 6:16 PM, Harondel J. Sibble wrote:

>
> ahh, it's zrtp_agent.lua which is why I didn't find it. I'll add  
> that into
> the mix and cross my fingers that zfone3 gets released soon along  
> with it's
> inclusion into the softphones I have on my smartphone devices.



From help at pdscc.com  Sun Aug 23 16:50:32 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 16:50:32 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090823231610.3398273E@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>,
	<20090823231610.3398273E@sinclaire.sibble.net>
Message-ID: <20090823235030.A95BF5FE@sinclaire.sibble.net>



On 23 Aug 2009 at 16:16, Harondel J. Sibble wrote:

> ahh, it's zrtp_agent.lua which is why I didn't find it. I'll add that into
> the mix and cross my fingers that zfone3 gets released soon along with it's
> inclusion into the softphones I have on my smartphone devices.

Well good news for the Tiviphone client

    Dear Harondel J. Sibble,

    We support zfone3 starting from Tiviphone release 2.0.6 it is available for
    Symbian try upgrading.

    -- 
    Harondel J. Sibble 
    Sibble Computer Consulting
    Creating Solutions for the small and medium business computer user.
    help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
    (604) 739-3709 (voice)



From brian at freeswitch.org  Sun Aug 23 16:50:27 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 18:50:27 -0500
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <4A91D07C.1080805@venturevoip.com>
References: <4A91D07C.1080805@venturevoip.com>
Message-ID: <6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>


On Aug 23, 2009, at 6:27 PM, Matt Riddell wrote:

> Hi,
>
> I don't see how I can read some responses to command using esl.
>
> I.E. esl_send_recv(&handle, "api show calls count\n\n");
>
> and
>
> printf("Header Test %s\n", esl_event_get_header(event, "API- 
> Command"));
> printf("Body Test %s\n", esl_event_get_body(event));
>
> the header details are returned.
>
> The body is null.

I'm not too sure about using ESL in C, I have used it pretty much  
exclusively in perl.

> Also, I can originate a call and set the account code for it, but  
> how do
> I get a list of calls with their account codes?

originate {account_code=1234}sofia/profile/target at ip ....

You can get the list of the channels via "show channels" or bridged  
calls with "show calls"

 From there you have the UUID's you can call uuid_dump on them to get  
all the variables.

> Do I get a list of calls then go through them one by one and get the
> variables for those calls by uuid?

You could do this or setup a listener to get the events as they happen  
and keep the info you need.

> Does anyone have any documentation for the esl api?

http://docs.freeswitch.org/ (this should help, its under files list  
see esl.h)

> Even if I could read some comments from a usage of it would be useful.

I just find it interesting you're doing this with C.

> -- 
> Cheers,
>
> Matt Riddell
> Director
> _______________________________________________
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Sun Aug 23 16:53:36 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 18:53:36 -0500
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090823235030.A95BF5FE@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>,
	<20090823231610.3398273E@sinclaire.sibble.net>
	<20090823235030.A95BF5FE@sinclaire.sibble.net>
Message-ID: <09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org>

Wish they would send me one for my E63 for testing... only been  
working with zfone 3 so far.

/b

On Aug 23, 2009, at 6:50 PM, Harondel J. Sibble wrote:

> Well good news for the Tiviphone client



From matt at venturevoip.com  Sun Aug 23 16:57:37 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Mon, 24 Aug 2009 11:57:37 +1200
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
References: <4A91D07C.1080805@venturevoip.com>
	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
Message-ID: <4A91D771.9080609@venturevoip.com>

On 24/08/09 11:50 AM, Brian West wrote:

>> Even if I could read some comments from a usage of it would be useful.
>
> I just find it interesting you're doing this with C.

:)  The rest of the application is in C, so it makes sense to use 
FreeSwitch's esl in C.

Thanks for your help man will let you know if I have any other questions :)

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)


From help at pdscc.com  Sun Aug 23 17:20:12 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 17:20:12 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090823235030.A95BF5FE@sinclaire.sibble.net>,
	<09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org>
Message-ID: <20090824002011.04DDE9CF@sinclaire.sibble.net>

They have a trial version of the full client available from their website, I 
think it's only for 10 days though. I suspect if you approached them, they'd 
probably give you a full client for permanent use for interoperability 
testing.

http://www.tivi.com/en/download/credit_paypal.php

Opps, my bad, it's only for 3 days :-(

You can bid on the price you want to pay if you are into buying a copy.

Just got the word back from them, the 2.0.6 client is available for Windows 
Mobile but not officially released since it has some issues with video 
encryption. Since I'm not using video yet, I am going to upgrade the windows 
mobile phones with this version.

So I popped the zrtp_agent.lua script into

/usr/local/freeswitch/scripts/

added following line to 

/usr/local/freeswitch/conf/autoload_configs/lua.conf.xml

   <!--<param name="startup-script" value="zrtp_agent.lua"/>-->

under this section

    <!--
        The following options identifies a lua script that is launched
        at startup and may live forever in the background.
        You can define multiple lines, one for each script you
        need to run.

And restarted FS, when I try to connect and type zrtp on the keypad it 
doesn't work and I see

2009-08-23 17:06:46.272135 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1001 at 10.12.13.45 [03265312-9042-11de-8c5d-d333d780ffc7]
2009-08-23 17:06:46.288730 [INFO] mod_dialplan_xml.c:315 Processing 1001-
>zrtp in context default
2009-08-23 17:06:46.301899 [NOTICE] switch_ivr.c:1349 Transfer 
sofia/internal/1001 at 10.12.13.45 to enum[zrtp at default]
2009-08-23 17:06:46.495944 [INFO] switch_core_state_machine.c:136 No Route, 
Aborting
2009-08-23 17:06:46.495944 [NOTICE] switch_core_state_machine.c:137 Hangup 
sofia/internal/1001 at 10.12.13.45 [CS_ROUTING] [NO_ROUTE_DESTINATION]
2009-08-23 17:06:46.503261 [NOTICE] switch_core_session.c:1086 Session 3 
(sofia/internal/1001 at 10.12.13.45) Ended
2009-08-23 17:06:46.503261 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/1001 at 10.12.13.45 [CS_DESTROY]

Were you being literal when you said to type in zrtp on the phones keypad? I 
assume you meant dial the phone number equivalent to that.

I also assume this is not working because 
1) I should put the zrtp_agent.lua script in one of the designated script 
dirs like 

    <!-- <param name="module-directory" value="/usr/lib/lua/5.1/?.so"/> -->
    <!-- <param name="module-directory" value="/usr/local/lib/lua/5.1/?.so"/> 
-->

or 

2) I need to wait until zfone3 is released and recompile FS with the new tar 
file from the zfoneproject site

or

3) both of the above apply?


On 23 Aug 2009 at 18:53, Brian West wrote:

> Wish they would send me one for my E63 for testing... only been  
> working with zfone 3 so far.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From brian at freeswitch.org  Sun Aug 23 17:53:14 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 19:53:14 -0500
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090824002011.04DDE9CF@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090823235030.A95BF5FE@sinclaire.sibble.net>,
	<09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org>
	<20090824002011.04DDE9CF@sinclaire.sibble.net>
Message-ID: <C9F0463C-F22E-4339-879C-FF5B35C3FE43@freeswitch.org>


On Aug 23, 2009, at 7:20 PM, Harondel J. Sibble wrote:

> added following line to
>
> /usr/local/freeswitch/conf/autoload_configs/lua.conf.xml

WRONG.

>
>
>   <!--<param name="startup-script" value="zrtp_agent.lua"/>-->
>

Don't touch this.

> under this section
>
>    <!--
>        The following options identifies a lua script that is launched
>        at startup and may live forever in the background.
>        You can define multiple lines, one for each script you
>        need to run.
>
> And restarted FS, when I try to connect and type zrtp on the keypad it
> doesn't work and I see

Just put it into the scripts folder and run it from the dialplan see  
default configs.

     <!-- install zrtp_agent.lua into scripts (ZRTP == 9787) -->
     <extension name="zrtp_enrollement">
       <condition field="destination_number" expression="^9787$">
	<action application="answer"/>
	<action application="lua" data="zrtp_agent.lua"/>
       </condition>
     </extension>


/b

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From brian at freeswitch.org  Sun Aug 23 17:54:54 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 19:54:54 -0500
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <4A91D771.9080609@venturevoip.com>
References: <4A91D07C.1080805@venturevoip.com>
	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
	<4A91D771.9080609@venturevoip.com>
Message-ID: <DE14F3F4-C7A7-481D-B537-2B9D283AF600@freeswitch.org>


On Aug 23, 2009, at 6:57 PM, Matt Riddell wrote:

> :)  The rest of the application is in C, so it makes sense to use
> FreeSwitch's esl in C.

I'm a perl monkey when it comes to using ESL  :P  I like reusing all  
the modules already out there plus I'm a little lazy :P

> Thanks for your help man will let you know if I have any other  
> questions :)

kewl.

/b



From help at pdscc.com  Sun Aug 23 19:05:25 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 19:05:25 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <C9F0463C-F22E-4339-879C-FF5B35C3FE43@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090824002011.04DDE9CF@sinclaire.sibble.net>,
	<C9F0463C-F22E-4339-879C-FF5B35C3FE43@freeswitch.org>
Message-ID: <20090824020523.A44439CF@sinclaire.sibble.net>



On 23 Aug 2009 at 19:53, Brian West wrote:

> Just put it into the scripts folder and run it from the dialplan see  
> default configs.
> 
>      <!-- install zrtp_agent.lua into scripts (ZRTP == 9787) -->
>      <extension name="zrtp_enrollement">
>        <condition field="destination_number" expression="^9787$">
> 	<action application="answer"/>
> 	<action application="lua" data="zrtp_agent.lua"/>
>        </condition>
>      </extension>

Okay, where do I get the audio files?

2009-08-23 18:54:08.991629 [INFO] mod_dialplan_xml.c:315 Processing 1001-
>9787 in context default
2009-08-23 18:54:09.10863 [NOTICE] mod_dptools.c:649 Channel 
[sofia/internal/1001 at 10.12.13.45] has been answered
2009-08-23 18:54:09.215288 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr                  tp/zrtp-
status_securing.wav] [System error : No such file or directory.]
2009-08-23 18:54:12.233267 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr                  tp/zrtp-
status_secure.wav] [System error : No such file or directory.]
2009-08-23 18:54:12.234565 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr                  tp/zrtp-
enroll_welcome.wav] [System error : No such file or directory.]
2009-08-23 18:54:13.252933 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr                  tp/zrtp-
enroll_confirmed.wav] [System error : No such file or directory.]
2009-08-23 18:54:15.293476 [WARNING] mod_sofia.c:810 We were told to use 
ptime 20 but what they meant to say was 80
This issue has so far been identified to happen on the following broken 
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who 
knows what will happen..
2009-08-23 18:54:17.333543 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr                  tp/zrtp-
thankyou_goodbye.wav] [System error : No such file or directory.]


my /usr/local/freeswitch/sounds is bereft of any files whatsoever, searching 
the full filesystem for callie only gets me

/home/sibbleh/freeswitch-1.0.4/debian/freeswitch-sounds-en-us-callie-
16000.install
/home/sibbleh/freeswitch-1.0.4/debian/freeswitch-sounds-en-us-callie-
32000.install
/home/sibbleh/freeswitch-1.0.4/debian/freeswitch-sounds-en-us-callie-
8000.install





-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From brian at freeswitch.org  Sun Aug 23 19:30:11 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 21:30:11 -0500
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090824020523.A44439CF@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090824002011.04DDE9CF@sinclaire.sibble.net>,
	<C9F0463C-F22E-4339-879C-FF5B35C3FE43@freeswitch.org>
	<20090824020523.A44439CF@sinclaire.sibble.net>
Message-ID: <CD6A6C1A-9CEC-4663-8C6C-C7A5237BBC30@freeswitch.org>

If you reinstall the latest sound files you'll have them.

/b

On Aug 23, 2009, at 9:05 PM, Harondel J. Sibble wrote:

>>     </extension>
>
> Okay, where do I get the audio files?



From help at pdscc.com  Sun Aug 23 19:38:51 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 19:38:51 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <CD6A6C1A-9CEC-4663-8C6C-C7A5237BBC30@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090824020523.A44439CF@sinclaire.sibble.net>,
	<CD6A6C1A-9CEC-4663-8C6C-C7A5237BBC30@freeswitch.org>
Message-ID: <20090824023850.05CEE79C@sinclaire.sibble.net>

Ahh, I didn't quite clue-in that I had to run the additional installers as 
below when the main compile finished, I thought it was saying it had already 
done that.

Makes perfect sense in hindsite ;-)

    +-------- FreeSWITCH install Complete ----------+
     + FreeSWITCH has been successfully installed.   +
     +                                               +
     +       Install sounds:                         +
     +       (uhd-sounds includes hd-sounds, sounds) +
     +       (hd-sounds includes sounds)             +
     +       ------------------------------------    +
     +               make cd-sounds-install          +
     +               make cd-moh-install             +
     +                                               +
     +               make uhd-sounds-install         +
     +               make uhd-moh-install            +
     +                                               +
     +               make hd-sounds-install          +
     +               make hd-moh-install             +
     +                                               +
     +               make sounds-install             +
     +               make moh-install                +
     +                                               +
     +       Install non english sounds:             +
     +       replace XX with language                +
     +       (ru : Russian)                          +
     +       ------------------------------------    +
     +               make cd-sounds-XX-install       +
     +               make uhd-sounds-XX-install      +
     +               make hd-sounds-XX-install       +
     +               make sounds-XX-install          +


On 23 Aug 2009 at 21:30, Brian West wrote:

> If you reinstall the latest sound files you'll have them.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From help at pdscc.com  Sun Aug 23 20:22:02 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 20:22:02 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <CD6A6C1A-9CEC-4663-8C6C-C7A5237BBC30@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090824020523.A44439CF@sinclaire.sibble.net>,
	<CD6A6C1A-9CEC-4663-8C6C-C7A5237BBC30@freeswitch.org>
Message-ID: <20090824032200.B213D79C@sinclaire.sibble.net>

Whoah.....

2009-08-23 20:07:52.583524 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1001 at 192.168.73.45 [4ff9d452-905b-11de-8c5d-d333d780ffc7]
2009-08-23 20:07:52.740094 [INFO] mod_dialplan_xml.c:315 Processing 1001-
>9787 in context default
2009-08-23 20:07:52.980164 [NOTICE] mod_dptools.c:649 Channel 
[sofia/internal/1001 at 192.168.73.45] has been answered
2009-08-23 20:07:54.333362 [WARNING] mod_sofia.c:810 We were told to use 
ptime 20 but what they meant to say was 80
This issue has so far been identified to happen on the following broken 
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who 
knows what will happen..
2009-08-23 20:08:16.912027 [NOTICE] sofia.c:322 Hangup 
sofia/internal/1001 at 192.168.73.45 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 20:08:22.54838 [NOTICE] switch_core_session.c:1086 Session 12 
(sofia/internal/1001 at 192.168.73.45) Ended
2009-08-23 20:08:22.54838 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/1001 at 192.168.73.45 [CS_DESTROY]

I get audio now, but it's running really slooooooooooooooowly. I'd say about 
1/4 to 1/8 normal speech speed

I did the following

 make cd-sounds-install;make uhd-sounds-install;make uhd-moh-install

Probably relevant, box this is running on is a Celeron 1ghz with 512mb ram.

Weird, it starts off okay, but when this pops up

2009-08-23 20:21:10.93004 [WARNING] mod_sofia.c:810 We were told to use ptime 
20 but what they meant to say was 80
This issue has so far been identified to happen on the following broken 
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who 
knows what will happen..

the audio starts to crawl...
On 23 Aug 2009 at 21:30, Brian West wrote:

> If you reinstall the latest sound files you'll have them.
> 
> /b
> 
> On Aug 23, 2009, at 9:05 PM, Harondel J. Sibble wrote:
> 
> >>     </extension>
> >
> > Okay, where do I get the audio files?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From scott.torr.fs at letterboxes.org  Sun Aug 23 20:26:17 2009
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Mon, 24 Aug 2009 13:26:17 +1000
Subject: [Freeswitch-users] Screaming monkeys on ext 5000
Message-ID: <1251084377.461.1331260223@webmail.messagingengine.com>

Just a quick note, and I'm sure why, but screaming monkeys does not play
on the the default installation.

I have not looked into why, but thought I would just quickly let you
know.

Perhaps I have not done something?

regards,
sbt 


From brian at freeswitch.org  Sun Aug 23 20:31:14 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 22:31:14 -0500
Subject: [Freeswitch-users] Screaming monkeys on ext 5000
In-Reply-To: <1251084377.461.1331260223@webmail.messagingengine.com>
References: <1251084377.461.1331260223@webmail.messagingengine.com>
Message-ID: <A7FF6EFA-5CBD-4419-BC0A-656B43B34686@freeswitch.org>


On Aug 23, 2009, at 10:26 PM, Scott Torr wrote:

> Just a quick note, and I'm sure why, but screaming monkeys does not  
> play
> on the the default installation.

It requires internet connectivity.  It calls a remote system to play  
which is out of our control.

> I have not looked into why, but thought I would just quickly let you
> know.

Thanks,

> Perhaps I have not done something?
>
> regards,
> sbt

/b



From jim at evolutiontel.net  Sun Aug 23 21:14:15 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 24 Aug 2009 14:14:15 +1000
Subject: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with
	same Gateway
In-Reply-To: <4256bf830908212341k1c0a9c9x7c8a9c70460965f7@mail.gmail.com>
References: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>
	<6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>
	<4256bf830908201021s7da0bff0wed8e060dbac0db2c@mail.gmail.com>
	<4256bf830908212341k1c0a9c9x7c8a9c70460965f7@mail.gmail.com>
Message-ID: <ddad80610908232114y159e70cew499138d0261f35b5@mail.gmail.com>

If I understand your issue correctly, it sounds to me like FS is not
set to anchor the RTP media stream.  Experience suggests that most
SBC's do not like trying to loopback RTP traffic to themselves.

Check to see what IP address's are getting used in the
c=xxx.xxx.xxx.xxx for the INVITE and 200OK messages.  If you see the
SBC IP address in messages on both sides of the FS box this will be
your issue.

You could try setting bypass_media=false in your dialplan.



On Sat, Aug 22, 2009 at 4:41 PM, Matthew Fong<mattdfong at gmail.com> wrote:
> So there seems to be some sort of error when bridging directly like
> originate
> {ignore_early_media=true}sofia/gateway/XXXX.com/91415992XXXX?&bridge(sofia/gateway/XXXX.com/91415465XXXX)
> BUT
> if I get FS to send media to leg A, and then bridge to leg B by using a lua
> script like
> session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/hh/hh-welcome.wav");
> session:execute("bridge", "sofia/gateway/epik.com/91415XXXXXXX");
> then the legs bridge together OK. This happens when trying to bridge two
> channels via the same Broadsoft SBC. Does this sound like a bug that should
> be submitted to JIRA?
> --matt
> http://www.hellohunter.com
>
> On Thu, Aug 20, 2009 at 10:21 AM, Matthew Fong <mattdfong at gmail.com> wrote:
>>
>> originate
>> {ignore_early_media=true}sofia/gateway/epik.com/914159927717?&bridge(sofia/gateway/epik.com/914154650027)
>>
>> is the string I was using from the console.
>>
>> On Thu, Aug 20, 2009 at 6:00 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>>>
>>> Hi
>>> How are you bridging the calls in FS (which api call or C function are
>>> you using)?
>>> Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> mrene at avgs.ca
>>>
>>>
>>>
>>> On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:
>>>
>>> I'm trying to get FreeSWITCH to bridge two channels together through the
>>> same external gateway, but I'm having issues hearing audio. Both legs if
>>> setup independently and forwarded to 5000 (test ivr) work fine for both
>>> inbound and outbound media, but when I try to bridge them together,
>>> everything seems fine in FreeSWITCH, but neither party can hear the other
>>> speak. I'm thinking the external gateway might be having some issues because
>>> I've been able to bridge 2 channels together through the same gateway on
>>> different providers, but thought I'd also try to seek some help here.
>>> FreeSWITCH should be handling the media for both channels, so I can't figure
>>> out why if Leg A and Leg B work independently, but not if they are bridged
>>> together. Is there a setting somewhere in FS that I'm missing?
>>> Below is a ngrep of the SIP interactions if it's useful. Thanks for the
>>> help.
>>> --matt
>>>
>>> interface: eth0 (172.24.200.0/255.255.255.0)
>>> filter: (ip or ip6) and ( port 5060 )
>>> U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060
>>> INVITE sip:914159927717 at 38.98.58.148 SIP/2.0.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>>> Max-Forwards: 70.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> To: <sip:914159927717 at 38.98.58.148>.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257811 INVITE.
>>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>>> Supported: timer, precondition, path, replaces.
>>> Allow-Events: talk, refer.
>>> Content-Type: application/sdp.
>>> Content-Disposition: session.
>>> Content-Length: 293.
>>> Remote-Party-ID: "FreeSWITCH"
>>> <sip:0000000000 at 216.81.56.198>;party=calling;screen=yes;privacy=off.
>>> .
>>> v=0.
>>> o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
>>> s=FreeSWITCH.
>>> c=IN IP4 216.81.56.198.
>>> t=0 0.
>>> m=audio 24700 RTP/AVP 0 8 3 101 13.
>>> a=rtpmap:0 PCMU/8000.
>>> a=rtpmap:8 PCMA/8000.
>>> a=rtpmap:3 GSM/8000.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-16.
>>> a=rtpmap:13 CN/8000.
>>> a=ptime:20.
>>>
>>> U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 100 Trying.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257811 INVITE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 183 Session Progress.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257811 INVITE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>>> Allow:
>>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>> Content-Type: application/sdp.
>>> Content-Length: 227.
>>> .
>>> v=0.
>>> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
>>> s=M6 Call.
>>> c=IN IP4 38.98.58.148.
>>> t=0 0.
>>> m=audio 42554 RTP/AVP 0 101.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-15.
>>> a=ptime:20.
>>> a=sendrecv.
>>> a=rtcp:6461 IN IP4 10.10.24.50.
>>>
>>> U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 200 OK.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257811 INVITE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>>> Session-Expires: 1800;refresher=uas.
>>> Allow:
>>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>> Supported: timer.
>>> Content-Type: application/sdp.
>>> Content-Length: 227.
>>> .
>>> v=0.
>>> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
>>> s=M6 Call.
>>> c=IN IP4 38.98.58.148.
>>> t=0 0.
>>> m=audio 42554 RTP/AVP 0 101.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-15.
>>> a=ptime:20.
>>> a=sendrecv.
>>> a=rtcp:6461 IN IP4 10.10.24.50.
>>>
>>> U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060
>>> ACK sip:914159927717 at 38.98.58.148:5060 SIP/2.0.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc.
>>> Max-Forwards: 70.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257811 ACK.
>>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060
>>> INVITE sip:914154650027 at 38.98.58.148 SIP/2.0.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>>> Max-Forwards: 70.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257815 INVITE.
>>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>>> Supported: timer, precondition, path, replaces.
>>> Allow-Events: talk, refer.
>>> Content-Type: application/sdp.
>>> Content-Disposition: session.
>>> Content-Length: 293.
>>> Remote-Party-ID: "FreeSWITCH"
>>> <sip:0000000000 at 216.81.56.198>;party=calling;screen=yes;privacy=off.
>>> .
>>> v=0.
>>> o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198.
>>> s=FreeSWITCH.
>>> c=IN IP4 216.81.56.198.
>>> t=0 0.
>>> m=audio 24798 RTP/AVP 0 8 3 101 13.
>>> a=rtpmap:0 PCMU/8000.
>>> a=rtpmap:8 PCMA/8000.
>>> a=rtpmap:3 GSM/8000.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-16.
>>> a=rtpmap:13 CN/8000.
>>> a=ptime:20.
>>>
>>> U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 100 Trying.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257815 INVITE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 183 Session Progress.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257815 INVITE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>>> Allow:
>>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>> Content-Type: application/sdp.
>>> Content-Length: 225.
>>> .
>>> v=0.
>>> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
>>> s=M6 Call.
>>> c=IN IP4 38.98.58.148.
>>> t=0 0.
>>> m=audio 46520 RTP/AVP 0 101.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-15.
>>> a=ptime:20.
>>> a=sendrecv.
>>> a=rtcp:6451 IN IP4 10.10.24.50.
>>>
>>> U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 200 OK.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257815 INVITE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>>> Session-Expires: 1800;refresher=uas.
>>> Allow:
>>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>> Supported: timer.
>>> Content-Type: application/sdp.
>>> Content-Length: 225.
>>> .
>>> v=0.
>>> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
>>> s=M6 Call.
>>> c=IN IP4 38.98.58.148.
>>> t=0 0.
>>> m=audio 46520 RTP/AVP 0 101.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-15.
>>> a=ptime:20.
>>> a=sendrecv.
>>> a=rtcp:6451 IN IP4 10.10.24.50.
>>>
>>> U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060
>>> ACK sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK.
>>> Max-Forwards: 70.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257815 ACK.
>>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080
>>> BYE sip:gw+epik.com at 216.81.56.198:5080 SIP/2.0.
>>> From: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
>>> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 4817 BYE.
>>> Max-Forwards: 70.
>>> Via: SIP/2.0/UDP
>>> 38.98.58.148:5060;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
>>> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
>>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>>> Allow:
>>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060
>>> SIP/2.0 200 OK.
>>> Via: SIP/2.0/UDP
>>> 38.98.58.148:5060;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
>>> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
>>> From: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
>>> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 4817 BYE.
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>>> Supported: timer, precondition, path, replaces.
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060
>>> BYE sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
>>> Max-Forwards: 70.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257816 BYE.
>>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>>> Supported: timer, precondition, path, replaces.
>>> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 200 OK.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257816 BYE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
>>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>>> Allow:
>>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>> Content-Length: 0.
>>> .
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From jim at evolutiontel.net  Sun Aug 23 21:19:55 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 24 Aug 2009 14:19:55 +1000
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
Message-ID: <ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>

In your SIP profiles this could be set.  I beleive 120 is the default setting.

param name="session-timeout" value="120"

On Fri, Aug 21, 2009 at 11:35 PM, bakko<asannucci at gmail.com> wrote:
> Do you have those lines in switch.conf file?
>
> ? ?<!--RTP port range -->
> ? ?<param name="rtp-start-port" value="10000"/>
> ? ?<param name="rtp-end-port" value="32767"/>
>
>
> BR
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From jmesquita at gmail.com  Sun Aug 23 21:27:16 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 24 Aug 2009 01:27:16 -0300
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
References: <4A91D07C.1080805@venturevoip.com>
	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
Message-ID: <5a8712120908232127n6cb2f0bdq7e05b3d7778f0e70@mail.gmail.com>

Hey there, FsGui uses ESL a lot and I had to go through the code to document
it so here is a few hints inline ...

Don't hesitate to keep the questions coming. I will fill in whenever I can.

jmesquita

On Sun, Aug 23, 2009 at 8:50 PM, Brian West <brian at freeswitch.org> wrote:

>
> On Aug 23, 2009, at 6:27 PM, Matt Riddell wrote:
>
> > Hi,
> >
> > I don't see how I can read some responses to command using esl.
> >
> > I.E. esl_send_recv(&handle, "api show calls count\n\n");
> >
> > and
> >
> > printf("Header Test %s\n", esl_event_get_header(event, "API-
> > Command"));
> > printf("Body Test %s\n", esl_event_get_body(event));
> >
> > the header details are returned.
> >
> > The body is null.
>

Body is null on every event that does not use headers to output information.
A good example would be console logs. I haven't seen too many default events
besides log that have body besides a application custom events.


>
>
> I'm not too sure about using ESL in C, I have used it pretty much
> exclusively in perl.
>
> > Also, I can originate a call and set the account code for it, but
> > how do
> > I get a list of calls with their account codes?
>
> originate {account_code=1234}sofia/profile/target at ip ....
>
> You can get the list of the channels via "show channels" or bridged
> calls with "show calls"
>
>  From there you have the UUID's you can call uuid_dump on them to get
> all the variables.


>
> > Do I get a list of calls then go through them one by one and get the
> > variables for those calls by uuid?
>

All ESL does is output events to socket and expose the API commands. It does
not maintain any kind of list of calls or anything like that so it is up to
you to maintain that yourself if you don't want to parse API output every
time.


>
>
> You could do this or setup a listener to get the events as they happen
> and keep the info you need.
>
> > Does anyone have any documentation for the esl api?
>
> http://docs.freeswitch.org/ (this should help, its under files list
> see esl.h)
>

I need to work a little bit more on that documentation as well.. I saw a few
conflicts with the core documentation too. Will get there once I have some
more time left.


>
> > Even if I could read some comments from a usage of it would be useful.
>
> I just find it interesting you're doing this with C.
>
> > --
> > Cheers,
> >
> > Matt Riddell
> > Director
> > _______________________________________________
> >
> > http://www.venturevoip.com/news.php (Daily Asterisk News)
> > http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> > http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jim at evolutiontel.net  Sun Aug 23 21:32:26 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 24 Aug 2009 14:32:26 +1000
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
Message-ID: <ddad80610908232132v3442c353s6cc09362cf71952c@mail.gmail.com>

X-Lite does support TCP, however you need to have NAPTR and SRV DNS
entries.  It used to support TLS, but this seems to have been removed
:(

sip.mydomain.net. IN NAPTR 0 0 "s" "SIPS+D2T"  "" _sips._tcp.sip.mydomain.net.
sip.mydomain.net. IN NAPTR 1 0 "s" "SIP+D2T"  "" _sip._tcp.sip.mydomain.net.
sip.mydomain.net. IN NAPTR 2 0 "s" "SIP+D2U"  "" _sip._udp.sip.mydomain.net.


_sip._udp.sip.mydomain.net.   43200 IN SRV 1 10 5060  sip.mydomain.net.
_sip._tcp.sip.mydomain.net.   43200 IN SRV 1 10 5060  sip.mydomain.net.
_sips._tcp.sip.mydomain.net.   43200 IN SRV 1 10 5061  sip.mydomain.net.


On Mon, Aug 17, 2009 at 10:18 PM, bakko<asannucci at gmail.com> wrote:
> Y make my tests with eyebeam.
>
> I thing X-lite dont't support TCP transport.
>
> BR
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From rogelio.perez at gmail.com  Sun Aug 23 21:45:06 2009
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Mon, 24 Aug 2009 01:45:06 -0300
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <191c3a030908211320s6feb5f10sf8670575069ec95a@mail.gmail.com>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
	<191c3a030908211320s6feb5f10sf8670575069ec95a@mail.gmail.com>
Message-ID: <9D12BA51-0C76-45B1-A8DF-3A1FB74E005B@gmail.com>

Thanks Andrew and Anthony,

I created a ramdisk for the db and log directories using tmpfs and now  
I see better performance times:

startup:		15.6 sec.
call extension: 	0 sec.
shutdown:	7.5 sec
reload config:	0 sec.

I have noticed that during the startup there is a 12 sec. pause while  
checking for UPnP:

2009-08-24 04:39:29.910459 [ERR] switch_nat.c:183 Error checking for  
PMP [general error]
2009-08-24 04:39:29.910694 [DEBUG] switch_nat.c:397 Checking for UPnP
2009-08-24 04:39:41.906029 [INFO] switch_nat.c:411 No PMP or UPnP NAT  
detected!

Is there any way to lower this time? Maybe disabling the check?

Thanks,
Rogelio

On Aug 21, 2009, at 5:20 PM, Anthony Minessale wrote:

> probably disk i/o.
>
> Is it some kind of flash drive?
>
> make a ramdisk and simlink in /usr/local/freeswitch/db and /usr/ 
> local/freeswitch/log to it
> the default configuration uses a lot of high level features that use  
> the sqlite db on the disk.
>
> We also offer commercial support where we could dig deeper into the  
> problem if you can't figure it out
> consulting at freeswitch.org
>
>
>
> On Fri, Aug 21, 2009 at 2:15 PM, Rogelio Perez <rogelio.perez at gmail.com 
> > wrote:
> Hi Everyone,
>
> I'm working on a PBX project for the Sheevaplug ARM based computer,  
> with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
> So far I've found a big difference between Freeswitch and Asterisk  
> performance times.
> This is a comparison of the time it takes them to perform different  
> actions:
>
> startup Freeswitch:			3 min.
> startup Asterisk:			2 sec.
>
> call extension Freeswitch:	6 sec.
> call extension Asterisk:		0 sec.
>
> shutdown Freeswitch:		6.5 sec
> shutdown Asterisk:			0 sec.
>
> reload config Freeswitch: 	1 sec.
> reload config Asterisk:		1 sec.
>
> Both were built from sources natively (no cross-compiling), and they  
> use the default startup configurations.
> I have managed to lower the Freeswitch times by disabling most of  
> the modules and recompiling, but it is still far away from Asterisk  
> (i.e. FS startup time 2.5 min).
>
> 1. Is there any way to further improve Freeswitch performance for  
> the ARM architecture?
> 2. Can this be related to the lack of a FPU (the Sheevalug emulates  
> the floating point operations).
> 3. On the startup I see this error repeated many times: [ERR]  
> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be  
> related?
>
> Thanks,
> Rogelio Perez
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jim at evolutiontel.net  Sun Aug 23 21:49:01 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 24 Aug 2009 14:49:01 +1000
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <9B65BFD4-C93F-4E26-82F1-AF8D687DCF52@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
	<CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>
	<10128ef10908170706p414098faq5d9d8024ba0cb3ef@mail.gmail.com>
	<9B65BFD4-C93F-4E26-82F1-AF8D687DCF52@freeswitch.org>
Message-ID: <ddad80610908232149k39ac88d6qee647b78f1d19078@mail.gmail.com>

"Well if you append ;transport=tcp on the bridge lines it will use TCP"

IMHO this statement needs some clarification based on the context of
this thread.

If the destination is another PBX or Freeswitch box then this is ok as
FS will be initiating the TCP connection.

For terminating calls to registered User Agents (UA) the decision to
use TCP or UDP should be made using information collected when the UA
registered.  i.e if the UA registers using TCP, FS should use TCP to
send and receive messages, if the UA registers with TLS, then FS
should use TLS, same story for UDP.

On Tue, Aug 18, 2009 at 12:10 AM, Brian West<brian at freeswitch.org> wrote:
> Well if you append ;transport=tcp on the bridge lines it will use TCP .
>
> /b
>
> On Aug 17, 2009, at 9:06 AM, Tzury Bar Yochay wrote:
>
>>> FreeSWITCH works very well as a client :P
>> I am currently porting it into iPhone and Symbian, I am almost
>> done ;-)
>>
>> anyway, seriously now, can one point to a wiki page about this?
>> How do I do that?
>> I would need 3 server instances to place a call, right?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From brian at freeswitch.org  Sun Aug 23 21:52:51 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 23:52:51 -0500
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <9D12BA51-0C76-45B1-A8DF-3A1FB74E005B@gmail.com>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
	<191c3a030908211320s6feb5f10sf8670575069ec95a@mail.gmail.com>
	<9D12BA51-0C76-45B1-A8DF-3A1FB74E005B@gmail.com>
Message-ID: <2439A4F0-2969-4C9F-ACBC-B56788B15410@freeswitch.org>

freeswitch -nonat

/b

On Aug 23, 2009, at 11:45 PM, Rogelio Perez wrote:

> 2009-08-24 04:39:29.910459 [ERR] switch_nat.c:183 Error checking for  
> PMP [general error]
> 2009-08-24 04:39:29.910694 [DEBUG] switch_nat.c:397 Checking for UPnP
> 2009-08-24 04:39:41.906029 [INFO] switch_nat.c:411 No PMP or UPnP  
> NAT detected!



From brian at freeswitch.org  Sun Aug 23 21:53:39 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 23:53:39 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <ddad80610908232149k39ac88d6qee647b78f1d19078@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
	<CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>
	<10128ef10908170706p414098faq5d9d8024ba0cb3ef@mail.gmail.com>
	<9B65BFD4-C93F-4E26-82F1-AF8D687DCF52@freeswitch.org>
	<ddad80610908232149k39ac88d6qee647b78f1d19078@mail.gmail.com>
Message-ID: <8A373389-2CE4-4606-8073-1D763BA2C644@freeswitch.org>

It already does exactly this.

/b

On Aug 23, 2009, at 11:49 PM, Jim Burke wrote:

> For terminating calls to registered User Agents (UA) the decision to
> use TCP or UDP should be made using information collected when the UA
> registered.  i.e if the UA registers using TCP, FS should use TCP to
> send and receive messages, if the UA registers with TLS, then FS
> should use TLS, same story for UDP.



From rogelio.perez at gmail.com  Sun Aug 23 22:03:40 2009
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Mon, 24 Aug 2009 02:03:40 -0300
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <2439A4F0-2969-4C9F-ACBC-B56788B15410@freeswitch.org>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
	<191c3a030908211320s6feb5f10sf8670575069ec95a@mail.gmail.com>
	<9D12BA51-0C76-45B1-A8DF-3A1FB74E005B@gmail.com>
	<2439A4F0-2969-4C9F-ACBC-B56788B15410@freeswitch.org>
Message-ID: <CC428124-7379-4C78-8122-88ED5573C464@gmail.com>

Thanks Brian, now the startup time is 3 sec.

On Aug 24, 2009, at 1:52 AM, Brian West wrote:

> freeswitch -nonat
>
> /b
>
> On Aug 23, 2009, at 11:45 PM, Rogelio Perez wrote:
>
>> 2009-08-24 04:39:29.910459 [ERR] switch_nat.c:183 Error checking for
>> PMP [general error]
>> 2009-08-24 04:39:29.910694 [DEBUG] switch_nat.c:397 Checking for UPnP
>> 2009-08-24 04:39:41.906029 [INFO] switch_nat.c:411 No PMP or UPnP
>> NAT detected!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From jim at evolutiontel.net  Sun Aug 23 22:05:42 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 24 Aug 2009 15:05:42 +1000
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <8A373389-2CE4-4606-8073-1D763BA2C644@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
	<CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>
	<10128ef10908170706p414098faq5d9d8024ba0cb3ef@mail.gmail.com>
	<9B65BFD4-C93F-4E26-82F1-AF8D687DCF52@freeswitch.org>
	<ddad80610908232149k39ac88d6qee647b78f1d19078@mail.gmail.com>
	<8A373389-2CE4-4606-8073-1D763BA2C644@freeswitch.org>
Message-ID: <ddad80610908232205h3d720c1ai35a68c749cde9cfa@mail.gmail.com>

I always expected it did :)

My point was that you cannot put transport=TCP on a bridge statement
line to an internal registered client and expect it to use a protocol
that was not used at registration.

Hence the clarification based on the context of the thread :)

On Mon, Aug 24, 2009 at 2:53 PM, Brian West<brian at freeswitch.org> wrote:
> It already does exactly this.
>
> /b
>
> On Aug 23, 2009, at 11:49 PM, Jim Burke wrote:
>
>> For terminating calls to registered User Agents (UA) the decision to
>> use TCP or UDP should be made using information collected when the UA
>> registered. ?i.e if the UA registers using TCP, FS should use TCP to
>> send and receive messages, if the UA registers with TLS, then FS
>> should use TLS, same story for UDP.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From help at pdscc.com  Sun Aug 23 22:49:03 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 22:49:03 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090824032200.B213D79C@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<CD6A6C1A-9CEC-4663-8C6C-C7A5237BBC30@freeswitch.org>,
	<20090824032200.B213D79C@sinclaire.sibble.net>
Message-ID: <20090824054901.A614079C@sinclaire.sibble.net>



On 23 Aug 2009 at 20:22, Harondel J. Sibble wrote:

> Whoah.....
> I get audio now, but it's running really slooooooooooooooowly. I'd say about
> 1/4 to 1/8 normal speech speed

Hmmm, using one of my hardphones, specifically the Integrated Networks IN-
1002

2009-08-23 20:46:07.334596 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1004 at 10.12.13.45 [a7c10746-9060-11de-8c5d-d333d780ffc7]
2009-08-23 20:46:07.348834 [INFO] mod_dialplan_xml.c:315 Processing 1004-
>5000 in context default
2009-08-23 20:46:07.364484 [NOTICE] mod_dptools.c:649 Channel 
[sofia/internal/1004 at 10.12.13.45] has been answered
2009-08-23 20:46:08.33173 [WARNING] mod_sofia.c:810 We were told to use ptime 
20 but what they meant to say was 10
This issue has so far been identified to happen on the following broken 
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who 
knows what will happen..
2009-08-23 20:47:30.512970 [NOTICE] switch_core_state_machine.c:179 Hangup 
sofia/internal/1004 at 10.12.13.45 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 20:47:30.541157 [NOTICE] switch_core_session.c:1086 Session 20 
(sofia/internal/1004 at 10.12.13.45) Ended
2009-08-23 20:47:30.541157 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/1004 at 10.12.13.45 [CS_DESTROY]

gets me much better results and no slowdown of audio, although there are some 
drop outs.  Running Ekiga on xp machine with zfone client works okay, but 
being it's an older verison of zfone, I'm guessing it doesn't support the 
enrollment packet from FS as the voice says I have to choose to enroll and 
then says goodbye ;-)

>From looking through the list archives, I am guessing I need to talk to the 
Tivi developers, question is what should I say to them, there's nothing 
really exposed in the client, other than adjusting the codecs, 3 choices
GSM, uLaw and aLaw can be enabled or disabled.

Both phones are running over a wifi link and there is (randomly) a fair bit 
of interference affecting speeds.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From juanbackson at gmail.com  Sun Aug 23 23:00:43 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Mon, 24 Aug 2009 14:00:43 +0800
Subject: [Freeswitch-users] unable to download non-US sound files
Message-ID: <27c25bc40908232300h7ae3cbbbkcd5ce5bbddb885f2@mail.gmail.com>

Hi

I tried to download non-US sound files, but I am getting this error:

--21:50:09--  http://files.freeswitch.org/freeswitch-sounds-fr-1.0.10.tar.gz
Resolving files.freeswitch.org... 69.174.57.101
Connecting to files.freeswitch.org|69.174.57.101|:80... connected.
HTTP request sent, awaiting response... 404 Not Found
21:50:12 ERROR 404: Not Found.

The command I used is make sounds-fr-install

Any idea what's wrong?

Thanks,
JB
-------------- next part --------------
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From matt at venturevoip.com  Sun Aug 23 23:07:19 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Mon, 24 Aug 2009 18:07:19 +1200
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <5a8712120908232127n6cb2f0bdq7e05b3d7778f0e70@mail.gmail.com>
References: <4A91D07C.1080805@venturevoip.com>	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
	<5a8712120908232127n6cb2f0bdq7e05b3d7778f0e70@mail.gmail.com>
Message-ID: <4A922E17.2040401@venturevoip.com>

On 24/08/09 4:27 PM, Jo?o Mesquita wrote:
>
> I need to work a little bit more on that documentation as well.. I saw a
> few conflicts with the core documentation too. Will get there once I
> have some more time left.

Awesome man - thanks for these.

So, one more question - in the show calls (or show channels) api 
response, which header would I use to retrieve the response?

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)


From matt at venturevoip.com  Sun Aug 23 23:15:06 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Mon, 24 Aug 2009 18:15:06 +1200
Subject: [Freeswitch-users] unable to download non-US sound files
In-Reply-To: <27c25bc40908232300h7ae3cbbbkcd5ce5bbddb885f2@mail.gmail.com>
References: <27c25bc40908232300h7ae3cbbbkcd5ce5bbddb885f2@mail.gmail.com>
Message-ID: <4A922FEA.3020203@venturevoip.com>

On 24/08/09 6:00 PM, Juan Backson wrote:
> Hi
>
> I tried to download non-US sound files, but I am getting this error:
>
> --21:50:09-- http://files.freeswitch.org/freeswitch-sounds-fr-1.0.10.tar.gz
> Resolving files.freeswitch.org... 69.174.57.101
> Connecting to files.freeswitch.org
> <http://files.freeswitch.org>|69.174.57.101|:80... connected.
> HTTP request sent, awaiting response... 404 Not Found
> 21:50:12 ERROR 404: Not Found.
>
> The command I used is make sounds-fr-install

It appears there are only sounds for en-us and ru-RU in:

http://files.freeswitch.org/

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)


From bruce.mcalister at blueface.ie  Mon Aug 24 01:59:24 2009
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Mon, 24 Aug 2009 09:59:24 +0100
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>
	<4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>
Message-ID: <4A92566C.3030200@blueface.ie>

Hi Vladimir,

Did you get an update on this at all?

Thanks


vmorales wrote:
> Hi Michal,
> 
> Just checking in to see if you've been able to take a stab at this.
> 
> Thanks,
> Vladimir
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michal Bielicki
> Sent: Tuesday, August 11, 2009 5:33 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
> 
> I'll retst it later today and give you a link with instructions
> 
> Am 10.08.2009 um 20:14 schrieb vmorales:
> 
>> By "./compile" I was referring to "./configure"
>>
>> Vladimir
>>
>> -----Original Message-----
>> From: vmorales [mailto:email.list.subscriber at gmail.com]
>> Sent: Monday, August 10, 2009 11:49 AM
>> To: 'freeswitch-users at lists.freeswitch.org'
>> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> Thanks for the response(s):
>>
>> I ran the "./compile" script with a set PREFIX.  This took a few
>> attempts with errors before it was able to complete error-free, as I
>> had to install libtool.
>>
>> Since then, I have tried running 'make', 'gmake', and
>> '/opt/gnu/bin/make', but each results with an error.  This is the
>> error when running 'make' or 'gmake':
>>
>> <snip>
>> make: Fatal error: Command failed for target `all-recursive'
>> Current working directory /home/vmorales/freeswitch-1.0.4
>> *** Error code 1
>> make: Fatal error: Command failed for target `all'
>> </snip>
>>
>>
>> This is the error when running '/opt/gnu/bin/make':
>>
>> <snip>
>> make[5]: *** [mod_amr.so] Error 1
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_amr-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> Making all in build
>> +-------- FreeSWITCH Build Complete -----------+
>> + FreeSWITCH has been successfully built.      +
>> + Install by running:                          +
>> +                                              +
>> +               /opt/gnu/bin/make install                   +
>> +----------------------------------------------+
>> make[1]: *** [all-recursive] Error 1
>> make: *** [all] Error 2
>> </snip>
>>
>>
>> I re-untar'd before each compile/make attempt.  Let me know if this
> is
>> something that I can resolve.
>>
>> Vladimir
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Michael Jerris
>> Sent: Saturday, August 08, 2009 12:37 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> This is not currently a supported platform, it only builds on 64 bit
>> right now I think on solaris.
>>
>> Mike
>>
>> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>>
>>> Hello,
>>>
>>> Does anyone have, or know where to get, a pre-compiled copy of
>>> FreeSwitch for Solaris 10/x86?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>> http://www.freeswitch.org
> 
> Michal Bielicki
> Leiter der Niederlassung
> HaloKwadrat Sp. z o.o.
> Niederlassung Kleinmachnow
> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
> Ust.Id.: DE261885536
> Geschaeftsfuehrer: Aleksander Wiercinski
> Meiereifeld 2b, 14532 Kleinmachnow
> t. +49 33203 263220
> f. +49 33203 263229 sip. info at halokwadrat.de
> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
> Hauptgesch?ftsstelle:
> Halo Kwadrat Sp. z o.o.
> ul. Polna 46/14
> 00-644 Warszawa, Polen
> EIngetragen im HRB Warszawa, KRS 0000153539
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 


From gmaruzz at celliax.org  Mon Aug 24 03:03:17 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 24 Aug 2009 12:03:17 +0200
Subject: [Freeswitch-users] Screaming monkeys on ext 5000
In-Reply-To: <A7FF6EFA-5CBD-4419-BC0A-656B43B34686@freeswitch.org>
References: <1251084377.461.1331260223@webmail.messagingengine.com> 
	<A7FF6EFA-5CBD-4419-BC0A-656B43B34686@freeswitch.org>
Message-ID: <7b197bef0908240303r61d89baar351450e488dc51a3@mail.gmail.com>

On Mon, Aug 24, 2009 at 5:31 AM, Brian West<brian at freeswitch.org> wrote:
> It requires internet connectivity. ?It calls a remote system to play
> which is out of our control.
>

Yeah, I noted this too, since a couple weeks at least...

Maybe let's Todd know it's monkeys are out of voice?

-giovanni


From Prometheus001 at gmx.net  Mon Aug 24 04:25:16 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 24 Aug 2009 13:25:16 +0200
Subject: [Freeswitch-users] XML-RPC on different ip than 0.0.0.0
Message-ID: <4A92789C.5040900@gmx.net>

Hello,

is there any chance to limit the listening ips of the xml-rpc server
(which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)?

Best regards
Peter


From asannucci at gmail.com  Mon Aug 24 06:15:49 2009
From: asannucci at gmail.com (bakko)
Date: Mon, 24 Aug 2009 15:15:49 +0200
Subject: [Freeswitch-users] zrtp endpoints have different sas through
	fs1.0.4
In-Reply-To: <09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>,
	<20090823231610.3398273E@sinclaire.sibble.net><20090823235030.A95BF5FE@sinclaire.sibble.net>
	<09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org>
Message-ID: <7541BAF571934E12B19428744AE5FB96@voztovoice>

Me too :)

----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, August 24, 2009 1:53 AM
Subject: Re: [Freeswitch-users] zrtp endpoints have different sas through 
fs1.0.4


> Wish they would send me one for my E63 for testing... only been
> working with zfone 3 so far.
>
> /b
>
> On Aug 23, 2009, at 6:50 PM, Harondel J. Sibble wrote:
>
>> Well good news for the Tiviphone client
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 



From m.krivushin at imarto.net  Sun Aug 23 22:57:19 2009
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Mon, 24 Aug 2009 12:57:19 +0700
Subject: [Freeswitch-users] CHANNEL_CREATE and full information
Message-ID: <5be734a50908232257s1950f82esf8d26cc6c12441be@mail.gmail.com>

I put some variable in originate command for identify created channel for my
app using event socket.

Is any ability to get all channel variables without uuid_show on
channel_create? Can I ask FS to take me all information on channel creating?
Can this be in trunk, if it cannt be get without source editing? May be we
can add custom event - CHANNEL_CREATE_FULL_INFO?

I just think, that is smarty to push channel information for controller in
channel create. I can call api uuid_call, but in some situations, why I call
api, channel destroyed. I can use bgapi result, BACKGROUND_JOB, but not in
all cases.

Sometimes we have situation when call initiated not from my caller app. In
this case we *cant* get information about destroyed channel.

Please, point me to right side - may be I need to write something module?
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From jmesquita at freeswitch.org  Mon Aug 24 00:10:33 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 24 Aug 2009 04:10:33 -0300
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <4A922E17.2040401@venturevoip.com>
References: <4A91D07C.1080805@venturevoip.com>
	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
	<5a8712120908232127n6cb2f0bdq7e05b3d7778f0e70@mail.gmail.com>
	<4A922E17.2040401@venturevoip.com>
Message-ID: <ebbc2d550908240010i17ac47b8r44a2465cab2f673@mail.gmail.com>

For api responses you have to use body. I should have reminded that as
well, sorry...

Jmesquita

On 8/24/09, Matt Riddell <matt at venturevoip.com> wrote:
> On 24/08/09 4:27 PM, Jo?o Mesquita wrote:
>>
>> I need to work a little bit more on that documentation as well.. I saw a
>> few conflicts with the core documentation too. Will get there once I
>> have some more time left.
>
> Awesome man - thanks for these.
>
> So, one more question - in the show calls (or show channels) api
> response, which header would I use to retrieve the response?
>
> --
> Cheers,
>
> Matt Riddell
> Director
> _______________________________________________
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device


From brian at freeswitch.org  Mon Aug 24 07:05:26 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 09:05:26 -0500
Subject: [Freeswitch-users] unable to download non-US sound files
In-Reply-To: <27c25bc40908232300h7ae3cbbbkcd5ce5bbddb885f2@mail.gmail.com>
References: <27c25bc40908232300h7ae3cbbbkcd5ce5bbddb885f2@mail.gmail.com>
Message-ID: <C426BDB1-7BB8-4FAE-BD9F-661D2F8A8849@freeswitch.org>

No french files exist yet. ;)

/b

On Aug 24, 2009, at 1:00 AM, Juan Backson wrote:

> Any idea what's wrong?



From anthony.minessale at gmail.com  Mon Aug 24 07:05:31 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 09:05:31 -0500
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <4A922E17.2040401@venturevoip.com>
References: <4A91D07C.1080805@venturevoip.com>
	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
	<5a8712120908232127n6cb2f0bdq7e05b3d7778f0e70@mail.gmail.com>
	<4A922E17.2040401@venturevoip.com>
Message-ID: <191c3a030908240705m5003a26bpd97b2c041d7b1526@mail.gmail.com>

I updated testclient.c so you can see how now.

#include <stdio.h>
#include <stdlib.h>
#include <esl.h>


int main(void)
{
    esl_handle_t handle = {{0}};

    esl_connect(&handle, "localhost", 8021, "ClueCon");

    esl_send_recv(&handle, "api status\n\n");

    if (handle.last_sr_event && handle.last_sr_event->body) {
        printf("%s\n", handle.last_sr_event->body);
    } else {
        // this is unlikely to happen with api or bgapi (which is hardcoded
above) but prefix but may be true for other commands
        printf("%s\n", handle.last_sr_reply);
    }

    esl_disconnect(&handle);

    return 0;
}


On Mon, Aug 24, 2009 at 1:07 AM, Matt Riddell <matt at venturevoip.com> wrote:

> On 24/08/09 4:27 PM, Jo?o Mesquita wrote:
> >
> > I need to work a little bit more on that documentation as well.. I saw a
> > few conflicts with the core documentation too. Will get there once I
> > have some more time left.
>
> Awesome man - thanks for these.
>
> So, one more question - in the show calls (or show channels) api
> response, which header would I use to retrieve the response?
>
> --
> Cheers,
>
> Matt Riddell
> Director
> _______________________________________________
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Aug 24 07:07:10 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 09:07:10 -0500
Subject: [Freeswitch-users] CHANNEL_CREATE and full information
In-Reply-To: <5be734a50908232257s1950f82esf8d26cc6c12441be@mail.gmail.com>
References: <5be734a50908232257s1950f82esf8d26cc6c12441be@mail.gmail.com>
Message-ID: <191c3a030908240707o2a8a534axf1ac789dd65d25a2@mail.gmail.com>

try looking for the channel_originate events which is fired as soon as any
outbound call is mature enough to have variables.


On Mon, Aug 24, 2009 at 12:57 AM, Mikhail Krivushin
<m.krivushin at imarto.net>wrote:

> I put some variable in originate command for identify created channel for
> my app using event socket.
>
> Is any ability to get all channel variables without uuid_show on
> channel_create? Can I ask FS to take me all information on channel creating?
> Can this be in trunk, if it cannt be get without source editing? May be we
> can add custom event - CHANNEL_CREATE_FULL_INFO?
>
> I just think, that is smarty to push channel information for controller in
> channel create. I can call api uuid_call, but in some situations, why I call
> api, channel destroyed. I can use bgapi result, BACKGROUND_JOB, but not in
> all cases.
>
> Sometimes we have situation when call initiated not from my caller app. In
> this case we *cant* get information about destroyed channel.
>
> Please, point me to right side - may be I need to write something module?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From juanbackson at gmail.com  Mon Aug 24 07:25:01 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Mon, 24 Aug 2009 22:25:01 +0800
Subject: [Freeswitch-users] fifo question
Message-ID: <27c25bc40908240725k5e0c92a2u2def5bd679b7b666@mail.gmail.com>

Hi,

Does anyone know the purpose of fifo_orbit_announce?

When does fifo_orbit_announce get played?

Thanks,
JB
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From houndd at ymail.com  Mon Aug 24 07:45:29 2009
From: houndd at ymail.com (Hound Dog)
Date: Mon, 24 Aug 2009 07:45:29 -0700 (PDT)
Subject: [Freeswitch-users] topology hiding leaking information in SDP data
Message-ID: <510213.5618.qm@web44709.mail.sp1.yahoo.com>

carriers need topology hiding , its an important feature for both security and also to hide you business partners from each other 

freeSwitch talks about it and also does a good job in hiding the signalling topology 

there is however a hole in the SDP manipulation that I am trying to plug and would love to get some help , obviously once resolved I am also happy to add documentation for all to use 


I was thinking that the best way would be to build the SDP message from scratch based on the incoming info , and maybe in special cases have the SDP copied over from the original message. 

is there a way to have FS build a clean SDP message ?





see and example of Bria softphone making a call via freeswitch , note that the SDP to leg B contains original addresses and even internal ones 


incoming Invite message 
-------------------
INVITE sip:442078562101 at pbx.rilcomm.com SIP/2.0
Via: SIP/2.0/UDP 82.80.130.222:26762;branch=z9hG4bK-d8754z-8fba35a96b6d949f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:44558678567378 at 82.80.130.222:26762>
To: <sip:442078562101 at pbx.rilcomm.com>
From: <sip:44558678567378 at pbx.rilcomm.com>;tag=47360c64
Call-ID: YTY4NjMwMjg4MWRmODY5NDlhOWQ4MDg5MWIwN2Y3MTY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: Bria release 2.5.4 stamp 53956
Content-Length: 325

v=0
o=- 0 2 IN IP4 82.80.130.222
s=CounterPath Bria
c=IN IP4 82.80.130.222
t=0 0
m=audio 27848 RTP/AVP 18 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 2 : XlhJjZhm LLmd1Kyz 192.168.1.50 24514
a=alt:2 1 : GOsgLipv VjRq6zYk 192.168.1.60 24514

-------------------


message to leg B see the original IP addresses in the SDP  ( anything that is not 81.89.136.231)

note 82.80.130.222 which is the original address being visible in the SDP fields

-------------------
INVITE sip:1001 at 82.80.130.222:20014 SIP/2.0
Via: SIP/2.0/UDP 81.89.136.231:5080;rport;branch=z9hG4bKBBgvK7Sap23tN
Max-Forwards: 67
From: "44558678567378" <sip:44558678567378 at 81.89.136.231>;tag=1FUrD0t1gF55a
To: <sip:1001 at 82.80.130.222:20014>
Call-ID: 8eed1cbc-9030-11de-87e7-27e3a7b5c9da
CSeq: 119414118 INVITE
Contact: <sip:mod_sofia at 81.89.136.231:5080>
User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 331
Remote-Party-ID: "44558678567378" <sip:44558678567378 at 81.89.136.231>;party=calling;screen=yes;privacy=off

v=0
o=- 2301954626585387485 2 IN IP4 82.80.130.222
s=CounterPath Bria
c=IN IP4 81.89.136.231
t=0 0
m=audio 25974 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 2 : XlhJjZhm LLmd1Kyz 192.168.1.50 24514
a=alt:2 1 : GOsgLipv VjRq6zYk 192.168.1.60 24514

-------------------



thank you
Ori


      
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From brian at freeswitch.org  Mon Aug 24 07:53:15 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 09:53:15 -0500
Subject: [Freeswitch-users] topology hiding leaking information in SDP
	data
In-Reply-To: <510213.5618.qm@web44709.mail.sp1.yahoo.com>
References: <510213.5618.qm@web44709.mail.sp1.yahoo.com>
Message-ID: <6D897008-2E65-4314-B890-18264A615EBC@freeswitch.org>

You're using Proxy Media and the only clean way to do this is not use  
proxy media that way a complete clean SDP will be generated for the B- 
Leg.

/b

On Aug 24, 2009, at 9:45 AM, Hound Dog wrote:

> is there a way to have FS build a clean SDP message ?

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From vhatz at kinetix.gr  Mon Aug 24 08:28:05 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Mon, 24 Aug 2009 18:28:05 +0300
Subject: [Freeswitch-users] topology hiding leaking information in SDP
 data
In-Reply-To: <6D897008-2E65-4314-B890-18264A615EBC@freeswitch.org>
References: <510213.5618.qm@web44709.mail.sp1.yahoo.com>
	<6D897008-2E65-4314-B890-18264A615EBC@freeswitch.org>
Message-ID: <4A92B185.7030401@kinetix.gr>

But that would allow tell FS to also do transcoding in some cases, correct?

Is there a way to avoid transcoding and still build the required SDP in 
a clean manner?


Best regards,
Vlasis Hatzistavrou.

Brian West wrote:
> You're using Proxy Media and the only clean way to do this is not use 
> proxy media that way a complete clean SDP will be generated for the B-Leg.
> 
> /b
> 
> On Aug 24, 2009, at 9:45 AM, Hound Dog wrote:
> 
>> is there a way to have FS build a clean SDP message ?
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


From brian at freeswitch.org  Mon Aug 24 08:37:57 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 10:37:57 -0500
Subject: [Freeswitch-users] topology hiding leaking information in SDP
	data
In-Reply-To: <4A92B185.7030401@kinetix.gr>
References: <510213.5618.qm@web44709.mail.sp1.yahoo.com>
	<6D897008-2E65-4314-B890-18264A615EBC@freeswitch.org>
	<4A92B185.7030401@kinetix.gr>
Message-ID: <057646D2-5A8D-4877-8C2B-E72D2D902446@freeswitch.org>

You can disable transcoding on the sofia profiles... see defaults.

/b

On Aug 24, 2009, at 10:28 AM, Vlasis Hatzistavrou (KTI) wrote:

> But that would allow tell FS to also do transcoding in some cases,  
> correct?
>
> Is there a way to avoid transcoding and still build the required SDP  
> in
> a clean manner?
>
>
> Best regards,
> Vlasis Hatzistavrou.



From d_hound at ymail.com  Mon Aug 24 09:33:14 2009
From: d_hound at ymail.com (Hound Dog)
Date: Mon, 24 Aug 2009 09:33:14 -0700 (PDT)
Subject: [Freeswitch-users]  topology hiding leaking information in SDP
Message-ID: <508126.46278.qm@web111920.mail.gq1.yahoo.com>

Brian,

I set proxy_media to false ad now getting clean SDP without all the incoming data

thank you 



this also means that the packets have to go all the way in and out of the codec chain  ( instead of being taken in and thrown out with no processing )   , is that the case ?

what is the performance hit of such change , on a modern strong machine - what is the estimated streams that FS can take ?



BTW - sorry for not answering the list as I should have , I think I registered wrong somewhere so now have a new user 


      
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From anthony.minessale at gmail.com  Mon Aug 24 09:51:29 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 11:51:29 -0500
Subject: [Freeswitch-users] XML-RPC on different ip than 0.0.0.0
In-Reply-To: <4A92789C.5040900@gmx.net>
References: <4A92789C.5040900@gmx.net>
Message-ID: <191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>

there is a configuration option in the xml file to control which ip it binds
to.



On Mon, Aug 24, 2009 at 6:25 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> is there any chance to limit the listening ips of the xml-rpc server
> (which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)?
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Aug 24 09:58:47 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 11:58:47 -0500
Subject: [Freeswitch-users] topology hiding leaking information in SDP
In-Reply-To: <508126.46278.qm@web111920.mail.gq1.yahoo.com>
References: <508126.46278.qm@web111920.mail.gq1.yahoo.com>
Message-ID: <191c3a030908240958q45f44fd6jb2eeaa6e6a95bbea@mail.gmail.com>

if you establish an audio path that is using the same codec there is no
difference in performance as the codecs
do not do anything when both sides are the same.


On Mon, Aug 24, 2009 at 11:33 AM, Hound Dog <d_hound at ymail.com> wrote:

> Brian,
>
> I set proxy_media to false ad now getting clean SDP without all the
> incoming data
>
> thank you
>
>
>
> this also means that the packets have to go all the way in and out of the
> codec chain  ( instead of being taken in and thrown out with no processing )
>   , is that the case ?
>
> what is the performance hit of such change , on a modern strong machine -
> what is the estimated streams that FS can take ?
>
>
>
> BTW - sorry for not answering the list as I should have , I think I
> registered wrong somewhere so now have a new user
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Mon Aug 24 10:19:01 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 24 Aug 2009 19:19:01 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK
	message
In-Reply-To: <65d96fc80908241006p7b5aef46n1475a1fc0710ca4@mail.gmail.com>
References: <65d96fc80908241006p7b5aef46n1475a1fc0710ca4@mail.gmail.com>
Message-ID: <65d96fc80908241019p365b563eu4507913d3fcb7ecd@mail.gmail.com>

Hello,

I've been with freeswittch for a while now.. and i can say it is worth
developing it.

anyhow i got into a strange issue... I'm tryng to see what load FS on my
server can take. The Call flow is like this:

SIPp                   FS

INVITE -------->
           <------- 100 Trying
           <------- 302 Moved Temporary
ACK    --------->



I use a dummy dialplan for that. All custom functions i've build are
disabled and i'm not using it here. Also custom modules are not loaded as
well.


   <extension name="ServiceLookup">
      <condition field="destination_number" expression="(^300030)(.*)">
         <!--action application="lookup_service_destination" data="in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, i
n $1, in pgw01.ot.hr:5060, out red_contact, out authResult"/-->
         <action application="log" data="INFO ########################
ServiceLookup ########################\n"/>
         <action application="log" data="INFO ########################
contact = '${red_contact}' ##############\n"/>
         <action application="log" data="INFO ########################
CallerNum = '${caller_id_number:6:16}' ##########\n"/>
         <action application="log" data="INFO ########################
RADIUS auth = '${authResult}' ##########\n"/>
         <action application="execute_extension" data="doRedirect XML
public"/>
        </condition>
   </extension>


   <extension name="doRedirect">
      <condition field="destination_number" expression="^doRedirect$"/>
      <condition field="${authResult}" expression="^0$|^60$">
         <action application="log" data="INFO ########################
RADIUS auth OK!!!' ##########\n"/>
         <!--action application="redirect" data="sip:${red_contact}"/-->
         <!--action application="answer"/-->
         <action application="redirect" data="
sip:12345616094191500 at pgw01.ot.hr:5060"/>
         <!--anti-action application="answer"/-->
         <!--anti-action application="sleep" data="2000"/-->
         <action application="hangup" data="USER_BUSY"/>
         <anti-action application="redirect" data="
sip:12345616094191500 at pgw01.ot.hr:5060"/>
         <anti-action application="log" data="INFO ########################
RADIUS auth NOK!! ##########\n"/>
         <!--anti-action application="respond" data="403 Forbidden"/-->
         <anti-action application="hangup" data="USER_BUSY"/>
      </condition>
   </extension>


When i place a call from x-lite everything works fine ... x-lite sends an
invite, gets SIP 302 and ACKs it correctly... FS is happy.

When i place a call from SIPp i have the same scenario except FS seems not
understand ACK message from SIPp and re-sends SIP 302 multiple times untill
it gives up.


I beleive this is due to 302 resend issue but; when i load FS with 100 CPS,
i can see high CPU usage (just one thread taking most load... the rest does
almost nothing) on FS. Also, starting from 40 CPS there is a big delay in
receiving SIP 302 messages meaning i've sent 6000 calls and so far only for
half of them got 302 response.


Does anybody have a clue ?





Here is a trace taken on FS for calls originated from SIPp (sipp -sn uac
10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100
-trace_msg -inf test.txt -m 1 -l 4000):

freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
16:44:26.527236:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Max-Forwards: 70
   Subject: Performance Test
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 ACK
   Contact: sip:sipp at 10.4.4.252:5060
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0


Tihomir.
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freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 16:44:26.527236:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Max-Forwards: 70
   Subject: Performance Test
   Content-Type: application/sdp
   Content-Length:   131
   
   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-08-24 18:44:26.525364 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22222238515000403 at 10.4.4.251 [62a395f8-90cd-11de-99d5-cb0c6bd8522b]
2009-08-24 18:44:26.533096 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->30003016094191500 in context public
2009-08-24 18:44:26.533096 [INFO] mod_dptools.c:932 ######################## ServiceLookup ########################\n
2009-08-24 18:44:26.533096 [INFO] mod_dptools.c:932 ######################## contact = '' ##############\n
2009-08-24 18:44:26.533096 [INFO] mod_dptools.c:932 ######################## CallerNum = '38515000403' ##########\n
2009-08-24 18:44:26.533096 [INFO] mod_dptools.c:932 ######################## RADIUS auth = '' ##########\n
2009-08-24 18:44:26.533096 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->doRedirect in context public
2009-08-24 18:44:26.533096 [NOTICE] switch_core_session.c:1576 Execute redirect(sip:12345616094191500 at pgw01.ot.hr:5060)
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 ACK
   Contact: sip:sipp at 10.4.4.252:5060
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-08-24 18:44:26.533096 [NOTICE] sofia.c:3863 Hangup sofia/internal/22222238515000403 at 10.4.4.251 [CS_EXECUTE] [NORMAL_UNSPECIFIED]
2009-08-24 18:44:26.537029 [NOTICE] switch_core_session.c:1086 Session 3 (sofia/internal/22222238515000403 at 10.4.4.251) Ended
2009-08-24 18:44:26.537029 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22222238515000403 at 10.4.4.251 [CS_DESTROY]
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:30.037065:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:34.037063:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:38.037068:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:42.037064:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:46.037064:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:50.037066:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:54.037067:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:58.037067:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
-------------- next part --------------
freeswitch at l01sipindir1> 
freeswitch at l01sipindir1> 
freeswitch at l01sipindir1> recv 855 bytes from udp/[10.1.14.30]:27454 at 16:42:56.522659:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251 SIP/2.0
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-b9056b2ac55b4e54-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.1.14.30:27454>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 1 INVITE
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
   Content-Type: application/sdp
   User-Agent: X-Lite release 1103k stamp 53621
   Content-Length: 258
   
   v=0
   o=- 0 2 IN IP4 10.1.14.30
   s=CounterPath X-Lite 3.0
   c=IN IP4 10.1.14.30
   t=0 0
   m=audio 44488 RTP/AVP 107 0 8 101
   a=alt:1 1 : 5KgV8ECY o9MWrE7h 10.1.14.30 44488
   a=fmtp:101 0-15
   a=rtpmap:107 BV32/16000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   ------------------------------------------------------------------------
send 396 bytes to udp/[10.1.14.30]:27454 at 16:42:56.523076:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-b9056b2ac55b4e54-1---d8754z-;rport=27454
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-08-24 18:42:56.521041 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22222238515000403 at 10.4.4.251 [2cfdfdbc-90cd-11de-99d5-cb0c6bd8522b]
2009-08-24 18:42:56.529053 [CONSOLE] sofia_presence.c:680 Event Thread Started
2009-08-24 18:42:56.529053 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->30003016094191500 in context public
2009-08-24 18:42:56.529053 [INFO] mod_dptools.c:932 ######################## ServiceLookup ########################\n
2009-08-24 18:42:56.529053 [INFO] mod_dptools.c:932 ######################## contact = '' ##############\n
2009-08-24 18:42:56.529053 [INFO] mod_dptools.c:932 ######################## CallerNum = '38515000403' ##########\n
2009-08-24 18:42:56.529053 [INFO] mod_dptools.c:932 ######################## RADIUS auth = '' ##########\n
2009-08-24 18:42:56.529053 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->doRedirect in context public
2009-08-24 18:42:56.529053 [NOTICE] switch_core_session.c:1576 Execute redirect(sip:12345616094191500 at pgw01.ot.hr:5060)
send 790 bytes to udp/[10.1.14.30]:27454 at 16:42:56.531535:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-b9056b2ac55b4e54-1---d8754z-;rport=27454
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=F6H3DQS7g78ra
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces2009-08-24 18:42:56.529053 [NOTICE] switch_core_session.c:1576 Execute log(INFO ######################## RADIUS auth NOK!! ##########\n)

   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-08-24 18:42:56.529053 [NOTICE] sofia.c:3863 Hangup sofia/internal/22222238515000403 at 10.4.4.251 [CS_EXECUTE] [NORMAL_UNSPECIFIED]
2009-08-24 18:42:56.529053 [INFO] mod_dptools.c:932 ######################## RADIUS auth NOK!! ##########\n
2009-08-24 18:42:56.529053 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/internal/22222238515000403 at 10.4.4.251) Ended
2009-08-24 18:42:56.529053 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22222238515000403 at 10.4.4.251 [CS_DESTROY]
recv 379 bytes from udp/[10.1.14.30]:27454 at 16:42:56.533966:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251 SIP/2.0
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-b9056b2ac55b4e54-1---d8754z-;rport
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=F6H3DQS7g78ra
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 1 ACK
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 861 bytes from udp/[10.1.14.30]:27454 at 16:42:56.638930:
   ------------------------------------------------------------------------
   INVITE sip:12345616094191500 at pgw01.ot.hr:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-213d2e12ad57b353-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.1.14.30:27454>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 2 INVITE
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
   Content-Type: application/sdp
   User-Agent: X-Lite release 1103k stamp 53621
   Content-Length: 258
   
   v=0
   o=- 0 2 IN IP4 10.1.14.30
   s=CounterPath X-Lite 3.0
   c=IN IP4 10.1.14.30
   t=0 0
   m=audio 44488 RTP/AVP 107 0 8 101
   a=alt:1 1 : 5KgV8ECY o9MWrE7h 10.1.14.30 44488
   a=fmtp:101 0-15
   a=rtpmap:107 BV32/16000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   ------------------------------------------------------------------------
send 396 bytes to udp/[10.1.14.30]:27454 at 16:42:56.639220:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-213d2e12ad57b353-1---d8754z-;rport=27454
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 2 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-08-24 18:42:56.637047 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22222238515000403 at 10.4.4.251 [2d0fb3fe-90cd-11de-99d5-cb0c6bd8522b]
2009-08-24 18:42:56.641071 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->12345616094191500 in context public
2009-08-24 18:42:56.641071 [INFO] switch_core_state_machine.c:136 No Route, Aborting
2009-08-24 18:42:56.641071 [NOTICE] switch_core_state_machine.c:137 Hangup sofia/internal/22222238515000403 at 10.4.4.251 [CS_ROUTING] [NO_ROUTE_DESTINATION]
send 782 bytes to udp/[10.1.14.30]:27454 at 16:42:56.642591:
   ------------------------------------------------------------------------
   SIP/2.0 404 Not Found
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-213d2e12ad57b353-1---d8754z-;rport=27454
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=gFBvFjaBegZBp
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 2 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Reason: Q.850;cause=3;text="NO_ROUTE_DESTINATION"
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-08-24 18:42:56.641071 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/internal/22222238515000403 at 10.4.4.251) Ended
2009-08-24 18:42:56.641071 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22222238515000403 at 10.4.4.251 [CS_DESTROY]
recv 385 bytes from udp/[10.1.14.30]:27454 at 16:42:56.646734:
   ------------------------------------------------------------------------
   ACK sip:12345616094191500 at pgw01.ot.hr:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-213d2e12ad57b353-1---d8754z-;rport
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=gFBvFjaBegZBp
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 2 ACK
   Content-Length: 0
   
   ------------------------------------------------------------------------
-------------- next part --------------
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-------------- next part --------------
SEQUENTIAL
22222238515000403;22222238515000403

From anthony.minessale at gmail.com  Mon Aug 24 10:32:27 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 12:32:27 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241019p365b563eu4507913d3fcb7ecd@mail.gmail.com>
References: <65d96fc80908241006p7b5aef46n1475a1fc0710ca4@mail.gmail.com>
	<65d96fc80908241019p365b563eu4507913d3fcb7ecd@mail.gmail.com>
Message-ID: <191c3a030908241032i7d8eef27v5baf67b909d01e95@mail.gmail.com>

Your ACK message must not be valid (dialog matching or something else)
so every 1 call will generate 30 retries that are queued up in the sip
stack.

at 100cps you will be generating this problem 100 times per second and queue
up countless unfinished dialogs thus
eating up the cpu.






On Mon, Aug 24, 2009 at 12:19 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> Hello,
>
> I've been with freeswittch for a while now.. and i can say it is worth
> developing it.
>
> anyhow i got into a strange issue... I'm tryng to see what load FS on my
> server can take. The Call flow is like this:
>
> SIPp                   FS
>
> INVITE -------->
>            <------- 100 Trying
>            <------- 302 Moved Temporary
> ACK    --------->
>
>
>
> I use a dummy dialplan for that. All custom functions i've build are
> disabled and i'm not using it here. Also custom modules are not loaded as
> well.
>
>
>    <extension name="ServiceLookup">
>       <condition field="destination_number" expression="(^300030)(.*)">
>          <!--action application="lookup_service_destination" data="in
> ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, i
> n $1, in pgw01.ot.hr:5060, out red_contact, out authResult"/-->
>          <action application="log" data="INFO ########################
> ServiceLookup ########################\n"/>
>          <action application="log" data="INFO ########################
> contact = '${red_contact}' ##############\n"/>
>          <action application="log" data="INFO ########################
> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
>          <action application="log" data="INFO ########################
> RADIUS auth = '${authResult}' ##########\n"/>
>          <action application="execute_extension" data="doRedirect XML
> public"/>
>         </condition>
>    </extension>
>
>
>    <extension name="doRedirect">
>       <condition field="destination_number" expression="^doRedirect$"/>
>       <condition field="${authResult}" expression="^0$|^60$">
>          <action application="log" data="INFO ########################
> RADIUS auth OK!!!' ##########\n"/>
>          <!--action application="redirect" data="sip:${red_contact}"/-->
>          <!--action application="answer"/-->
>          <action application="redirect" data="
> sip:12345616094191500 at pgw01.ot.hr:5060"/>
>          <!--anti-action application="answer"/-->
>          <!--anti-action application="sleep" data="2000"/-->
>          <action application="hangup" data="USER_BUSY"/>
>          <anti-action application="redirect" data="
> sip:12345616094191500 at pgw01.ot.hr:5060"/>
>          <anti-action application="log" data="INFO ########################
> RADIUS auth NOK!! ##########\n"/>
>          <!--anti-action application="respond" data="403 Forbidden"/-->
>          <anti-action application="hangup" data="USER_BUSY"/>
>       </condition>
>    </extension>
>
>
> When i place a call from x-lite everything works fine ... x-lite sends an
> invite, gets SIP 302 and ACKs it correctly... FS is happy.
>
> When i place a call from SIPp i have the same scenario except FS seems not
> understand ACK message from SIPp and re-sends SIP 302 multiple times untill
> it gives up.
>
>
> I beleive this is due to 302 resend issue but; when i load FS with 100 CPS,
> i can see high CPU usage (just one thread taking most load... the rest does
> almost nothing) on FS. Also, starting from 40 CPS there is a big delay in
> receiving SIP 302 messages meaning i've sent 6000 calls and so far only for
> half of them got 302 response.
>
>
> Does anybody have a clue ?
>
>
>
>
>
> Here is a trace taken on FS for calls originated from SIPp (sipp -sn uac
> 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100
> -trace_msg -inf test.txt -m 1 -l 4000):
>
> freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
> 16:44:26.527236:
>    ------------------------------------------------------------------------
>    INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
>    Max-Forwards: 70
>    Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
> >
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
> >
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
> >;tag=1
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Type: application/sdp
>    Content-Length:   131
>
>    v=0
>    o=user1 53655765 2353687637 IN IP4 10.4.4.252
>    s=-
>    c=IN IP4 10.4.4.252
>    t=0 0
>    m=audio 6000 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
>    ------------------------------------------------------------------------
> send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
> >;tag=1
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
> >
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
>    ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
> >;tag=1
>    To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
> >;tag=Hr4mHDUeBSNyH
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
>    ------------------------------------------------------------------------
>    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
> >
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
> >;tag=1
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 ACK
>    Contact: sip:sipp at 10.4.4.252:5060
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070:
>    ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
> >;tag=1
>    To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
> >;tag=Hr4mHDUeBSNyH
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063:
>    ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
> >;tag=1
>    To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
> >;tag=Hr4mHDUeBSNyH
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>
> Tihomir.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From raffaele.p.guidi at gmail.com  Mon Aug 24 11:24:28 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Mon, 24 Aug 2009 20:24:28 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
Message-ID: <b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>

Actually I did that and it worked fine. I had the problem the SECOND time I
run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were
not

On Sun, Aug 16, 2009 at 16:04, Carlos Talbot <carlos.talbot at gmail.com>wrote:

> When you configure FreePBX for the first time it wipes out the sip_profiles
> directory. If you follow the FreePBX shortcut on your desktop it'll mention
> this on the last screen of the configuration. You might see something such
> as the following below. If you plan to use FreePBX you'll need to define
> trunk groups, trunks, etc in order to have the sip_profiles directory
> populated.
> regards,
>
> Carlos
>
>
>  Incompatible Configuration WARNING: THE FOLLOWING FILES WILL BE DELETED!
>
>    - D:/FreeSWITCH/conf/sip_profiles/external.xml
>    - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
>    - D:/FreeSWITCH/conf/sip_profiles/internal.xml
>
>
> On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> I had the sweet surprise to find the installer packaged with FreePBX...
>> really great! Why it has not been advertised as it deserves? It worked like
>> a breeze once launched, with the automatic configuration and all of that.,
>> Only thing: once stopped I cannot get it to load sofia profiles anymore -
>> issueing sofia status doesn't show anything. I had to copy internal.xml and
>> default.xml from a previous installation and everything got to work again -
>> but no FreePBX anymore :( I'm sure I'm missing something important.
>> Can you give me a hint? Should sofia profiles be served by curl or
>> something?
>>
>> Thanks,
>>    Raffaele
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From raffaele.p.guidi at gmail.com  Mon Aug 24 11:27:12 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Mon, 24 Aug 2009 20:27:12 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
Message-ID: <b0a8792f0908241127k54ac14bfo8f441244ddc838b9@mail.gmail.com>

actually I meant "the SECOND time I *RAN* FS and freepbx."

On Mon, Aug 24, 2009 at 20:24, Raffaele P. Guidi <raffaele.p.guidi at gmail.com
> wrote:

> Actually I did that and it worked fine. I had the problem the SECOND time I
> run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were
> not
>
>
> On Sun, Aug 16, 2009 at 16:04, Carlos Talbot <carlos.talbot at gmail.com>wrote:
>
>> When you configure FreePBX for the first time it wipes out the
>> sip_profiles directory. If you follow the FreePBX shortcut on your desktop
>> it'll mention this on the last screen of the configuration. You might see
>> something such as the following below. If you plan to use FreePBX you'll
>> need to define trunk groups, trunks, etc in order to have the sip_profiles
>> directory populated.
>> regards,
>>
>> Carlos
>>
>>
>>  Incompatible Configuration WARNING: THE FOLLOWING FILES WILL BE DELETED!
>>
>>    - D:/FreeSWITCH/conf/sip_profiles/external.xml
>>    - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
>>    - D:/FreeSWITCH/conf/sip_profiles/internal.xml
>>
>>
>> On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi <
>> raffaele.p.guidi at gmail.com> wrote:
>>
>>> I had the sweet surprise to find the installer packaged with FreePBX...
>>> really great! Why it has not been advertised as it deserves? It worked like
>>> a breeze once launched, with the automatic configuration and all of that.,
>>> Only thing: once stopped I cannot get it to load sofia profiles anymore -
>>> issueing sofia status doesn't show anything. I had to copy internal.xml and
>>> default.xml from a previous installation and everything got to work again -
>>> but no FreePBX anymore :( I'm sure I'm missing something important.
>>> Can you give me a hint? Should sofia profiles be served by curl or
>>> something?
>>>
>>> Thanks,
>>>    Raffaele
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From brian at freeswitch.org  Mon Aug 24 11:30:43 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 13:30:43 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
Message-ID: <22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>

If you installed FreePBX then it would be that softwares job to manage  
the sofia profiles... wouldn't it?

/b

On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:

> Actually I did that and it worked fine. I had the problem the SECOND  
> time I run FS and freepbx. And (@Brian) mod_sofia was loaded but  
> sip_profiles were not



From d_hound at ymail.com  Mon Aug 24 11:37:53 2009
From: d_hound at ymail.com (Hound Dog)
Date: Mon, 24 Aug 2009 11:37:53 -0700 (PDT)
Subject: [Freeswitch-users] topology hiding leaking information in SDP
In-Reply-To: <191c3a030908240958q45f44fd6jb2eeaa6e6a95bbea@mail.gmail.com>
References: <508126.46278.qm@web111920.mail.gq1.yahoo.com>
	<191c3a030908240958q45f44fd6jb2eeaa6e6a95bbea@mail.gmail.com>
Message-ID: <920686.47099.qm@web111920.mail.gq1.yahoo.com>

so problem solved , Thanks you anthony and Brian



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Monday, August 24, 2009 7:58:47 PM
Subject: Re: [Freeswitch-users] topology hiding leaking information in SDP

if you establish an audio path that is using the same codec there is no difference in performance as the codecs
do not do anything when both sides are the same.



On Mon, Aug 24, 2009 at 11:33 AM, Hound Dog <d_hound at ymail.com> wrote:

Brian,
>
>
>I set proxy_media to false ad now getting clean SDP without all the incoming data
>
>
>thank you 
>
>
>
>
>
>
>this also means that the packets have to go all the way in and out of the codec chain  ( instead of being taken in and thrown out with no processing )   , is that the case ?
>
>
>what is the performance hit of such change , on a modern strong machine - what is the estimated streams that FS can take ?
>
>
>
>
>
>
>BTW - sorry for not answering the list as I should have , I think I registered wrong somewhere so now have a new user 
>
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



      
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From tculjaga at gmail.com  Mon Aug 24 12:03:01 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 24 Aug 2009 21:03:01 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK
	message
Message-ID: <65d96fc80908241203m29abc2c6t7678689a3402c31f@mail.gmail.com>

Hi Anthony,

I'm aware it is generating 30 retries per a call and this is killing me ...

I lost my entire working day to figure out what is missing in the damn ACK
message SIPp is sending back... ACK looks quite ok to me.

pls can you help ?


freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
16:44:26.527236:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Max-Forwards: 70
   Subject: Performance Test
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0


   ------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 ACK
   Contact: sip:sipp at 10.4.4.252:5060
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0




What m'i missing ?



> Your ACK message must not be valid (dialog matching or something else)
> so every 1 call will generate 30 retries that are queued up in the sip
> stack.
>
> at 100cps you will be generating this problem 100 times per second and
> queue up countless unfinished dialogs thus
> eating up the cpu.
>
>
>
>
>
>
> On Mon, Aug 24, 2009 at 12:19 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> Hello,
>>
>> I've been with freeswittch for a while now.. and i can say it is worth
>> developing it.
>>
>> anyhow i got into a strange issue... I'm tryng to see what load FS on my
>> server can take. The Call flow is like this:
>>
>> SIPp                   FS
>>
>> INVITE -------->
>>            <------- 100 Trying
>>            <------- 302 Moved Temporary
>> ACK    --------->
>>
>>
>>
>> I use a dummy dialplan for that. All custom functions i've build are
>> disabled and i'm not using it here. Also custom modules are not loaded as
>> well.
>>
>>
>>    <extension name="ServiceLookup">
>>       <condition field="destination_number" expression="(^300030)(.*)">
>>          <!--action application="lookup_service_destination" data="in
>> ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, i
>> n $1, in pgw01.ot.hr:5060, out red_contact, out authResult"/-->
>>          <action application="log" data="INFO ########################
>> ServiceLookup ########################\n"/>
>>          <action application="log" data="INFO ########################
>> contact = '${red_contact}' ##############\n"/>
>>          <action application="log" data="INFO ########################
>> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
>>          <action application="log" data="INFO ########################
>> RADIUS auth = '${authResult}' ##########\n"/>
>>          <action application="execute_extension" data="doRedirect XML
>> public"/>
>>         </condition>
>>    </extension>
>>
>>
>>    <extension name="doRedirect">
>>       <condition field="destination_number" expression="^doRedirect$"/>
>>       <condition field="${authResult}" expression="^0$|^60$">
>>          <action application="log" data="INFO ########################
>> RADIUS auth OK!!!' ##########\n"/>
>>          <!--action application="redirect" data="sip:${red_contact}"/-->
>>          <!--action application="answer"/-->
>>          <action application="redirect" data="
>> sip:12345616094191500 at pgw01.ot.hr:5060"/>
>>          <!--anti-action application="answer"/-->
>>          <!--anti-action application="sleep" data="2000"/-->
>>          <action application="hangup" data="USER_BUSY"/>
>>          <anti-action application="redirect" data="
>> sip:12345616094191500 at pgw01.ot.hr:5060"/>
>>          <anti-action application="log" data="INFO
>> ######################## RADIUS auth NOK!! ##########\n"/>
>>          <!--anti-action application="respond" data="403 Forbidden"/-->
>>          <anti-action application="hangup" data="USER_BUSY"/>
>>       </condition>
>>    </extension>
>>
>>
>> When i place a call from x-lite everything works fine ... x-lite sends an
>> invite, gets SIP 302 and ACKs it correctly... FS is happy.
>>
>> When i place a call from SIPp i have the same scenario except FS seems not
>> understand ACK message from SIPp and re-sends SIP 302 multiple times untill
>> it gives up.
>>
>>
>> I beleive this is due to 302 resend issue but; when i load FS with 100
>> CPS, i can see high CPU usage (just one thread taking most load... the rest
>> does almost nothing) on FS. Also, starting from 40 CPS there is a big delay
>> in receiving SIP 302 messages meaning i've sent 6000 calls and so far only
>> for half of them got 302 response.
>>
>>
>> Does anybody have a clue ?
>>
>>
>>
>>
>>
>> Here is a trace taken on FS for calls originated from SIPp (sipp -sn uac
>> 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100
>> -trace_msg -inf test.txt -m 1 -l 4000):
>>
>> freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
>> 16:44:26.527236:
>>
>> ------------------------------------------------------------------------
>>    INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
>>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
>>    Max-Forwards: 70
>>    Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>> >
>>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >
>>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>    Call-ID: 1-6962 at 10.4.4.252
>>    CSeq: 1 INVITE
>>    Max-Forwards: 70
>>    Subject: Performance Test
>>    Content-Type: application/sdp
>>    Content-Length:   131
>>
>>    v=0
>>    o=user1 53655765 2353687637 IN IP4 10.4.4.252
>>    s=-
>>    c=IN IP4 10.4.4.252
>>    t=0 0
>>    m=audio 6000 RTP/AVP 0
>>    a=rtpmap:0 PCMU/8000
>>
>> ------------------------------------------------------------------------
>> send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 100 Trying
>>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >
>>    Call-ID: 1-6962 at 10.4.4.252
>>    CSeq: 1 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 302 Moved Temporarily
>>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>    To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >;tag=Hr4mHDUeBSNyH
>>    Call-ID: 1-6962 at 10.4.4.252
>>    CSeq: 1 INVITE
>>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>>    Supported: timer, precondition, path, replaces
>>    Allow-Events: talk, presence, dialog, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
>>
>> ------------------------------------------------------------------------
>>    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>>    Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >
>>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>    Call-ID: 1-6962 at 10.4.4.252
>>    CSeq: 1 ACK
>>    Contact: sip:sipp at 10.4.4.252:5060
>>    Max-Forwards: 70
>>    Subject: Performance Test
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 302 Moved Temporarily
>>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>    To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >;tag=Hr4mHDUeBSNyH
>>    Call-ID: 1-6962 at 10.4.4.252
>>    CSeq: 1 INVITE
>>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>>    Supported: timer, precondition, path, replaces
>>    Allow-Events: talk, presence, dialog, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 302 Moved Temporarily
>>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>    To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >;tag=Hr4mHDUeBSNyH
>>    Call-ID: 1-6962 at 10.4.4.252
>>    CSeq: 1 INVITE
>>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>>    Supported: timer, precondition, path, replaces
>>    Allow-Events: talk, presence, dialog, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>>    Content-Length: 0
>>
>>
>> Tihomir.
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
> ---------- Forwarded message ----------
> From: "Raffaele P. Guidi" <raffaele.p.guidi at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 20:24:28 +0200
> Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
> but I have a little problem
> Actually I did that and it worked fine. I had the problem the SECOND time I
> run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were
> not
>
> On Sun, Aug 16, 2009 at 16:04, Carlos Talbot <carlos.talbot at gmail.com>wrote:
>
>> When you configure FreePBX for the first time it wipes out the
>> sip_profiles directory. If you follow the FreePBX shortcut on your desktop
>> it'll mention this on the last screen of the configuration. You might see
>> something such as the following below. If you plan to use FreePBX you'll
>> need to define trunk groups, trunks, etc in order to have the sip_profiles
>> directory populated.
>> regards,
>>
>> Carlos
>>
>>
>>  Incompatible Configuration WARNING: THE FOLLOWING FILES WILL BE DELETED!
>>
>>    - D:/FreeSWITCH/conf/sip_profiles/external.xml
>>    - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
>>    - D:/FreeSWITCH/conf/sip_profiles/internal.xml
>>
>>
>> On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi <
>> raffaele.p.guidi at gmail.com> wrote:
>>
>>> I had the sweet surprise to find the installer packaged with FreePBX...
>>> really great! Why it has not been advertised as it deserves? It worked like
>>> a breeze once launched, with the automatic configuration and all of that.,
>>> Only thing: once stopped I cannot get it to load sofia profiles anymore -
>>> issueing sofia status doesn't show anything. I had to copy internal.xml and
>>> default.xml from a previous installation and everything got to work again -
>>> but no FreePBX anymore :( I'm sure I'm missing something important.
>>> Can you give me a hint? Should sofia profiles be served by curl or
>>> something?
>>>
>>> Thanks,
>>>    Raffaele
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Mon Aug 24 12:05:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 24 Aug 2009 15:05:58 -0400
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4A91A480.7080307@gmx.net>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>	<4A8DB808.1030903@gmx.net>	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>	<4A8EBF77.1080204@gmx.net>	<BD3DE8E0-3363-465C-985D-EC593AE45694@freeswitch.org>
	<191c3a030908211309j60c16b26h1e6445891136ae7f@mail.gmail.com>
	<4A91A480.7080307@gmx.net>
Message-ID: <D8BD24FC-9B07-4045-B6FD-A6D00156F877@jerris.com>

That is the remote sdp, not the local sdp.  They are sending ptime 20,  
not us.  Are they actually sending 20 ms packets or are they sending 30?

MIke

On Aug 23, 2009, at 4:20 PM, Peter P GMX wrote:

> Hello Anthony,
>
> I set PCMA at 30i,PCMU at 30i and I can see in the logs that PCMA is used.
> However ptime is set to 20 msec as shown in the Logs:
>
> 2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP:
> v=0
> o=user 2075230 2075230 IN IP4 217.xx.xx.xxx
> s=call
> c=IN IP4 217.xx.xx.xxx
> t=0 0
> m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
> a=rtpmap:2 G726-32/8000
> a=rtpmap:102 G726-32/8000
> a=rtpmap:100 G726-40/8000
> a=rtpmap:99 G726-24/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
> a=rtcp:7079
>
> 2009-08-23 22:11:29.530301 [DEBUG] switch_core_state_machine.c:404
> (sofia/internal/02xxxxxxxxx at fs1.my.domain) State NEW
> 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec  
> Compare
> [PCMA:8:8000:0]/[G722:9:8000:20]
> 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec  
> Compare
> [PCMA:8:8000:0]/[PCMU:0:8000:20]
> 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec  
> Compare
> [PCMA:8:8000:0]/[PCMA:8:8000:20]
> 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:2090 Set Codec
> sofia/internal/02xxxxxxxxx at fs1.my.domain PCMA/8000 20 ms 160 samples
> 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
> payload to 101
>
> Sound from FS to Fritzbox is fine. Sound from Fritzbox to FS is  
> horrible.
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
>> try setting FS to 30ms too
>>
>> edit vars.xml and add @30i to everywhere you see  PCMU or PCMA so it
>> looks like PCMU at 30i
>>
>> from:
>>
>>  <X-PRE-PROCESS cmd="set"
>> data 
>> ="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>>
>> to:
>>
>>  <X-PRE-PROCESS cmd="set"
>> data 
>> = 
>> "global_codec_prefs 
>> =G7221 at 32000h,G7221 at 16000h,G722,PCMU at 30i,PCMA at 30i,GSM"/>
>>  <X-PRE-PROCESS cmd="set"
>> data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i,GSM"/>
>>
>>
>> On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian at freeswitch.org
>> <mailto:brian at freeswitch.org>> wrote:
>>
>>    You can ship me one whois bkw.org <http://bkw.org>, I can add it
>>    to my lab.
>>
>>    /b
>>
>>    On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
>>
>>>
>>> BTW: We can ship you a FritzBox if you need one for testing.
>>
>>
>>    _______________________________________________
>>    FreeSWITCH-users mailing list
>>    FreeSWITCH-users at lists.freeswitch.org
>>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>> freeswitch-users
>>    http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> <mailto:sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> <http://iax:guest at conference.freeswitch.org/888>
>> googletalk:conf+888 at conference.freeswitch.org
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From brian at freeswitch.org  Mon Aug 24 12:15:40 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 14:15:40 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241203m29abc2c6t7678689a3402c31f@mail.gmail.com>
References: <65d96fc80908241203m29abc2c6t7678689a3402c31f@mail.gmail.com>
Message-ID: <A972ED9E-9EA2-481C-818F-98437DF5FC09@freeswitch.org>

In your scenario you need to add [peer_tag_param] at the end of the to  
on the Ack.

/b

On Aug 24, 2009, at 2:03 PM, Tihomir Culjaga wrote:

>
>     
> ------------------------------------------------------------------------
> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
>     
> ------------------------------------------------------------------------
>    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 ACK
>    Contact: sip:sipp at 10.4.4.252:5060
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Length: 0



From jerry.richards at teotech.com  Mon Aug 24 12:24:42 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 24 Aug 2009 12:24:42 -0700
Subject: [Freeswitch-users] Cannot create outgoing channel type [error]
	cause: [FACILITY_NOT_SUBSCRIBED]
Message-ID: <16C552FB75474E8D82974FCF182B06D3@greyhawk.tonecommander.com>

Hello All,

I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP machine
for the first time using the Getting Started Guide.  I can register three
lines (1000, 1001, and 1002), but when I attempt to call one phone to the
other I hear the operator say:

"The person at extension 1000 is not available..."

Also, the Freeswitch log shows:

Cannot create outgoing channel type [error] cause:
[FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause:
[FACILITY_NOT_SUBSCRIBED]

Does anyone know why I get this error?

Best Regards,
Jerry



From brian at freeswitch.org  Mon Aug 24 12:33:22 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 14:33:22 -0500
Subject: [Freeswitch-users] Cannot create outgoing channel type [error]
	cause: [FACILITY_NOT_SUBSCRIBED]
In-Reply-To: <16C552FB75474E8D82974FCF182B06D3@greyhawk.tonecommander.com>
References: <16C552FB75474E8D82974FCF182B06D3@greyhawk.tonecommander.com>
Message-ID: <C61B5EA8-345C-471E-BBA5-534A1726F396@freeswitch.org>

Are you trying to test everything on the same machine?

/b

On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote:

> Hello All,
>
> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP  
> machine
> for the first time using the Getting Started Guide.  I can register  
> three
> lines (1000, 1001, and 1002), but when I attempt to call one phone  
> to the
> other I hear the operator say:
>
> "The person at extension 1000 is not available..."
>
> Also, the Freeswitch log shows:
>
> Cannot create outgoing channel type [error] cause:
> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user]  
> cause:
> [FACILITY_NOT_SUBSCRIBED]
>
> Does anyone know why I get this error?
>
> Best Regards,
> Jerry



From mike at jerris.com  Mon Aug 24 12:44:18 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 24 Aug 2009 15:44:18 -0400
Subject: [Freeswitch-users] Problem with cnam.js?
In-Reply-To: <00b901ca2347$950feee0$bf2fcca0$@com>
References: <00b901ca2347$950feee0$bf2fcca0$@com>
Message-ID: <2FCFC745-86FC-4A66-B919-200FC865C019@jerris.com>

Every page on the wiki should be editable.  If you don't already have  
an account, go to:

http://wiki.freeswitch.org/index.php?title=Special:UserLogin&type=signup

Mike

On Aug 22, 2009, at 12:42 PM, Lars Zeb wrote:

> I think there?s something wrong with the script at http://wiki.freeswitch.org/wiki/Examples_cnam.js 
> .
>
> If you use it as is, it displays ?Content-type: text/html? for the  
> effective_caller_id_name. In cnam.pl, the first two output lines are  
> generated by:
>
> if (!$debug) {print "Content-type: text/html\n\n";}
>
> with the actual name in the third line.
>
> So I changed:
>
> fd.open("read");
> buff = fd.readln();
>
> if(buff) {
>    logger(buff, "info");
>    session.setVariable("effective_caller_id_name", buff);
> }
>
> To:
>
> fd.open("read");
> buff = fd.readAll();
>
> if(buff[2]) {
>    logger(buff, "info");
>    session.setVariable("effective_caller_id_name", buff[2]);
> }
>
> Or remove the print statement from cnam.pl.
>
> Sorry for the code, but the page was not editable.
>
> Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mike at jerris.com  Mon Aug 24 12:46:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 24 Aug 2009 15:46:58 -0400
Subject: [Freeswitch-users] Yet another question about A500 + FS
In-Reply-To: <8a9b664c0908222321r20c5c4d8wd08721dc8955f62c@mail.gmail.com>
References: <8a9b664c0908222321r20c5c4d8wd08721dc8955f62c@mail.gmail.com>
Message-ID: <56CFD144-94D0-42CE-B59D-7DC524C1919D@jerris.com>

Do you have an answer in the dialplan for that extension?  Also, check  
out the ignore_early_media variable.

Mike

On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote:

> Hi,
>
> I managed to get our A500 running with FreeSWITCH 1.0.4 stable using  
> wanpipe 3.4.4 drivers. But now I have another problem...
> I want to originate calls through event socket, and I only want to  
> receive ANSWERED(+OK) reply when the user actually answers.
>
> Now the situation is:
>
> ====================================
> originate openzap/1/a/123456 023
> 2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT:  
> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456] Ci= 
> [0000000000]
> 2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT  
> (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4
> 2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel  
> OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082]
> 2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer  
> OpenZAP/1:1/123456!
> API CALL [originate(openzap/1/a/123456 023)] output:
> +OK f8fca2be-8fa7-11de-9076-511e29dfc082
>
> 2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer  
> OpenZAP/1:1/123456 to XML[023 at default]
> freeswitch at emo-voip> 2009-08-23 08:44:06.743475 [INFO]  
> mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default
> 2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel  
> [OpenZAP/1:1/123456] has been answered
> 2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT  
> (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5
> 2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT  
> (N): CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6
> 2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup  
> OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT  
> (N): CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3
> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086  
> Session 2 (OpenZAP/1:1/123456) Ended
> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close  
> Channel OpenZAP/1:1/123456 [CS_DESTROY]
> ====================================
>
> Extension 023 is an IVR. As you can see FreeSWITCH answers the call  
> (2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel  
> [OpenZAP/1:1/123456] has been answered) 20 seconds before user  
> actually pick up the phone (2009-08-23 08:44:20.206010 [WARNING]  
> ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0  
> CSid=2 Seq=5).
>
> So Sangoma drivers/daemons report the events correctly.
> How can I set FreeSWITCH to answer after receiving RX EVENT (N):  
> CALL_ANSWERED from the driver?
>
> Thank you,
> V. Panayotov
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Mon Aug 24 12:50:57 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 14:50:57 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <D8BD24FC-9B07-4045-B6FD-A6D00156F877@jerris.com>
References: <4A8D5132.7010807@gmx.net> <4A8D5D43.3030300@gmx.net>
	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>
	<4A8DB808.1030903@gmx.net>
	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
	<4A8EBF77.1080204@gmx.net>
	<BD3DE8E0-3363-465C-985D-EC593AE45694@freeswitch.org>
	<191c3a030908211309j60c16b26h1e6445891136ae7f@mail.gmail.com>
	<4A91A480.7080307@gmx.net>
	<D8BD24FC-9B07-4045-B6FD-A6D00156F877@jerris.com>
Message-ID: <191c3a030908241250o5284e986hc9cfa6093495a42c@mail.gmail.com>

mr fritz is lying somewhere

get a pcap of the traffic from fritz to FS and look at the size of the audio
packets
if they are 160(172 with headers) bytes then it's 20ms if it's 240 (252)
then it's 30ms

if it's saying 20 but it means 30 you should leave the last change in place
and also add in

<param name="rtp-autofix-timing" value="true" />
and
<param name="inbound-codec-negotiation" value="scrooge"/>

to overcome the bug on their end.


On Mon, Aug 24, 2009 at 2:05 PM, Michael Jerris <mike at jerris.com> wrote:

> That is the remote sdp, not the local sdp.  They are sending ptime 20,
> not us.  Are they actually sending 20 ms packets or are they sending 30?
>
> MIke
>
> On Aug 23, 2009, at 4:20 PM, Peter P GMX wrote:
>
> > Hello Anthony,
> >
> > I set PCMA at 30i,PCMU at 30i and I can see in the logs that PCMA is used.
> > However ptime is set to 20 msec as shown in the Logs:
> >
> > 2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP:
> > v=0
> > o=user 2075230 2075230 IN IP4 217.xx.xx.xxx
> > s=call
> > c=IN IP4 217.xx.xx.xxx
> > t=0 0
> > m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
> > a=rtpmap:2 G726-32/8000
> > a=rtpmap:102 G726-32/8000
> > a=rtpmap:100 G726-40/8000
> > a=rtpmap:99 G726-24/8000
> > a=rtpmap:97 iLBC/8000
> > a=fmtp:97 mode=30
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-11
> > a=rtcp:7079
> >
> > 2009-08-23 22:11:29.530301 [DEBUG] switch_core_state_machine.c:404
> > (sofia/internal/02xxxxxxxxx at fs1.my.domain) State NEW
> > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec
> > Compare
> > [PCMA:8:8000:0]/[G722:9:8000:20]
> > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec
> > Compare
> > [PCMA:8:8000:0]/[PCMU:0:8000:20]
> > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec
> > Compare
> > [PCMA:8:8000:0]/[PCMA:8:8000:20]
> > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:2090 Set Codec
> > sofia/internal/02xxxxxxxxx at fs1.my.domain PCMA/8000 20 ms 160 samples
> > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
> > payload to 101
> >
> > Sound from FS to Fritzbox is fine. Sound from Fritzbox to FS is
> > horrible.
> >
> > Best regards
> > Peter
> >
> > Anthony Minessale schrieb:
> >> try setting FS to 30ms too
> >>
> >> edit vars.xml and add @30i to everywhere you see  PCMU or PCMA so it
> >> looks like PCMU at 30i
> >>
> >> from:
> >>
> >>  <X-PRE-PROCESS cmd="set"
> >> data
> >> ="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
> >>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
> >>
> >> to:
> >>
> >>  <X-PRE-PROCESS cmd="set"
> >> data
> >> =
> >> "global_codec_prefs
> >> =G7221 at 32000h,G7221 at 16000h,G722,PCMU at 30i,PCMA at 30i,GSM"/>
> >>  <X-PRE-PROCESS cmd="set"
> >> data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i,GSM"/>
> >>
> >>
> >> On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian at freeswitch.org
> >> <mailto:brian at freeswitch.org>> wrote:
> >>
> >>    You can ship me one whois bkw.org <http://bkw.org>, I can add it
> >>    to my lab.
> >>
> >>    /b
> >>
> >>    On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
> >>
> >>>
> >>> BTW: We can ship you a FritzBox if you need one for testing.
> >>
> >>
> >>    _______________________________________________
> >>    FreeSWITCH-users mailing list
> >>    FreeSWITCH-users at lists.freeswitch.org
> >>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
> >> freeswitch-users
> >>    http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >> iax:guest at conference.freeswitch.org/888
> >> <http://iax:guest at conference.freeswitch.org/888>
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >> pstn:213-799-1400
> >> ------------------------------------------------------------------------
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jerry.richards at teotech.com  Mon Aug 24 13:25:59 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 24 Aug 2009 13:25:59 -0700
Subject: [Freeswitch-users] Cannot create outgoing channel type
	[error]cause: [FACILITY_NOT_SUBSCRIBED]
In-Reply-To: <C61B5EA8-345C-471E-BBA5-534A1726F396@freeswitch.org>
References: <16C552FB75474E8D82974FCF182B06D3@greyhawk.tonecommander.com>
	<C61B5EA8-345C-471E-BBA5-534A1726F396@freeswitch.org>
Message-ID: <12329F13B58D4DE2AC8AC895EBA94074@greyhawk.tonecommander.com>

Yes.  This a stand-alone Windows XP machine.

Jerry 

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Monday, August 24, 2009 12:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cannot create outgoing channel type
[error]cause: [FACILITY_NOT_SUBSCRIBED]

Are you trying to test everything on the same machine?

/b

On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote:

> Hello All,
>
> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP 
> machine for the first time using the Getting Started Guide.  I can 
> register three lines (1000, 1001, and 1002), but when I attempt to 
> call one phone to the other I hear the operator say:
>
> "The person at extension 1000 is not available..."
>
> Also, the Freeswitch log shows:
>
> Cannot create outgoing channel type [error] cause:
> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user]
> cause:
> [FACILITY_NOT_SUBSCRIBED]
>
> Does anyone know why I get this error?
>
> Best Regards,
> Jerry






From brian at freeswitch.org  Mon Aug 24 13:29:09 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 15:29:09 -0500
Subject: [Freeswitch-users] Cannot create outgoing channel type
	[error]cause: [FACILITY_NOT_SUBSCRIBED]
In-Reply-To: <12329F13B58D4DE2AC8AC895EBA94074@greyhawk.tonecommander.com>
References: <16C552FB75474E8D82974FCF182B06D3@greyhawk.tonecommander.com>
	<C61B5EA8-345C-471E-BBA5-534A1726F396@freeswitch.org>
	<12329F13B58D4DE2AC8AC895EBA94074@greyhawk.tonecommander.com>
Message-ID: <E4FC66FB-73A5-48F0-BEC3-E9A71403B31B@freeswitch.org>

Make sure your x-lite is not on port 5060

/b

On Aug 24, 2009, at 3:25 PM, Jerry Richards wrote:

> Yes.  This a stand-alone Windows XP machine.
>
> Jerry



From tculjaga at gmail.com  Mon Aug 24 13:37:24 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 24 Aug 2009 22:37:24 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
Message-ID: <65d96fc80908241337m3c2ea93bv935a5b1cfc3f0c9a@mail.gmail.com>

Hello Brian,

it doesn't work .. tried this today as well:



freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
20:28:09.367300:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 INVITE
   Max-Forwards: 70
   Subject: Performance Test
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 20:28:09.367634:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.371759:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=ygQBtp6QpKtcD
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 401 bytes from udp/[10.4.4.252]:5060 at 20:28:09.371989:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7019-1-3;rport
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=ygQBtp6QpKtcD
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 ACK
   Contact: sip:sipp at 10.4.4.252:5060
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.873045:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=ygQBtp6QpKtcD
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0



This thing is driving me crazy, pls help.

T.



>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 14:15:40 -0500
> Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand
> ACK message
> In your scenario you need to add [peer_tag_param] at the end of the to on
> the Ack.
>
> /b
>
> On Aug 24, 2009, at 2:03 PM, Tihomir Culjaga wrote:
>
>
>>   ------------------------------------------------------------------------
>> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
>>   ------------------------------------------------------------------------
>>   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>>   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>>   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >
>>   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>   Call-ID: 1-6962 at 10.4.4.252
>>   CSeq: 1 ACK
>>   Contact: sip:sipp at 10.4.4.252:5060
>>   Max-Forwards: 70
>>   Subject: Performance Test
>>   Content-Length: 0
>>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: "Jerry Richards" <jerry.richards at teotech.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 24 Aug 2009 12:24:42 -0700
> Subject: [Freeswitch-users] Cannot create outgoing channel type [error]
> cause: [FACILITY_NOT_SUBSCRIBED]
> Hello All,
>
> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP
> machine
> for the first time using the Getting Started Guide.  I can register three
> lines (1000, 1001, and 1002), but when I attempt to call one phone to the
> other I hear the operator say:
>
> "The person at extension 1000 is not available..."
>
> Also, the Freeswitch log shows:
>
> Cannot create outgoing channel type [error] cause:
> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause:
> [FACILITY_NOT_SUBSCRIBED]
>
> Does anyone know why I get this error?
>
> Best Regards,
> Jerry
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 14:33:22 -0500
> Subject: Re: [Freeswitch-users] Cannot create outgoing channel type [error]
> cause: [FACILITY_NOT_SUBSCRIBED]
> Are you trying to test everything on the same machine?
>
> /b
>
> On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote:
>
>  Hello All,
>>
>> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP
>> machine
>> for the first time using the Getting Started Guide.  I can register three
>> lines (1000, 1001, and 1002), but when I attempt to call one phone to the
>> other I hear the operator say:
>>
>> "The person at extension 1000 is not available..."
>>
>> Also, the Freeswitch log shows:
>>
>> Cannot create outgoing channel type [error] cause:
>> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause:
>> [FACILITY_NOT_SUBSCRIBED]
>>
>> Does anyone know why I get this error?
>>
>> Best Regards,
>> Jerry
>>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 15:44:18 -0400
> Subject: Re: [Freeswitch-users] Problem with cnam.js?
> Every page on the wiki should be editable.  If you don't already have an
> account, go to:
> http://wiki.freeswitch.org/index.php?title=Special:UserLogin&type=signup
>
> Mike
>
> On Aug 22, 2009, at 12:42 PM, Lars Zeb wrote:
>
> I think there?s something wrong with the script at
> http://wiki.freeswitch.org/wiki/Examples_cnam.js.
>
> If you use it as is, it displays ?Content-type: text/html? for the
> effective_caller_id_name. In cnam.pl, the first two output lines are
> generated by:
>
> if (!$debug) {print "Content-type: text/html\n\n";}
>
> with the actual name in the third line.
>
> So I changed:
>
> fd.open("read");
> buff = fd.readln();
>
> if(buff) {
>    logger(buff, "info");
>    session.setVariable("effective_caller_id_name", buff);
> }
>
> To:
>
> fd.open("read");
> buff = fd.readAll();
>
> if(buff[2]) {
>    logger(buff, "info");
>    session.setVariable("effective_caller_id_name", buff[2]);
> }
>
> Or remove the print statement from cnam.pl.
>
> Sorry for the code, but the page was not editable.
>
> Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 15:46:58 -0400
> Subject: Re: [Freeswitch-users] Yet another question about A500 + FS
> Do you have an answer in the dialplan for that extension?  Also, check out
> the ignore_early_media variable.
>
> Mike
>
> On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote:
>
>  Hi,
>>
>> I managed to get our A500 running with FreeSWITCH 1.0.4 stable using
>> wanpipe 3.4.4 drivers. But now I have another problem...
>> I want to originate calls through event socket, and I only want to receive
>> ANSWERED(+OK) reply when the user actually answers.
>>
>> Now the situation is:
>>
>> ====================================
>> originate openzap/1/a/123456 023
>> 2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT:
>> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456]
>> Ci=[0000000000]
>> 2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
>> CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4
>> 2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel
>> OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082]
>> 2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer
>> OpenZAP/1:1/123456!
>> API CALL [originate(openzap/1/a/123456 023)] output:
>> +OK f8fca2be-8fa7-11de-9076-511e29dfc082
>>
>> 2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer
>> OpenZAP/1:1/123456 to XML[023 at default]
>> freeswitch at emo-voip> 2009-08-23 08:44:06.743475 [INFO]
>> mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default
>> 2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
>> [OpenZAP/1:1/123456] has been answered
>> 2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
>> CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5
>> 2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
>> CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6
>> 2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup
>> OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT (N):
>> CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3
>> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086 Session 2
>> (OpenZAP/1:1/123456) Ended
>> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close
>> Channel OpenZAP/1:1/123456 [CS_DESTROY]
>> ====================================
>>
>> Extension 023 is an IVR. As you can see FreeSWITCH answers the call
>> (2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
>> [OpenZAP/1:1/123456] has been answered) 20 seconds before user actually pick
>> up the phone (2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX
>> EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5).
>>
>> So Sangoma drivers/daemons report the events correctly.
>> How can I set FreeSWITCH to answer after receiving RX EVENT (N):
>> CALL_ANSWERED from the driver?
>>
>> Thank you,
>> V. Panayotov
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dave at 3c.co.uk  Mon Aug 24 13:41:23 2009
From: dave at 3c.co.uk (David Knell)
Date: Mon, 24 Aug 2009 23:41:23 +0300
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
 ACK message
In-Reply-To: <65d96fc80908241203m29abc2c6t7678689a3402c31f@mail.gmail.com>
References: <65d96fc80908241203m29abc2c6t7678689a3402c31f@mail.gmail.com>
Message-ID: <1251146483.16312.36.camel@dk-d820>

Hi Tihomir -

I'm no SIP guru, but the things which look suspicious about the ACK to
me are:
- Via header - different branch
- Contact header - differs from INVITE

--Dave

> Hi Anthony,
> 
> I'm aware it is generating 30 retries per a call and this is killing
> me ...
> 
> I lost my entire working day to figure out what is missing in the damn
> ACK message SIPp is sending back... ACK looks quite ok to me.
> 
> pls can you help ?
> 
> 
> freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
> 16:44:26.527236:
>    ------------------------------
> ------------------------------------------
>    INVITE sip:30003016094191500 at 10.4.4.251 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
>    Max-Forwards: 70
>    Contact: <sip:22222238515000403 at 10.4.4.252>
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Type: application/sdp
>    Content-Length:   131
>    
>    v=0
>    o=user1 53655765 2353687637 IN IP4 10.4.4.252
>    s=-
>    c=IN IP4 10.4.4.252
>    t=0 0
>    m=audio 6000 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
> 
> ------------------------------------------------------------------------
> send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
> 
> ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Content-Length: 0
>    
> 
> ------------------------------------------------------------------------
> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
> 
> ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    To: "30003016094191500"
> <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
> 
> 
> 
> ------------------------------------------------------------------------
> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
> 
> ------------------------------------------------------------------------
>    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 ACK
>    Contact: sip:sipp at 10.4.4.252:5060
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Length: 0
> 
> 
> 
> 
> 
> What m'i missing ?
> 
> 
> 
>         
>         Your ACK message must not be valid (dialog matching or
>         something else)
>         so every 1 call will generate 30 retries that are queued up in
>         the sip stack.
>         
>         at 100cps you will be generating this problem 100 times per
>         second and queue up countless unfinished dialogs thus
>         eating up the cpu.
>         
>         
>         
>         
>         
>         
>         On Mon, Aug 24, 2009 at 12:19 PM, Tihomir Culjaga
>         <tculjaga at gmail.com> wrote:
>                 Hello,
>                 
>                 I've been with freeswittch for a while now.. and i can
>                 say it is worth developing it.
>                 
>                 anyhow i got into a strange issue... I'm tryng to see
>                 what load FS on my server can take. The Call flow is
>                 like this:
>                 
>                 SIPp                   FS
>                 
>                 INVITE --------> 
>                            <------- 100 Trying
>                            <------- 302 Moved Temporary
>                 ACK    --------->
>                 
>                 
>                 
>                 I use a dummy dialplan for that. All custom functions
>                 i've build are disabled and i'm not using it here.
>                 Also custom modules are not loaded as well.
>                 
>                 
>                    <extension name="ServiceLookup">
>                       <condition field="destination_number"
>                 expression="(^300030)(.*)">
>                          <!--action
>                 application="lookup_service_destination" data="in
>                 ${caller_id_number:6:16}, in ${caller_id_number:0:6},
>                 in $2, i
>                 n $1, in pgw01.ot.hr:5060, out red_contact, out
>                 authResult"/-->
>                          <action application="log" data="INFO
>                 ######################## ServiceLookup
>                 ########################\n"/>
>                          <action application="log" data="INFO
>                 ######################## contact = '${red_contact}'
>                 ##############\n"/>
>                          <action application="log" data="INFO
>                 ######################## CallerNum =
>                 '${caller_id_number:6:16}' ##########\n"/>
>                          <action application="log" data="INFO
>                 ######################## RADIUS auth = '${authResult}'
>                 ##########\n"/>
>                          <action application="execute_extension"
>                 data="doRedirect XML public"/>
>                         </condition>
>                    </extension>
>                 
>                 
>                    <extension name="doRedirect">
>                       <condition field="destination_number"
>                 expression="^doRedirect$"/>
>                       <condition field="${authResult}" expression="^0
>                 $|^60$">
>                          <action application="log" data="INFO
>                 ######################## RADIUS auth OK!!!' ##########
>                 \n"/>
>                          <!--action application="redirect" data="sip:
>                 ${red_contact}"/-->
>                          <!--action application="answer"/-->
>                          <action application="redirect"
>                 data="sip:12345616094191500 at pgw01.ot.hr:5060"/>
>                          <!--anti-action application="answer"/-->
>                          <!--anti-action application="sleep"
>                 data="2000"/-->
>                          <action application="hangup"
>                 data="USER_BUSY"/>
>                          <anti-action application="redirect"
>                 data="sip:12345616094191500 at pgw01.ot.hr:5060"/>
>                          <anti-action application="log" data="INFO
>                 ######################## RADIUS auth NOK!! ##########
>                 \n"/>
>                          <!--anti-action application="respond"
>                 data="403 Forbidden"/-->
>                          <anti-action application="hangup"
>                 data="USER_BUSY"/>
>                       </condition>
>                    </extension>
>                 
>                 
>                 When i place a call from x-lite everything works
>                 fine ... x-lite sends an invite, gets SIP 302 and ACKs
>                 it correctly... FS is happy.
>                 
>                 When i place a call from SIPp i have the same scenario
>                 except FS seems not understand ACK message from SIPp
>                 and re-sends SIP 302 multiple times untill it gives
>                 up.
>                 
>                 
>                 I beleive this is due to 302 resend issue but; when i
>                 load FS with 100 CPS, i can see high CPU usage (just
>                 one thread taking most load... the rest does almost
>                 nothing) on FS. Also, starting from 40 CPS there is a
>                 big delay in receiving SIP 302 messages meaning i've
>                 sent 6000 calls and so far only for half of them got
>                 302 response.
>                 
>                 
>                 Does anybody have a clue ?
>                 
>                 
>                 
>                 
>                 
>                 Here is a trace taken on FS for calls originated from
>                 SIPp (sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
>                 30003016094191500 -trace_err -r 1 -rp 100 -trace_msg
>                 -inf test.txt -m 1 -l 4000):
>                 
>                 freeswitch at l01sipindir1> recv 573 bytes from
>                 udp/[10.4.4.252]:5060 at 16:44:26.527236:
>                 
>                 ------------------------------------------------------------------------
>                    INVITE sip:30003016094191500 at 10.4.4.251 SIP/2.0
>                    Via: SIP/2.0/UDP
>                 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
>                    Max-Forwards: 70
>                    Contact: <sip:22222238515000403 at 10.4.4.252>
>                    To:
>                 "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>                    From:
>                 "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>                    Call-ID: 1-6962 at 10.4.4.252
>                    CSeq: 1 INVITE
>                    Max-Forwards: 70
>                    Subject: Performance Test
>                    Content-Type: application/sdp
>                    Content-Length:   131
>                    
>                    v=0
>                    o=user1 53655765 2353687637 IN IP4 10.4.4.252
>                    s=-
>                    c=IN IP4 10.4.4.252
>                    t=0 0
>                    m=audio 6000 RTP/AVP 0
>                    a=rtpmap:0 PCMU/8000
>                 
>                 ------------------------------------------------------------------------
>                 send 328 bytes to udp/[10.4.4.252]:5060 at
>                 16:44:26.527566:
>                 
>                 ------------------------------------------------------------------------
>                    SIP/2.0 100 Trying
>                    Via: SIP/2.0/UDP
>                 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>                    From:
>                 "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>                    To:
>                 "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>                    Call-ID: 1-6962 at 10.4.4.252
>                    CSeq: 1 INVITE
>                    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>                    Content-Length: 0
>                    
>                 
>                 ------------------------------------------------------------------------
>                 send 722 bytes to udp/[10.4.4.252]:5060 at
>                 16:44:26.535582:
>                 
>                 ------------------------------------------------------------------------
>                    SIP/2.0 302 Moved Temporarily
>                    Via: SIP/2.0/UDP
>                 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>                    From:
>                 "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>                    To: "30003016094191500"
>                 <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
>                    Call-ID: 1-6962 at 10.4.4.252
>                    CSeq: 1 INVITE
>                    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>                    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>                    Accept: application/sdp
>                    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
>                 MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER,
>                 INFO, PUBLISH
>                    Supported: timer, precondition, path, replaces
>                    Allow-Events: talk, presence, dialog, call-info,
>                 sla, include-session-description, presence.winfo,
>                 message-summary, refer
>                    Content-Length: 0
>                    
>                 
>                 ------------------------------------------------------------------------
>                 recv 383 bytes from udp/[10.4.4.252]:5060 at
>                 16:44:26.535809:
>                 
>                 ------------------------------------------------------------------------
>                    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>                    Via: SIP/2.0/UDP
>                 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>                    To:
>                 "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>                    From:
>                 "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>                    Call-ID: 1-6962 at 10.4.4.252
>                    CSeq: 1 ACK
>                    Contact: sip:sipp at 10.4.4.252:5060
>                    Max-Forwards: 70
>                    Subject: Performance Test
>                    Content-Length: 0
>                    
>                 
>                 ------------------------------------------------------------------------
>                 send 722 bytes to udp/[10.4.4.252]:5060 at
>                 16:44:27.037070:
>                 
>                 ------------------------------------------------------------------------
>                    SIP/2.0 302 Moved Temporarily
>                    Via: SIP/2.0/UDP
>                 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>                    From:
>                 "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>                    To: "30003016094191500"
>                 <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
>                    Call-ID: 1-6962 at 10.4.4.252
>                    CSeq: 1 INVITE
>                    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>                    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>                    Accept: application/sdp
>                    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
>                 MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER,
>                 INFO, PUBLISH
>                    Supported: timer, precondition, path, replaces
>                    Allow-Events: talk, presence, dialog, call-info,
>                 sla, include-session-description, presence.winfo,
>                 message-summary, refer
>                    Content-Length: 0
>                    
>                 
>                 ------------------------------------------------------------------------
>                 send 722 bytes to udp/[10.4.4.252]:5060 at
>                 16:44:28.037063:
>                 
>                 ------------------------------------------------------------------------
>                    SIP/2.0 302 Moved Temporarily
>                    Via: SIP/2.0/UDP
>                 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>                    From:
>                 "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>                    To: "30003016094191500"
>                 <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
>                    Call-ID: 1-6962 at 10.4.4.252
>                    CSeq: 1 INVITE
>                    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>                    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>                    Accept: application/sdp
>                    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
>                 MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER,
>                 INFO, PUBLISH
>                    Supported: timer, precondition, path, replaces
>                    Allow-Events: talk, presence, dialog, call-info,
>                 sla, include-session-description, presence.winfo,
>                 message-summary, refer
>                    Content-Length: 0
>                 
>                 
>                 Tihomir.
>                 
>                 
>                 
>                 
>                 
>                 
>                 _______________________________________________
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at lists.freeswitch.org
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 http://www.freeswitch.org
>                 
>         
>         
>         
>         -- 
>         Anthony Minessale II
>         
>         FreeSWITCH http://www.freeswitch.org/
>         ClueCon http://www.cluecon.com/
>         Twitter: http://twitter.com/FreeSWITCH_wire
>         
>         AIM: anthm
>         MSN:anthony_minessale at hotmail.com
>         GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>         IRC: irc.freenode.net #freeswitch
>         
>         FreeSWITCH Developer Conference
>         sip:888 at conference.freeswitch.org
>         iax:guest at conference.freeswitch.org/888
>         googletalk:conf+888 at conference.freeswitch.org
>         pstn:213-799-1400
>         
>         
>         ---------- Forwarded message ----------
>         From: "Raffaele P. Guidi" <raffaele.p.guidi at gmail.com>
>         To: freeswitch-users at lists.freeswitch.org
>         Date: Mon, 24 Aug 2009 20:24:28 +0200
>         Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows
>         installer - great but I have a little problem
>         Actually I did that and it worked fine. I had the problem the
>         SECOND time I run FS and freepbx. And (@Brian) mod_sofia was
>         loaded but sip_profiles were not
>         
>         On Sun, Aug 16, 2009 at 16:04, Carlos Talbot
>         <carlos.talbot at gmail.com> wrote:
>                 When you configure FreePBX for the first time it wipes
>                 out the sip_profiles directory. If you follow the
>                 FreePBX shortcut on your desktop it'll mention this on
>                 the last screen of the configuration. You might see
>                 something such as the following below. If you plan to
>                 use FreePBX you'll need to define trunk groups,
>                 trunks, etc in order to have the sip_profiles
>                 directory populated.
>                 
>                 
>                 regards,
>                 
>                 
>                 Carlos
>                 
>                 
>                 
>                 
>                               Incompatible Configuration
>                     WARNING: THE FOLLOWING FILES WILL BE DELETED!
>                 
>                       * D:/FreeSWITCH/conf/sip_profiles/external.xml
>                       * D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
>                       * D:/FreeSWITCH/conf/sip_profiles/internal.xml
>                 
>                 
>                 On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi
>                 <raffaele.p.guidi at gmail.com> wrote:
>                 
>                         
>                         I had the sweet surprise to find the installer
>                         packaged with FreePBX... really great! Why it
>                         has not been advertised as it deserves? It
>                         worked like a breeze once launched, with the
>                         automatic configuration and all of that., Only
>                         thing: once stopped I cannot get it to load
>                         sofia profiles anymore - issueing sofia status
>                         doesn't show anything. I had to copy
>                         internal.xml and default.xml from a previous
>                         installation and everything got to work again
>                         - but no FreePBX anymore :( I'm sure I'm
>                         missing something important.
>                         
>                         
>                         Can you give me a hint? Should sofia profiles
>                         be served by curl or something?
>                         
>                         
>                         Thanks,
>                            Raffaele
>                         
>                         
>                         _______________________________________________
>                         FreeSWITCH-users mailing list
>                         FreeSWITCH-users at lists.freeswitch.org
>                         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                         http://www.freeswitch.org
>                         
>                 
>                 
>                 
>                 _______________________________________________
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at lists.freeswitch.org
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 http://www.freeswitch.org
>                 
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From brian at freeswitch.org  Mon Aug 24 13:42:31 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 15:42:31 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241337m3c2ea93bv935a5b1cfc3f0c9a@mail.gmail.com>
References: <65d96fc80908241337m3c2ea93bv935a5b1cfc3f0c9a@mail.gmail.com>
Message-ID: <FEB84EFF-305E-4964-BBAB-2C95B2EAB865@freeswitch.org>

ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

Use that.. your scenario has some hard coded IP's in the fields that  
shouldn't be there.

/b

On Aug 24, 2009, at 3:37 PM, Tihomir Culjaga wrote:

> Hello Brian,
>
> it doesn't work .. tried this today as well:
>
>
>
> freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060  
> at 20:28:09.367300:
>     
> ------------------------------------------------------------------------
>    INVITE sip:30003016094191500 at 10.4.4.251 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport
>    Max-Forwards: 70
>    Contact: <sip:22222238515000403 at 10.4.4.252>
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    Call-ID: 1-7019 at 10.4.4.252
>    CSeq: 1 INVITE
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Type: application/sdp
>    Content-Length:   131
>
>    v=0
>    o=user1 53655765 2353687637 IN IP4 10.4.4.252
>    s=-
>    c=IN IP4 10.4.4.252
>    t=0 0
>    m=audio 6000 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
>     
> ------------------------------------------------------------------------
> send 328 bytes to udp/[10.4.4.252]:5060 at 20:28:09.367634:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>    Call-ID: 1-7019 at 10.4.4.252
>    CSeq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
> send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.371759:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    To: "30003016094191500" <sip: 
> 30003016094191500 at 10.4.4.251>;tag=ygQBtp6QpKtcD
>    Call-ID: 1-7019 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla, include- 
> session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
> recv 401 bytes from udp/[10.4.4.252]:5060 at 20:28:09.371989:
>     
> ------------------------------------------------------------------------
>    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7019-1-3;rport
>    To: "30003016094191500"<sip: 
> 30003016094191500 at 10.4.4.251>;tag=ygQBtp6QpKtcD
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    Call-ID: 1-7019 at 10.4.4.252
>    CSeq: 1 ACK
>    Contact: sip:sipp at 10.4.4.252:5060
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
> send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.873045:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    To: "30003016094191500" <sip: 
> 30003016094191500 at 10.4.4.251>;tag=ygQBtp6QpKtcD
>    Call-ID: 1-7019 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla, include- 
> session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>
>
> This thing is driving me crazy, pls help.
>
> T.
>
>
>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 14:15:40 -0500
> Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't  
> understand ACK message
> In your scenario you need to add [peer_tag_param] at the end of the  
> to on the Ack.
>
> /b
>
> On Aug 24, 2009, at 2:03 PM, Tihomir Culjaga wrote:
>
>
>    
> ------------------------------------------------------------------------
> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
>    
> ------------------------------------------------------------------------
>   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>   Call-ID: 1-6962 at 10.4.4.252
>   CSeq: 1 ACK
>   Contact: sip:sipp at 10.4.4.252:5060
>   Max-Forwards: 70
>   Subject: Performance Test
>   Content-Length: 0
>
>
>
>
>
> ---------- Forwarded message ----------
> From: "Jerry Richards" <jerry.richards at teotech.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 24 Aug 2009 12:24:42 -0700
> Subject: [Freeswitch-users] Cannot create outgoing channel type  
> [error] cause: [FACILITY_NOT_SUBSCRIBED]
> Hello All,
>
> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP  
> machine
> for the first time using the Getting Started Guide.  I can register  
> three
> lines (1000, 1001, and 1002), but when I attempt to call one phone  
> to the
> other I hear the operator say:
>
> "The person at extension 1000 is not available..."
>
> Also, the Freeswitch log shows:
>
> Cannot create outgoing channel type [error] cause:
> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user]  
> cause:
> [FACILITY_NOT_SUBSCRIBED]
>
> Does anyone know why I get this error?
>
> Best Regards,
> Jerry
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 14:33:22 -0500
> Subject: Re: [Freeswitch-users] Cannot create outgoing channel type  
> [error] cause: [FACILITY_NOT_SUBSCRIBED]
> Are you trying to test everything on the same machine?
>
> /b
>
> On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote:
>
> Hello All,
>
> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP  
> machine
> for the first time using the Getting Started Guide.  I can register  
> three
> lines (1000, 1001, and 1002), but when I attempt to call one phone  
> to the
> other I hear the operator say:
>
> "The person at extension 1000 is not available..."
>
> Also, the Freeswitch log shows:
>
> Cannot create outgoing channel type [error] cause:
> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user]  
> cause:
> [FACILITY_NOT_SUBSCRIBED]
>
> Does anyone know why I get this error?
>
> Best Regards,
> Jerry
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 15:44:18 -0400
> Subject: Re: [Freeswitch-users] Problem with cnam.js?
> Every page on the wiki should be editable.  If you don't already  
> have an account, go to:
>
> http://wiki.freeswitch.org/index.php?title=Special:UserLogin&type=signup
>
> Mike
>
> On Aug 22, 2009, at 12:42 PM, Lars Zeb wrote:
>
>> I think there?s something wrong with the script at http://wiki.freeswitch.org/wiki/Examples_cnam.js 
>> .
>>
>> If you use it as is, it displays ?Content-type: text/html? for the  
>> effective_caller_id_name. In cnam.pl, the first two output lines  
>> are generated by:
>>
>> if (!$debug) {print "Content-type: text/html\n\n";}
>>
>> with the actual name in the third line.
>>
>> So I changed:
>>
>> fd.open("read");
>> buff = fd.readln();
>>
>> if(buff) {
>>    logger(buff, "info");
>>    session.setVariable("effective_caller_id_name", buff);
>> }
>>
>> To:
>>
>> fd.open("read");
>> buff = fd.readAll();
>>
>> if(buff[2]) {
>>    logger(buff, "info");
>>    session.setVariable("effective_caller_id_name", buff[2]);
>> }
>>
>> Or remove the print statement from cnam.pl.
>>
>> Sorry for the code, but the page was not editable.
>>
>> Lars
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 15:46:58 -0400
> Subject: Re: [Freeswitch-users] Yet another question about A500 + FS
> Do you have an answer in the dialplan for that extension?  Also,  
> check out the ignore_early_media variable.
>
> Mike
>
> On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote:
>
> Hi,
>
> I managed to get our A500 running with FreeSWITCH 1.0.4 stable using  
> wanpipe 3.4.4 drivers. But now I have another problem...
> I want to originate calls through event socket, and I only want to  
> receive ANSWERED(+OK) reply when the user actually answers.
>
> Now the situation is:
>
> ====================================
> originate openzap/1/a/123456 023
> 2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT:  
> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456] Ci= 
> [0000000000]
> 2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT  
> (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4
> 2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel  
> OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082]
> 2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer  
> OpenZAP/1:1/123456!
> API CALL [originate(openzap/1/a/123456 023)] output:
> +OK f8fca2be-8fa7-11de-9076-511e29dfc082
>
> 2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer  
> OpenZAP/1:1/123456 to XML[023 at default]
> freeswitch at emo-voip> 2009-08-23 08:44:06.743475 [INFO]  
> mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default
> 2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel  
> [OpenZAP/1:1/123456] has been answered
> 2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT  
> (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5
> 2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT  
> (N): CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6
> 2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup  
> OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT  
> (N): CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3
> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086  
> Session 2 (OpenZAP/1:1/123456) Ended
> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close  
> Channel OpenZAP/1:1/123456 [CS_DESTROY]
> ====================================
>
> Extension 023 is an IVR. As you can see FreeSWITCH answers the call  
> (2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel  
> [OpenZAP/1:1/123456] has been answered) 20 seconds before user  
> actually pick up the phone (2009-08-23 08:44:20.206010 [WARNING]  
> ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0  
> CSid=2 Seq=5).
>
> So Sangoma drivers/daemons report the events correctly.
> How can I set FreeSWITCH to answer after receiving RX EVENT (N):  
> CALL_ANSWERED from the driver?
>
> Thank you,
> V. Panayotov
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From MPeace at edcogroupinc.com  Mon Aug 24 13:42:10 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Mon, 24 Aug 2009 15:42:10 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>


I just installed FS on a fresh Centos 5.3 install, everything went perfect. I loaded x0lite up on a couple of hosts and they registered and worked fine. Then I moved the Centos server to a new subnet on out network that we have tons of boxes on and have all the routes already configured on and now the Sip phones will no longer register. No other changes have been made on the FreeSwitch server other than changing the IP.  BTW I can ping to and from the FS server and the hosts and the firewall is off on the FS server and the hosts.

I plugged a laptop with X-lite into a switch on the same network so the packets don't even need to route and it still won't register!

BTW  I also downloaded 3CX VoIP phone and it behaves the same way.

Any ideas?

TIA

=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From brian at freeswitch.org  Mon Aug 24 13:45:30 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 15:45:30 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
Message-ID: <8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>

Did you happen to hard code an IP in your vars.xml?

/b

On Aug 24, 2009, at 3:42 PM, Mike Peace wrote:

>
> I just installed FS on a fresh Centos 5.3 install, everything went  
> perfect. I loaded x0lite up on a couple of hosts and they registered  
> and worked fine. Then I moved the Centos server to a new subnet on  
> out network that we have tons of boxes on and have all the routes  
> already configured on and now the Sip phones will no longer  
> register. No other changes have been made on the FreeSwitch server  
> other than changing the IP.  BTW I can ping to and from the FS  
> server and the hosts and the firewall is off on the FS server and  
> the hosts.
>
> I plugged a laptop with X-lite into a switch on the same network so  
> the packets don?t even need to route and it still won?t register!
>
> BTW  I also downloaded 3CX VoIP phone and it behaves the same way.
>
> Any ideas?
>
> TIA
>

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From javieraristizabal at gmail.com  Mon Aug 24 13:46:09 2009
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 24 Aug 2009 15:46:09 -0500
Subject: [Freeswitch-users] pass-thru issue (G729)
Message-ID: <b6f11f300908241346k7526b910rf21a7cc30db363fa@mail.gmail.com>

Hello All,
I'm having an issue with g.729 pass-thru. On an older 1.0.3 install it's all
good, but on a newer machine, more cores, etc. the quality of voice on g.729
is very poor. Anyone here dealt with something like this?

With all other codecs are perfect, even in pass-thru...


Thanks in ddvance
Javier
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From matt at venturevoip.com  Mon Aug 24 13:47:42 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Tue, 25 Aug 2009 08:47:42 +1200
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <191c3a030908240705m5003a26bpd97b2c041d7b1526@mail.gmail.com>
References: <4A91D07C.1080805@venturevoip.com>	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>	<5a8712120908232127n6cb2f0bdq7e05b3d7778f0e70@mail.gmail.com>	<4A922E17.2040401@venturevoip.com>
	<191c3a030908240705m5003a26bpd97b2c041d7b1526@mail.gmail.com>
Message-ID: <4A92FC6E.5050005@venturevoip.com>

On 25/08/09 2:05 AM, Anthony Minessale wrote:
> I updated testclient.c so you can see how now.

Awesome awesome awesome!

Thanks heaps man - everybody here has been great!

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)


From MPeace at edcogroupinc.com  Mon Aug 24 14:01:00 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Mon, 24 Aug 2009 16:01:00 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>

I haven't changed any of the conf files.

Mike Peace
Network Analyst
EDCO, The Document People
Direct 417-447-3367

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, August 24, 2009 3:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

Did you happen to hard code an IP in your vars.xml?

/b

On Aug 24, 2009, at 3:42 PM, Mike Peace wrote:



I just installed FS on a fresh Centos 5.3 install, everything went perfect. I loaded x0lite up on a couple of hosts and they registered and worked fine. Then I moved the Centos server to a new subnet on out network that we have tons of boxes on and have all the routes already configured on and now the Sip phones will no longer register. No other changes have been made on the FreeSwitch server other than changing the IP.  BTW I can ping to and from the FS server and the hosts and the firewall is off on the FS server and the hosts.

I plugged a laptop with X-lite into a switch on the same network so the packets don't even need to route and it still won't register!

BTW  I also downloaded 3CX VoIP phone and it behaves the same way.

Any ideas?

TIA



=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From Prometheus001 at gmx.net  Mon Aug 24 14:17:15 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 24 Aug 2009 23:17:15 +0200
Subject: [Freeswitch-users] XML-RPC on different ip than 0.0.0.0
In-Reply-To: <191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
Message-ID: <4A93035B.5010907@gmx.net>

Any clue which one? I could not identify it, also looked into the sources.

Best regards
Peter

Anthony Minessale schrieb:
> there is a configuration option in the xml file to control which ip it
> binds to.
>
>
>
> On Mon, Aug 24, 2009 at 6:25 AM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Hello,
>
>     is there any chance to limit the listening ips of the xml-rpc server
>     (which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)?
>
>     Best regards
>     Peter
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From msc at freeswitch.org  Mon Aug 24 14:18:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Aug 2009 14:18:10 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
Message-ID: <87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>

On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com> wrote:

>  I haven?t changed any of the conf files.
>
>
What happens when you try to register? Do you get 480? (timeout) Or
something else?
-MC
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From MPeace at edcogroupinc.com  Mon Aug 24 14:26:46 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Mon, 24 Aug 2009 16:26:46 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>

"Registration error: 408-Request Timeout"

Mike Peace
Network Analyst
EDCO, The Document People
Direct 417-447-3367

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

I haven't changed any of the conf files.

What happens when you try to register? Do you get 480? (timeout) Or something else?
-MC

=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From msc at freeswitch.org  Mon Aug 24 14:34:29 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Aug 2009 14:34:29 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
Message-ID: <87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>

On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com> wrote:

>  ?Registration error: 408-Request Timeout?
>
Sorry for the the typo - 408 = timeout. (480 = temp unavail)

Try stopping iptables in Linux and try again. Sounds like something is
interfering with your packets getting from here to there...
Try:

/etc/init.d/iptables stop

And then see if your packets can move again.

-MC

>
>
> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:18 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> I haven?t changed any of the conf files.
>
>
>
> What happens when you try to register? Do you get 480? (timeout) Or
> something else?
> -MC
>
> ------------------------------
>
>  *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>  Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Aug 24 14:41:05 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 16:41:05 -0500
Subject: [Freeswitch-users] XML-RPC on different ip than 0.0.0.0
In-Reply-To: <4A93035B.5010907@gmx.net>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
Message-ID: <191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>

oh yeah, it's only the port they let you pick

it looks like they don't let you pick ip in the abyss code, it would require
an intrusive patch into that depend lib
to allow you to set that.


On Mon, Aug 24, 2009 at 4:17 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Any clue which one? I could not identify it, also looked into the sources.
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
> > there is a configuration option in the xml file to control which ip it
> > binds to.
> >
> >
> >
> > On Mon, Aug 24, 2009 at 6:25 AM, Peter P GMX <Prometheus001 at gmx.net
> > <mailto:Prometheus001 at gmx.net>> wrote:
> >
> >     Hello,
> >
> >     is there any chance to limit the listening ips of the xml-rpc server
> >     (which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)?
> >
> >     Best regards
> >     Peter
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From MPeace at edcogroupinc.com  Mon Aug 24 14:49:37 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Mon, 24 Aug 2009 16:49:37 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>

It does the same thing, does anything get set during the install that would remember or cache the old network settings? I can access anything from the FS server on any of several networks and vice-versa but the SIP will not register, again no firewalls are upon any of the test hosts.
Doesn't make sense to me.

Mike Peace
Network Analyst
EDCO, The Document People
Direct 417-447-3367

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

"Registration error: 408-Request Timeout"
Sorry for the the typo - 408 = timeout. (480 = temp unavail)

Try stopping iptables in Linux and try again. Sounds like something is interfering with your packets getting from here to there...
Try:

/etc/init.d/iptables stop

And then see if your packets can move again.

-MC



Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:18 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.





On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

I haven't changed any of the conf files.



What happens when you try to register? Do you get 480? (timeout) Or something else?
-MC

________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From telles-listas at devel-it.com.br  Mon Aug 24 15:00:11 2009
From: telles-listas at devel-it.com.br (Rodrigo P. Telles)
Date: Mon, 24 Aug 2009 19:00:11 -0300
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
In-Reply-To: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
References: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
Message-ID: <4A930D6B.2040202@devel-it.com.br>

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From raffaele.p.guidi at gmail.com  Mon Aug 24 15:06:18 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Tue, 25 Aug 2009 00:06:18 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
Message-ID: <b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>

This is what I was asking! :D When the installer finished it started the
whole thing and everything got loaded fine, but when I restarted my system
it didn't (and did not anymore). Well, I will try to install everything from
scratch again and see...

On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org> wrote:

> If you installed FreePBX then it would be that softwares job to manage
> the sofia profiles... wouldn't it?
>
> /b
>
> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>
> > Actually I did that and it worked fine. I had the problem the SECOND
> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
> > sip_profiles were not
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Mon Aug 24 15:13:47 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 00:13:47 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
Message-ID: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>

Hello Brian, Dave


Still nothing... i've changed ip_addresses (remote_ip, local_ip) and changed
branch within ACK message to meet INVITE's one....but it is still not
enough...
Also i checked RFC and this is how should it be ... (ACK without contact
taking care to have correct TAGs and branch)... what can it be?


   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 325 bytes to udp/[10.4.4.252]:5060 at 21:56:08.152812:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
send 718 bytes to udp/[10.4.4.252]:5060 at 21:56:08.159929:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=cFS6jHj9DgjjF
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 342 bytes from udp/[10.4.4.252]:5060 at 21:56:08.160166:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7079-1-0
   From: 22222238515000403 <sip:22222238515000403 at 110.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=cFS6jHj9DgjjF
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 ACK
   Max-Forwards: 70
   Content-Length: 0

   ------------------------------------------------------------------------
send 718 bytes to udp/[10.4.4.252]:5060 at 21:56:08.661299:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=cFS6jHj9DgjjF
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0





here is a scenario i use:


<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">


<scenario name="Basic Sipstone UAC">
  <send retrans="500">
    <![CDATA[

      INVITE sip:[service]@[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip];branch=[branch]
      Max-Forwards: 70
      Contact: <sip:[field1]@[local_ip]>
      From: [field1]
<sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
      To: [service] <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000

    ]]>
  </send>

  <recv response="100"
        optional="true">
  </recv>


  <recv response="302" rtd="true">
  </recv>

  <send>
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
      From: [field1]
<sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]
      To: [service]
<sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Max-Forwards: 70
      Content-Length: 0

    ]]>
  </send>

  <!-- definition of the response time repartition table (unit is ms)   -->
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

  <!-- definition of the call length repartition table (unit is ms)     -->
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>






---------- Forwarded message ----------
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Date: Mon, 24 Aug 2009 15:42:31 -0500
Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand
ACK message
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

Use that.. your scenario has some hard coded IP's in the fields that
shouldn't be there.

/b

On Aug 24, 2009, at 3:37 PM, Tihomir Culjaga wrote:

Hello Brian,

it doesn't work .. tried this today as well:
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From anthony.minessale at gmail.com  Mon Aug 24 15:20:45 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 17:20:45 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
Message-ID: <191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>

What is your exact sipp scenerio file and dialplan to this point after the
changes you were suggested to use.
please send both.

If we have to stop what we are doing to prove this works are you prepared to
offer your soul to help
document and other project maintenance?


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Mon Aug 24 15:31:11 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 00:31:11 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
Message-ID: <65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>

sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l
4000
scenario file:      uac_redirect.xml
FS dialplan:       public.xml
SIP trace:          trace.log


Here it is... sorry for not including at first...

T.



On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> What is your exact sipp scenerio file and dialplan to this point after the
> changes you were suggested to use.
> please send both.
>
> If we have to stop what we are doing to prove this works are you prepared
> to offer your soul to help
> document and other project maintenance?
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Aug 24 15:31:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Aug 2009 15:31:54 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
Message-ID: <87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>

On Mon, Aug 24, 2009 at 2:49 PM, Mike Peace <MPeace at edcogroupinc.com> wrote:

>  It does the same thing, does anything get set during the install that
> would remember or cache the old network settings? I can access anything from
> the FS server on any of several networks and vice-versa but the SIP will not
> register, again no firewalls are upon any of the test hosts.
>
> Doesn?t make sense to me.
>
>
>
Time to bust out tcpdump and/or wireshark to make 100% certain you know
what's happening with all those SIP packets.

> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:34 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> ?Registration error: 408-Request Timeout?
>
> Sorry for the the typo - 408 = timeout. (480 = temp unavail)
>
> Try stopping iptables in Linux and try again. Sounds like something is
> interfering with your packets getting from here to there...
> Try:
>
> /etc/init.d/iptables stop
>
> And then see if your packets can move again.
>
> -MC
>
>
>
> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:18 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> I haven?t changed any of the conf files.
>
>
>
> What happens when you try to register? Do you get 480? (timeout) Or
> something else?
> -MC
>
>
>  ------------------------------
>
>
>
> *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>
>
> Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
>
>  *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>  Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon Aug 24 15:48:42 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 00:48:42 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
Message-ID: <65d96fc80908241548g2a83b718y550efd87e431a700@mail.gmail.com>

... documentation hate that :)) ... but thats my life actually... thats what
i do for living :)

of course i can go forward and document a lot of things... a lot of
docummentation is pending to completing the project i currently have using
FS.

well ... my soul? .. is it really necessary?

T.


On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> What is your exact sipp scenerio file and dialplan to this point after the
> changes you were suggested to use.
> please send both.
>
> If we have to stop what we are doing to prove this works are you prepared
> to offer your soul to help
> document and other project maintenance?
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Aug 24 15:55:22 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 17:55:22 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241548g2a83b718y550efd87e431a700@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241548g2a83b718y550efd87e431a700@mail.gmail.com>
Message-ID: <191c3a030908241555x14edc34bu41bda4d3a9a48f4e@mail.gmail.com>

Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]

shouldn't that be

Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]


On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> ... documentation hate that :)) ... but thats my life actually... thats
> what i do for living :)
>
> of course i can go forward and document a lot of things... a lot of
> docummentation is pending to completing the project i currently have using
> FS.
>
> well ... my soul? .. is it really necessary?
>
> T.
>
>
> On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> What is your exact sipp scenerio file and dialplan to this point after the
>> changes you were suggested to use.
>> please send both.
>>
>> If we have to stop what we are doing to prove this works are you prepared
>> to offer your soul to help
>> document and other project maintenance?
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Mon Aug 24 16:14:23 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 01:14:23 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <191c3a030908241555x14edc34bu41bda4d3a9a48f4e@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241548g2a83b718y550efd87e431a700@mail.gmail.com>
	<191c3a030908241555x14edc34bu41bda4d3a9a48f4e@mail.gmail.com>
Message-ID: <65d96fc80908241614l60a9b6bbq3c77ae5caa2c11e5@mail.gmail.com>

this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0



http://sipp.sourceforge.net/doc3.0/reference.html

*[branch]* - Provide a branch value which is a concatenation of magic cookie
(z9hG4bK) + call number + message index in scenario.
An offset (like [branch-N]) can be appended if you need to have the same
branch value as a previous message.



http://www.ietf.org/rfc/rfc3665.txt :according to examples in RFC i se
branch being same for INVITE, 302 and ACK messages...



[branch-3] makes branch of ACK message same as INVITE....



here is a test with just [branch] ... pls note: Via: SIP/2.0/UDP
10.4.4.252:5060;branch=z9hG4bK-7093-1-3 from ACK message .... the 3rd
message...



freeswitch at l01sipindir1> recv 530 bytes from udp/[10.4.4.252]:5060 at
23:11:34.506681:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
   Call-ID: 1-7093 at 10.4.4.252
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 325 bytes to udp/[10.4.4.252]:5060 at 23:11:34.506957:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
   Call-ID: 1-7093 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
2009-08-25 01:11:34.505272 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/22222238515000403 at 10.4.4.252:5060[7797f5ae-9103-11de-95e3-cda626a03c4b]
2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing
22222238515000403->30003016094191500 in context public
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ########################
ServiceLookup ########################\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ########################
contact = '' ##############\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ########################
CallerNum = '38515000403' ##########\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ########################
RADIUS auth = '' ##########\n
2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing
22222238515000403->doRedirect in context public
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute
redirect(sip:12345616094191500 at pgw01.ot.hr:5060)
send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:34.515143:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 342 bytes from udp/[10.4.4.252]:5060 at 23:11:34.515369:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3
   From: 22222238515000403 <sip:22222238515000403 at 110.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093 at 10.4.4.252
   CSeq: 1 ACK
   Max-Forwards: 70
   Content-Length: 0

   ------------------------------------------------------------------------
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute
log(INFO ######################## RADIUS auth NOK!! ##########\n)
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ########################
RADIUS auth NOK!! ##########\n
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute
hangup(USER_BUSY)
2009-08-25 01:11:34.513275 [NOTICE] mod_dptools.c:633 Hangup sofia/internal/
22222238515000403 at 10.4.4.252:5060 [CS_EXECUTE] [USER_BUSY]
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1086 Session 3
(sofia/internal/22222238515000403 at 10.4.4.252:5060) Ended
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1088 Close Channel
sofia/internal/22222238515000403 at 10.4.4.252:5060 [CS_DESTROY]
send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:35.017059:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0




On Tue, Aug 25, 2009 at 12:55 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>
> shouldn't that be
>
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>
>
>
> On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> ... documentation hate that :)) ... but thats my life actually... thats
>> what i do for living :)
>>
>> of course i can go forward and document a lot of things... a lot of
>> docummentation is pending to completing the project i currently have using
>> FS.
>>
>> well ... my soul? .. is it really necessary?
>>
>> T.
>>
>>
>> On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> What is your exact sipp scenerio file and dialplan to this point after
>>> the changes you were suggested to use.
>>> please send both.
>>>
>>> If we have to stop what we are doing to prove this works are you prepared
>>> to offer your soul to help
>>> document and other project maintenance?
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon Aug 24 16:17:18 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 01:17:18 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241614l60a9b6bbq3c77ae5caa2c11e5@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241548g2a83b718y550efd87e431a700@mail.gmail.com>
	<191c3a030908241555x14edc34bu41bda4d3a9a48f4e@mail.gmail.com>
	<65d96fc80908241614l60a9b6bbq3c77ae5caa2c11e5@mail.gmail.com>
Message-ID: <65d96fc80908241617g169f261r706ef2fec4514650@mail.gmail.com>

Is there any way to make FS complain about what header is wrong ?

T.

On Tue, Aug 25, 2009 at 1:14 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0
>
>
>
> http://sipp.sourceforge.net/doc3.0/reference.html
>
> *[branch]* - Provide a branch value which is a concatenation of magic
> cookie (z9hG4bK) + call number + message index in scenario.
> An offset (like [branch-N]) can be appended if you need to have the same
> branch value as a previous message.
>
>
>
> http://www.ietf.org/rfc/rfc3665.txt :according to examples in RFC i se
> branch being same for INVITE, 302 and ACK messages...
>
>
>
> [branch-3] makes branch of ACK message same as INVITE....
>
>
>
> here is a test with just [branch] ... pls note: Via: SIP/2.0/UDP
> 10.4.4.252:5060;branch=z9hG4bK-7093-1-3 from ACK message .... the 3rd
> message...
>
>
>
> freeswitch at l01sipindir1> recv 530 bytes from udp/[10.4.4.252]:5060 at
> 23:11:34.506681:
>    ------------------------------------------------------------------------
>    INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
>    Max-Forwards: 70
>    Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
> >
>    From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
>    To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
>    Call-ID: 1-7093 at 10.4.4.252
>    CSeq: 1 INVITE
>    Content-Type: application/sdp
>    Content-Length:   131
>
>    v=0
>    o=user1 53655765 2353687637 IN IP4 10.4.4.252
>    s=-
>    c=IN IP4 10.4.4.252
>    t=0 0
>    m=audio 6000 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
>    ------------------------------------------------------------------------
> send 325 bytes to udp/[10.4.4.252]:5060 at 23:11:34.506957:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
>    From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
>    To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
>    Call-ID: 1-7093 at 10.4.4.252
>    CSeq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2009-08-25 01:11:34.505272 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/22222238515000403 at 10.4.4.252:5060[7797f5ae-9103-11de-95e3-cda626a03c4b]
> 2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing
> 22222238515000403->30003016094191500 in context public
> 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932
> ######################## ServiceLookup ########################\n
> 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932
> ######################## contact = '' ##############\n
> 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932
> ######################## CallerNum = '38515000403' ##########\n
> 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932
> ######################## RADIUS auth = '' ##########\n
> 2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing
> 22222238515000403->doRedirect in context public
> 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute
> redirect(sip:12345616094191500 at pgw01.ot.hr:5060)
> send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:34.515143:
>    ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
>    From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
>    To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
> >;tag=y38ac8m1m6e0K
>    Call-ID: 1-7093 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 342 bytes from udp/[10.4.4.252]:5060 at 23:11:34.515369:
>    ------------------------------------------------------------------------
>    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3
>    From: 22222238515000403 <sip:22222238515000403 at 110.4.4.252:5060>;tag=1
>    To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
> >;tag=y38ac8m1m6e0K
>    Call-ID: 1-7093 at 10.4.4.252
>    CSeq: 1 ACK
>    Max-Forwards: 70
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute
> log(INFO ######################## RADIUS auth NOK!! ##########\n)
> 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932
> ######################## RADIUS auth NOK!! ##########\n
> 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute
> hangup(USER_BUSY)
> 2009-08-25 01:11:34.513275 [NOTICE] mod_dptools.c:633 Hangup
> sofia/internal/22222238515000403 at 10.4.4.252:5060 [CS_EXECUTE] [USER_BUSY]
> 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1086 Session 3
> (sofia/internal/22222238515000403 at 10.4.4.252:5060) Ended
> 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1088 Close
> Channel sofia/internal/22222238515000403 at 10.4.4.252:5060 [CS_DESTROY]
> send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:35.017059:
>    ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
>    From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
>    To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
> >;tag=y38ac8m1m6e0K
>    Call-ID: 1-7093 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>
>
>
> On Tue, Aug 25, 2009 at 12:55 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>>
>> shouldn't that be
>>
>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>
>>
>>
>> On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> ... documentation hate that :)) ... but thats my life actually... thats
>>> what i do for living :)
>>>
>>> of course i can go forward and document a lot of things... a lot of
>>> docummentation is pending to completing the project i currently have using
>>> FS.
>>>
>>> well ... my soul? .. is it really necessary?
>>>
>>> T.
>>>
>>>
>>> On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> What is your exact sipp scenerio file and dialplan to this point after
>>>> the changes you were suggested to use.
>>>> please send both.
>>>>
>>>> If we have to stop what we are doing to prove this works are you
>>>> prepared to offer your soul to help
>>>> document and other project maintenance?
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From rogelio.perez at gmail.com  Mon Aug 24 16:52:47 2009
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Mon, 24 Aug 2009 20:52:47 -0300
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <3BF035D5-ED50-406E-8D19-FAAD314BB045@freeswitch.org>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
	<3BF035D5-ED50-406E-8D19-FAAD314BB045@freeswitch.org>
Message-ID: <9C2E3DC8-3A13-4D2F-A55C-4AAA88FC90CC@gmail.com>

Hi Raymond,

I'm not planning to have more than 10 concurrent calls on this  
devices, but I'm also curious as you about how many calls can it handle.
When I get to that point I will post the test results on this list.

Regards,
Rogelio


On Aug 23, 2009, at 8:23 PM, Raymond Chandler wrote:

> Could you load freeswitch with a couple hundred calls then run the  
> test again.. and do the same to asterisk and see how the numbers  
> stack up then? I'm just curious to see what happens at that point.
>
> -Ray
>
> On Aug 21, 2009, at 3:15 PM, Rogelio Perez wrote:
>
>> Hi Everyone,
>>
>> I'm working on a PBX project for the Sheevaplug ARM based computer,  
>> with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
>> So far I've found a big difference between Freeswitch and Asterisk  
>> performance times.
>> This is a comparison of the time it takes them to perform different  
>> actions:
>>
>> startup Freeswitch:			3 min.
>> startup Asterisk:			2 sec.
>>
>> call extension Freeswitch:	6 sec.
>> call extension Asterisk:		0 sec.
>>
>> shutdown Freeswitch:		6.5 sec
>> shutdown Asterisk:			0 sec.
>>
>> reload config Freeswitch: 	1 sec.
>> reload config Asterisk:		1 sec.
>>
>> Both were built from sources natively (no cross-compiling), and  
>> they use the default startup configurations.
>> I have managed to lower the Freeswitch times by disabling most of  
>> the modules and recompiling, but it is still far away from Asterisk  
>> (i.e. FS startup time 2.5 min).
>>
>> 1. Is there any way to further improve Freeswitch performance for  
>> the ARM architecture?
>> 2. Can this be related to the lack of a FPU (the Sheevalug emulates  
>> the floating point operations).
>> 3. On the startup I see this error repeated many times: [ERR]  
>> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be  
>> related?
>>
>> Thanks,
>> Rogelio Perez
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mayamatakeshi at gmail.com  Mon Aug 24 18:52:27 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 25 Aug 2009 10:52:27 +0900
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
Message-ID: <15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>

On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com> wrote:
>
> sipp_cmd:???????? sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l
> 4000
> scenario file:????? uac_redirect.xml
> FS dialplan:?????? public.xml
> SIP trace:????????? trace.log

The Via definition in your SIPp scenario differs between the INVITE and the ACK:

INVITE:
Via: SIP/2.0/[transport] [local_ip];branch=[branch]

ACK:
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]


In the INVITE, you are not adding the [local_port] as you do in the ACK.
Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK.
So it seems FS is not checking just the ACK's branch against the
INVITE's; it seems it is checking the whole Via header.
I don't know if this is in accordance to SIP specs.

Another thing, about the way you are calling SIPp: do no use "-sn uac"
and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
means "use the internal (embedded) scenario named xxx". So this
conflicts with the other parameter "-sf" which specifies an external
profile. It seems this doesn't cause any problem (probably because in
the sipp startup, -sf overrides -sn), but it is misleading.

regards,
takeshi


From mayamatakeshi at gmail.com  Mon Aug 24 18:57:36 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 25 Aug 2009 10:57:36 +0900
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
	<15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>
Message-ID: <15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com>

On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com> wrote:
> On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com> wrote:
>>
>> sipp_cmd:???????? sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
>> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l
>> 4000
>> scenario file:????? uac_redirect.xml
>> FS dialplan:?????? public.xml
>> SIP trace:????????? trace.log
>
> The Via definition in your SIPp scenario differs between the INVITE and the ACK:
>
> INVITE:
> Via: SIP/2.0/[transport] [local_ip];branch=[branch]
>
> ACK:
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>
>
> In the INVITE, you are not adding the [local_port] as you do in the ACK.
> Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK.
> So it seems FS is not checking just the ACK's branch against the
> INVITE's; it seems it is checking the whole Via header.
> I don't know if this is in accordance to SIP specs.
> Another thing, about the way you are calling SIPp: do no use "-sn uac"
> and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
> means "use the internal (embedded) scenario named xxx". So this
> conflicts with the other parameter "-sf" which specifies an external
> profile.

I mean, an external scenario (file).

 It seems this doesn't cause any problem (probably because in
> the sipp startup, -sf overrides -sn), but it is misleading.
>
> regards,
> takeshi
>


From gmaruzz at celliax.org  Mon Aug 24 19:56:45 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 04:56:45 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com> 
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com> 
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com> 
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
Message-ID: <7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>

Windows installer does not work for me.

I've reinstalled various times, same results.

I can correctly create a number, but when I try to create a device for
that number, it tells me that cannot locate the device, and the
password for vicemail will be invalid.

After that, it begins to give the php error page, it cannot find the
start < tag in directory/default.xml

Also for me there are no sofia profiles...

So, I cannot start to test it (eg: I would like to add mod_skypiax
support to it).


Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
Guidi<raffaele.p.guidi at gmail.com> wrote:
> This is what I was asking! :D When the installer finished it started the
> whole thing and everything got loaded fine, but when I restarted my system
> it didn't (and did not anymore). Well, I will try to install everything from
> scratch again and see...
>
> On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org> wrote:
>>
>> If you installed FreePBX then it would be that softwares job to manage
>> the sofia profiles... wouldn't it?
>>
>> /b
>>
>> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>>
>> > Actually I did that and it worked fine. I had the problem the SECOND
>> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
>> > sip_profiles were not
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From tparikh at gmail.com  Mon Aug 24 21:56:45 2009
From: tparikh at gmail.com (Tapan Parikh)
Date: Mon, 24 Aug 2009 21:56:45 -0700
Subject: [Freeswitch-users] compile freeswitch, lua, luasql for 64-bit arch
Message-ID: <1ecdcb6a0908242156l65cb60fk6c096760da0a2d8e@mail.gmail.com>

Hi -

Does anyone know how to compile freeswitch for a 64-bit architecture
on mac os x?

MySQL on MAC OS X only comes pre-compiled for 64-bit.  I compiled
luasql against this libmysqlclient.dylib.  Now when I call lua from
freeswitch I get:

2009-08-24 21:06:44.370439 [ERR] mod_lua.cpp:182 error loading module
'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
	dlopen(/usr/local/lib/lua/5.1/luasql/mysql.so, 2): no suitable image
found.  Did find:
	/usr/local/lib/lua/5.1/luasql/mysql.so: mach-o, but wrong architecture
stack traceback:
	[C]: ?
	[C]: in function 'require'
	/usr/local/freeswitch/scripts/otalo.lua:3: in main chun

Any pointers?


From panayotov.vd at gmail.com  Mon Aug 24 23:22:37 2009
From: panayotov.vd at gmail.com (Vassil Panayotov)
Date: Tue, 25 Aug 2009 09:22:37 +0300
Subject: [Freeswitch-users] Yet another question about A500 + FS
In-Reply-To: <56CFD144-94D0-42CE-B59D-7DC524C1919D@jerris.com>
References: <8a9b664c0908222321r20c5c4d8wd08721dc8955f62c@mail.gmail.com>
	<56CFD144-94D0-42CE-B59D-7DC524C1919D@jerris.com>
Message-ID: <8a9b664c0908242322t652c7120ob76bb984597402d@mail.gmail.com>

Thank you for you reply Mike!

'ignore_early_media=true' variable setting is the solution, but I figured it
out shortly after posting to the ML.

Best regards,
V. Panayotov

On Mon, Aug 24, 2009 at 10:46 PM, Michael Jerris <mike at jerris.com> wrote:

> Do you have an answer in the dialplan for that extension?  Also, check
> out the ignore_early_media variable.
>
> Mike
>
> On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote:
>
> > Hi,
> >
> > I managed to get our A500 running with FreeSWITCH 1.0.4 stable using
> > wanpipe 3.4.4 drivers. But now I have another problem...
> > I want to originate calls through event socket, and I only want to
> > receive ANSWERED(+OK) reply when the user actually answers.
> >
> > Now the situation is:
> >
> > ====================================
> > originate openzap/1/a/123456 023
> > 2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT:
> > CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456] Ci=
> > [0000000000]
> > 2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT
> > (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4
> > 2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel
> > OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082]
> > 2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer
> > OpenZAP/1:1/123456!
> > API CALL [originate(openzap/1/a/123456 023)] output:
> > +OK f8fca2be-8fa7-11de-9076-511e29dfc082
> >
> > 2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer
> > OpenZAP/1:1/123456 to XML[023 at default]
> > freeswitch at emo-voip> 2009-08-23 08:44:06.743475 [INFO]
> > mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default
> > 2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
> > [OpenZAP/1:1/123456] has been answered
> > 2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT
> > (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5
> > 2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT
> > (N): CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6
> > 2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup
> > OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING]
> > 2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT
> > (N): CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3
> > 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086
> > Session 2 (OpenZAP/1:1/123456) Ended
> > 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close
> > Channel OpenZAP/1:1/123456 [CS_DESTROY]
> > ====================================
> >
> > Extension 023 is an IVR. As you can see FreeSWITCH answers the call
> > (2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
> > [OpenZAP/1:1/123456] has been answered) 20 seconds before user
> > actually pick up the phone (2009-08-23 08:44:20.206010 [WARNING]
> > ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0
> > CSid=2 Seq=5).
> >
> > So Sangoma drivers/daemons report the events correctly.
> > How can I set FreeSWITCH to answer after receiving RX EVENT (N):
> > CALL_ANSWERED from the driver?
> >
> > Thank you,
> > V. Panayotov
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Mon Aug 24 23:51:22 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 08:51:22 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
	<15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>
	<15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com>
Message-ID: <65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com>

Hello Takeshi,

Thanks for your hint... it worked out... so to be precise:

VIA header of both INVITE and ACK messages MUST be identical (IP:PORT +
branch)... and you are right... it might not be according to SIP
specification. Anyhow, i get FS understand my ACK message.


Finally, here is what i used and I'm getting some poor results .. but this
is another topic :)


Thanks for your help.
Tihomir.


sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1
-rp 100 -trace_msg -inf test.txt -m 20000 -l 4000


<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">


<scenario name="Basic Sipstone UAC">
  <send retrans="500" start_rtd="1" start_rtd="2">

    <![CDATA[

      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      Max-Forwards: 70
      Contact: <sip:[field1]@[local_ip]>
      From: [field1]
<sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
      To: [service] <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000

    ]]>
  </send>

  <recv response="100"
        optional="true" rtd="1">
  </recv>


  <recv response="302" rtd="2">
  </recv>

  <send>
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
      From: [field1]
<sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]
      To: [service]
<sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Max-Forwards: 70
      Content-Length: 0

    ]]>
  </send>

  <!-- definition of the response time repartition table (unit is ms)   -->
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

  <!-- definition of the call length repartition table (unit is ms)     -->
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>



On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com>
> wrote:
> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com>
> wrote:
> >>
> >> sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1
> -l
> >> 4000
> >> scenario file:      uac_redirect.xml
> >> FS dialplan:       public.xml
> >> SIP trace:          trace.log
> >
> > The Via definition in your SIPp scenario differs between the INVITE and
> the ACK:
> >
> > INVITE:
> > Via: SIP/2.0/[transport] [local_ip];branch=[branch]
> >
> > ACK:
> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
> >
> >
> > In the INVITE, you are not adding the [local_port] as you do in the ACK.
> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept the
> ACK.
> > So it seems FS is not checking just the ACK's branch against the
> > INVITE's; it seems it is checking the whole Via header.
> > I don't know if this is in accordance to SIP specs.
> > Another thing, about the way you are calling SIPp: do no use "-sn uac"
> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
> > means "use the internal (embedded) scenario named xxx". So this
> > conflicts with the other parameter "-sf" which specifies an external
> > profile.
>
> I mean, an external scenario (file).
>
>  It seems this doesn't cause any problem (probably because in
> > the sipp startup, -sf overrides -sn), but it is misleading.
> >
> > regards,
> > takeshi
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From juanbackson at gmail.com  Tue Aug 25 01:21:58 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Tue, 25 Aug 2009 16:21:58 +0800
Subject: [Freeswitch-users] reload user data
Message-ID: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>

Hello,

I would like to dynamically add user to freeswitch.  If I add a new file to
the directory dir, is there anyway to have freeswitch to read the new user
xml file without having to restart freeswitch?

Other than using flat file, is there anyway to add user to freeswitch user
api command?

Thanks,
JB
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From krice at freeswitch.org  Tue Aug 25 01:30:17 2009
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 25 Aug 2009 03:30:17 -0500
Subject: [Freeswitch-users] reload user data
In-Reply-To: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>
Message-ID: <C6B90B49.5D5AC%krice@freeswitch.org>

You just need to reloadxml you don?t have to restart the whole thing. You
can also use xml_curl to feed the users from a database see my contrib
directory (contrib/swk) for some example scripts and db code



From: Juan Backson <juanbackson at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Tue, 25 Aug 2009 16:21:58 +0800
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] reload user data

Hello,
?
I would like to dynamically add user to freeswitch.? If I add a new file to
the directory dir, is there anyway to have freeswitch to read the new user
xml file without having to restart freeswitch?
?
Other than using flat file, is there anyway to add user to freeswitch user
api command?
?
Thanks,
JB


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From tculjaga at gmail.com  Tue Aug 25 01:54:07 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 10:54:07 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting server
Message-ID: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>

Hello,

i'm trying to use freeswitch as a redirecting server meaning FS has to
receive an INVITE and according to some rules it will redirect calls to
other destinations.


CALLING_USER                FREESWITCH                        SOMEWHERE

INVITE ------------------------------->
           <------------------------------ 100 Trying
           <------------------------------ 302 Moved Temporary
ACK    ------------------------------->
INVITE--------------------------------------------------------------------------------->



Well, wverything works well except i have perfromance issues .... on my HW
FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary). When
i increase the rate, FS starts delaying 302 response. Right at 50 CPS i see
"calls" being build up in FS and the delay begining to grow.

When i observe the machine, load average is almost nothing (load average:
1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread taking
most load... all others are just sitting there with 1-5 % CPU time.
This looks to me as FS handles 302 messages in a single thread?!?!


tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H

top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41, 0.61, 0.60
Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,  0.2%si,
0.0%st
Mem:   2074520k total,   571244k used,  1503276k free,   259604k buffers
Swap:  2650684k total,     3020k used,  2647664k free,   153868k cached

  PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
COMMAND
 4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
freeswitch
 4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
freeswitch
 4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
freeswitch
 4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
freeswitch
 4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
freeswitch
 4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
freeswitch
 4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
freeswitch
 4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
freeswitch
 4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
freeswitch
 4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50 freeswitch



cat /proc/cpuinfo
processor       : 0
vendor_id       : GenuineIntel
cpu family      : 6
model           : 15
model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
stepping        : 6
cpu MHz         : 2333.560
cache size      : 4096 KB
physical id     : 0
siblings        : 2
core id         : 0
cpu cores       : 2
apicid          : 0
initial apicid  : 0
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16
xtpr dca lahf_lm
bogomips        : 4670.78
clflush size    : 64
power management:

processor       : 1
vendor_id       : GenuineIntel
cpu family      : 6
model           : 15
model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
stepping        : 6
cpu MHz         : 2333.560
cache size      : 4096 KB
physical id     : 0
siblings        : 2
core id         : 1
cpu cores       : 2
apicid          : 1
initial apicid  : 1
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16
xtpr dca lahf_lm
bogomips        : 4666.82
clflush size    : 64
power management:



uname -a
Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009 i686
GNU/Linux



Of course, i've tuned the machine up

ulimit -c unlimited
ulimit -d unlimited
ulimit -f unlimited
ulimit -i unlimited
ulimit -n 999999
ulimit -q unlimited
ulimit -u unlimited
ulimit -v unlimited
ulimit -x unlimited
ulimit -s 240
ulimit -l unlimited
ulimit -a


Started FS with minimum modules but still 40 CPS seems to be the limit.


So, is there any way to improve performance?


Tihomir.
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From ivanov.maxim at gmail.com  Tue Aug 25 01:57:08 2009
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Tue, 25 Aug 2009 12:57:08 +0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
Message-ID: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>

Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
,
the whole sting is tooo long, is there any shorter way to write same thing? Like
"sofia/gateway/panas*/1000" will try all gateways matching the pattern.


From gmaruzz at celliax.org  Tue Aug 25 02:00:06 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 11:00:06 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
Message-ID: <7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>

Maybe your load comes from disk access?

Try putting the sql and log directories on a ramdisk.

OTH,

-giovanni

On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com> wrote:
> Hello,
>
> i'm trying to use freeswitch as a redirecting server meaning FS has to
> receive an INVITE and according to some rules it will redirect calls to
> other destinations.
>
>
> CALLING_USER??????????????? FREESWITCH??????????????????????? SOMEWHERE
>
> INVITE ------------------------------->
> ?????????? <------------------------------ 100 Trying
> ?????????? <------------------------------ 302 Moved Temporary
> ACK ?? ------------------------------->
> INVITE--------------------------------------------------------------------------------->
>
>
>
> Well, wverything works well except i have perfromance issues .... on my HW
> FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary). When
> i increase the rate, FS starts delaying 302 response. Right at 50 CPS i see
> "calls" being build up in FS and the delay begining to grow.
>
> When i observe the machine, load average is almost nothing (load average:
> 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread taking
> most load... all others are just sitting there with 1-5 % CPU time.
> This looks to me as FS handles 302 messages in a single thread?!?!
>
>
> tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>
> top - 10:41:37 up 167 days, 20:42,? 3 users,? load average: 1.41, 0.61, 0.60
> Tasks:? 83 total,?? 2 running,? 81 sleeping,?? 0 stopped,?? 0 zombie
> Cpu(s): 25.3%us,? 1.5%sy,? 0.0%ni, 30.3%id, 42.7%wa,? 0.0%hi,? 0.2%si,
> 0.0%st
> Mem:?? 2074520k total,?? 571244k used,? 1503276k free,?? 259604k buffers
> Swap:? 2650684k total,???? 3020k used,? 2647664k free,?? 153868k cached
>
> ? PID USER????? PR? NI? VIRT? RES? SHR S %CPU %MEM??? TIME+
> COMMAND
> ?4814 root????? 20?? 0 34188? 20m 3780 S?? 38? 1.0?? 3:10.29
> freeswitch
> ?4800 root????? 20?? 0 34188? 20m 3780 S??? 6? 1.0?? 0:08.26
> freeswitch
> ?4798 root????? 20?? 0 34188? 20m 3780 R??? 5? 1.0?? 0:24.46
> freeswitch
> ?4787 root????? 20?? 0 34188? 20m 3780 S??? 2? 1.0?? 0:11.24
> freeswitch
> ?4794 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.42
> freeswitch
> ?4803 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.74
> freeswitch
> ?4788 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.96
> freeswitch
> ?4804 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.64
> freeswitch
> ?4807 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.68
> freeswitch
> ?4811 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.50 freeswitch
>
>
>
> cat /proc/cpuinfo
> processor?????? : 0
> vendor_id?????? : GenuineIntel
> cpu family????? : 6
> model?????????? : 15
> model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
> stepping??????? : 6
> cpu MHz???????? : 2333.560
> cache size????? : 4096 KB
> physical id???? : 0
> siblings??????? : 2
> core id???????? : 0
> cpu cores?????? : 2
> apicid????????? : 0
> initial apicid? : 0
> fdiv_bug??????? : no
> hlt_bug???????? : no
> f00f_bug??????? : no
> coma_bug??????? : no
> fpu???????????? : yes
> fpu_exception?? : yes
> cpuid level???? : 10
> wp????????????? : yes
> flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
> cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16
> xtpr dca lahf_lm
> bogomips??????? : 4670.78
> clflush size??? : 64
> power management:
>
> processor?????? : 1
> vendor_id?????? : GenuineIntel
> cpu family????? : 6
> model?????????? : 15
> model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
> stepping??????? : 6
> cpu MHz???????? : 2333.560
> cache size????? : 4096 KB
> physical id???? : 0
> siblings??????? : 2
> core id???????? : 1
> cpu cores?????? : 2
> apicid????????? : 1
> initial apicid? : 1
> fdiv_bug??????? : no
> hlt_bug???????? : no
> f00f_bug??????? : no
> coma_bug??????? : no
> fpu???????????? : yes
> fpu_exception?? : yes
> cpuid level???? : 10
> wp????????????? : yes
> flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
> cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16
> xtpr dca lahf_lm
> bogomips??????? : 4666.82
> clflush size??? : 64
> power management:
>
>
>
> uname -a
> Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009 i686
> GNU/Linux
>
>
>
> Of course, i've tuned the machine up
>
> ulimit -c unlimited
> ulimit -d unlimited
> ulimit -f unlimited
> ulimit -i unlimited
> ulimit -n 999999
> ulimit -q unlimited
> ulimit -u unlimited
> ulimit -v unlimited
> ulimit -x unlimited
> ulimit -s 240
> ulimit -l unlimited
> ulimit -a
>
>
> Started FS with minimum modules but still 40 CPS seems to be the limit.
>
>
> So, is there any way to improve performance?
>
>
> Tihomir.
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From juanbackson at gmail.com  Tue Aug 25 02:22:54 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Tue, 25 Aug 2009 17:22:54 +0800
Subject: [Freeswitch-users] reload user data
In-Reply-To: <C6B90B49.5D5AC%krice@freeswitch.org>
References: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>
	<C6B90B49.5D5AC%krice@freeswitch.org>
Message-ID: <27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>

Hi Ken,

xml_curl is a great idea.  Is there anyway to not having to setup another
HTTP server?  For instance, can I have freeswitch to call an api or call a
lua or php or c script that will return the xml response?  That way, I don't
need to maintain yet another service.

Thanks,
JB

On Tue, Aug 25, 2009 at 4:30 PM, Ken Rice <krice at freeswitch.org> wrote:

>  You just need to reloadxml you don?t have to restart the whole thing. You
> can also use xml_curl to feed the users from a database see my contrib
> directory (contrib/swk) for some example scripts and db code
>
>
> ------------------------------
> *From: *Juan Backson <juanbackson at gmail.com>
> *Reply-To: *<freeswitch-users at lists.freeswitch.org>
> *Date: *Tue, 25 Aug 2009 16:21:58 +0800
> *To: *<freeswitch-users at lists.freeswitch.org>
> *Subject: *[Freeswitch-users] reload user data
>
>
> Hello,
>
> I would like to dynamically add user to freeswitch.  If I add a new file to
> the directory dir, is there anyway to have freeswitch to read the new user
> xml file without having to restart freeswitch?
>
> Other than using flat file, is there anyway to add user to freeswitch user
> api command?
>
> Thanks,
> JB
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jim at evolutiontel.net  Tue Aug 25 02:30:37 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Tue, 25 Aug 2009 19:30:37 +1000
Subject: [Freeswitch-users] reload user data
In-Reply-To: <27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>
References: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>
	<C6B90B49.5D5AC%krice@freeswitch.org>
	<27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>
Message-ID: <ddad80610908250230y373207d3j85f201e73a834dc5@mail.gmail.com>

Is this what you are after?

http://wiki.freeswitch.org/wiki/Mod_xml_odbc

Cheers,
Jim

On Tue, Aug 25, 2009 at 7:22 PM, Juan Backson<juanbackson at gmail.com> wrote:
> Hi Ken,
>
> xml_curl is a great idea.? Is there anyway to not having to setup another
> HTTP server?? For instance, can I have freeswitch to call an api or call a
> lua or php or c script that will return the xml response?? That way, I don't
> need to maintain yet another service.
>
> Thanks,
> JB
>
> On Tue, Aug 25, 2009 at 4:30 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> You just need to reloadxml you don?t have to restart the whole thing. You
>> can also use xml_curl to feed the users from a database see my contrib
>> directory (contrib/swk) for some example scripts and db code
>>
>>
>> ________________________________
>> From: Juan Backson <juanbackson at gmail.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Tue, 25 Aug 2009 16:21:58 +0800
>> To: <freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] reload user data
>>
>> Hello,
>>
>> I would like to dynamically add user to freeswitch.? If I add a new file
>> to the directory dir, is there anyway to have freeswitch to read the new
>> user xml file without having to restart freeswitch?
>>
>> Other than using flat file, is there anyway to add user to freeswitch user
>> api command?
>>
>> Thanks,
>> JB
>>
>> ________________________________
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From leon at scarlet-internet.nl  Tue Aug 25 03:00:57 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Tue, 25 Aug 2009 12:00:57 +0200
Subject: [Freeswitch-users] reload user data
In-Reply-To: <ddad80610908250230y373207d3j85f201e73a834dc5@mail.gmail.com>
References: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>
	<C6B90B49.5D5AC%krice@freeswitch.org>
	<27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>
	<ddad80610908250230y373207d3j85f201e73a834dc5@mail.gmail.com>
Message-ID: <8685DDD8-F246-4F3C-91FF-420BA108680F@scarlet-internet.nl>

Hi,

I wrote that module, but been on vacation for a while :-) It's not  
really finished yet, but it worked well for generating user directory  
xml..

Some things that still need to be done:
- Fix it so that reloadxml works
- Don't write the generated xml always to disk before returning it to fs
- Surely there are bugs that need to be fixed

Did anyone try it yet ? If so, what is your experience ?

Regards,

Leon




On Aug 25, 2009, at 11:30 AM, Jim Burke wrote:

> Is this what you are after?
>
> http://wiki.freeswitch.org/wiki/Mod_xml_odbc
>
> Cheers,
> Jim
>
> On Tue, Aug 25, 2009 at 7:22 PM, Juan Backson<juanbackson at gmail.com>  
> wrote:
>> Hi Ken,
>>
>> xml_curl is a great idea.  Is there anyway to not having to setup  
>> another
>> HTTP server?  For instance, can I have freeswitch to call an api or  
>> call a
>> lua or php or c script that will return the xml response?  That  
>> way, I don't
>> need to maintain yet another service.
>>
>> Thanks,
>> JB
>>
>> On Tue, Aug 25, 2009 at 4:30 PM, Ken Rice <krice at freeswitch.org>  
>> wrote:
>>>
>>> You just need to reloadxml you don?t have to restart the whole  
>>> thing. You
>>> can also use xml_curl to feed the users from a database see my  
>>> contrib
>>> directory (contrib/swk) for some example scripts and db code
>>>
>>>
>>> ________________________________
>>> From: Juan Backson <juanbackson at gmail.com>
>>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>>> Date: Tue, 25 Aug 2009 16:21:58 +0800
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Subject: [Freeswitch-users] reload user data
>>>
>>> Hello,
>>>
>>> I would like to dynamically add user to freeswitch.  If I add a  
>>> new file
>>> to the directory dir, is there anyway to have freeswitch to read  
>>> the new
>>> user xml file without having to restart freeswitch?
>>>
>>> Other than using flat file, is there anyway to add user to  
>>> freeswitch user
>>> api command?
>>>
>>> Thanks,
>>> JB
>>>
>>> ________________________________
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From tculjaga at gmail.com  Tue Aug 25 06:19:41 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 15:19:41 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
Message-ID: <65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>

Hey Giovanni,

thanks for the tip... indeed the db files were heavily used regardless if i
started freeswitch with nosql option (freeswitch -nosql)... FS was not
writing anything into that files ... instead it was just accessing it....
This behaviour leads to a waste of 40% CPU time... waiting for other
processes (mainly disk access) to finish!!!

I moved freeswitch/db/ to a ramdisk and the performance got a boost to 140
CPS with a CPU load of 80%. I was keeping the machine for a while (20 - 30
minutes) on that rate when i sow CPU suddenly went to 100% and FS becoming
irresponsive :).


What can be wrong?
What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
cross?
What fine tuning do we need in order to asure a long high load run?



Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have sense
to move my OS to 64 bit? ... will FS gain more preformance ?... I mean will
FS perofomr drastically better 20%+ ?


Tihomir.


On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> Maybe your load comes from disk access?
>
> Try putting the sql and log directories on a ramdisk.
>
> OTH,
>
> -giovanni
>
> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com>
> wrote:
> > Hello,
> >
> > i'm trying to use freeswitch as a redirecting server meaning FS has to
> > receive an INVITE and according to some rules it will redirect calls to
> > other destinations.
> >
> >
> > CALLING_USER                FREESWITCH                        SOMEWHERE
> >
> > INVITE ------------------------------->
> >            <------------------------------ 100 Trying
> >            <------------------------------ 302 Moved Temporary
> > ACK    ------------------------------->
> >
> INVITE--------------------------------------------------------------------------------->
> >
> >
> >
> > Well, wverything works well except i have perfromance issues .... on my
> HW
> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary).
> When
> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS i
> see
> > "calls" being build up in FS and the delay begining to grow.
> >
> > When i observe the machine, load average is almost nothing (load average:
> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
> taking
> > most load... all others are just sitting there with 1-5 % CPU time.
> > This looks to me as FS handles 302 messages in a single thread?!?!
> >
> >
> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
> >
> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41, 0.61,
> 0.60
> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,  0.2%si,
> > 0.0%st
> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k buffers
> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k cached
> >
> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
> > COMMAND
> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
> > freeswitch
> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
> > freeswitch
> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
> > freeswitch
> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
> > freeswitch
> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
> > freeswitch
> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
> > freeswitch
> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
> > freeswitch
> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
> > freeswitch
> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
> > freeswitch
> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50 freeswitch
> >
> >
> >
> > cat /proc/cpuinfo
> > processor       : 0
> > vendor_id       : GenuineIntel
> > cpu family      : 6
> > model           : 15
> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> > stepping        : 6
> > cpu MHz         : 2333.560
> > cache size      : 4096 KB
> > physical id     : 0
> > siblings        : 2
> > core id         : 0
> > cpu cores       : 2
> > apicid          : 0
> > initial apicid  : 0
> > fdiv_bug        : no
> > hlt_bug         : no
> > f00f_bug        : no
> > coma_bug        : no
> > fpu             : yes
> > fpu_exception   : yes
> > cpuid level     : 10
> > wp              : yes
> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
> mca
> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
> cx16
> > xtpr dca lahf_lm
> > bogomips        : 4670.78
> > clflush size    : 64
> > power management:
> >
> > processor       : 1
> > vendor_id       : GenuineIntel
> > cpu family      : 6
> > model           : 15
> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> > stepping        : 6
> > cpu MHz         : 2333.560
> > cache size      : 4096 KB
> > physical id     : 0
> > siblings        : 2
> > core id         : 1
> > cpu cores       : 2
> > apicid          : 1
> > initial apicid  : 1
> > fdiv_bug        : no
> > hlt_bug         : no
> > f00f_bug        : no
> > coma_bug        : no
> > fpu             : yes
> > fpu_exception   : yes
> > cpuid level     : 10
> > wp              : yes
> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
> mca
> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
> cx16
> > xtpr dca lahf_lm
> > bogomips        : 4666.82
> > clflush size    : 64
> > power management:
> >
> >
> >
> > uname -a
> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009 i686
> > GNU/Linux
> >
> >
> >
> > Of course, i've tuned the machine up
> >
> > ulimit -c unlimited
> > ulimit -d unlimited
> > ulimit -f unlimited
> > ulimit -i unlimited
> > ulimit -n 999999
> > ulimit -q unlimited
> > ulimit -u unlimited
> > ulimit -v unlimited
> > ulimit -x unlimited
> > ulimit -s 240
> > ulimit -l unlimited
> > ulimit -a
> >
> >
> > Started FS with minimum modules but still 40 CPS seems to be the limit.
> >
> >
> > So, is there any way to improve performance?
> >
> >
> > Tihomir.
> >
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jaybinks at gmail.com  Tue Aug 25 06:39:07 2009
From: jaybinks at gmail.com (Jay Binks)
Date: Tue, 25 Aug 2009 23:39:07 +1000
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
Message-ID: <BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>

Everytime someone asks this ,  the resounding answer is use a 64bit os..

No question

Jay



On 25/08/2009, at 23:19, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Hey Giovanni,
>
> thanks for the tip... indeed the db files were heavily used  
> regardless if i started freeswitch with nosql option (freeswitch - 
> nosql)... FS was not writing anything into that files ... instead it  
> was just accessing it....
> This behaviour leads to a waste of 40% CPU time... waiting for other  
> processes (mainly disk access) to finish!!!
>
> I moved freeswitch/db/ to a ramdisk and the performance got a boost  
> to 140 CPS with a CPU load of 80%. I was keeping the machine for a  
> while (20 - 30 minutes) on that rate when i sow CPU suddenly went to  
> 100% and FS becoming irresponsive :).
>
>
> What can be wrong?
> What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should  
> not cross?
> What fine tuning do we need in order to asure a long high load run?
>
>
>
> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it  
> have sense to move my OS to 64 bit? ... will FS gain more  
> preformance ?... I mean will FS perofomr drastically better 20%+ ?
>
>
> Tihomir.
>
>
> On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli <gmaruzz at celliax.org 
> > wrote:
> Maybe your load comes from disk access?
>
> Try putting the sql and log directories on a ramdisk.
>
> OTH,
>
> -giovanni
>
> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir  
> Culjaga<tculjaga at gmail.com> wrote:
> > Hello,
> >
> > i'm trying to use freeswitch as a redirecting server meaning FS  
> has to
> > receive an INVITE and according to some rules it will redirect  
> calls to
> > other destinations.
> >
> >
> > CALLING_USER                FREESWITCH                         
> SOMEWHERE
> >
> > INVITE ------------------------------->
> >            <------------------------------ 100 Trying
> >            <------------------------------ 302 Moved Temporary
> > ACK    ------------------------------->
> >  
> INVITE- 
> --- 
> --- 
> --- 
> --- 
> -------------------------------------------------------------------->
> >
> >
> >
> > Well, wverything works well except i have perfromance issues ....  
> on my HW
> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved  
> Temporary). When
> > i increase the rate, FS starts delaying 302 response. Right at 50  
> CPS i see
> > "calls" being build up in FS and the delay begining to grow.
> >
> > When i observe the machine, load average is almost nothing (load  
> average:
> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one  
> thread taking
> > most load... all others are just sitting there with 1-5 % CPU time.
> > This looks to me as FS handles 302 messages in a single thread?!?!
> >
> >
> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
> >
> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41,  
> 0.61, 0.60
> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,   
> 0.2%si,
> > 0.0%st
> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k  
> buffers
> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k  
> cached
> >
> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
> > COMMAND
> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
> > freeswitch
> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
> > freeswitch
> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
> > freeswitch
> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
> > freeswitch
> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
> > freeswitch
> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
> > freeswitch
> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
> > freeswitch
> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
> > freeswitch
> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
> > freeswitch
> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50  
> freeswitch
> >
> >
> >
> > cat /proc/cpuinfo
> > processor       : 0
> > vendor_id       : GenuineIntel
> > cpu family      : 6
> > model           : 15
> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> > stepping        : 6
> > cpu MHz         : 2333.560
> > cache size      : 4096 KB
> > physical id     : 0
> > siblings        : 2
> > core id         : 0
> > cpu cores       : 2
> > apicid          : 0
> > initial apicid  : 0
> > fdiv_bug        : no
> > hlt_bug         : no
> > f00f_bug        : no
> > coma_bug        : no
> > fpu             : yes
> > fpu_exception   : yes
> > cpuid level     : 10
> > wp              : yes
> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr  
> pge mca
> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2  
> ssse3 cx16
> > xtpr dca lahf_lm
> > bogomips        : 4670.78
> > clflush size    : 64
> > power management:
> >
> > processor       : 1
> > vendor_id       : GenuineIntel
> > cpu family      : 6
> > model           : 15
> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> > stepping        : 6
> > cpu MHz         : 2333.560
> > cache size      : 4096 KB
> > physical id     : 0
> > siblings        : 2
> > core id         : 1
> > cpu cores       : 2
> > apicid          : 1
> > initial apicid  : 1
> > fdiv_bug        : no
> > hlt_bug         : no
> > f00f_bug        : no
> > coma_bug        : no
> > fpu             : yes
> > fpu_exception   : yes
> > cpuid level     : 10
> > wp              : yes
> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr  
> pge mca
> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2  
> ssse3 cx16
> > xtpr dca lahf_lm
> > bogomips        : 4666.82
> > clflush size    : 64
> > power management:
> >
> >
> >
> > uname -a
> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC  
> 2009 i686
> > GNU/Linux
> >
> >
> >
> > Of course, i've tuned the machine up
> >
> > ulimit -c unlimited
> > ulimit -d unlimited
> > ulimit -f unlimited
> > ulimit -i unlimited
> > ulimit -n 999999
> > ulimit -q unlimited
> > ulimit -u unlimited
> > ulimit -v unlimited
> > ulimit -x unlimited
> > ulimit -s 240
> > ulimit -l unlimited
> > ulimit -a
> >
> >
> > Started FS with minimum modules but still 40 CPS seems to be the  
> limit.
> >
> >
> > So, is there any way to improve performance?
> >
> >
> > Tihomir.
> >
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From gmaruzz at celliax.org  Tue Aug 25 06:41:56 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 15:41:56 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com> 
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com> 
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
Message-ID: <7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com>

Definitely go for 64 bit OS.

If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
used both for development and for heavy duty production.

Also Ubuntu 8.04 is good.

Other versions/distros are less used by the community.

Adding RAM and CPUs helps to scale up.

-gm



Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com> wrote:
> Hey Giovanni,
>
> thanks for the tip... indeed the db files were heavily used regardless if i
> started freeswitch with nosql option (freeswitch -nosql)... FS was not
> writing anything into that files ... instead it was just accessing it....
> This behaviour leads to a waste of 40% CPU time... waiting for other
> processes (mainly disk access) to finish!!!
>
> I moved freeswitch/db/ to a ramdisk and the performance got a boost to 140
> CPS with a CPU load of 80%. I was keeping the machine for a while (20 - 30
> minutes) on that rate when i sow CPU suddenly went to 100% and FS becoming
> irresponsive :).
>
>
> What can be wrong?
> What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
> cross?
> What fine tuning do we need in order to asure a long high load run?
>
>
>
> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have sense
> to move my OS to 64 bit? ... will FS gain more preformance ?... I mean will
> FS perofomr drastically better 20%+ ?
>
>
> Tihomir.
>
>
> On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> Maybe your load comes from disk access?
>>
>> Try putting the sql and log directories on a ramdisk.
>>
>> OTH,
>>
>> -giovanni
>>
>> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com>
>> wrote:
>> > Hello,
>> >
>> > i'm trying to use freeswitch as a redirecting server meaning FS has to
>> > receive an INVITE and according to some rules it will redirect calls to
>> > other destinations.
>> >
>> >
>> > CALLING_USER??????????????? FREESWITCH??????????????????????? SOMEWHERE
>> >
>> > INVITE ------------------------------->
>> > ?????????? <------------------------------ 100 Trying
>> > ?????????? <------------------------------ 302 Moved Temporary
>> > ACK ?? ------------------------------->
>> >
>> > INVITE--------------------------------------------------------------------------------->
>> >
>> >
>> >
>> > Well, wverything works well except i have perfromance issues .... on my
>> > HW
>> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary).
>> > When
>> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS i
>> > see
>> > "calls" being build up in FS and the delay begining to grow.
>> >
>> > When i observe the machine, load average is almost nothing (load
>> > average:
>> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
>> > taking
>> > most load... all others are just sitting there with 1-5 % CPU time.
>> > This looks to me as FS handles 302 messages in a single thread?!?!
>> >
>> >
>> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>> >
>> > top - 10:41:37 up 167 days, 20:42,? 3 users,? load average: 1.41, 0.61,
>> > 0.60
>> > Tasks:? 83 total,?? 2 running,? 81 sleeping,?? 0 stopped,?? 0 zombie
>> > Cpu(s): 25.3%us,? 1.5%sy,? 0.0%ni, 30.3%id, 42.7%wa,? 0.0%hi,? 0.2%si,
>> > 0.0%st
>> > Mem:?? 2074520k total,?? 571244k used,? 1503276k free,?? 259604k buffers
>> > Swap:? 2650684k total,???? 3020k used,? 2647664k free,?? 153868k cached
>> >
>> > ? PID USER????? PR? NI? VIRT? RES? SHR S %CPU %MEM??? TIME+
>> > COMMAND
>> > ?4814 root????? 20?? 0 34188? 20m 3780 S?? 38? 1.0?? 3:10.29
>> > freeswitch
>> > ?4800 root????? 20?? 0 34188? 20m 3780 S??? 6? 1.0?? 0:08.26
>> > freeswitch
>> > ?4798 root????? 20?? 0 34188? 20m 3780 R??? 5? 1.0?? 0:24.46
>> > freeswitch
>> > ?4787 root????? 20?? 0 34188? 20m 3780 S??? 2? 1.0?? 0:11.24
>> > freeswitch
>> > ?4794 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.42
>> > freeswitch
>> > ?4803 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.74
>> > freeswitch
>> > ?4788 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.96
>> > freeswitch
>> > ?4804 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.64
>> > freeswitch
>> > ?4807 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.68
>> > freeswitch
>> > ?4811 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.50 freeswitch
>> >
>> >
>> >
>> > cat /proc/cpuinfo
>> > processor?????? : 0
>> > vendor_id?????? : GenuineIntel
>> > cpu family????? : 6
>> > model?????????? : 15
>> > model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
>> > stepping??????? : 6
>> > cpu MHz???????? : 2333.560
>> > cache size????? : 4096 KB
>> > physical id???? : 0
>> > siblings??????? : 2
>> > core id???????? : 0
>> > cpu cores?????? : 2
>> > apicid????????? : 0
>> > initial apicid? : 0
>> > fdiv_bug??????? : no
>> > hlt_bug???????? : no
>> > f00f_bug??????? : no
>> > coma_bug??????? : no
>> > fpu???????????? : yes
>> > fpu_exception?? : yes
>> > cpuid level???? : 10
>> > wp????????????? : yes
>> > flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
>> > mca
>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
>> > cx16
>> > xtpr dca lahf_lm
>> > bogomips??????? : 4670.78
>> > clflush size??? : 64
>> > power management:
>> >
>> > processor?????? : 1
>> > vendor_id?????? : GenuineIntel
>> > cpu family????? : 6
>> > model?????????? : 15
>> > model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
>> > stepping??????? : 6
>> > cpu MHz???????? : 2333.560
>> > cache size????? : 4096 KB
>> > physical id???? : 0
>> > siblings??????? : 2
>> > core id???????? : 1
>> > cpu cores?????? : 2
>> > apicid????????? : 1
>> > initial apicid? : 1
>> > fdiv_bug??????? : no
>> > hlt_bug???????? : no
>> > f00f_bug??????? : no
>> > coma_bug??????? : no
>> > fpu???????????? : yes
>> > fpu_exception?? : yes
>> > cpuid level???? : 10
>> > wp????????????? : yes
>> > flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
>> > mca
>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
>> > cx16
>> > xtpr dca lahf_lm
>> > bogomips??????? : 4666.82
>> > clflush size??? : 64
>> > power management:
>> >
>> >
>> >
>> > uname -a
>> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009 i686
>> > GNU/Linux
>> >
>> >
>> >
>> > Of course, i've tuned the machine up
>> >
>> > ulimit -c unlimited
>> > ulimit -d unlimited
>> > ulimit -f unlimited
>> > ulimit -i unlimited
>> > ulimit -n 999999
>> > ulimit -q unlimited
>> > ulimit -u unlimited
>> > ulimit -v unlimited
>> > ulimit -x unlimited
>> > ulimit -s 240
>> > ulimit -l unlimited
>> > ulimit -a
>> >
>> >
>> > Started FS with minimum modules but still 40 CPS seems to be the limit.
>> >
>> >
>> > So, is there any way to improve performance?
>> >
>> >
>> > Tihomir.
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From tculjaga at gmail.com  Tue Aug 25 06:42:24 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 15:42:24 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
Message-ID: <65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>

thanks for the feedback... this is something im going to do tomorrow...

what about other things?


On Tue, Aug 25, 2009 at 3:39 PM, Jay Binks <jaybinks at gmail.com> wrote:

> Everytime someone asks this ,  the resounding answer is use a 64bit os..
>
> No question
>
> Jay
>
>
>
> On 25/08/2009, at 23:19, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
> Hey Giovanni,
>
> thanks for the tip... indeed the db files were heavily used regardless if i
> started freeswitch with nosql option (freeswitch -nosql)... FS was not
> writing anything into that files ... instead it was just accessing it....
> This behaviour leads to a waste of 40% CPU time... waiting for other
> processes (mainly disk access) to finish!!!
>
> I moved freeswitch/db/ to a ramdisk and the performance got a boost to 140
> CPS with a CPU load of 80%. I was keeping the machine for a while (20 - 30
> minutes) on that rate when i sow CPU suddenly went to 100% and FS becoming
> irresponsive :).
>
>
> What can be wrong?
> What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
> cross?
> What fine tuning do we need in order to asure a long high load run?
>
>
>
> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
> sense to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
> will FS perofomr drastically better 20%+ ?
>
>
> Tihomir.
>
>
> On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli <<gmaruzz at celliax.org>
> gmaruzz at celliax.org> wrote:
>
>> Maybe your load comes from disk access?
>>
>> Try putting the sql and log directories on a ramdisk.
>>
>> OTH,
>>
>> -giovanni
>>
>> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga< <tculjaga at gmail.com>
>> tculjaga at gmail.com> wrote:
>> > Hello,
>> >
>> > i'm trying to use freeswitch as a redirecting server meaning FS has to
>> > receive an INVITE and according to some rules it will redirect calls to
>> > other destinations.
>> >
>> >
>> > CALLING_USER                FREESWITCH                        SOMEWHERE
>> >
>> > INVITE ------------------------------->
>> >            <------------------------------ 100 Trying
>> >            <------------------------------ 302 Moved Temporary
>> > ACK    ------------------------------->
>> >
>> INVITE--------------------------------------------------------------------------------->
>> >
>> >
>> >
>> > Well, wverything works well except i have perfromance issues .... on my
>> HW
>> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary).
>> When
>> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS i
>> see
>> > "calls" being build up in FS and the delay begining to grow.
>> >
>> > When i observe the machine, load average is almost nothing (load
>> average:
>> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
>> taking
>> > most load... all others are just sitting there with 1-5 % CPU time.
>> > This looks to me as FS handles 302 messages in a single thread?!?!
>> >
>> >
>> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>> >
>> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41, 0.61,
>> 0.60
>> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
>> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,  0.2%si,
>> > 0.0%st
>> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k buffers
>> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k cached
>> >
>> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
>> > COMMAND
>> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
>> > freeswitch
>> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
>> > freeswitch
>> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
>> > freeswitch
>> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
>> > freeswitch
>> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
>> > freeswitch
>> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
>> > freeswitch
>> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
>> > freeswitch
>> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
>> > freeswitch
>> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
>> > freeswitch
>> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50 freeswitch
>> >
>> >
>> >
>> > cat /proc/cpuinfo
>> > processor       : 0
>> > vendor_id       : GenuineIntel
>> > cpu family      : 6
>> > model           : 15
>> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
>> > stepping        : 6
>> > cpu MHz         : 2333.560
>> > cache size      : 4096 KB
>> > physical id     : 0
>> > siblings        : 2
>> > core id         : 0
>> > cpu cores       : 2
>> > apicid          : 0
>> > initial apicid  : 0
>> > fdiv_bug        : no
>> > hlt_bug         : no
>> > f00f_bug        : no
>> > coma_bug        : no
>> > fpu             : yes
>> > fpu_exception   : yes
>> > cpuid level     : 10
>> > wp              : yes
>> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
>> mca
>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
>> cx16
>> > xtpr dca lahf_lm
>> > bogomips        : 4670.78
>> > clflush size    : 64
>> > power management:
>> >
>> > processor       : 1
>> > vendor_id       : GenuineIntel
>> > cpu family      : 6
>> > model           : 15
>> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
>> > stepping        : 6
>> > cpu MHz         : 2333.560
>> > cache size      : 4096 KB
>> > physical id     : 0
>> > siblings        : 2
>> > core id         : 1
>> > cpu cores       : 2
>> > apicid          : 1
>> > initial apicid  : 1
>> > fdiv_bug        : no
>> > hlt_bug         : no
>> > f00f_bug        : no
>> > coma_bug        : no
>> > fpu             : yes
>> > fpu_exception   : yes
>> > cpuid level     : 10
>> > wp              : yes
>> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
>> mca
>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
>> cx16
>> > xtpr dca lahf_lm
>> > bogomips        : 4666.82
>> > clflush size    : 64
>> > power management:
>> >
>> >
>> >
>> > uname -a
>> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009 i686
>> > GNU/Linux
>> >
>> >
>> >
>> > Of course, i've tuned the machine up
>> >
>> > ulimit -c unlimited
>> > ulimit -d unlimited
>> > ulimit -f unlimited
>> > ulimit -i unlimited
>> > ulimit -n 999999
>> > ulimit -q unlimited
>> > ulimit -u unlimited
>> > ulimit -v unlimited
>> > ulimit -x unlimited
>> > ulimit -s 240
>> > ulimit -l unlimited
>> > ulimit -a
>> >
>> >
>> > Started FS with minimum modules but still 40 CPS seems to be the limit.
>> >
>> >
>> > So, is there any way to improve performance?
>> >
>> >
>> > Tihomir.
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > <http://www.freeswitch.org>http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Tue Aug 25 06:47:37 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 15:47:37 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com> 
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com> 
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com> 
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
Message-ID: <7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>

is a heavely multithreaded software, it benefits from number of CPUs
(or cores), RAM, and heavy duty kernel features (found in 64bit
kernels)

put all accesses on ramdisk, leave out the modules you don't use...

experiment, test, and find the best for your specific application/workload

test not only with sipp, but with real load too (often they're very different)

-gm


On Tue, Aug 25, 2009 at 3:42 PM, Tihomir Culjaga<tculjaga at gmail.com> wrote:
> thanks for the feedback... this is something im going to do tomorrow...
>
> what about other things?
>
>
> On Tue, Aug 25, 2009 at 3:39 PM, Jay Binks <jaybinks at gmail.com> wrote:
>>
>> Everytime someone asks this , ?the resounding answer is use a 64bit os..
>> No question
>> Jay
>>
>>
>>
>> On 25/08/2009, at 23:19, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>
>> Hey Giovanni,
>>
>> thanks for the tip... indeed the db files were heavily used regardless if
>> i started freeswitch with nosql option (freeswitch -nosql)... FS was not
>> writing anything into that files ... instead it was just accessing it....
>> This behaviour leads to a waste of 40% CPU time... waiting for other
>> processes (mainly disk access) to finish!!!
>>
>> I moved freeswitch/db/ to a ramdisk and the performance got a boost to 140
>> CPS with a CPU load of 80%. I was keeping the machine for a while (20 - 30
>> minutes) on that rate when i sow CPU suddenly went to 100% and FS becoming
>> irresponsive :).
>>
>>
>> What can be wrong?
>> What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
>> cross?
>> What fine tuning do we need in order to asure a long high load run?
>>
>>
>>
>> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
>> sense to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
>> will FS perofomr drastically better 20%+ ?
>>
>>
>> Tihomir.
>>
>>
>> On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
>> <gmaruzz at celliax.org> wrote:
>>>
>>> Maybe your load comes from disk access?
>>>
>>> Try putting the sql and log directories on a ramdisk.
>>>
>>> OTH,
>>>
>>> -giovanni
>>>
>>> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com>
>>> wrote:
>>> > Hello,
>>> >
>>> > i'm trying to use freeswitch as a redirecting server meaning FS has to
>>> > receive an INVITE and according to some rules it will redirect calls to
>>> > other destinations.
>>> >
>>> >
>>> > CALLING_USER??????????????? FREESWITCH??????????????????????? SOMEWHERE
>>> >
>>> > INVITE ------------------------------->
>>> > ?????????? <------------------------------ 100 Trying
>>> > ?????????? <------------------------------ 302 Moved Temporary
>>> > ACK ?? ------------------------------->
>>> >
>>> > INVITE--------------------------------------------------------------------------------->
>>> >
>>> >
>>> >
>>> > Well, wverything works well except i have perfromance issues .... on my
>>> > HW
>>> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary).
>>> > When
>>> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS i
>>> > see
>>> > "calls" being build up in FS and the delay begining to grow.
>>> >
>>> > When i observe the machine, load average is almost nothing (load
>>> > average:
>>> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
>>> > taking
>>> > most load... all others are just sitting there with 1-5 % CPU time.
>>> > This looks to me as FS handles 302 messages in a single thread?!?!
>>> >
>>> >
>>> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>>> >
>>> > top - 10:41:37 up 167 days, 20:42,? 3 users,? load average: 1.41, 0.61,
>>> > 0.60
>>> > Tasks:? 83 total,?? 2 running,? 81 sleeping,?? 0 stopped,?? 0 zombie
>>> > Cpu(s): 25.3%us,? 1.5%sy,? 0.0%ni, 30.3%id, 42.7%wa,? 0.0%hi,? 0.2%si,
>>> > 0.0%st
>>> > Mem:?? 2074520k total,?? 571244k used,? 1503276k free,?? 259604k
>>> > buffers
>>> > Swap:? 2650684k total,???? 3020k used,? 2647664k free,?? 153868k cached
>>> >
>>> > ? PID USER????? PR? NI? VIRT? RES? SHR S %CPU %MEM??? TIME+
>>> > COMMAND
>>> > ?4814 root????? 20?? 0 34188? 20m 3780 S?? 38? 1.0?? 3:10.29
>>> > freeswitch
>>> > ?4800 root????? 20?? 0 34188? 20m 3780 S??? 6? 1.0?? 0:08.26
>>> > freeswitch
>>> > ?4798 root????? 20?? 0 34188? 20m 3780 R??? 5? 1.0?? 0:24.46
>>> > freeswitch
>>> > ?4787 root????? 20?? 0 34188? 20m 3780 S??? 2? 1.0?? 0:11.24
>>> > freeswitch
>>> > ?4794 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.42
>>> > freeswitch
>>> > ?4803 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.74
>>> > freeswitch
>>> > ?4788 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.96
>>> > freeswitch
>>> > ?4804 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.64
>>> > freeswitch
>>> > ?4807 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.68
>>> > freeswitch
>>> > ?4811 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.50 freeswitch
>>> >
>>> >
>>> >
>>> > cat /proc/cpuinfo
>>> > processor?????? : 0
>>> > vendor_id?????? : GenuineIntel
>>> > cpu family????? : 6
>>> > model?????????? : 15
>>> > model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
>>> > stepping??????? : 6
>>> > cpu MHz???????? : 2333.560
>>> > cache size????? : 4096 KB
>>> > physical id???? : 0
>>> > siblings??????? : 2
>>> > core id???????? : 0
>>> > cpu cores?????? : 2
>>> > apicid????????? : 0
>>> > initial apicid? : 0
>>> > fdiv_bug??????? : no
>>> > hlt_bug???????? : no
>>> > f00f_bug??????? : no
>>> > coma_bug??????? : no
>>> > fpu???????????? : yes
>>> > fpu_exception?? : yes
>>> > cpuid level???? : 10
>>> > wp????????????? : yes
>>> > flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
>>> > mca
>>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
>>> > cx16
>>> > xtpr dca lahf_lm
>>> > bogomips??????? : 4670.78
>>> > clflush size??? : 64
>>> > power management:
>>> >
>>> > processor?????? : 1
>>> > vendor_id?????? : GenuineIntel
>>> > cpu family????? : 6
>>> > model?????????? : 15
>>> > model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
>>> > stepping??????? : 6
>>> > cpu MHz???????? : 2333.560
>>> > cache size????? : 4096 KB
>>> > physical id???? : 0
>>> > siblings??????? : 2
>>> > core id???????? : 1
>>> > cpu cores?????? : 2
>>> > apicid????????? : 1
>>> > initial apicid? : 1
>>> > fdiv_bug??????? : no
>>> > hlt_bug???????? : no
>>> > f00f_bug??????? : no
>>> > coma_bug??????? : no
>>> > fpu???????????? : yes
>>> > fpu_exception?? : yes
>>> > cpuid level???? : 10
>>> > wp????????????? : yes
>>> > flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
>>> > mca
>>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
>>> > cx16
>>> > xtpr dca lahf_lm
>>> > bogomips??????? : 4666.82
>>> > clflush size??? : 64
>>> > power management:
>>> >
>>> >
>>> >
>>> > uname -a
>>> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009
>>> > i686
>>> > GNU/Linux
>>> >
>>> >
>>> >
>>> > Of course, i've tuned the machine up
>>> >
>>> > ulimit -c unlimited
>>> > ulimit -d unlimited
>>> > ulimit -f unlimited
>>> > ulimit -i unlimited
>>> > ulimit -n 999999
>>> > ulimit -q unlimited
>>> > ulimit -u unlimited
>>> > ulimit -v unlimited
>>> > ulimit -x unlimited
>>> > ulimit -s 240
>>> > ulimit -l unlimited
>>> > ulimit -a
>>> >
>>> >
>>> > Started FS with minimum modules but still 40 CPS seems to be the limit.
>>> >
>>> >
>>> > So, is there any way to improve performance?
>>> >
>>> >
>>> > Tihomir.
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From csa at nowthor.com  Tue Aug 25 07:06:25 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Tue, 25 Aug 2009 10:06:25 -0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
Message-ID: <4A93EFE1.6010007@nowthor.com>

Max,

I would like to see something similar too. For example, it would be 
wonderful if one could specify multiple gateways to try like this or 
something similar:

<action application="bridge">
    <!-- set some kind of *bridge-specific* parameter/variable -->
    <gateway order=1 data="sofia/gateway/gw-1/1$1"/>
    <gateway order=1 data="sofia/gateway/gw-2/1$1"/>
    <gateway order=2 data="sofia/gateway/gw-3/1$1">
        <!-- set some kind of *gw-specific* parameter/variable -->
    </gateway>
    <gateway order=3 data="sofia/gateway/gw-4/1$1"/>
</action>

One would be able to avoid the "[]" and "{}" hacks and combine 
sequential and simultaneous trying of gateways.

What do the developers think of this?

Carlos

Max Ivanov wrote:
> Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
> sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
> ,
> the whole sting is tooo long, is there any shorter way to write same thing? Like
> "sofia/gateway/panas*/1000" will try all gateways matching the pattern.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From rs at runsolutions.com  Tue Aug 25 02:06:37 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Tue, 25 Aug 2009 11:06:37 +0200
Subject: [Freeswitch-users] using sipp to test call transfers
Message-ID: <84D97940-68D0-402F-A7E2-B5ADD287DC73@runsolutions.com>

Hi List,

i have some scripts to test our lab:

I have scripts to create sipp instances which act as individual  
agents, which log on, take a random amount of calls, log off, wait a  
bit, log on again, etc.
I have scripts to generate call traffic for our queues to saturate them

But what I have difficulties is how to create a good scenario xml file  
for call transfers? Has anybody here a sample for me to sink my teeth  
in?

thanks and best regards

-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares



From aep.lists at it46.se  Mon Aug 24 23:54:24 2009
From: aep.lists at it46.se (Alberto Escudero-Pascual (lists))
Date: Tue, 25 Aug 2009 08:54:24 +0200
Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
In-Reply-To: <191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
	<191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
Message-ID: <6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>

Dear all,

In one of the applications I am writing I need to convert a recorded wav
to mp3.

After using

session.recordFile() and obtaining a foo.wav file, I am calling

session.execute("system",lmLameCmd);

to invoke lame for the conversion.

The system command looks like this:
lmLameCmd = "/usr/local/freeswitch/bin/lame -V2 foo.wav foo.mp3 -S";


Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
appears, if I use version 3.97 (older version), it does!.

If I execute the conversion from the command line, i get the mp3 with both
3.97 and 3.98.2

In fact, i am considering doing the conversions as background job, but I
am very curious to hear if this behavior has a pseudo-scientific
explanation

/aep

-- 
Stopping junk mailers is good for the environment




From ederwander at gmail.com  Tue Aug 25 06:06:40 2009
From: ederwander at gmail.com (Eder Souza)
Date: Tue, 25 Aug 2009 10:06:40 -0300
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
In-Reply-To: <4A930D6B.2040202@devel-it.com.br>
References: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
	<4A930D6B.2040202@devel-it.com.br>
Message-ID: <622bedea0908250606w57d4eba8q3045eea272bf012f@mail.gmail.com>

Very Nice my friends i'm from Brazil i'm crazy for the tests ...


Eng Eder de Souza

On Mon, Aug 24, 2009 at 7:00 PM, Rodrigo P. Telles <
telles-listas at devel-it.com.br> wrote:

> It is very nice to hear that, great work!
> Thanks to support FS.
>
> Telles
>
>
> Moises Silva wrote:
>
> So, I finally took some days to put up OpenR2 working with OpenZAP, which
> means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
> support for. Including Mexico, Brazil, Argentina and others. The stack has
> been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most
> countries that users may be interested in, support for new variants will be
> added on-demand only (in any case users can always tweak the advanced
> configuration file to create their own variants as a last resort).
> I created a wiki page to illustrate the basic setup:
> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
>
> Now is time for testing. I just did minimal testing on my development
> environment, no serious testing, and I know that some stuff is not working
> at this point (I had some issues with variable length DNIS and ANI) which
> should be fixed soon.
>
> If anyone around happens to have an R2 link and wants to test R2 support in
> OpenZAP, I can give them a hand with the configuration and any issues you
> may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
> #openzap as moy.
>
> --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rs at runsolutions.com  Tue Aug 25 06:35:13 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Tue, 25 Aug 2009 15:35:13 +0200
Subject: [Freeswitch-users] Freeswitch & DAHDI Kernel drivers / Hardware
	questions
Message-ID: <DD5C7209-7041-4D21-869D-2F3BABEADC78@runsolutions.com>

Hello,

i was reading through the openzap wiki page and searched the wiki, the  
only thing I found about dahdi is a note from January that freeswitch  
does not work with dahdi right now.

Is this current information? Can't I use dahdi kernel drivers for  
freeswitch? As the last update to the zaptel drivers is now nearly a  
year old (1.4.12) and development has, AFAIK, ceased and new stuff  
only gehts to dahdi, correct me please if I am wrong.

Second question,

is it possible to use this hardware:

00:0d.0 Network controller: Cologne Chip Designs GmbH ISDN network  
controller [HFC-PCI] (rev 02)
	Subsystem: Cologne Chip Designs GmbH ISDN Board
	Flags: bus master, medium devsel, latency 16, IRQ 10
	I/O ports at e000 [disabled] [size=8]
	Memory at ee001000 (32-bit, non-prefetchable) [size=256]
	Capabilities: [40] Power Management version 1
	Kernel modules: hisax

with openZap? I read in the wiki (openZap.conf_Examples) something  
about HFC 4 BRI, but i do not know if this is the same as the above  
mentioned card.


I have access to some pcmcia fxo, and pcmcia isdn cards, could they be  
used? I have as well an soekris board with two pcmcia slots, could i  
use this with an fxo pcmcia card to actually drive a small home - pbx  
with freeswitch?

I did not find answers for this questions in the wiki :-)

best regards


-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-
Email: rs at runsolutions.com
tel: 625 40 32 08

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

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From brian at freeswitch.org  Tue Aug 25 07:16:04 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 25 Aug 2009 09:16:04 -0500
Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
In-Reply-To: <6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
	<191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
	<6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>
Message-ID: <5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org>

Try running it at the CLI and see if you see any errors.  Also please  
do not hijack threads.  The original thread "[Freeswitch-users] XML- 
RPC on different ip than 0.0.0.0" which was hijacked by clicking  
reply, changing the subject and clicking send.  Please in the future  
do not do that as it clutters up the threading and could get your  
query lost in the noise.

Thanks,
Brian

On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote:

> Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
> appears, if I use version 3.97 (older version), it does!.



From MPeace at edcogroupinc.com  Tue Aug 25 07:20:41 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 09:20:41 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>

I fired up Wireshark on each side and I can see the SIP register request coming from the laptop, the Freeswitch server replies with a Destination Unreachable (Port unreachable) message.

I rebooted the Server and now I get a "Registration error; 405 Method not allowed" on the softphone and the Wiresharp capture shows: "Status 405 Method Not Allowerd (0 bindings).

It does this even after I stop and restart the Freeswitch service.

Shouldn't all the ports be open if the firewall and SELunix are disabled?

Mike Peace
Network Analyst
EDCO, The Document People
Direct 417-447-3367

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 5:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


On Mon, Aug 24, 2009 at 2:49 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

It does the same thing, does anything get set during the install that would remember or cache the old network settings? I can access anything from the FS server on any of several networks and vice-versa but the SIP will not register, again no firewalls are upon any of the test hosts.

Doesn't make sense to me.


Time to bust out tcpdump and/or wireshark to make 100% certain you know what's happening with all those SIP packets.

Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:34 PM

To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.





On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

"Registration error: 408-Request Timeout"

Sorry for the the typo - 408 = timeout. (480 = temp unavail)

Try stopping iptables in Linux and try again. Sounds like something is interfering with your packets getting from here to there...
Try:

/etc/init.d/iptables stop

And then see if your packets can move again.

-MC



Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:18 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.





On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

I haven't changed any of the conf files.



What happens when you try to register? Do you get 480? (timeout) Or something else?
-MC



________________________________



EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/



Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From moises.silva at gmail.com  Tue Aug 25 07:24:50 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 25 Aug 2009 10:24:50 -0400
Subject: [Freeswitch-users] Freeswitch & DAHDI Kernel drivers / Hardware
	questions
In-Reply-To: <DD5C7209-7041-4D21-869D-2F3BABEADC78@runsolutions.com>
References: <DD5C7209-7041-4D21-869D-2F3BABEADC78@runsolutions.com>
Message-ID: <c4d05cbe0908250724j407636e0o33f0df4bbfbb461@mail.gmail.com>

On Tue, Aug 25, 2009 at 9:35 AM, Raimund Sacherer <rs at runsolutions.com>wrote:

> Hello,
> i was reading through the openzap wiki page and searched the wiki, the only
> thing I found about dahdi is a note from January that freeswitch does not
> work with dahdi right now.
>
>
You did not search well enough.
http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 , read note 5 there.


> Is this current information? Can't I use dahdi kernel drivers for
> freeswitch? As the last update to the zaptel drivers is now nearly a year
> old (1.4.12) and development has, AFAIK, ceased and new stuff only gehts to
> dahdi, correct me please if I am wrong.
>
both dahdi and zaptel work for FreeSWITCH and no headers are needed, we have
our own.


>
> 00:0d.0 Network controller: Cologne Chip Designs GmbH ISDN network
> controller [HFC-PCI] (rev 02)
> Subsystem: Cologne Chip Designs GmbH ISDN Board
> Flags: bus master, medium devsel, latency 16, IRQ 10
> I/O ports at e000 [disabled] [size=8]
> Memory at ee001000 (32-bit, non-prefetchable) [size=256]
> Capabilities: [40] Power Management version 1
> Kernel modules: hisax
>
> with openZap? I read in the wiki (openZap.conf_Examples) something about
> HFC 4 BRI, but i do not know if this is the same as the above mentioned
> card.
>
>
> I have access to some pcmcia fxo, and pcmcia isdn cards, could they be
> used? I have as well an soekris board with two pcmcia slots, could i use
> this with an fxo pcmcia card to actually drive a small home - pbx with
> freeswitch?
>

I don't have experience with such hardware, cannot tell, may be someone else
around here may know.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From rs at runsolutions.com  Tue Aug 25 07:35:05 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Tue, 25 Aug 2009 16:35:05 +0200
Subject: [Freeswitch-users] Freeswitch & DAHDI Kernel drivers / Hardware
	questions
In-Reply-To: <c4d05cbe0908250724j407636e0o33f0df4bbfbb461@mail.gmail.com>
References: <DD5C7209-7041-4D21-869D-2F3BABEADC78@runsolutions.com>
	<c4d05cbe0908250724j407636e0o33f0df4bbfbb461@mail.gmail.com>
Message-ID: <7CEF2F0A-0735-41E8-ABA5-FBCB2A0AC0DF@runsolutions.com>


On Aug 25, 2009, at 4:24 PM, Moises Silva wrote:

> On Tue, Aug 25, 2009 at 9:35 AM, Raimund Sacherer  
> <rs at runsolutions.com> wrote:
> Hello,
>
> i was reading through the openzap wiki page and searched the wiki,  
> the only thing I found about dahdi is a note from January that  
> freeswitch does not work with dahdi right now.
>
>
> You did not search well enough. http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 
>  , read note 5 there.
Thank you, i did not see any link to OpenZAP_OpenR2, but this page  
explains a lot for me!
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From pjintheusa at gmail.com  Tue Aug 25 07:35:45 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 25 Aug 2009 10:35:45 -0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <4A93EFE1.6010007@nowthor.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
	<4A93EFE1.6010007@nowthor.com>
Message-ID: <367751820908250735t35cd96f6kfcf46c2c39e8d4d@mail.gmail.com>

Take a look at http://jira.freeswitch.org/browse/FSCORE-422. This a
feature request I submitted. This problem it solves is different - but
the solution is the same. Perhaps you add your take to the comments
there.

On Tue, Aug 25, 2009 at 10:06 AM, Carlos S. Antunes<csa at nowthor.com> wrote:
> Max,
>
> I would like to see something similar too. For example, it would be
> wonderful if one could specify multiple gateways to try like this or
> something similar:
>
> <action application="bridge">
> ? ?<!-- set some kind of *bridge-specific* parameter/variable -->
> ? ?<gateway order=1 data="sofia/gateway/gw-1/1$1"/>
> ? ?<gateway order=1 data="sofia/gateway/gw-2/1$1"/>
> ? ?<gateway order=2 data="sofia/gateway/gw-3/1$1">
> ? ? ? ?<!-- set some kind of *gw-specific* parameter/variable -->
> ? ?</gateway>
> ? ?<gateway order=3 data="sofia/gateway/gw-4/1$1"/>
> </action>
>
> One would be able to avoid the "[]" and "{}" hacks and combine
> sequential and simultaneous trying of gateways.
>
> What do the developers think of this?
>
> Carlos
>
> Max Ivanov wrote:
>> Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
>> sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
>> ,
>> the whole sting is tooo long, is there any shorter way to write same thing? Like
>> "sofia/gateway/panas*/1000" will try all gateways matching the pattern.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From anthony.minessale at gmail.com  Tue Aug 25 07:45:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Aug 2009 09:45:04 -0500
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <4A93EFE1.6010007@nowthor.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
	<4A93EFE1.6010007@nowthor.com>
Message-ID: <191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>

This suggestion violates the scope boundaries.

gateways are specific concept to mod_sofia so a <gateway> tag in <action>
(part of agnostic xml dialplan)
does not flow properly.

you can also use combinations of continue_on_fail and hangup_after bridge so
you can
just put each bridge statement in it's own action.


On Tue, Aug 25, 2009 at 9:06 AM, Carlos S. Antunes <csa at nowthor.com> wrote:

> Max,
>
> I would like to see something similar too. For example, it would be
> wonderful if one could specify multiple gateways to try like this or
> something similar:
>
> <action application="bridge">
>    <!-- set some kind of *bridge-specific* parameter/variable -->
>    <gateway order=1 data="sofia/gateway/gw-1/1$1"/>
>    <gateway order=1 data="sofia/gateway/gw-2/1$1"/>
>    <gateway order=2 data="sofia/gateway/gw-3/1$1">
>        <!-- set some kind of *gw-specific* parameter/variable -->
>    </gateway>
>    <gateway order=3 data="sofia/gateway/gw-4/1$1"/>
> </action>
>
> One would be able to avoid the "[]" and "{}" hacks and combine
> sequential and simultaneous trying of gateways.
>
> What do the developers think of this?
>
> Carlos
>
> Max Ivanov wrote:
> > Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
> >
> sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
> > ,
> > the whole sting is tooo long, is there any shorter way to write same
> thing? Like
> > "sofia/gateway/panas*/1000" will try all gateways matching the pattern.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Tue Aug 25 08:00:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 11:00:27 -0400
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
	<15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>
	<15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com>
	<65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com>
Message-ID: <84597815-246D-45BB-9C9D-755C2C74D4C4@jerris.com>

I beleive this is following the right rfc rules for dialog matching.   
If it is not, please open up a bug on jira.freeswitch.org with  
references of what exactly is not right.

Mike

On Aug 25, 2009, at 2:51 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Hello Takeshi,
>
> Thanks for your hint... it worked out... so to be precise:
>
> VIA header of both INVITE and ACK messages MUST be identical  
> (IP:PORT + branch)... and you are right... it might not be according  
> to SIP specification. Anyhow, i get FS understand my ACK message.
>
>
> Finally, here is what i used and I'm getting some poor results ..  
> but this is another topic :)
>
>
> Thanks for your help.
> Tihomir.
>
>
> sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err  
> -r 1 -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
>
>
> <?xml version="1.0" encoding="ISO-8859-1" ?>
> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>
>
> <scenario name="Basic Sipstone UAC">
>   <send retrans="500" start_rtd="1" start_rtd="2">
>
>     <![CDATA[
>
>       INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>       Max-Forwards: 70
>       Contact: <sip:[field1]@[local_ip]>
>       From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag= 
> [call_number]
>       To: [service] <sip:[service]@[remote_ip]:[remote_port]>
>       Call-ID: [call_id]
>       CSeq: 1 INVITE
>       Content-Type: application/sdp
>       Content-Length: [len]
>
>       v=0
>       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>       s=-
>       c=IN IP[media_ip_type] [media_ip]
>       t=0 0
>       m=audio [media_port] RTP/AVP 0
>       a=rtpmap:0 PCMU/8000
>
>     ]]>
>   </send>
>
>   <recv response="100"
>         optional="true" rtd="1">
>   </recv>
>
>
>   <recv response="302" rtd="2">
>   </recv>
>
>   <send>
>     <![CDATA[
>
>       ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch= 
> [branch-3]
>       From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag= 
> [call_number]
>       To: [service] <sip:[service]@[remote_ip]:[remote_port]> 
> [peer_tag_param]
>       Call-ID: [call_id]
>       CSeq: 1 ACK
>       Max-Forwards: 70
>       Content-Length: 0
>
>     ]]>
>   </send>
>
>   <!-- definition of the response time repartition table (unit is  
> ms)   -->
>   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
>
>   <!-- definition of the call length repartition table (unit is  
> ms)     -->
>   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
>
> </scenario>
>
>
>
> On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi at gmail.com 
> > wrote:
> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com 
> > wrote:
> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir  
> Culjaga<tculjaga at gmail.com> wrote:
> >>
> >> sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf  
> test.txt -m 1 -l
> >> 4000
> >> scenario file:      uac_redirect.xml
> >> FS dialplan:       public.xml
> >> SIP trace:          trace.log
> >
> > The Via definition in your SIPp scenario differs between the  
> INVITE and the ACK:
> >
> > INVITE:
> > Via: SIP/2.0/[transport] [local_ip];branch=[branch]
> >
> > ACK:
> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
> >
> >
> > In the INVITE, you are not adding the [local_port] as you do in  
> the ACK.
> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept  
> the ACK.
> > So it seems FS is not checking just the ACK's branch against the
> > INVITE's; it seems it is checking the whole Via header.
> > I don't know if this is in accordance to SIP specs.
> > Another thing, about the way you are calling SIPp: do no use "-sn  
> uac"
> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
> > means "use the internal (embedded) scenario named xxx". So this
> > conflicts with the other parameter "-sf" which specifies an external
> > profile.
>
> I mean, an external scenario (file).
>
>  It seems this doesn't cause any problem (probably because in
> > the sipp startup, -sf overrides -sn), but it is misleading.
> >
> > regards,
> > takeshi
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From sdjernes at gmail.com  Tue Aug 25 07:30:49 2009
From: sdjernes at gmail.com (Shawn L. Djernes)
Date: Tue, 25 Aug 2009 09:30:49 -0500
Subject: [Freeswitch-users] Group Call
Message-ID: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>

Hello,
I am trying to get group calling to work.  AKA: dial an extension (7300 in
this case) and it ring all phones at once.  We are moving a server from
Asterisk to FS

Here is the conf/directory/default.xml section:

        <group name="sdjernes">
                <users>
                        <user id="7301" type="pointer"/>
                        <user id="7302" type="pointer"/>
                        <user id="7303" type="pointer"/>
                        <user id="7304" type="pointer"/>
                </users>
        </group>

Here is the conf/dialplan/default/7300_group.xml extension statement:

     <extension name="Group 7300">
       <condition field="destination_number" expression="(7300)">
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" data="continue_on_fail=true"/>
         <action application="set" data="call_timeout=15"/>
         <action application="bridge"
data="{ignore_early_media=true}${group(call:sdjernes+A@${domain_name})}"/>
<!--         <action application="transfer" data="7301 XML default"/> -->
         <action application="hangup"/>
       </condition>
     </extension>

The log output shows the following:

2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3289 Channel
sofia/internal/7305 at ewr.
djernes.net entering state [received][100]
2009-08-25 14:27:45.282619 [DEBUG] switch_core_state_machine.c:398
(sofia/intern
al/7305 at ewr.djernes.net) Running State Change CS_NEW
2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3296 Remote SDP:
v=0
o=root 628695665 628695665 IN IP4 98.188.201.109
s=call
c=IN IP4 98.188.201.109
t=0 0
m=audio 61012 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[CELT:114:48000:10]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[CELT:114:32000:10]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[SPEEX:98:8000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[SPEEX:98:8000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[SPEEX:98:8000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[G7221:115:32000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[G7221:107:16000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[G722:9:8000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[PCMU:0:8000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:2029 Set Codec
sofia/internal/73
05 at ewr.djernes.net PCMU/8000 20 ms 160 samples
2009-08-25 14:27:45.282619 [DEBUG] switch_core_state_machine.c:404
(sofia/intern
al/7305 at ewr.djernes.net) State NEW
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload
to 10
1
2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3455
(sofia/internal/7305 at ewr.djernes
.net) State Change CS_NEW -> CS_INIT
2009-08-25 14:27:45.282619 [DEBUG] switch_core_session.c:932 Send signal
sofia/i
nternal/7305 at ewr.djernes.net [BREAK]
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
(sofia/intern
al/7305 at ewr.djernes.net) Running State Change CS_INIT
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:481
(sofia/intern
al/7305 at ewr.djernes.net) State INIT
2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:83
sofia/internal/7305 at ewr.djerne
s.net SOFIA INIT
2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:111
(sofia/internal/7305 at ewr.djer
nes.net) State Change CS_INIT -> CS_ROUTING
2009-08-25 14:27:45.287037 [DEBUG] switch_core_session.c:932 Send signal
sofia/i
nternal/7305 at ewr.djernes.net [BREAK]
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:481
(sofia/intern
al/7305 at ewr.djernes.net) State INIT going to sleep
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
(sofia/intern
al/7305 at ewr.djernes.net) Running State Change CS_ROUTING
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:484
(sofia/intern
al/7305 at ewr.djernes.net) State ROUTING
2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:130
sofia/internal/7305 at ewr.djern
es.net SOFIA ROUTING
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:78
sofia/internal
/7305 at ewr.djernes.net Standard ROUTING
2009-08-25 14:27:45.287037 [INFO] mod_dialplan_xml.c:315 Processing Sdjernes
TOI
 Phone->7300 in context default
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unloop]
continue
=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unloop]
${unroll_loo
ps}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unloop]
${sip_looped
_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->tod_example]
con
tinue=true
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [tod_example]
${strft
ime(%w)}(2) =~ /^([1-5])$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [tod_example]
${strft
ime(%H%M)}(1427) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Action set(open=true)
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->global-intercept
] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[global-intercept] de
stination_number(7300) =~ /^886$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->group-intercept]
 continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [group-intercept]
des
tination_number(7300) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->intercept-ext] c
ontinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [intercept-ext]
desti
nation_number(7300) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->redial]
continue
=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [redial]
destination_
number(7300) =~ /^870$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->global]
continue
=true
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [global]
${call_debug
}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [global]
${sip_has_cr
ypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/internal/7305 at ewr.djernes.net Absolute Condition [global]
Dialplan: sofia/internal/7305 at ewr.djernes.net Action
hash(insert/${domain_name}-
spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/7305 at ewr.djernes.net Action
hash(insert/${domain_name}-
last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/internal/7305 at ewr.djernes.net Action
hash(insert/${domain_name}-
last_dial/global/${uuid})
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->snom-demo-2]
con
tinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [snom-demo-2]
destina
tion_number(7300) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->snom-demo-1]
con
tinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [snom-demo-1]
destina
tion_number(7300) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->eavesdrop]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [eavesdrop]
destinati
on_number(7300) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->eavesdrop]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [eavesdrop]
destinati
on_number(7300) =~ /^779$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->call_return]
con
tinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call_return]
destina
tion_number(7300) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->del-group]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [del-group]
destinati
on_number(7300) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->add-group]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [add-group]
destinati
on_number(7300) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->call-group-simo]
 continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call-group-simo]
des
tination_number(7300) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->call-group-order
] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[call-group-order] de
stination_number(7300) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->extension-interc
om] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[extension-intercom]
destination_number(7300) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->Local_Extension]
 continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [Local_Extension]
des
tination_number(7300) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->group_dial_sales
] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[group_dial_sales] de
stination_number(7300) =~ /^2000$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->group_dial_suppo
rt] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[group_dial_support]
destination_number(7300) =~ /^2001$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->group_dial_billi
ng] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[group_dial_billing]
destination_number(7300) =~ /^2002$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->operator]
contin
ue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [operator]
destinatio
n_number(7300) =~ /^(operator|0)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->vmain]
continue=
false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [vmain]
destination_n
umber(7300) =~ /^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->sip_uri]
continu
e=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [sip_uri]
destination
_number(7300) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->nb_conferences]
continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [nb_conferences]
dest
ination_number(7300) =~ /^(30\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->wb_conferences]
continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [wb_conferences]
dest
ination_number(7300) =~ /^(31\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->uwb_conferences]
 continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [uwb_conferences]
des
tination_number(7300) =~ /^(32\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->cdquality_confer
ences] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[cdquality_conference
s] destination_number(7300) =~ /^(33\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->freeswitch_publi
c_conf_via_sip] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[freeswitch_public_co
nf_via_sip] destination_number(7300) =~ /^9(888|1616|3232)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->mad_boss_interco
m] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[mad_boss_intercom] d
estination_number(7300) =~ /^0911$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->mad_boss_interco
m] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[mad_boss_intercom] d
estination_number(7300) =~ /^0912$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->mad_boss]
contin
ue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [mad_boss]
destinatio
n_number(7300) =~ /^0913$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->ivr_demo]
contin
ue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ivr_demo]
destinatio
n_number(7300) =~ /^5000$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->dynamic_conferen
ce] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[dynamic_conference]
destination_number(7300) =~ /^5001$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->rtp_multicast_pa
ge] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[rtp_multicast_page]
destination_number(7300) =~ /^pagegroup$|^7243$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
continue=f
alse
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
destination_nu
mber(7300) =~ /^5900$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
continue
=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
destination_
number(7300) =~ /^5901$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
continue=f
alse
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [park]
source(mod_sof
ia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
destination_nu
mber(7300) =~ /park\+(\d+)/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
continue
=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unpark]
source(mod_s
ofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
destination_
number(7300) =~ /^parking$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
continue=f
alse
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [park]
source(mod_sof
ia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
destination_nu
mber(7300) =~ /callpark/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
continue
=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unpark]
source(mod_s
ofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
destination_
number(7300) =~ /pickup/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->wait]
continue=f
alse
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [wait]
destination_nu
mber(7300) =~ /^wait$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->fax_receive]
con
tinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [fax_receive]
destina
tion_number(7300) =~ /^9978$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->fax_transmit] co
ntinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [fax_transmit]
destin
ation_number(7300) =~ /^9979$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->ringback_180] co
ntinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ringback_180]
destin
ation_number(7300) =~ /^9980$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->ringback_183_uk_
ring] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[ringback_183_uk_ring
] destination_number(7300) =~ /^9981$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->ringback_183_mus
ic_ring] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[ringback_183_music_r
ing] destination_number(7300) =~ /^9982$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->ringback_post_an
swer_uk_ring] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[ringback_post_answer
_uk_ring] destination_number(7300) =~ /^9983$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->ringback_post_an
swer_music] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[ringback_post_answer
_music] destination_number(7300) =~ /^9984$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->ClueCon]
continu
e=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ClueCon]
destination
_number(7300) =~ /^9991$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->show_info]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [show_info]
destinati
on_number(7300) =~ /^9992$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->video_record] co
ntinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [video_record]
destin
ation_number(7300) =~ /^9993$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->video_playback]
continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [video_playback]
dest
ination_number(7300) =~ /^9994$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->delay_echo]
cont
inue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [delay_echo]
destinat
ion_number(7300) =~ /^9995$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->echo]
continue=f
alse
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [echo]
destination_nu
mber(7300) =~ /^9996$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->milliwatt]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [milliwatt]
destinati
on_number(7300) =~ /^9997$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->tone_stream]
con
tinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [tone_stream]
destina
tion_number(7300) =~ /^9998$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->zrtp_enrollement
] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[zrtp_enrollement] de
stination_number(7300) =~ /^9787$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->hold_music]
cont
inue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [hold_music]
destinat
ion_number(7300) =~ /^9999$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->Rednote]
continu
e=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [Rednote]
destination
_number(7300) =~ /^0220(\d+)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->carmickle]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [carmickle]
destinati
on_number(7300) =~ /^0352(\d+)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->call_jason]
cont
inue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call_jason]
destinat
ion_number(7300) =~ /^059(\d+)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->TOI]
continue=fa
lse
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [TOI]
destination_num
ber(7300) =~ /^08740(\d+)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->Group 7300]
cont
inue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [Group 7300]
destinat
ion_number(7300) =~ /(7300)/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Action
set(hangup_after_bridge=tru
e)
Dialplan: sofia/internal/7305 at ewr.djernes.net Action
set(continue_on_fail=true)
Dialplan: sofia/internal/7305 at ewr.djernes.net Action set(call_timeout=15)
Dialplan: sofia/internal/7305 at ewr.djernes.net Action
bridge({ignore_early_media=
true}${group_call(sdjernes+A@${domain_name})})
Dialplan: sofia/internal/7305 at ewr.djernes.net Action hangup()
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:114
(sofia/intern
al/7305 at ewr.djernes.net) State Change CS_ROUTING -> CS_EXECUTE
2009-08-25 14:27:45.287037 [DEBUG] switch_core_session.c:932 Send signal
sofia/i
nternal/7305 at ewr.djernes.net [BREAK]
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:484
(sofia/intern
al/7305 at ewr.djernes.net) State ROUTING going to sleep
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
(sofia/intern
al/7305 at ewr.djernes.net) Running State Change CS_EXECUTE
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:491
(sofia/intern
al/7305 at ewr.djernes.net) State EXECUTE
2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:173
sofia/internal/7305 at ewr.djern
es.net SOFIA EXECUTE
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:151
sofia/interna
l/7305 at ewr.djernes.net Standard EXECUTE
EXECUTE sofia/internal/7305 at ewr.djernes.net set(open=true)
2009-08-25 14:27:45.287037 [DEBUG] mod_dptools.c:748
sofia/internal/7305 at ewr.dje
rnes.net SET [open]=[true]
EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-s
pymap/7305/733dbed2-9183-11de-95f0-07acd7038fc1)
EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-l
ast_dial/7305/7300)
EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-l
ast_dial/global/733dbed2-9183-11de-95f0-07acd7038fc1)
EXECUTE sofia/internal/7305 at ewr.djernes.net set(hangup_after_bridge=true)
2009-08-25 14:27:45.291899 [DEBUG] mod_dptools.c:748
sofia/internal/7305 at ewr.dje
rnes.net SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/7305 at ewr.djernes.net set(continue_on_fail=true)
2009-08-25 14:27:45.291899 [DEBUG] mod_dptools.c:748
sofia/internal/7305 at ewr.dje
rnes.net SET [continue_on_fail]=[true]
EXECUTE sofia/internal/7305 at ewr.djernes.net set(call_timeout=15)
2009-08-25 14:27:45.294755 [DEBUG] mod_dptools.c:748
sofia/internal/7305 at ewr.dje
rnes.net SET [call_timeout]=[15]
EXECUTE sofia/internal/7305 at ewr.djernes.netbridge({ignore_early_media=true})
2009-08-25 14:27:45.318579 [WARNING] switch_ivr_originate.c:1001 No
origination
URL specified!
2009-08-25 14:27:45.318579 [DEBUG] switch_ivr_originate.c:2138 Originate
Resulte
d in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2009-08-25 14:27:45.318579 [INFO] mod_dptools.c:2093 Originate Failed.
 Cause: D
ESTINATION_OUT_OF_ORDER
2009-08-25 14:27:45.318579 [DEBUG] mod_dptools.c:2120 Continue on fail
[true]:
Cause: DESTINATION_OUT_OF_ORDER
EXECUTE sofia/internal/7305 at ewr.djernes.net hangup()
2009-08-25 14:27:45.318579 [NOTICE] mod_dptools.c:633 Hangup
sofia/internal/7305
@ewr.djernes.net [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-25 14:27:45.318579 [DEBUG] switch_channel.c:1683 Send signal
sofia/inter
nal/7305 at ewr.djernes.net [KILL]
2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:932 Send signal
sofia/i
nternal/7305 at ewr.djernes.net [BREAK]
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:491
(sofia/intern
al/7305 at ewr.djernes.net) State EXECUTE going to sleep
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:398
(sofia/intern
al/7305 at ewr.djernes.net) Running State Change CS_HANGUP
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:434
(sofia/intern
al/7305 at ewr.djernes.net) State HANGUP
2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:338 Channel
sofia/internal/7305 at e
wr.djernes.net hanging up, cause: NORMAL_CLEARING
2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:417 Responding to INVITE
with: 48
0
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:46
sofia/internal
/7305 at ewr.djernes.net Standard HANGUP, cause: NORMAL_CLEARING
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:434
(sofia/intern
al/7305 at ewr.djernes.net) State HANGUP going to sleep
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:476
(sofia/intern
al/7305 at ewr.djernes.net) State Change CS_HANGUP -> CS_REPORTING
2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:932 Send signal
sofia/i
nternal/7305 at ewr.djernes.net [BREAK]
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:398
(sofia/intern
al/7305 at ewr.djernes.net) Running State Change CS_REPORTING
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:612
(sofia/intern
al/7305 at ewr.djernes.net) State REPORTING
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:53
sofia/internal
/7305 at ewr.djernes.net Standard REPORTING, cause: NORMAL_CLEARING
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:612
(sofia/intern
al/7305 at ewr.djernes.net) State REPORTING going to sleep
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:411
(sofia/intern
al/7305 at ewr.djernes.net) State Change CS_REPORTING -> CS_DESTROY
2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:1068 Session 117
(sofia
/internal/7305 at ewr.djernes.net) Locked, Waiting on external entities
2009-08-25 14:27:45.318579 [NOTICE] switch_core_session.c:1086 Session 117
(sofi
a/internal/7305 at ewr.djernes.net) Ended
2009-08-25 14:27:45.318579 [NOTICE] switch_core_session.c:1088 Close Channel
sof
ia/internal/7305 at ewr.djernes.net [CS_DESTROY]
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:564
(sofia/intern
al/7305 at ewr.djernes.net) State DESTROY
2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:255
sofia/internal/7305 at ewr.djern
es.net SOFIA DESTROY
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:60
sofia/internal
/7305 at ewr.djernes.net Standard DESTROY
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:564
(sofia/intern
al/7305 at ewr.djernes.net) State DESTROY going to sleep


Any Help would be much appreciated.
-- 
Shawn L. Djernes
SD Consulting
shawn at djernes.org | sdjernes at gmail.com
MSN: wizardwlf at hotmail.com
402.345.7734 | 402.350.6973 Cell
-------------- next part --------------
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From mike at jerris.com  Tue Aug 25 08:29:26 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 11:29:26 -0400
Subject: [Freeswitch-users] reload user data
In-Reply-To: <27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>
References: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>
	<C6B90B49.5D5AC%krice@freeswitch.org>
	<27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>
Message-ID: <9FE7AA03-C09B-4DB3-A15A-CF99D2052726@jerris.com>

You can also do xml config hooks in perl and some of the other  
embedded languages.

Mike

On Aug 25, 2009, at 5:22 AM, Juan Backson wrote:

> Hi Ken,
>
> xml_curl is a great idea.  Is there anyway to not having to setup  
> another HTTP server?  For instance, can I have freeswitch to call an  
> api or call a lua or php or c script that will return the xml  
> response?  That way, I don't need to maintain yet another service.
>
> Thanks,
> JB
>
> On Tue, Aug 25, 2009 at 4:30 PM, Ken Rice <krice at freeswitch.org>  
> wrote:
> You just need to reloadxml you don?t have to restart the whole  
> thing. You can also use xml_curl to feed the users from a database  
> see my contrib directory (contrib/swk) for some example scripts and  
> db code
>
>
> From: Juan Backson <juanbackson at gmail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 25 Aug 2009 16:21:58 +0800
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] reload user data
>
>
> Hello,
>
> I would like to dynamically add user to freeswitch.  If I add a new  
> file to the directory dir, is there anyway to have freeswitch to  
> read the new user xml file without having to restart freeswitch?
>
> Other than using flat file, is there anyway to add user to  
> freeswitch user api command?
>
> Thanks,
> JB

-------------- next part --------------
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From mike at jerris.com  Tue Aug 25 08:31:49 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 11:31:49 -0400
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
Message-ID: <2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>

Actually in this case, we are bound to one thread in sofia.

Mike

On Aug 25, 2009, at 9:47 AM, Giovanni Maruzzelli wrote:

> is a heavely multithreaded software, it benefits from number of CPUs
> (or cores), RAM, and heavy duty kernel features (found in 64bit
> kernels)
>
> put all accesses on ramdisk, leave out the modules you don't use...
>
> experiment, test, and find the best for your specific application/ 
> workload
>
> test not only with sipp, but with real load too (often they're very  
> different)
>
> -gm
>



From msc at freeswitch.org  Tue Aug 25 08:55:29 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 08:55:29 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
Message-ID: <87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>

On Tue, Aug 25, 2009 at 7:20 AM, Mike Peace <MPeace at edcogroupinc.com> wrote:

>  I fired up Wireshark on each side and I can see the SIP register request
> coming from the laptop, the Freeswitch server replies with a Destination
> Unreachable (Port unreachable) message.
>
>
>
> I rebooted the Server and now I get a ?Registration error; 405 Method not
> allowed? on the softphone and the Wiresharp capture shows: ?Status 405
> Method Not Allowerd (0 bindings).
>
>
>
Hmm... can you fire up FS and then do "sofia status" and tell us if the
internal profile looks okay? Also, do "sofia status profile internal" and
capture the output. Something fishy is going on...
-MC

> It does this even after I stop and restart the Freeswitch service.
>
>
>
> Shouldn?t all the ports be open if the firewall and SELunix are disabled?
>
>
>
> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 5:32 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:49 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> It does the same thing, does anything get set during the install that would
> remember or cache the old network settings? I can access anything from the
> FS server on any of several networks and vice-versa but the SIP will not
> register, again no firewalls are upon any of the test hosts.
>
> Doesn?t make sense to me.
>
>
>
> Time to bust out tcpdump and/or wireshark to make 100% certain you know
> what's happening with all those SIP packets.
>
>   *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:34 PM
>
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> ?Registration error: 408-Request Timeout?
>
> Sorry for the the typo - 408 = timeout. (480 = temp unavail)
>
> Try stopping iptables in Linux and try again. Sounds like something is
> interfering with your packets getting from here to there...
> Try:
>
> /etc/init.d/iptables stop
>
> And then see if your packets can move again.
>
> -MC
>
>
>
> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:18 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> I haven?t changed any of the conf files.
>
>
>
> What happens when you try to register? Do you get 480? (timeout) Or
> something else?
> -MC
>
>
>  ------------------------------
>
>
>
> *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>
>
> Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>  ------------------------------
>
>
>
> *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>
>
> Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
>
>  *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>  Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Aug 25 08:59:28 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 08:59:28 -0700
Subject: [Freeswitch-users] using sipp to test call transfers
In-Reply-To: <84D97940-68D0-402F-A7E2-B5ADD287DC73@runsolutions.com>
References: <84D97940-68D0-402F-A7E2-B5ADD287DC73@runsolutions.com>
Message-ID: <87f2f3b90908250859w5f5c8f25qd1ec1a83c9a706fd@mail.gmail.com>

On Tue, Aug 25, 2009 at 2:06 AM, Raimund Sacherer <rs at runsolutions.com>wrote:

> Hi List,
>
> i have some scripts to test our lab:
>
> I have scripts to create sipp instances which act as individual
> agents, which log on, take a random amount of calls, log off, wait a
> bit, log on again, etc.
> I have scripts to generate call traffic for our queues to saturate them
>
> But what I have difficulties is how to create a good scenario xml file
> for call transfers? Has anybody here a sample for me to sink my teeth
> in?


Start with this...
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations

-MC


>
>
> thanks and best regards
>
> --
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Aug 25 09:05:14 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 09:05:14 -0700
Subject: [Freeswitch-users] Freeswitch & DAHDI Kernel drivers / Hardware
	questions
In-Reply-To: <7CEF2F0A-0735-41E8-ABA5-FBCB2A0AC0DF@runsolutions.com>
References: <DD5C7209-7041-4D21-869D-2F3BABEADC78@runsolutions.com>
	<c4d05cbe0908250724j407636e0o33f0df4bbfbb461@mail.gmail.com>
	<7CEF2F0A-0735-41E8-ABA5-FBCB2A0AC0DF@runsolutions.com>
Message-ID: <87f2f3b90908250905y4516550ag11b92d0642179f04@mail.gmail.com>

On Tue, Aug 25, 2009 at 7:35 AM, Raimund Sacherer <rs at runsolutions.com>wrote:

>
> On Aug 25, 2009, at 4:24 PM, Moises Silva wrote:
>
> On Tue, Aug 25, 2009 at 9:35 AM, Raimund Sacherer <rs at runsolutions.com>wrote:
>
>> Hello,
>> i was reading through the openzap wiki page and searched the wiki, the
>> only thing I found about dahdi is a note from January that freeswitch does
>> not work with dahdi right now.
>>
>>
> You did not search well enough.
> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 , read note 5 there.
>
> Thank you, i did not see any link to OpenZAP_OpenR2, but this page explains
> a lot for me!
>

FYI,
I added a link to the R2 page on the main OpenZAP page.
-MC
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From msc at freeswitch.org  Tue Aug 25 09:11:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 09:11:51 -0700
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
Message-ID: <87f2f3b90908250911m31ef0914t136f7511b59ecb79@mail.gmail.com>

> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
> sense to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
> will FS perofomr drastically better 20%+ ?
>

If you really want to get on the same page as the developers then get the
64bit CentOS 5.3 loaded on your machine. Also, we've seen reports of really
weird things happening when people run 32bit OS on 64bit hardware. That's a
big no-no. :)

Let us know how it goes.
-MC
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From msc at freeswitch.org  Tue Aug 25 09:14:15 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 09:14:15 -0700
Subject: [Freeswitch-users] reload user data
In-Reply-To: <9FE7AA03-C09B-4DB3-A15A-CF99D2052726@jerris.com>
References: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>
	<C6B90B49.5D5AC%krice@freeswitch.org>
	<27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>
	<9FE7AA03-C09B-4DB3-A15A-CF99D2052726@jerris.com>
Message-ID: <87f2f3b90908250914r4a864f25o677e1735d1856ff4@mail.gmail.com>

On Tue, Aug 25, 2009 at 8:29 AM, Michael Jerris <mike at jerris.com> wrote:

> You can also do xml config hooks in perl and some of the other embedded
> languages.
>

FYI, here's a page to go you started:
http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML

-MC
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From tculjaga at gmail.com  Tue Aug 25 09:34:42 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 18:34:42 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
Message-ID: <65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>

Exactly... the scenario i use seems operating on a single thread... why is
that ? can it be changed?

T.

On Tue, Aug 25, 2009 at 5:31 PM, Michael Jerris <mike at jerris.com> wrote:

> Actually in this case, we are bound to one thread in sofia.
>
> Mike
>
> On Aug 25, 2009, at 9:47 AM, Giovanni Maruzzelli wrote:
>
> > is a heavely multithreaded software, it benefits from number of CPUs
> > (or cores), RAM, and heavy duty kernel features (found in 64bit
> > kernels)
> >
> > put all accesses on ramdisk, leave out the modules you don't use...
> >
> > experiment, test, and find the best for your specific application/
> > workload
> >
> > test not only with sipp, but with real load too (often they're very
> > different)
> >
> > -gm
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Tue Aug 25 09:35:36 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 18:35:36 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <87f2f3b90908250911m31ef0914t136f7511b59ecb79@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<87f2f3b90908250911m31ef0914t136f7511b59ecb79@mail.gmail.com>
Message-ID: <65d96fc80908250935j4be3574ala817f9823204b462@mail.gmail.com>

well :) ... this is something we are going to change tomorrow.... of course
will let you posted.

T.

On Tue, Aug 25, 2009 at 6:11 PM, Michael Collins <msc at freeswitch.org> wrote:

>
> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
>> sense to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
>> will FS perofomr drastically better 20%+ ?
>>
>
> If you really want to get on the same page as the developers then get the
> 64bit CentOS 5.3 loaded on your machine. Also, we've seen reports of really
> weird things happening when people run 32bit OS on 64bit hardware. That's a
> big no-no. :)
>
> Let us know how it goes.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Tue Aug 25 09:41:29 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 25 Aug 2009 13:41:29 -0300
Subject: [Freeswitch-users] Group Call
In-Reply-To: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
Message-ID: <ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>

According to example dialplan you hould do this:

<action application="bridge" data="{ignore_early_media=true}${group(call:$1@
${domain_name})}"/>

According to wiki:

group,[insert|delete|call]:<group name>:<url>,group [insert|delete|call


Only difference I see here is the "+A" on your bridge statement.


jmesquita


On Tue, Aug 25, 2009 at 11:30 AM, Shawn L. Djernes <sdjernes at gmail.com>wrote:

> Hello,
> I am trying to get group calling to work.  AKA: dial an extension (7300 in
> this case) and it ring all phones at once.  We are moving a server from
> Asterisk to FS
>
> Here is the conf/directory/default.xml section:
>
>         <group name="sdjernes">
>                 <users>
>                         <user id="7301" type="pointer"/>
>                         <user id="7302" type="pointer"/>
>                         <user id="7303" type="pointer"/>
>                         <user id="7304" type="pointer"/>
>                 </users>
>         </group>
>
> Here is the conf/dialplan/default/7300_group.xml extension statement:
>
>      <extension name="Group 7300">
>        <condition field="destination_number" expression="(7300)">
>          <action application="set" data="hangup_after_bridge=true"/>
>          <action application="set" data="continue_on_fail=true"/>
>          <action application="set" data="call_timeout=15"/>
>          <action application="bridge"
> data="{ignore_early_media=true}${group(call:sdjernes+A@${domain_name})}"/>
> <!--         <action application="transfer" data="7301 XML default"/> -->
>          <action application="hangup"/>
>        </condition>
>      </extension>
>
> The log output shows the following:
>
> 2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3289 Channel
> sofia/internal/7305 at ewr.
> djernes.net entering state [received][100]
> 2009-08-25 14:27:45.282619 [DEBUG] switch_core_state_machine.c:398
> (sofia/intern
> al/7305 at ewr.djernes.net) Running State Change CS_NEW
> 2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3296 Remote SDP:
> v=0
> o=root 628695665 628695665 IN IP4 98.188.201.109
> s=call
> c=IN IP4 98.188.201.109
> t=0 0
> m=audio 61012 RTP/AVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[CELT:114:48000:10]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[CELT:114:32000:10]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[SPEEX:98:8000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[SPEEX:98:8000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[SPEEX:98:8000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[G7221:115:32000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[G7221:107:16000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[G722:9:8000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[PCMU:0:8000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:2029 Set Codec
> sofia/internal/73
> 05 at ewr.djernes.net PCMU/8000 20 ms 160 samples
> 2009-08-25 14:27:45.282619 [DEBUG] switch_core_state_machine.c:404
> (sofia/intern
> al/7305 at ewr.djernes.net) State NEW
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload
> to 10
> 1
> 2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3455
> (sofia/internal/7305 at ewr.djernes
> .net) State Change CS_NEW -> CS_INIT
> 2009-08-25 14:27:45.282619 [DEBUG] switch_core_session.c:932 Send signal
> sofia/i
> nternal/7305 at ewr.djernes.net [BREAK]
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
> (sofia/intern
> al/7305 at ewr.djernes.net) Running State Change CS_INIT
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:481
> (sofia/intern
> al/7305 at ewr.djernes.net) State INIT
> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:83
> sofia/internal/7305 at ewr.djerne
> s.net SOFIA INIT
> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:111
> (sofia/internal/7305 at ewr.djer
> nes.net) State Change CS_INIT -> CS_ROUTING
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_session.c:932 Send signal
> sofia/i
> nternal/7305 at ewr.djernes.net [BREAK]
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:481
> (sofia/intern
> al/7305 at ewr.djernes.net) State INIT going to sleep
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
> (sofia/intern
> al/7305 at ewr.djernes.net) Running State Change CS_ROUTING
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:484
> (sofia/intern
> al/7305 at ewr.djernes.net) State ROUTING
> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:130
> sofia/internal/7305 at ewr.djern
> es.net SOFIA ROUTING
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:78
> sofia/internal
> /7305 at ewr.djernes.net Standard ROUTING
> 2009-08-25 14:27:45.287037 [INFO] mod_dialplan_xml.c:315 Processing
> Sdjernes TOI
>  Phone->7300 in context default
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unloop]
> continue
> =false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unloop]
> ${unroll_loo
> ps}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unloop]
> ${sip_looped
> _call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->tod_example] con
> tinue=true
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [tod_example]
> ${strft
> ime(%w)}(2) =~ /^([1-5])$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [tod_example]
> ${strft
> ime(%H%M)}(1427) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action set(open=true)
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->global-intercept
> ] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [global-intercept] de
> stination_number(7300) =~ /^886$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->group-intercept]
>  continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [group-intercept] des
> tination_number(7300) =~ /^\*8$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->intercept-ext] c
> ontinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [intercept-ext]
> desti
> nation_number(7300) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->redial]
> continue
> =false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [redial]
> destination_
> number(7300) =~ /^870$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->global]
> continue
> =true
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [global]
> ${call_debug
> }(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [global]
> ${sip_has_cr
> ypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
> break=never
> Dialplan: sofia/internal/7305 at ewr.djernes.net Absolute Condition [global]
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
> hash(insert/${domain_name}-
> spymap/${caller_id_number}/${uuid})
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
> hash(insert/${domain_name}-
> last_dial/${caller_id_number}/${destination_number})
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
> hash(insert/${domain_name}-
> last_dial/global/${uuid})
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->snom-demo-2] con
> tinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [snom-demo-2]
> destina
> tion_number(7300) =~ /^9001$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->snom-demo-1] con
> tinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [snom-demo-1]
> destina
> tion_number(7300) =~ /^9000$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->eavesdrop]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [eavesdrop]
> destinati
> on_number(7300) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->eavesdrop]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [eavesdrop]
> destinati
> on_number(7300) =~ /^779$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->call_return] con
> tinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call_return]
> destina
> tion_number(7300) =~ /^\*69$|^869$|^lcr$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->del-group]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [del-group]
> destinati
> on_number(7300) =~ /^80(\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->add-group]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [add-group]
> destinati
> on_number(7300) =~ /^81(\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->call-group-simo]
>  continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [call-group-simo] des
> tination_number(7300) =~ /^82(\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->call-group-order
> ] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [call-group-order] de
> stination_number(7300) =~ /^83(\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->extension-interc
> om] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [extension-intercom]
> destination_number(7300) =~ /^8(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->Local_Extension]
>  continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [Local_Extension] des
> tination_number(7300) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->group_dial_sales
> ] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [group_dial_sales] de
> stination_number(7300) =~ /^2000$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->group_dial_suppo
> rt] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [group_dial_support]
> destination_number(7300) =~ /^2001$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->group_dial_billi
> ng] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [group_dial_billing]
> destination_number(7300) =~ /^2002$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->operator]
> contin
> ue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [operator]
> destinatio
> n_number(7300) =~ /^(operator|0)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->vmain]
> continue=
> false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [vmain]
> destination_n
> umber(7300) =~ /^vmain$|^4000$|^\*98$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->sip_uri]
> continu
> e=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [sip_uri]
> destination
> _number(7300) =~ /^sip:(.*)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->nb_conferences]
> continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [nb_conferences] dest
> ination_number(7300) =~ /^(30\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->wb_conferences]
> continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [wb_conferences] dest
> ination_number(7300) =~ /^(31\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->uwb_conferences]
>  continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [uwb_conferences] des
> tination_number(7300) =~ /^(32\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->cdquality_confer
> ences] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [cdquality_conference
> s] destination_number(7300) =~ /^(33\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->freeswitch_publi
> c_conf_via_sip] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [freeswitch_public_co
> nf_via_sip] destination_number(7300) =~ /^9(888|1616|3232)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->mad_boss_interco
> m] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [mad_boss_intercom] d
> estination_number(7300) =~ /^0911$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->mad_boss_interco
> m] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [mad_boss_intercom] d
> estination_number(7300) =~ /^0912$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->mad_boss]
> contin
> ue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [mad_boss]
> destinatio
> n_number(7300) =~ /^0913$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->ivr_demo]
> contin
> ue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ivr_demo]
> destinatio
> n_number(7300) =~ /^5000$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->dynamic_conferen
> ce] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [dynamic_conference]
> destination_number(7300) =~ /^5001$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->rtp_multicast_pa
> ge] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [rtp_multicast_page]
> destination_number(7300) =~ /^pagegroup$|^7243$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
> continue=f
> alse
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
> destination_nu
> mber(7300) =~ /^5900$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
> continue
> =false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
> destination_
> number(7300) =~ /^5901$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
> continue=f
> alse
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [park]
> source(mod_sof
> ia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
> destination_nu
> mber(7300) =~ /park\+(\d+)/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
> continue
> =false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unpark]
> source(mod_s
> ofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
> destination_
> number(7300) =~ /^parking$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
> continue=f
> alse
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [park]
> source(mod_sof
> ia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
> destination_nu
> mber(7300) =~ /callpark/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
> continue
> =false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unpark]
> source(mod_s
> ofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
> destination_
> number(7300) =~ /pickup/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->wait]
> continue=f
> alse
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [wait]
> destination_nu
> mber(7300) =~ /^wait$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->fax_receive] con
> tinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [fax_receive]
> destina
> tion_number(7300) =~ /^9978$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->fax_transmit] co
> ntinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [fax_transmit]
> destin
> ation_number(7300) =~ /^9979$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->ringback_180] co
> ntinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ringback_180]
> destin
> ation_number(7300) =~ /^9980$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->ringback_183_uk_
> ring] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [ringback_183_uk_ring
> ] destination_number(7300) =~ /^9981$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->ringback_183_mus
> ic_ring] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [ringback_183_music_r
> ing] destination_number(7300) =~ /^9982$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->ringback_post_an
> swer_uk_ring] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [ringback_post_answer
> _uk_ring] destination_number(7300) =~ /^9983$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->ringback_post_an
> swer_music] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [ringback_post_answer
> _music] destination_number(7300) =~ /^9984$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->ClueCon]
> continu
> e=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ClueCon]
> destination
> _number(7300) =~ /^9991$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->show_info]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [show_info]
> destinati
> on_number(7300) =~ /^9992$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->video_record] co
> ntinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [video_record]
> destin
> ation_number(7300) =~ /^9993$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->video_playback]
> continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [video_playback] dest
> ination_number(7300) =~ /^9994$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->delay_echo] cont
> inue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [delay_echo]
> destinat
> ion_number(7300) =~ /^9995$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->echo]
> continue=f
> alse
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [echo]
> destination_nu
> mber(7300) =~ /^9996$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->milliwatt]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [milliwatt]
> destinati
> on_number(7300) =~ /^9997$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->tone_stream] con
> tinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [tone_stream]
> destina
> tion_number(7300) =~ /^9998$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->zrtp_enrollement
> ] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [zrtp_enrollement] de
> stination_number(7300) =~ /^9787$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->hold_music] cont
> inue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [hold_music]
> destinat
> ion_number(7300) =~ /^9999$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->Rednote]
> continu
> e=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [Rednote]
> destination
> _number(7300) =~ /^0220(\d+)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->carmickle]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [carmickle]
> destinati
> on_number(7300) =~ /^0352(\d+)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->call_jason] cont
> inue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call_jason]
> destinat
> ion_number(7300) =~ /^059(\d+)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->TOI]
> continue=fa
> lse
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [TOI]
> destination_num
> ber(7300) =~ /^08740(\d+)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->Group
> 7300] cont
> inue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [Group 7300]
> destinat
> ion_number(7300) =~ /(7300)/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
> set(hangup_after_bridge=tru
> e)
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
> set(continue_on_fail=true)
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action set(call_timeout=15)
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
> bridge({ignore_early_media=
> true}${group_call(sdjernes+A@${domain_name})})
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action hangup()
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:114
> (sofia/intern
> al/7305 at ewr.djernes.net) State Change CS_ROUTING -> CS_EXECUTE
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_session.c:932 Send signal
> sofia/i
> nternal/7305 at ewr.djernes.net [BREAK]
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:484
> (sofia/intern
> al/7305 at ewr.djernes.net) State ROUTING going to sleep
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
> (sofia/intern
> al/7305 at ewr.djernes.net) Running State Change CS_EXECUTE
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:491
> (sofia/intern
> al/7305 at ewr.djernes.net) State EXECUTE
> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:173
> sofia/internal/7305 at ewr.djern
> es.net SOFIA EXECUTE
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:151
> sofia/interna
> l/7305 at ewr.djernes.net Standard EXECUTE
> EXECUTE sofia/internal/7305 at ewr.djernes.net set(open=true)
> 2009-08-25 14:27:45.287037 [DEBUG] mod_dptools.c:748
> sofia/internal/7305 at ewr.dje
> rnes.net SET [open]=[true]
> EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-s
> pymap/7305/733dbed2-9183-11de-95f0-07acd7038fc1)
> EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-l
> ast_dial/7305/7300)
> EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-l
> ast_dial/global/733dbed2-9183-11de-95f0-07acd7038fc1)
> EXECUTE sofia/internal/7305 at ewr.djernes.net set(hangup_after_bridge=true)
> 2009-08-25 14:27:45.291899 [DEBUG] mod_dptools.c:748
> sofia/internal/7305 at ewr.dje
> rnes.net SET [hangup_after_bridge]=[true]
> EXECUTE sofia/internal/7305 at ewr.djernes.net set(continue_on_fail=true)
> 2009-08-25 14:27:45.291899 [DEBUG] mod_dptools.c:748
> sofia/internal/7305 at ewr.dje
> rnes.net SET [continue_on_fail]=[true]
> EXECUTE sofia/internal/7305 at ewr.djernes.net set(call_timeout=15)
> 2009-08-25 14:27:45.294755 [DEBUG] mod_dptools.c:748
> sofia/internal/7305 at ewr.dje
> rnes.net SET [call_timeout]=[15]
> EXECUTE sofia/internal/7305 at ewr.djernes.netbridge({ignore_early_media=true})
> 2009-08-25 14:27:45.318579 [WARNING] switch_ivr_originate.c:1001 No
> origination
> URL specified!
> 2009-08-25 14:27:45.318579 [DEBUG] switch_ivr_originate.c:2138 Originate
> Resulte
> d in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
> 2009-08-25 14:27:45.318579 [INFO] mod_dptools.c:2093 Originate Failed.
>  Cause: D
> ESTINATION_OUT_OF_ORDER
> 2009-08-25 14:27:45.318579 [DEBUG] mod_dptools.c:2120 Continue on fail
> [true]:
> Cause: DESTINATION_OUT_OF_ORDER
> EXECUTE sofia/internal/7305 at ewr.djernes.net hangup()
> 2009-08-25 14:27:45.318579 [NOTICE] mod_dptools.c:633 Hangup
> sofia/internal/7305
> @ewr.djernes.net [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-08-25 14:27:45.318579 [DEBUG] switch_channel.c:1683 Send signal
> sofia/inter
> nal/7305 at ewr.djernes.net [KILL]
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:932 Send signal
> sofia/i
> nternal/7305 at ewr.djernes.net [BREAK]
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:491
> (sofia/intern
> al/7305 at ewr.djernes.net) State EXECUTE going to sleep
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:398
> (sofia/intern
> al/7305 at ewr.djernes.net) Running State Change CS_HANGUP
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:434
> (sofia/intern
> al/7305 at ewr.djernes.net) State HANGUP
> 2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:338 Channel
> sofia/internal/7305 at e
> wr.djernes.net hanging up, cause: NORMAL_CLEARING
> 2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:417 Responding to INVITE
> with: 48
> 0
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:46
> sofia/internal
> /7305 at ewr.djernes.net Standard HANGUP, cause: NORMAL_CLEARING
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:434
> (sofia/intern
> al/7305 at ewr.djernes.net) State HANGUP going to sleep
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:476
> (sofia/intern
> al/7305 at ewr.djernes.net) State Change CS_HANGUP -> CS_REPORTING
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:932 Send signal
> sofia/i
> nternal/7305 at ewr.djernes.net [BREAK]
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:398
> (sofia/intern
> al/7305 at ewr.djernes.net) Running State Change CS_REPORTING
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:612
> (sofia/intern
> al/7305 at ewr.djernes.net) State REPORTING
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:53
> sofia/internal
> /7305 at ewr.djernes.net Standard REPORTING, cause: NORMAL_CLEARING
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:612
> (sofia/intern
> al/7305 at ewr.djernes.net) State REPORTING going to sleep
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:411
> (sofia/intern
> al/7305 at ewr.djernes.net) State Change CS_REPORTING -> CS_DESTROY
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:1068 Session 117
> (sofia
> /internal/7305 at ewr.djernes.net) Locked, Waiting on external entities
> 2009-08-25 14:27:45.318579 [NOTICE] switch_core_session.c:1086 Session 117
> (sofi
> a/internal/7305 at ewr.djernes.net) Ended
> 2009-08-25 14:27:45.318579 [NOTICE] switch_core_session.c:1088 Close
> Channel sof
> ia/internal/7305 at ewr.djernes.net [CS_DESTROY]
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:564
> (sofia/intern
> al/7305 at ewr.djernes.net) State DESTROY
> 2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:255
> sofia/internal/7305 at ewr.djern
> es.net SOFIA DESTROY
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:60
> sofia/internal
> /7305 at ewr.djernes.net Standard DESTROY
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:564
> (sofia/intern
> al/7305 at ewr.djernes.net) State DESTROY going to sleep
>
>
> Any Help would be much appreciated.
> --
> Shawn L. Djernes
> SD Consulting
> shawn at djernes.org | sdjernes at gmail.com
> MSN: wizardwlf at hotmail.com
> 402.345.7734 | 402.350.6973 Cell
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From MPeace at edcogroupinc.com  Tue Aug 25 10:04:50 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 12:04:50 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>

I get a bash: sofia: command not found. Is there something I need to add to my config to use these commands?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 25, 2009 10:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


On Tue, Aug 25, 2009 at 7:20 AM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

I fired up Wireshark on each side and I can see the SIP register request coming from the laptop, the Freeswitch server replies with a Destination Unreachable (Port unreachable) message.



I rebooted the Server and now I get a "Registration error; 405 Method not allowed" on the softphone and the Wiresharp capture shows: "Status 405 Method Not Allowerd (0 bindings).


Hmm... can you fire up FS and then do "sofia status" and tell us if the internal profile looks okay? Also, do "sofia status profile internal" and capture the output. Something fishy is going on...
-MC

It does this even after I stop and restart the Freeswitch service.



Shouldn't all the ports be open if the firewall and SELunix are disabled?



Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 5:32 PM

To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.





On Mon, Aug 24, 2009 at 2:49 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

It does the same thing, does anything get set during the install that would remember or cache the old network settings? I can access anything from the FS server on any of several networks and vice-versa but the SIP will not register, again no firewalls are upon any of the test hosts.

Doesn't make sense to me.



Time to bust out tcpdump and/or wireshark to make 100% certain you know what's happening with all those SIP packets.

Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:34 PM

To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.





On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

"Registration error: 408-Request Timeout"

Sorry for the the typo - 408 = timeout. (480 = temp unavail)

Try stopping iptables in Linux and try again. Sounds like something is interfering with your packets getting from here to there...
Try:

/etc/init.d/iptables stop

And then see if your packets can move again.

-MC



Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:18 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.





On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

I haven't changed any of the conf files.



What happens when you try to register? Do you get 480? (timeout) Or something else?
-MC



________________________________



EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/



Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





________________________________



EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/



Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From anthony.minessale at gmail.com  Tue Aug 25 10:05:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Aug 2009 12:05:14 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
	<15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>
	<15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com>
	<65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com>
Message-ID: <191c3a030908251005v6657943l5cdd7645446fcbc5@mail.gmail.com>

I wish I had a nickel for every guy struggling with sipp load testing vs
real world traffic.


On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Hello Takeshi,
>
> Thanks for your hint... it worked out... so to be precise:
>
> VIA header of both INVITE and ACK messages MUST be identical (IP:PORT +
> branch)... and you are right... it might not be according to SIP
> specification. Anyhow, i get FS understand my ACK message.
>
>
> Finally, here is what i used and I'm getting some poor results .. but this
> is another topic :)
>
>
> Thanks for your help.
> Tihomir.
>
>
> sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1
> -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
>
>
> <?xml version="1.0" encoding="ISO-8859-1" ?>
> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>
>
> <scenario name="Basic Sipstone UAC">
>   <send retrans="500" start_rtd="1" start_rtd="2">
>
>     <![CDATA[
>
>       INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>       Max-Forwards: 70
>       Contact: <sip:[field1]@[local_ip]>
>       From: [field1]
> <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
>       To: [service] <sip:[service]@[remote_ip]:[remote_port]>
>       Call-ID: [call_id]
>       CSeq: 1 INVITE
>       Content-Type: application/sdp
>       Content-Length: [len]
>
>       v=0
>       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>       s=-
>       c=IN IP[media_ip_type] [media_ip]
>       t=0 0
>       m=audio [media_port] RTP/AVP 0
>       a=rtpmap:0 PCMU/8000
>
>     ]]>
>   </send>
>
>   <recv response="100"
>         optional="true" rtd="1">
>   </recv>
>
>
>   <recv response="302" rtd="2">
>   </recv>
>
>   <send>
>     <![CDATA[
>
>       ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>       From: [field1]
> <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]
>       To: [service]
> <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
>       Call-ID: [call_id]
>       CSeq: 1 ACK
>       Max-Forwards: 70
>       Content-Length: 0
>
>     ]]>
>   </send>
>
>   <!-- definition of the response time repartition table (unit is ms)   -->
>   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
>
>   <!-- definition of the call length repartition table (unit is ms)     -->
>   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
>
> </scenario>
>
>
>
> On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>
>> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com>
>> wrote:
>> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com>
>> wrote:
>> >>
>> >> sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
>> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m
>> 1 -l
>> >> 4000
>> >> scenario file:      uac_redirect.xml
>> >> FS dialplan:       public.xml
>> >> SIP trace:          trace.log
>> >
>> > The Via definition in your SIPp scenario differs between the INVITE and
>> the ACK:
>> >
>> > INVITE:
>> > Via: SIP/2.0/[transport] [local_ip];branch=[branch]
>> >
>> > ACK:
>> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>> >
>> >
>> > In the INVITE, you are not adding the [local_port] as you do in the ACK.
>> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept the
>> ACK.
>> > So it seems FS is not checking just the ACK's branch against the
>> > INVITE's; it seems it is checking the whole Via header.
>> > I don't know if this is in accordance to SIP specs.
>> > Another thing, about the way you are calling SIPp: do no use "-sn uac"
>> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
>> > means "use the internal (embedded) scenario named xxx". So this
>> > conflicts with the other parameter "-sf" which specifies an external
>> > profile.
>>
>> I mean, an external scenario (file).
>>
>>  It seems this doesn't cause any problem (probably because in
>> > the sipp startup, -sf overrides -sn), but it is misleading.
>> >
>> > regards,
>> > takeshi
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Tue Aug 25 10:10:54 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Aug 2009 12:10:54 -0500
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
Message-ID: <191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>

You also should put your extension first in your dialplan ahead of the
default extensions which match every call and do a lot of db access for
record keeping etc.

The single thread in sofia is part of their concurrency model.
The single thread acts as a scheduler and indicates to us that an invite is
requested
we react to that and spawn a new thread for that call as soon as  the data
is received.

like I said above you are probably hitting all the default stuff to save
that last dialed number etc
that exists in the default config.

sigh, we go through this every new guy doing load testing.


On Tue, Aug 25, 2009 at 11:34 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> Exactly... the scenario i use seems operating on a single thread... why is
> that ? can it be changed?
>
> T.
>
>
> On Tue, Aug 25, 2009 at 5:31 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Actually in this case, we are bound to one thread in sofia.
>>
>> Mike
>>
>> On Aug 25, 2009, at 9:47 AM, Giovanni Maruzzelli wrote:
>>
>> > is a heavely multithreaded software, it benefits from number of CPUs
>> > (or cores), RAM, and heavy duty kernel features (found in 64bit
>> > kernels)
>> >
>> > put all accesses on ramdisk, leave out the modules you don't use...
>> >
>> > experiment, test, and find the best for your specific application/
>> > workload
>> >
>> > test not only with sipp, but with real load too (often they're very
>> > different)
>> >
>> > -gm
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Tue Aug 25 10:13:05 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 13:13:05 -0400
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
Message-ID: <278E22F4-46BE-4FB0-8FD1-52874B900C91@jerris.com>

This requires invasive changes in the sofia-sip stack to get thread- 
pooling working again.  I am sure they would accept patches if you can  
provide some that fully address any issues that may come up from  
adding this such as race conditions.

Mike

On Aug 25, 2009, at 12:34 PM, Tihomir Culjaga wrote:

> Exactly... the scenario i use seems operating on a single thread...  
> why is that ? can it be changed?
>
> T.
>
> On Tue, Aug 25, 2009 at 5:31 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> Actually in this case, we are bound to one thread in sofia.
>
> Mike
>
> On Aug 25, 2009, at 9:47 AM, Giovanni Maruzzelli wrote:
>
> > is a heavely multithreaded software, it benefits from number of CPUs
> > (or cores), RAM, and heavy duty kernel features (found in 64bit
> > kernels)
> >
> > put all accesses on ramdisk, leave out the modules you don't use...
> >
> > experiment, test, and find the best for your specific application/
> > workload
> >
> > test not only with sipp, but with real load too (often they're very
> > different)
> >
> > -gm
> >
>

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From sdjernes at gmail.com  Tue Aug 25 10:15:49 2009
From: sdjernes at gmail.com (Shawn L. Djernes)
Date: Tue, 25 Aug 2009 12:15:49 -0500
Subject: [Freeswitch-users] Group Call
In-Reply-To: <ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
Message-ID: <5ff3e3a70908251015ue88b8b9p7a5e091d15a6b5ae@mail.gmail.com>

Woops,
That line was an attempt to try something from an older diaplan example.
 Copied and pasted from a file on my computer not the server.  The actual
line is:

<action application="bridge"
data="{ignore_early_media=true}${group_call(sdjernes+A@${domain_name})}"/>

The +A is to ring all phones in the group at the same time.



2009/8/25 Jo?o Mesquita <jmesquita at freeswitch.org>

> According to example dialplan you hould do this:
>
> <action application="bridge"
> data="{ignore_early_media=true}${group(call:$1@${domain_name})}"/>
>
> According to wiki:
>
> group,[insert|delete|call]:<group name>:<url>,group [insert|delete|call
>
>
> Only difference I see here is the "+A" on your bridge statement.
>
>
> jmesquita
>
>
> On Tue, Aug 25, 2009 at 11:30 AM, Shawn L. Djernes <sdjernes at gmail.com>wrote:
>
>> Hello,
>> I am trying to get group calling to work.  AKA: dial an extension (7300 in
>> this case) and it ring all phones at once.  We are moving a server from
>> Asterisk to FS
>>
>> Here is the conf/directory/default.xml section:
>>
>>         <group name="sdjernes">
>>                 <users>
>>                         <user id="7301" type="pointer"/>
>>                         <user id="7302" type="pointer"/>
>>                         <user id="7303" type="pointer"/>
>>                         <user id="7304" type="pointer"/>
>>                 </users>
>>         </group>
>>
>> Here is the conf/dialplan/default/7300_group.xml extension statement:
>>
>>      <extension name="Group 7300">
>>        <condition field="destination_number" expression="(7300)">
>>          <action application="set" data="hangup_after_bridge=true"/>
>>          <action application="set" data="continue_on_fail=true"/>
>>          <action application="set" data="call_timeout=15"/>
>>          <action application="bridge"
>> data="{ignore_early_media=true}${group(call:sdjernes+A@
>> ${domain_name})}"/>
>> <!--         <action application="transfer" data="7301 XML default"/> -->
>>          <action application="hangup"/>
>>        </condition>
>>      </extension>
>>
>> The log output shows the following:
>>
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3289 Channel
>> sofia/internal/7305 at ewr.
>> djernes.net entering state [received][100]
>> 2009-08-25 14:27:45.282619 [DEBUG] switch_core_state_machine.c:398
>> (sofia/intern
>> al/7305 at ewr.djernes.net) Running State Change CS_NEW
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3296 Remote SDP:
>> v=0
>> o=root 628695665 628695665 IN IP4 98.188.201.109
>> s=call
>> c=IN IP4 98.188.201.109
>> t=0 0
>> m=audio 61012 RTP/AVP 0 8 9 99 3 18 4 101
>> a=rtpmap:0 pcmu/8000
>> a=rtpmap:8 pcma/8000
>> a=rtpmap:9 g722/8000
>> a=rtpmap:99 g726-32/8000
>> a=rtpmap:3 gsm/8000
>> a=rtpmap:18 g729/8000
>> a=rtpmap:4 g723/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[CELT:114:48000:10]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[CELT:114:32000:10]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[SPEEX:98:8000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[SPEEX:98:8000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[SPEEX:98:8000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[G7221:115:32000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[G7221:107:16000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[G722:9:8000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[PCMU:0:8000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:2029 Set Codec
>> sofia/internal/73
>> 05 at ewr.djernes.net PCMU/8000 20 ms 160 samples
>> 2009-08-25 14:27:45.282619 [DEBUG] switch_core_state_machine.c:404
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State NEW
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload
>> to 10
>> 1
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3455
>> (sofia/internal/7305 at ewr.djernes
>> .net) State Change CS_NEW -> CS_INIT
>> 2009-08-25 14:27:45.282619 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/i
>> nternal/7305 at ewr.djernes.net [BREAK]
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
>> (sofia/intern
>> al/7305 at ewr.djernes.net) Running State Change CS_INIT
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:481
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State INIT
>> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:83
>> sofia/internal/7305 at ewr.djerne
>> s.net SOFIA INIT
>> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:111
>> (sofia/internal/7305 at ewr.djer
>> nes.net) State Change CS_INIT -> CS_ROUTING
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/i
>> nternal/7305 at ewr.djernes.net [BREAK]
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:481
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State INIT going to sleep
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
>> (sofia/intern
>> al/7305 at ewr.djernes.net) Running State Change CS_ROUTING
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:484
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State ROUTING
>> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:130
>> sofia/internal/7305 at ewr.djern
>> es.net SOFIA ROUTING
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:78
>> sofia/internal
>> /7305 at ewr.djernes.net Standard ROUTING
>> 2009-08-25 14:27:45.287037 [INFO] mod_dialplan_xml.c:315 Processing
>> Sdjernes TOI
>>  Phone->7300 in context default
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unloop]
>> continue
>> =false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unloop]
>> ${unroll_loo
>> ps}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unloop]
>> ${sip_looped
>> _call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->tod_example] con
>> tinue=true
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [tod_example]
>> ${strft
>> ime(%w)}(2) =~ /^([1-5])$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [tod_example]
>> ${strft
>> ime(%H%M)}(1427) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action set(open=true)
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->global-intercept
>> ] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [global-intercept] de
>> stination_number(7300) =~ /^886$/ break=on-false
>>  Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->group-intercept]
>>  continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [group-intercept] des
>> tination_number(7300) =~ /^\*8$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->intercept-ext] c
>> ontinue=false
>>  Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [intercept-ext] desti
>> nation_number(7300) =~ /^\*\*(\d+)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->redial]
>> continue
>> =false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [redial]
>> destination_
>> number(7300) =~ /^870$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->global]
>> continue
>> =true
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [global]
>> ${call_debug
>> }(false) =~ /^true$/ break=never
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [global]
>> ${sip_has_cr
>> ypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
>> break=never
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Absolute Condition [global]
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
>> hash(insert/${domain_name}-
>> spymap/${caller_id_number}/${uuid})
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
>> hash(insert/${domain_name}-
>> last_dial/${caller_id_number}/${destination_number})
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
>> hash(insert/${domain_name}-
>> last_dial/global/${uuid})
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->snom-demo-2] con
>> tinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [snom-demo-2]
>> destina
>> tion_number(7300) =~ /^9001$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->snom-demo-1] con
>> tinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [snom-demo-1]
>> destina
>> tion_number(7300) =~ /^9000$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->eavesdrop] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [eavesdrop]
>> destinati
>> on_number(7300) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
>>  Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->eavesdrop] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [eavesdrop]
>> destinati
>> on_number(7300) =~ /^779$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->call_return] con
>> tinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call_return]
>> destina
>> tion_number(7300) =~ /^\*69$|^869$|^lcr$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->del-group] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [del-group]
>> destinati
>> on_number(7300) =~ /^80(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->add-group] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [add-group]
>> destinati
>> on_number(7300) =~ /^81(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->call-group-simo]
>>  continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [call-group-simo] des
>> tination_number(7300) =~ /^82(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->call-group-order
>> ] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [call-group-order] de
>> stination_number(7300) =~ /^83(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->extension-interc
>> om] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [extension-intercom]
>> destination_number(7300) =~ /^8(10[01][0-9])$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->Local_Extension]
>>  continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [Local_Extension] des
>> tination_number(7300) =~ /^(10[01][0-9])$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->group_dial_sales
>> ] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [group_dial_sales] de
>> stination_number(7300) =~ /^2000$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->group_dial_suppo
>> rt] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [group_dial_support]
>> destination_number(7300) =~ /^2001$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->group_dial_billi
>> ng] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [group_dial_billing]
>> destination_number(7300) =~ /^2002$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->operator]
>> contin
>> ue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [operator]
>> destinatio
>> n_number(7300) =~ /^(operator|0)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->vmain]
>> continue=
>> false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [vmain]
>> destination_n
>> umber(7300) =~ /^vmain$|^4000$|^\*98$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->sip_uri]
>> continu
>> e=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [sip_uri]
>> destination
>> _number(7300) =~ /^sip:(.*)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->nb_conferences]
>> continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [nb_conferences] dest
>> ination_number(7300) =~ /^(30\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->wb_conferences]
>> continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [wb_conferences] dest
>> ination_number(7300) =~ /^(31\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->uwb_conferences]
>>  continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [uwb_conferences] des
>> tination_number(7300) =~ /^(32\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->cdquality_confer
>> ences] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [cdquality_conference
>> s] destination_number(7300) =~ /^(33\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->freeswitch_publi
>> c_conf_via_sip] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [freeswitch_public_co
>> nf_via_sip] destination_number(7300) =~ /^9(888|1616|3232)$/
>> break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->mad_boss_interco
>> m] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [mad_boss_intercom] d
>> estination_number(7300) =~ /^0911$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->mad_boss_interco
>> m] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [mad_boss_intercom] d
>> estination_number(7300) =~ /^0912$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->mad_boss]
>> contin
>> ue=false
>>  Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [mad_boss]
>> destinatio
>> n_number(7300) =~ /^0913$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->ivr_demo]
>> contin
>> ue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ivr_demo]
>> destinatio
>> n_number(7300) =~ /^5000$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->dynamic_conferen
>> ce] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [dynamic_conference]
>> destination_number(7300) =~ /^5001$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->rtp_multicast_pa
>> ge] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [rtp_multicast_page]
>> destination_number(7300) =~ /^pagegroup$|^7243$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
>> continue=f
>> alse
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
>> destination_nu
>> mber(7300) =~ /^5900$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
>> continue
>> =false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
>> destination_
>> number(7300) =~ /^5901$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
>> continue=f
>> alse
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [park]
>> source(mod_sof
>> ia) =~ /mod_sofia/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
>> destination_nu
>> mber(7300) =~ /park\+(\d+)/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
>> continue
>> =false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unpark]
>> source(mod_s
>> ofia) =~ /mod_sofia/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
>> destination_
>> number(7300) =~ /^parking$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
>> continue=f
>> alse
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [park]
>> source(mod_sof
>> ia) =~ /mod_sofia/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
>> destination_nu
>> mber(7300) =~ /callpark/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
>> continue
>> =false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unpark]
>> source(mod_s
>> ofia) =~ /mod_sofia/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
>> destination_
>> number(7300) =~ /pickup/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->wait]
>> continue=f
>> alse
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [wait]
>> destination_nu
>> mber(7300) =~ /^wait$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->fax_receive] con
>> tinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [fax_receive]
>> destina
>> tion_number(7300) =~ /^9978$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->fax_transmit] co
>> ntinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [fax_transmit]
>> destin
>> ation_number(7300) =~ /^9979$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->ringback_180] co
>> ntinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ringback_180]
>> destin
>> ation_number(7300) =~ /^9980$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->ringback_183_uk_
>> ring] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [ringback_183_uk_ring
>> ] destination_number(7300) =~ /^9981$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->ringback_183_mus
>> ic_ring] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [ringback_183_music_r
>> ing] destination_number(7300) =~ /^9982$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->ringback_post_an
>> swer_uk_ring] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [ringback_post_answer
>> _uk_ring] destination_number(7300) =~ /^9983$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->ringback_post_an
>> swer_music] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [ringback_post_answer
>> _music] destination_number(7300) =~ /^9984$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->ClueCon]
>> continu
>> e=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ClueCon]
>> destination
>> _number(7300) =~ /^9991$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->show_info] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [show_info]
>> destinati
>> on_number(7300) =~ /^9992$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->video_record] co
>> ntinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [video_record]
>> destin
>> ation_number(7300) =~ /^9993$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->video_playback]
>> continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [video_playback] dest
>> ination_number(7300) =~ /^9994$/ break=on-false
>>  Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->delay_echo] cont
>> inue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [delay_echo]
>> destinat
>> ion_number(7300) =~ /^9995$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->echo]
>> continue=f
>> alse
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [echo]
>> destination_nu
>> mber(7300) =~ /^9996$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->milliwatt] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [milliwatt]
>> destinati
>> on_number(7300) =~ /^9997$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->tone_stream] con
>> tinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [tone_stream]
>> destina
>> tion_number(7300) =~ /^9998$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->zrtp_enrollement
>> ] continue=false
>>  Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [zrtp_enrollement] de
>> stination_number(7300) =~ /^9787$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->hold_music] cont
>> inue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [hold_music]
>> destinat
>> ion_number(7300) =~ /^9999$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->Rednote]
>> continu
>> e=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [Rednote]
>> destination
>> _number(7300) =~ /^0220(\d+)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->carmickle] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [carmickle]
>> destinati
>> on_number(7300) =~ /^0352(\d+)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->call_jason] cont
>> inue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call_jason]
>> destinat
>> ion_number(7300) =~ /^059(\d+)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->TOI]
>> continue=fa
>> lse
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [TOI]
>> destination_num
>> ber(7300) =~ /^08740(\d+)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->Group
>> 7300] cont
>> inue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [Group 7300]
>> destinat
>> ion_number(7300) =~ /(7300)/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
>> set(hangup_after_bridge=tru
>> e)
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
>> set(continue_on_fail=true)
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action set(call_timeout=15)
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
>> bridge({ignore_early_media=
>> true}${group_call(sdjernes+A@${domain_name})})
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action hangup()
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:114
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State Change CS_ROUTING -> CS_EXECUTE
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/i
>> nternal/7305 at ewr.djernes.net [BREAK]
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:484
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State ROUTING going to sleep
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
>> (sofia/intern
>> al/7305 at ewr.djernes.net) Running State Change CS_EXECUTE
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:491
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State EXECUTE
>> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:173
>> sofia/internal/7305 at ewr.djern
>> es.net SOFIA EXECUTE
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:151
>> sofia/interna
>> l/7305 at ewr.djernes.net Standard EXECUTE
>> EXECUTE sofia/internal/7305 at ewr.djernes.net set(open=true)
>> 2009-08-25 14:27:45.287037 [DEBUG] mod_dptools.c:748
>> sofia/internal/7305 at ewr.dje
>> rnes.net SET [open]=[true]
>> EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-s
>> pymap/7305/733dbed2-9183-11de-95f0-07acd7038fc1)
>> EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-l
>> ast_dial/7305/7300)
>> EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-l
>> ast_dial/global/733dbed2-9183-11de-95f0-07acd7038fc1)
>> EXECUTE sofia/internal/7305 at ewr.djernes.net set(hangup_after_bridge=true)
>> 2009-08-25 14:27:45.291899 [DEBUG] mod_dptools.c:748
>> sofia/internal/7305 at ewr.dje
>> rnes.net SET [hangup_after_bridge]=[true]
>> EXECUTE sofia/internal/7305 at ewr.djernes.net set(continue_on_fail=true)
>> 2009-08-25 14:27:45.291899 [DEBUG] mod_dptools.c:748
>> sofia/internal/7305 at ewr.dje
>> rnes.net SET [continue_on_fail]=[true]
>> EXECUTE sofia/internal/7305 at ewr.djernes.net set(call_timeout=15)
>> 2009-08-25 14:27:45.294755 [DEBUG] mod_dptools.c:748
>> sofia/internal/7305 at ewr.dje
>> rnes.net SET [call_timeout]=[15]
>> EXECUTE sofia/internal/7305 at ewr.djernes.netbridge({ignore_early_media=true})
>> 2009-08-25 14:27:45.318579 [WARNING] switch_ivr_originate.c:1001 No
>> origination
>> URL specified!
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_ivr_originate.c:2138 Originate
>> Resulte
>> d in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>> 2009-08-25 14:27:45.318579 [INFO] mod_dptools.c:2093 Originate Failed.
>>  Cause: D
>> ESTINATION_OUT_OF_ORDER
>> 2009-08-25 14:27:45.318579 [DEBUG] mod_dptools.c:2120 Continue on fail
>> [true]:
>> Cause: DESTINATION_OUT_OF_ORDER
>> EXECUTE sofia/internal/7305 at ewr.djernes.net hangup()
>> 2009-08-25 14:27:45.318579 [NOTICE] mod_dptools.c:633 Hangup
>> sofia/internal/7305
>> @ewr.djernes.net [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_channel.c:1683 Send signal
>> sofia/inter
>> nal/7305 at ewr.djernes.net [KILL]
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/i
>> nternal/7305 at ewr.djernes.net [BREAK]
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:491
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State EXECUTE going to sleep
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:398
>> (sofia/intern
>> al/7305 at ewr.djernes.net) Running State Change CS_HANGUP
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:434
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State HANGUP
>> 2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:338 Channel
>> sofia/internal/7305 at e
>> wr.djernes.net hanging up, cause: NORMAL_CLEARING
>> 2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:417 Responding to INVITE
>> with: 48
>> 0
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:46
>> sofia/internal
>> /7305 at ewr.djernes.net Standard HANGUP, cause: NORMAL_CLEARING
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:434
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State HANGUP going to sleep
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:476
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State Change CS_HANGUP -> CS_REPORTING
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/i
>> nternal/7305 at ewr.djernes.net [BREAK]
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:398
>> (sofia/intern
>> al/7305 at ewr.djernes.net) Running State Change CS_REPORTING
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:612
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State REPORTING
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal
>> /7305 at ewr.djernes.net Standard REPORTING, cause: NORMAL_CLEARING
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:612
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State REPORTING going to sleep
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:411
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State Change CS_REPORTING -> CS_DESTROY
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:1068 Session 117
>> (sofia
>> /internal/7305 at ewr.djernes.net) Locked, Waiting on external entities
>> 2009-08-25 14:27:45.318579 [NOTICE] switch_core_session.c:1086 Session 117
>> (sofi
>> a/internal/7305 at ewr.djernes.net) Ended
>> 2009-08-25 14:27:45.318579 [NOTICE] switch_core_session.c:1088 Close
>> Channel sof
>> ia/internal/7305 at ewr.djernes.net [CS_DESTROY]
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:564
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State DESTROY
>> 2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:255
>> sofia/internal/7305 at ewr.djern
>> es.net SOFIA DESTROY
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:60
>> sofia/internal
>> /7305 at ewr.djernes.net Standard DESTROY
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:564
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State DESTROY going to sleep
>>
>>
>> Any Help would be much appreciated.
>> --
>> Shawn L. Djernes
>> SD Consulting
>> shawn at djernes.org | sdjernes at gmail.com
>> MSN: wizardwlf at hotmail.com
>> 402.345.7734 | 402.350.6973 Cell
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Shawn L. Djernes
SD Consulting
shawn at djernes.org | sdjernes at gmail.com
MSN: wizardwlf at hotmail.com
402.345.7734 | 402.350.6973 Cell
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From tculjaga at gmail.com  Tue Aug 25 10:31:13 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 19:31:13 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
	<191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
Message-ID: <65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>

Of course i removed everytihng from teh dialplan except my extension :)

when exactly do you react and bring up a new thread ? ... is it on INVITE or
on 1st 1xx response ?

i beleive i can have several lets call it SIP interfaces ... on different
ports 5060, 5070, 5080 ... every interface will have it's own sip profile.

does it mean i will have one thread per profile?

T.


On Tue, Aug 25, 2009 at 7:10 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You also should put your extension first in your dialplan ahead of the
> default extensions which match every call and do a lot of db access for
> record keeping etc.
>
> The single thread in sofia is part of their concurrency model.
> The single thread acts as a scheduler and indicates to us that an invite is
> requested
> we react to that and spawn a new thread for that call as soon as  the data
> is received.
>
> like I said above you are probably hitting all the default stuff to save
> that last dialed number etc
> that exists in the default config.
>
> sigh, we go through this every new guy doing load testing.
>
>
>
> On Tue, Aug 25, 2009 at 11:34 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> Exactly... the scenario i use seems operating on a single thread... why is
>> that ? can it be changed?
>>
>> T.
>>
>>
>> On Tue, Aug 25, 2009 at 5:31 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Actually in this case, we are bound to one thread in sofia.
>>>
>>> Mike
>>>
>>> On Aug 25, 2009, at 9:47 AM, Giovanni Maruzzelli wrote:
>>>
>>> > is a heavely multithreaded software, it benefits from number of CPUs
>>> > (or cores), RAM, and heavy duty kernel features (found in 64bit
>>> > kernels)
>>> >
>>> > put all accesses on ramdisk, leave out the modules you don't use...
>>> >
>>> > experiment, test, and find the best for your specific application/
>>> > workload
>>> >
>>> > test not only with sipp, but with real load too (often they're very
>>> > different)
>>> >
>>> > -gm
>>> >
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Aug 25 10:39:22 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 13:39:22 -0400
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
Message-ID: <34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>

if you want to run that at your prompt instead of at the fs_cli you  
can do this:

function sofia() { fs_cli -x "$(echo "sofia $@")"; }

(thanks ray for the bash foo)

Mike

On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:

> I get a bash: sofia: command not found. Is there something I need to  
> add to my config to use these commands?
>
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
> Michael Collins
> Sent: Tuesday, August 25, 2009 10:55 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> Hmm... can you fire up FS and then do "sofia status" and tell us if  
> the internal profile looks okay? Also, do "sofia status profile  
> internal" and capture the output. Something fishy is going on...
> -MC
> Mike Peace
>
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>

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From msc at freeswitch.org  Tue Aug 25 10:40:36 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 10:40:36 -0700
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
	<191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
	<65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>
Message-ID: <87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>

On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> Of course i removed everytihng from teh dialplan except my extension :)
>
> when exactly do you react and bring up a new thread ? ... is it on INVITE
> or on 1st 1xx response ?
>
> i beleive i can have several lets call it SIP interfaces ... on different
> ports 5060, 5070, 5080 ... every interface will have it's own sip profile.
>
> does it mean i will have one thread per profile?

Yes.
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From tculjaga at gmail.com  Tue Aug 25 10:44:07 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 19:44:07 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
Message-ID: <65d96fc80908251044u9dfa12bi717cccd4e91f1560@mail.gmail.com>

Hi Giovanny,

thanks for your help,
everything that heavyly accesses the disk is on ramdisk now...
hopefully will get some real traffic pretty soon...


On Tue, Aug 25, 2009 at 3:47 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> is a heavely multithreaded software, it benefits from number of CPUs
> (or cores), RAM, and heavy duty kernel features (found in 64bit
> kernels)
>
> put all accesses on ramdisk, leave out the modules you don't use...
>
> experiment, test, and find the best for your specific application/workload
>
> test not only with sipp, but with real load too (often they're very
> different)
>
> -gm
>
>
> On Tue, Aug 25, 2009 at 3:42 PM, Tihomir Culjaga<tculjaga at gmail.com>
> wrote:
> > thanks for the feedback... this is something im going to do tomorrow...
> >
> > what about other things?
> >
> >
> > On Tue, Aug 25, 2009 at 3:39 PM, Jay Binks <jaybinks at gmail.com> wrote:
> >>
> >> Everytime someone asks this ,  the resounding answer is use a 64bit os..
> >> No question
> >> Jay
> >>
> >>
> >>
> >> On 25/08/2009, at 23:19, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> >>
> >> Hey Giovanni,
> >>
> >> thanks for the tip... indeed the db files were heavily used regardless
> if
> >> i started freeswitch with nosql option (freeswitch -nosql)... FS was not
> >> writing anything into that files ... instead it was just accessing
> it....
> >> This behaviour leads to a waste of 40% CPU time... waiting for other
> >> processes (mainly disk access) to finish!!!
> >>
> >> I moved freeswitch/db/ to a ramdisk and the performance got a boost to
> 140
> >> CPS with a CPU load of 80%. I was keeping the machine for a while (20 -
> 30
> >> minutes) on that rate when i sow CPU suddenly went to 100% and FS
> becoming
> >> irresponsive :).
> >>
> >>
> >> What can be wrong?
> >> What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
> >> cross?
> >> What fine tuning do we need in order to asure a long high load run?
> >>
> >>
> >>
> >> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
> >> sense to move my OS to 64 bit? ... will FS gain more preformance ?... I
> mean
> >> will FS perofomr drastically better 20%+ ?
> >>
> >>
> >> Tihomir.
> >>
> >>
> >> On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
> >> <gmaruzz at celliax.org> wrote:
> >>>
> >>> Maybe your load comes from disk access?
> >>>
> >>> Try putting the sql and log directories on a ramdisk.
> >>>
> >>> OTH,
> >>>
> >>> -giovanni
> >>>
> >>> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com>
> >>> wrote:
> >>> > Hello,
> >>> >
> >>> > i'm trying to use freeswitch as a redirecting server meaning FS has
> to
> >>> > receive an INVITE and according to some rules it will redirect calls
> to
> >>> > other destinations.
> >>> >
> >>> >
> >>> > CALLING_USER                FREESWITCH
> SOMEWHERE
> >>> >
> >>> > INVITE ------------------------------->
> >>> >            <------------------------------ 100 Trying
> >>> >            <------------------------------ 302 Moved Temporary
> >>> > ACK    ------------------------------->
> >>> >
> >>> >
> INVITE--------------------------------------------------------------------------------->
> >>> >
> >>> >
> >>> >
> >>> > Well, wverything works well except i have perfromance issues .... on
> my
> >>> > HW
> >>> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
> Temporary).
> >>> > When
> >>> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS
> i
> >>> > see
> >>> > "calls" being build up in FS and the delay begining to grow.
> >>> >
> >>> > When i observe the machine, load average is almost nothing (load
> >>> > average:
> >>> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
> >>> > taking
> >>> > most load... all others are just sitting there with 1-5 % CPU time.
> >>> > This looks to me as FS handles 302 messages in a single thread?!?!
> >>> >
> >>> >
> >>> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
> >>> >
> >>> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41,
> 0.61,
> >>> > 0.60
> >>> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
> >>> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,
> 0.2%si,
> >>> > 0.0%st
> >>> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k
> >>> > buffers
> >>> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k
> cached
> >>> >
> >>> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
> >>> > COMMAND
> >>> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
> >>> > freeswitch
> >>> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
> >>> > freeswitch
> >>> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
> >>> > freeswitch
> >>> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
> >>> > freeswitch
> >>> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
> >>> > freeswitch
> >>> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
> >>> > freeswitch
> >>> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
> >>> > freeswitch
> >>> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
> >>> > freeswitch
> >>> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
> >>> > freeswitch
> >>> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50
> freeswitch
> >>> >
> >>> >
> >>> >
> >>> > cat /proc/cpuinfo
> >>> > processor       : 0
> >>> > vendor_id       : GenuineIntel
> >>> > cpu family      : 6
> >>> > model           : 15
> >>> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> >>> > stepping        : 6
> >>> > cpu MHz         : 2333.560
> >>> > cache size      : 4096 KB
> >>> > physical id     : 0
> >>> > siblings        : 2
> >>> > core id         : 0
> >>> > cpu cores       : 2
> >>> > apicid          : 0
> >>> > initial apicid  : 0
> >>> > fdiv_bug        : no
> >>> > hlt_bug         : no
> >>> > f00f_bug        : no
> >>> > coma_bug        : no
> >>> > fpu             : yes
> >>> > fpu_exception   : yes
> >>> > cpuid level     : 10
> >>> > wp              : yes
> >>> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
> pge
> >>> > mca
> >>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> >>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
> ssse3
> >>> > cx16
> >>> > xtpr dca lahf_lm
> >>> > bogomips        : 4670.78
> >>> > clflush size    : 64
> >>> > power management:
> >>> >
> >>> > processor       : 1
> >>> > vendor_id       : GenuineIntel
> >>> > cpu family      : 6
> >>> > model           : 15
> >>> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> >>> > stepping        : 6
> >>> > cpu MHz         : 2333.560
> >>> > cache size      : 4096 KB
> >>> > physical id     : 0
> >>> > siblings        : 2
> >>> > core id         : 1
> >>> > cpu cores       : 2
> >>> > apicid          : 1
> >>> > initial apicid  : 1
> >>> > fdiv_bug        : no
> >>> > hlt_bug         : no
> >>> > f00f_bug        : no
> >>> > coma_bug        : no
> >>> > fpu             : yes
> >>> > fpu_exception   : yes
> >>> > cpuid level     : 10
> >>> > wp              : yes
> >>> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
> pge
> >>> > mca
> >>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> >>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
> ssse3
> >>> > cx16
> >>> > xtpr dca lahf_lm
> >>> > bogomips        : 4666.82
> >>> > clflush size    : 64
> >>> > power management:
> >>> >
> >>> >
> >>> >
> >>> > uname -a
> >>> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009
> >>> > i686
> >>> > GNU/Linux
> >>> >
> >>> >
> >>> >
> >>> > Of course, i've tuned the machine up
> >>> >
> >>> > ulimit -c unlimited
> >>> > ulimit -d unlimited
> >>> > ulimit -f unlimited
> >>> > ulimit -i unlimited
> >>> > ulimit -n 999999
> >>> > ulimit -q unlimited
> >>> > ulimit -u unlimited
> >>> > ulimit -v unlimited
> >>> > ulimit -x unlimited
> >>> > ulimit -s 240
> >>> > ulimit -l unlimited
> >>> > ulimit -a
> >>> >
> >>> >
> >>> > Started FS with minimum modules but still 40 CPS seems to be the
> limit.
> >>> >
> >>> >
> >>> > So, is there any way to improve performance?
> >>> >
> >>> >
> >>> > Tihomir.
> >>> >
> >>> >
> >>> >
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Aug 25 10:44:50 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 10:44:50 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
Message-ID: <87f2f3b90908251044p4f2ccf6cif79f5b86956708bc@mail.gmail.com>

On Tue, Aug 25, 2009 at 10:04 AM, Mike Peace <MPeace at edcogroupinc.com>wrote:

>  I get a bash: sofia: command not found. Is there something I need to add
> to my config to use these commands?
>
Hehe, sorry. The "sofia" command is for use at the FreeSWITCH command line.
If you have FS running as a daemon (that is, you launched it with the -nc
option) then you'll need to use the fs_cli program to connect to your
running FS daemon. On a typical install it would be something like this:

/usr/local/freeswitch/bin/fs_cli

Once you're there, then issue the two commands:
sofia status
sofia status profile internal

Have fun!
-MC

BTW, you should really get a copy of the September issue of Linux-Pro
magazine because it has a nice article that is a gentle introduction to
using FreeSWITCH. If you're in Western Europe it's already on the shelves.
In the US it should be out any day. In Australia/NZ at the end of the month.
Hope you don't mind the shameless plug.

>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, August 25, 2009 10:55 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Tue, Aug 25, 2009 at 7:20 AM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> I fired up Wireshark on each side and I can see the SIP register request
> coming from the laptop, the Freeswitch server replies with a Destination
> Unreachable (Port unreachable) message.
>
>
>
> I rebooted the Server and now I get a ?Registration error; 405 Method not
> allowed? on the softphone and the Wiresharp capture shows: ?Status 405
> Method Not Allowerd (0 bindings).
>
>
>
> Hmm... can you fire up FS and then do "sofia status" and tell us if the
> internal profile looks okay? Also, do "sofia status profile internal" and
> capture the output. Something fishy is going on...
> -MC
>
>  It does this even after I stop and restart the Freeswitch service.
>
>
>
> Shouldn?t all the ports be open if the firewall and SELunix are disabled?
>
>
>
> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 5:32 PM
>
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:49 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> It does the same thing, does anything get set during the install that would
> remember or cache the old network settings? I can access anything from the
> FS server on any of several networks and vice-versa but the SIP will not
> register, again no firewalls are upon any of the test hosts.
>
> Doesn?t make sense to me.
>
>
>
> Time to bust out tcpdump and/or wireshark to make 100% certain you know
> what's happening with all those SIP packets.
>
>   *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:34 PM
>
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> ?Registration error: 408-Request Timeout?
>
> Sorry for the the typo - 408 = timeout. (480 = temp unavail)
>
> Try stopping iptables in Linux and try again. Sounds like something is
> interfering with your packets getting from here to there...
> Try:
>
> /etc/init.d/iptables stop
>
> And then see if your packets can move again.
>
> -MC
>
>
>
> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:18 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> I haven?t changed any of the conf files.
>
>
>
> What happens when you try to register? Do you get 480? (timeout) Or
> something else?
> -MC
>
>
>  ------------------------------
>
>
>
> *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>
>
> Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>  ------------------------------
>
>
>
> *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>
>
> Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>  ------------------------------
>
>
>
> *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>
>
> Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
>
>  *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>  Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From MPeace at edcogroupinc.com  Tue Aug 25 10:53:00 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 12:53:00 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>

Ok, figured out from your post to load the FS-CLI in order to run the Sophia status Michael mentioned.

Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error] when running ./fs_cli from a terminal window in /usr/local/freeswitch/bin where fs_cli is located.

Mike
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, August 25, 2009 12:39 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

if you want to run that at your prompt instead of at the fs_cli you can do this:

function sofia() { fs_cli -x "$(echo "sofia $@")"; }

(thanks ray for the bash foo)

Mike

On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:


I get a bash: sofia: command not found. Is there something I need to add to my config to use these commands?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 25, 2009 10:55 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

Hmm... can you fire up FS and then do "sofia status" and tell us if the internal profile looks okay? Also, do "sofia status profile internal" and capture the output. Something fishy is going on...
-MC

Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From tculjaga at gmail.com  Tue Aug 25 10:53:54 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 19:53:54 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
	<191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
	<65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>
	<87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>
Message-ID: <65d96fc80908251053s53f45f62vc58656f24b5ddc6a@mail.gmail.com>

nice, this is one way to go...

btw: when do you exactly bring up a new thread ?

T.

On Tue, Aug 25, 2009 at 7:40 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> Of course i removed everytihng from teh dialplan except my extension :)
>>
>> when exactly do you react and bring up a new thread ? ... is it on INVITE
>> or on 1st 1xx response ?
>>
>> i beleive i can have several lets call it SIP interfaces ... on different
>> ports 5060, 5070, 5080 ... every interface will have it's own sip profile.
>>
>> does it mean i will have one thread per profile?
>
> Yes.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Tue Aug 25 10:54:36 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 25 Aug 2009 13:54:36 -0400
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
Message-ID: <D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>

It means freeswitch is not running. You need to start it with /usr/ 
local/freeswitch/bin/freeswitch -nc

@MikeJ: Nice bash scripting :D

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 25-Aug-09, at 1:53 PM, Mike Peace wrote:

> Ok, figured out from your post to load the FS-CLI in order to run  
> the Sophia status Michael mentioned.
>
> Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting  
> [Socket Connection Error] when running ./fs_cli from a terminal  
> window in /usr/local/freeswitch/bin where fs_cli is located.
>
> Mike
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Jerris
> Sent: Tuesday, August 25, 2009 12:39 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> if you want to run that at your prompt instead of at the fs_cli you  
> can do this:
>
> function sofia() { fs_cli -x "$(echo "sofia $@")"; }
>
> (thanks ray for the bash foo)
>
> Mike
>
> On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:
>
>
> I get a bash: sofia: command not found. Is there something I need to  
> add to my config to use these commands?
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Tuesday, August 25, 2009 10:55 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> Hmm... can you fire up FS and then do "sofia status" and tell us if  
> the internal profile looks okay? Also, do "sofia status profile  
> internal" and capture the output. Something fishy is going on...
> -MC
> Mike Peace
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
>
> EDCO Group, Inc.
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
> Confidentiality Notice: The information contained in this e-mail  
> message (including any attachments) may contain confidential and  
> privileged information, and is for the sole use of the intended  
> recipient(s). If you are not the intended recipient, any  
> unauthorized review, use, or disclosure or distribution is strictly  
> prohibited. If you have received this message in error, please  
> notify the sender by replying to this e-mail message or by telephone  
> at (800) 999-3456 and permanently destroy all copies of the original  
> message (including any attachments), along with any reply, and  
> delete them from your system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Aug 25 10:55:39 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 13:55:39 -0400
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
	<191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
	<65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>
	<87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>
Message-ID: <40411BDB-A19E-42CD-8F2B-70D45A490AFC@jerris.com>


On Aug 25, 2009, at 1:40 PM, Michael Collins wrote:

>
>
> On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga  
> <tculjaga at gmail.com> wrote:
> Of course i removed everytihng from teh dialplan except my  
> extension :)
>
> when exactly do you react and bring up a new thread ? ... is it on  
> INVITE or on 1st 1xx response ?

Before it hits the dialplan.

>
> i beleive i can have several lets call it SIP interfaces ... on  
> different ports 5060, 5070, 5080 ... every interface will have it's  
> own sip profile.
>
> does it mean i will have one thread per profile?
> Yes.

And no. There are still parts of the stack that are in one thread.

Mike
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From mrene_lists at avgs.ca  Tue Aug 25 10:57:50 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 25 Aug 2009 13:57:50 -0400
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908251053s53f45f62vc58656f24b5ddc6a@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
	<191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
	<65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>
	<87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>
	<65d96fc80908251053s53f45f62vc58656f24b5ddc6a@mail.gmail.com>
Message-ID: <B53FBFB4-F319-4DFE-9AEA-BBC949E014C1@avgs.ca>

mod_sofia will take care of spawning the session thread once it  
authenticated the call and loaded all the variables related to the  
call such as the caller profile (callerid, destination number, etc).

If you want to check the source, this is done in  
sofia_handle_sip_i_invite() (sofia.c:5261) with  
switch_core_session_thread_launch().

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 25-Aug-09, at 1:53 PM, Tihomir Culjaga wrote:

> nice, this is one way to go...
>
> btw: when do you exactly bring up a new thread ?
>
> T.
>
> On Tue, Aug 25, 2009 at 7:40 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga  
> <tculjaga at gmail.com> wrote:
> Of course i removed everytihng from teh dialplan except my  
> extension :)
>
> when exactly do you react and bring up a new thread ? ... is it on  
> INVITE or on 1st 1xx response ?
>
> i beleive i can have several lets call it SIP interfaces ... on  
> different ports 5060, 5070, 5080 ... every interface will have it's  
> own sip profile.
>
> does it mean i will have one thread per profile?
> Yes.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tculjaga at gmail.com  Tue Aug 25 10:59:14 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 19:59:14 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <40411BDB-A19E-42CD-8F2B-70D45A490AFC@jerris.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
	<191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
	<65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>
	<87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>
	<40411BDB-A19E-42CD-8F2B-70D45A490AFC@jerris.com>
Message-ID: <65d96fc80908251059x5ec061dfm60ae8fe9b45aec43@mail.gmail.com>

clear... thanks!

On Tue, Aug 25, 2009 at 7:55 PM, Michael Jerris <mike at jerris.com> wrote:

>
> On Aug 25, 2009, at 1:40 PM, Michael Collins wrote:
>
>
>
> On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> Of course i removed everytihng from teh dialplan except my extension :)
>>
>> when exactly do you react and bring up a new thread ? ... is it on INVITE
>> or on 1st 1xx response ?
>>
>
> Before it hits the dialplan.
>
>
>> i beleive i can have several lets call it SIP interfaces ... on different
>> ports 5060, 5070, 5080 ... every interface will have it's own sip profile.
>>
>> does it mean i will have one thread per profile?
>
> Yes.
>
>
> And no. There are still parts of the stack that are in one thread.
>
> Mike
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From carlos.talbot at gmail.com  Tue Aug 25 11:06:56 2009
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Tue, 25 Aug 2009 13:06:56 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
Message-ID: <5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>

Giovanni,
you mean like this message?

"Unable to determine location for device. Voicemail password set via FreePBX
will not be valid."

This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36

Let's keep in ming FreePBX v3 is a developer release and as such many
features are in flux and might not work. That being said there some features
in the Windows build that still do not work. The biggest one right now is
the lack of the php ESL library for Windows which affects the voicemail app.
I'm trying to get this to compile but it's been difficult.

I do include the .svn files with the FreePBX install so you can freely
install TortoiseSVN and update FreePBX at your leisure.

With regards to the sip_profiles, did you create a trunk group and trunk?

regards,

Carlos

On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Windows installer does not work for me.
>
> I've reinstalled various times, same results.
>
> I can correctly create a number, but when I try to create a device for
> that number, it tells me that cannot locate the device, and the
> password for vicemail will be invalid.
>
> After that, it begins to give the php error page, it cannot find the
> start < tag in directory/default.xml
>
> Also for me there are no sofia profiles...
>
> So, I cannot start to test it (eg: I would like to add mod_skypiax
> support to it).
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
>
> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
> Guidi<raffaele.p.guidi at gmail.com> wrote:
> > This is what I was asking! :D When the installer finished it started the
> > whole thing and everything got loaded fine, but when I restarted my
> system
> > it didn't (and did not anymore). Well, I will try to install everything
> from
> > scratch again and see...
> >
> > On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org> wrote:
> >>
> >> If you installed FreePBX then it would be that softwares job to manage
> >> the sofia profiles... wouldn't it?
> >>
> >> /b
> >>
> >> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
> >>
> >> > Actually I did that and it worked fine. I had the problem the SECOND
> >> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
> >> > sip_profiles were not
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jerry.richards at teotech.com  Tue Aug 25 11:12:39 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 25 Aug 2009 11:12:39 -0700
Subject: [Freeswitch-users] Scalability
Message-ID: <B43123F543104F3E894E15B06D705571@greyhawk.tonecommander.com>

Hello All,

Does anyone know what the capacity of a stand-alone Freeswitch, in terms of
how many users?

Also, when that number is exceeded, how can Freeswitch server be distributed
to accommodate a larger installation?

Best Regards,
Jerry 



From mdm at openaccessinc.com  Tue Aug 25 11:20:16 2009
From: mdm at openaccessinc.com (Michael Di Martino)
Date: Tue, 25 Aug 2009 13:20:16 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer -
	great	but I have a little problem
In-Reply-To: <5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
Message-ID: <5EA32FE3006E2740884F5A09AD98A5A718CDCB415A@34093-MBX-C03.mex07a.mlsrvr.com>

Is FreePBX V3 based on Freeswitch?

Michael DiMartino | Director of IT | Open Access, Inc.
115 Bi County Blvd | Farmingdale, NY 11735
631.227.1034| 631.694.6730 FAX |631.988.6060 MOBILE
www.openaccessinc.com

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Carlos Talbot
Sent: Tuesday, August 25, 2009 2:07 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

Giovanni,

you mean like this message?

"Unable to determine location for device. Voicemail password set via FreePBX will not be valid."

This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36

Let's keep in ming FreePBX v3 is a developer release and as such many features are in flux and might not work. That being said there some features in the Windows build that still do not work. The biggest one right now is the lack of the php ESL library for Windows which affects the voicemail app. I'm trying to get this to compile but it's been difficult.

I do include the .svn files with the FreePBX install so you can freely install TortoiseSVN and update FreePBX at your leisure.

With regards to the sip_profiles, did you create a trunk group and trunk?

regards,

Carlos
On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <gmaruzz at celliax.org<mailto:gmaruzz at celliax.org>> wrote:
Windows installer does not work for me.

I've reinstalled various times, same results.

I can correctly create a number, but when I try to create a device for
that number, it tells me that cannot locate the device, and the
password for vicemail will be invalid.

After that, it begins to give the php error page, it cannot find the
start < tag in directory/default.xml

Also for me there are no sofia profiles...

So, I cannot start to test it (eg: I would like to add mod_skypiax
support to it).


Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
Guidi<raffaele.p.guidi at gmail.com<mailto:raffaele.p.guidi at gmail.com>> wrote:
> This is what I was asking! :D When the installer finished it started the
> whole thing and everything got loaded fine, but when I restarted my system
> it didn't (and did not anymore). Well, I will try to install everything from
> scratch again and see...
>
> On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
>>
>> If you installed FreePBX then it would be that softwares job to manage
>> the sofia profiles... wouldn't it?
>>
>> /b
>>
>> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>>
>> > Actually I did that and it worked fine. I had the problem the SECOND
>> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
>> > sip_profiles were not
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From gmaruzz at celliax.org  Tue Aug 25 11:28:34 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 20:28:34 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com> 
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com> 
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com> 
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com> 
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com> 
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
Message-ID: <7b197bef0908251128y39de0bc8tcf123ee6343bfb33@mail.gmail.com>

Carlos,

you're very kind, as always.

I'm aware that this is a dev preview, and I'm interested just in that,
to begin getting acquainted with the framework (and adding support to
the endpoints/trunk I take care of).

I probably have not got the logic right :-) (I tried both Windows
Installer and Linux ISO)

I started fpbx with fs running, it works. I create a number, then I
create a device and I connect it to that number. It works.

If I restart FS, do not works anymore, complaining no sofia profiles.

>From the front page of FPBX is not clear you *must* create a trunk/trunk group.

I was thinking trunks were for outgoing calls, or for receiving from external.

I was just testing internal phones, trying an IVR, so I was thinking
trunks were not needed.

Can you explain to me?

Thanks again,

-giovanni

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 8:06 PM, Carlos Talbot<carlos.talbot at gmail.com> wrote:
> Giovanni,
> you mean like this message?
> "Unable to determine location for device. Voicemail password set via FreePBX
> will not be valid."
> This is a known FreePBX issue.?http://www.freepbx.org/v3/ticket/36
> Let's keep in ming FreePBX v3 is a developer release and as such many
> features are in flux and might not work.?That being said there some features
> in the Windows build that still do not work. The biggest one right now is
> the lack of the php ESL library for Windows which affects the voicemail app.
> I'm trying to get this to compile but it's been difficult.
> I do include the .svn files with the FreePBX install so you can freely
> install TortoiseSVN and update FreePBX at your leisure.
> With regards to the sip_profiles, did you create a trunk group and trunk?
> regards,
> Carlos
>
> On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> Windows installer does not work for me.
>>
>> I've reinstalled various times, same results.
>>
>> I can correctly create a number, but when I try to create a device for
>> that number, it tells me that cannot locate the device, and the
>> password for vicemail will be invalid.
>>
>> After that, it begins to give the php error page, it cannot find the
>> start < tag in directory/default.xml
>>
>> Also for me there are no sofia profiles...
>>
>> So, I cannot start to test it (eg: I would like to add mod_skypiax
>> support to it).
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>
>>
>>
>> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
>> Guidi<raffaele.p.guidi at gmail.com> wrote:
>> > This is what I was asking! :D When the installer finished it started the
>> > whole thing and everything got loaded fine, but when I restarted my
>> > system
>> > it didn't (and did not anymore). Well, I will try to install everything
>> > from
>> > scratch again and see...
>> >
>> > On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org> wrote:
>> >>
>> >> If you installed FreePBX then it would be that softwares job to manage
>> >> the sofia profiles... wouldn't it?
>> >>
>> >> /b
>> >>
>> >> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>> >>
>> >> > Actually I did that and it worked fine. I had the problem the SECOND
>> >> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
>> >> > sip_profiles were not
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From gmaruzz at celliax.org  Tue Aug 25 11:31:09 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 20:31:09 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <5EA32FE3006E2740884F5A09AD98A5A718CDCB415A@34093-MBX-C03.mex07a.mlsrvr.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com> 
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com> 
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com> 
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com> 
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com> 
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com> 
	<5EA32FE3006E2740884F5A09AD98A5A718CDCB415A@34093-MBX-C03.mex07a.mlsrvr.com>
Message-ID: <7b197bef0908251131r63d4f3a2t9d02724adcba7ea0@mail.gmail.com>

yes, and will support Asterisk and other engines in the future.

On Tue, Aug 25, 2009 at 8:20 PM, Michael Di
Martino<mdm at openaccessinc.com> wrote:
> Is FreePBX V3 based on Freeswitch?
>
>
>
> Michael DiMartino?| Director of IT | Open Access, Inc.
> 115 Bi County Blvd | Farmingdale, NY 11735
> 631.227.1034| 631.694.6730 FAX |631.988.6060?MOBILE
>
> www.openaccessinc.com
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Carlos
> Talbot
> Sent: Tuesday, August 25, 2009 2:07 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
> but I have a little problem
>
>
>
> Giovanni,
>
>
>
> you mean like this message?
>
>
>
> "Unable to determine location for device. Voicemail password set via FreePBX
> will not be valid."
>
>
>
> This is a known FreePBX issue.?http://www.freepbx.org/v3/ticket/36
>
>
>
> Let's keep in ming FreePBX v3 is a developer release and as such many
> features are in flux and might not work.?That being said there some features
> in the Windows build that still do not work. The biggest one right now is
> the lack of the php ESL library for Windows which affects the voicemail app.
> I'm trying to get this to compile but it's been difficult.
>
>
>
> I do include the .svn files with the FreePBX install so you can freely
> install TortoiseSVN and update FreePBX at your leisure.
>
>
>
> With regards to the sip_profiles, did you create a trunk group and trunk?
>
>
>
> regards,
>
>
>
> Carlos
>
> On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>
> Windows installer does not work for me.
>
> I've reinstalled various times, same results.
>
> I can correctly create a number, but when I try to create a device for
> that number, it tells me that cannot locate the device, and the
> password for vicemail will be invalid.
>
> After that, it begins to give the php error page, it cannot find the
> start < tag in directory/default.xml
>
> Also for me there are no sofia profiles...
>
> So, I cannot start to test it (eg: I would like to add mod_skypiax
> support to it).
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
>
> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
>
> Guidi<raffaele.p.guidi at gmail.com> wrote:
>
>> This is what I was asking! :D When the installer finished it started the
>> whole thing and everything got loaded fine, but when I restarted my system
>> it didn't (and did not anymore). Well, I will try to install everything
>> from
>> scratch again and see...
>>
>> On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org> wrote:
>>>
>>> If you installed FreePBX then it would be that softwares job to manage
>>> the sofia profiles... wouldn't it?
>>>
>>> /b
>>>
>>> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>>>
>>> > Actually I did that and it worked fine. I had the problem the SECOND
>>> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
>>> > sip_profiles were not
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From mike at jerris.com  Tue Aug 25 11:33:44 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 14:33:44 -0400
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer -
	great	but I have a little problem
In-Reply-To: <5EA32FE3006E2740884F5A09AD98A5A718CDCB415A@34093-MBX-C03.mex07a.mlsrvr.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
	<5EA32FE3006E2740884F5A09AD98A5A718CDCB415A@34093-MBX-C03.mex07a.mlsrvr.com>
Message-ID: <C46DBBA8-88CA-4B3D-935F-0E0A3D0F7382@jerris.com>

It supports FreeSWITCH and soon other engines as well.  More  
information is at:

http://www.freepbx.org/freepbx-v3

Mike

On Aug 25, 2009, at 2:20 PM, Michael Di Martino wrote:

> Is FreePBX V3 based on Freeswitch?
>
> Michael DiMartino | Director of IT | Open Access, Inc.
> 115 Bi County Blvd | Farmingdale, NY 11735
> 631.227.1034| 631.694.6730 FAX |631.988.6060 MOBILE
> www.openaccessinc.com
>
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From jerry.richards at teotech.com  Tue Aug 25 11:35:16 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 25 Aug 2009 11:35:16 -0700
Subject: [Freeswitch-users] RTP Packet Routing
Message-ID: <0003D215849D415591E531947383F082@greyhawk.tonecommander.com>

Hello All,

I noticed Freeswitch becomes the "middle-man", handling RTP traffic for an
active call.  How do I configure it so it allows the two SIP endpoints to
send RTP packet to each other directly?

Best Regards,
Jerry



From msc at freeswitch.org  Tue Aug 25 11:42:53 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 11:42:53 -0700
Subject: [Freeswitch-users] RTP Packet Routing
In-Reply-To: <0003D215849D415591E531947383F082@greyhawk.tonecommander.com>
References: <0003D215849D415591E531947383F082@greyhawk.tonecommander.com>
Message-ID: <87f2f3b90908251142w772a4a5ai47f1b100908da373@mail.gmail.com>

On Tue, Aug 25, 2009 at 11:35 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

> Hello All,
>
> I noticed Freeswitch becomes the "middle-man", handling RTP traffic for an
> active call.  How do I configure it so it allows the two SIP endpoints to
> send RTP packet to each other directly?
>

Check out bypass media mode:
http://wiki.freeswitch.org/wiki/Bypass_Media

-MC
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From msc at freeswitch.org  Tue Aug 25 11:47:35 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 11:47:35 -0700
Subject: [Freeswitch-users] Scalability
In-Reply-To: <B43123F543104F3E894E15B06D705571@greyhawk.tonecommander.com>
References: <B43123F543104F3E894E15B06D705571@greyhawk.tonecommander.com>
Message-ID: <87f2f3b90908251147h48b80dex384ff30912a902f6@mail.gmail.com>

On Tue, Aug 25, 2009 at 11:12 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

> Hello All,
>
> Does anyone know what the capacity of a stand-alone Freeswitch, in terms of
> how many users?
>

You mean concurrent registered users? I suppose the answer is "a lot"! :)

>
> Also, when that number is exceeded, how can Freeswitch server be
> distributed
> to accommodate a larger installation?


FreeSWITCH supports a number of ways of spreading things around. Check this
out:
http://wiki.freeswitch.org/wiki/Mod_xml_curl

-MC
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From krice at freeswitch.org  Tue Aug 25 11:53:22 2009
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 25 Aug 2009 13:53:22 -0500
Subject: [Freeswitch-users] Scalability
In-Reply-To: <B43123F543104F3E894E15B06D705571@greyhawk.tonecommander.com>
Message-ID: <C6B99D52.5D691%krice@freeswitch.org>

Deployable scalability varies based on a number of things... Number of users
registering, how often they register, concurrent call volume, call rate
(calls/second) etc...  Defining that a little better may illicit a better
response... But generally FS can scale into the 1000s of concurrent calls at
that 100s of calls/sec with 100s of registered users... The however as with
any software how well it actually scales for you will be dependant on the
hardware you deploy it on and its actual configuration...


> From: Jerry Richards <jerry.richards at teotech.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 25 Aug 2009 11:12:39 -0700
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Scalability
> 
> Hello All,
> 
> Does anyone know what the capacity of a stand-alone Freeswitch, in terms of
> how many users?
> 
> Also, when that number is exceeded, how can Freeswitch server be distributed
> to accommodate a larger installation?
> 
> Best Regards,
> Jerry 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From krice at freeswitch.org  Tue Aug 25 11:54:27 2009
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 25 Aug 2009 13:54:27 -0500
Subject: [Freeswitch-users] RTP Packet Routing
In-Reply-To: <87f2f3b90908251142w772a4a5ai47f1b100908da373@mail.gmail.com>
Message-ID: <C6B99D93.5D692%krice@freeswitch.org>

Just remember that bypass_media is a special mode and not everything will
work when a call is in that mode... Don?t expect to do anything that
remotely relies on media being in the FS box...



From: Michael Collins <msc at freeswitch.org>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Tue, 25 Aug 2009 11:42:53 -0700
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] RTP Packet Routing



On Tue, Aug 25, 2009 at 11:35 AM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> Hello All,
> 
> I noticed Freeswitch becomes the "middle-man", handling RTP traffic for an
> active call. ?How do I configure it so it allows the two SIP endpoints to
> send RTP packet to each other directly?

Check out bypass media mode:
http://wiki.freeswitch.org/wiki/Bypass_Media

-MC



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From MPeace at edcogroupinc.com  Tue Aug 25 11:58:07 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 13:58:07 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>

It was started from the services configuration window first and I still get the error. I set the service not to load rebooted then ran it from the prompt as you suggested. It stated "3585 Backgrounding" I then ran ps -eaf from the prompt and that PID didn't show up. Then when I attempted to run the fs_cli I received the same error as before  [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error]


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 12:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

It means freeswitch is not running. You need to start it with /usr/local/freeswitch/bin/freeswitch -nc

@MikeJ: Nice bash scripting :D

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>




On 25-Aug-09, at 1:53 PM, Mike Peace wrote:


Ok, figured out from your post to load the FS-CLI in order to run the Sophia status Michael mentioned.

Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error] when running ./fs_cli from a terminal window in /usr/local/freeswitch/bin where fs_cli is located.

Mike
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, August 25, 2009 12:39 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

if you want to run that at your prompt instead of at the fs_cli you can do this:

function sofia() { fs_cli -x "$(echo "sofia $@")"; }

(thanks ray for the bash foo)

Mike

On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:



I get a bash: sofia: command not found. Is there something I need to add to my config to use these commands?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 25, 2009 10:55 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

Hmm... can you fire up FS and then do "sofia status" and tell us if the internal profile looks okay? Also, do "sofia status profile internal" and capture the output. Something fishy is going on...
-MC

Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367


________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From mrene_lists at avgs.ca  Tue Aug 25 12:01:43 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 25 Aug 2009 15:01:43 -0400
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
Message-ID: <FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>

You can start it without the -nc switch and it'll stay active on the  
current terminal so you can look for any errors.

Just do:

/usr/local/freeswitch/bin/freeswitch


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca



On 25-Aug-09, at 2:58 PM, Mike Peace wrote:

> It was started from the services configuration window first and I  
> still get the error. I set the service not to load rebooted then ran  
> it from the prompt as you suggested. It stated ?3585 Backgrounding?  
> I then ran ps ?eaf from the prompt and that PID didn?t show up. Then  
> when I attempted to run the fs_cli I received the same error as  
> before  [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting  
> [Socket Connection Error]
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Mathieu Rene
> Sent: Tuesday, August 25, 2009 12:55 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> It means freeswitch is not running. You need to start it with /usr/ 
> local/freeswitch/bin/freeswitch -nc
>
> @MikeJ: Nice bash scripting :D
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 25-Aug-09, at 1:53 PM, Mike Peace wrote:
>
>
> Ok, figured out from your post to load the FS-CLI in order to run  
> the Sophia status Michael mentioned.
>
> Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting  
> [Socket Connection Error] when running ./fs_cli from a terminal  
> window in /usr/local/freeswitch/bin where fs_cli is located.
>
> Mike
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Jerris
> Sent: Tuesday, August 25, 2009 12:39 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> if you want to run that at your prompt instead of at the fs_cli you  
> can do this:
>
> function sofia() { fs_cli -x "$(echo "sofia $@")"; }
>
> (thanks ray for the bash foo)
>
> Mike
>
> On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:
>
>
>
> I get a bash: sofia: command not found. Is there something I need to  
> add to my config to use these commands?
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Tuesday, August 25, 2009 10:55 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> Hmm... can you fire up FS and then do "sofia status" and tell us if  
> the internal profile looks okay? Also, do "sofia status profile  
> internal" and capture the output. Something fishy is going on...
> -MC
> Mike Peace
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
>
> EDCO Group, Inc.
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
> Confidentiality Notice: The information contained in this e-mail  
> message (including any attachments) may contain confidential and  
> privileged information, and is for the sole use of the intended  
> recipient(s). If you are not the intended recipient, any  
> unauthorized review, use, or disclosure or distribution is strictly  
> prohibited. If you have received this message in error, please  
> notify the sender by replying to this e-mail message or by telephone  
> at (800) 999-3456 and permanently destroy all copies of the original  
> message (including any attachments), along with any reply, and  
> delete them from your system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> EDCO Group, Inc.
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
> Confidentiality Notice: The information contained in this e-mail  
> message (including any attachments) may contain confidential and  
> privileged information, and is for the sole use of the intended  
> recipient(s). If you are not the intended recipient, any  
> unauthorized review, use, or disclosure or distribution is strictly  
> prohibited. If you have received this message in error, please  
> notify the sender by replying to this e-mail message or by telephone  
> at (800) 999-3456 and permanently destroy all copies of the original  
> message (including any attachments), along with any reply, and  
> delete them from your system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From carlos.talbot at gmail.com  Tue Aug 25 12:02:02 2009
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Tue, 25 Aug 2009 14:02:02 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <7b197bef0908251128y39de0bc8tcf123ee6343bfb33@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
	<7b197bef0908251128y39de0bc8tcf123ee6343bfb33@mail.gmail.com>
Message-ID: <5800526b0908251202o730d2d18nb6948bad3766b705@mail.gmail.com>

This would be a question for Darren and the FreePBX group. :) I guess it
does not help that the *User Documentation* link on this page is currently
empty: http://www.freepbx.org/v3/wiki/

If you note the message during FreePBX initialization *all* files in the
sip_profiles directory are removed (including internal*.xml). This causes
'sofia status' to come back empty.


Incompatible ConfigurationWARNING: THE FOLLOWING FILES WILL BE DELETED!

   - D:/FreeSWITCH/conf/sip_profiles/external.xml
   - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
   - D:/FreeSWITCH/conf/sip_profiles/internal.xml


regards,

Carlos

On Tue, Aug 25, 2009 at 1:28 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
 wrote:

>
> >From the front page of FPBX is not clear you *must* create a trunk/trunk
> group.
>
> I was thinking trunks were for outgoing calls, or for receiving from
> external.
>
> I was just testing internal phones, trying an IVR, so I was thinking
> trunks were not needed.
>
> Can you explain to me?


>
> Thanks again,
>
> -giovanni
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
>
> On Tue, Aug 25, 2009 at 8:06 PM, Carlos Talbot<carlos.talbot at gmail.com>
> wrote:
> > Giovanni,
> > you mean like this message?
> > "Unable to determine location for device. Voicemail password set via
> FreePBX
> > will not be valid."
> > This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36
> > Let's keep in ming FreePBX v3 is a developer release and as such many
> > features are in flux and might not work. That being said there some
> features
> > in the Windows build that still do not work. The biggest one right now is
> > the lack of the php ESL library for Windows which affects the voicemail
> app.
> > I'm trying to get this to compile but it's been difficult.
> > I do include the .svn files with the FreePBX install so you can freely
> > install TortoiseSVN and update FreePBX at your leisure.
> > With regards to the sip_profiles, did you create a trunk group and trunk?
> > regards,
> > Carlos
> >
> > On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <
> gmaruzz at celliax.org>
> > wrote:
> >>
> >> Windows installer does not work for me.
> >>
> >> I've reinstalled various times, same results.
> >>
> >> I can correctly create a number, but when I try to create a device for
> >> that number, it tells me that cannot locate the device, and the
> >> password for vicemail will be invalid.
> >>
> >> After that, it begins to give the php error page, it cannot find the
> >> start < tag in directory/default.xml
> >>
> >> Also for me there are no sofia profiles...
> >>
> >> So, I cannot start to test it (eg: I would like to add mod_skypiax
> >> support to it).
> >>
> >>
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >>
> >>
> >>
> >> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
> >> Guidi<raffaele.p.guidi at gmail.com> wrote:
> >> > This is what I was asking! :D When the installer finished it started
> the
> >> > whole thing and everything got loaded fine, but when I restarted my
> >> > system
> >> > it didn't (and did not anymore). Well, I will try to install
> everything
> >> > from
> >> > scratch again and see...
> >> >
> >> > On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org>
> wrote:
> >> >>
> >> >> If you installed FreePBX then it would be that softwares job to
> manage
> >> >> the sofia profiles... wouldn't it?
> >> >>
> >> >> /b
> >> >>
> >> >> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
> >> >>
> >> >> > Actually I did that and it worked fine. I had the problem the
> SECOND
> >> >> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
> >> >> > sip_profiles were not
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From MPeace at edcogroupinc.com  Tue Aug 25 12:41:23 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 14:41:23 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
	<FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A688B@ESNEXS2.edcogroup.net>


OK,

Received a different error: Cannot Initialize [[error near line 294]: unclosed <!--]

?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 2:02 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

You can start it without the -nc switch and it'll stay active on the current terminal so you can look for any errors.

Just do:

/usr/local/freeswitch/bin/freeswitch


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>



On 25-Aug-09, at 2:58 PM, Mike Peace wrote:


It was started from the services configuration window first and I still get the error. I set the service not to load rebooted then ran it from the prompt as you suggested. It stated "3585 Backgrounding" I then ran ps -eaf from the prompt and that PID didn't show up. Then when I attempted to run the fs_cli I received the same error as before  [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error]


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 12:55 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

It means freeswitch is not running. You need to start it with /usr/local/freeswitch/bin/freeswitch -nc

@MikeJ: Nice bash scripting :D

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>




On 25-Aug-09, at 1:53 PM, Mike Peace wrote:



Ok, figured out from your post to load the FS-CLI in order to run the Sophia status Michael mentioned.

Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error] when running ./fs_cli from a terminal window in /usr/local/freeswitch/bin where fs_cli is located.

Mike
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, August 25, 2009 12:39 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

if you want to run that at your prompt instead of at the fs_cli you can do this:

function sofia() { fs_cli -x "$(echo "sofia $@")"; }

(thanks ray for the bash foo)

Mike

On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:




I get a bash: sofia: command not found. Is there something I need to add to my config to use these commands?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 25, 2009 10:55 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

Hmm... can you fire up FS and then do "sofia status" and tell us if the internal profile looks okay? Also, do "sofia status profile internal" and capture the output. Something fishy is going on...
-MC

Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367


________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From mrene_lists at avgs.ca  Tue Aug 25 12:45:01 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 25 Aug 2009 15:45:01 -0400
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A688B@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
	<FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A688B@ESNEXS2.edcogroup.net>
Message-ID: <9C4244D1-EC2C-4B3C-BD5E-16643BA5D0D4@avgs.ca>

It means you have an error in your XML configuration. You can open / 
usr/local/freeswitch/log/freeswitch.xml.fsxml to figure out where it  
is (but do not modify this file, as it is generated from all the other  
xml files).

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 25-Aug-09, at 3:41 PM, Mike Peace wrote:

>
> OK,
>
> Received a different error: Cannot Initialize [[error near line  
> 294]: unclosed <!--]
>
> ?
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Mathieu Rene
> Sent: Tuesday, August 25, 2009 2:02 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> You can start it without the -nc switch and it'll stay active on the  
> current terminal so you can look for any errors.
>
> Just do:
>
> /usr/local/freeswitch/bin/freeswitch
>
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
> On 25-Aug-09, at 2:58 PM, Mike Peace wrote:
>
>
> It was started from the services configuration window first and I  
> still get the error. I set the service not to load rebooted then ran  
> it from the prompt as you suggested. It stated ?3585 Backgrounding?  
> I then ran ps ?eaf from the prompt and that PID didn?t show up. Then  
> when I attempted to run the fs_cli I received the same error as  
> before  [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting  
> [Socket Connection Error]
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Mathieu Rene
> Sent: Tuesday, August 25, 2009 12:55 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> It means freeswitch is not running. You need to start it with /usr/ 
> local/freeswitch/bin/freeswitch -nc
>
> @MikeJ: Nice bash scripting :D
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 25-Aug-09, at 1:53 PM, Mike Peace wrote:
>
>
>
> Ok, figured out from your post to load the FS-CLI in order to run  
> the Sophia status Michael mentioned.
>
> Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting  
> [Socket Connection Error] when running ./fs_cli from a terminal  
> window in /usr/local/freeswitch/bin where fs_cli is located.
>
> Mike
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Jerris
> Sent: Tuesday, August 25, 2009 12:39 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> if you want to run that at your prompt instead of at the fs_cli you  
> can do this:
>
> function sofia() { fs_cli -x "$(echo "sofia $@")"; }
>
> (thanks ray for the bash foo)
>
> Mike
>
> On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:
>
>
>
>
> I get a bash: sofia: command not found. Is there something I need to  
> add to my config to use these commands?
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Tuesday, August 25, 2009 10:55 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> Hmm... can you fire up FS and then do "sofia status" and tell us if  
> the internal profile looks okay? Also, do "sofia status profile  
> internal" and capture the output. Something fishy is going on...
> -MC
> Mike Peace
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
>
> EDCO Group, Inc.
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
> Confidentiality Notice: The information contained in this e-mail  
> message (including any attachments) may contain confidential and  
> privileged information, and is for the sole use of the intended  
> recipient(s). If you are not the intended recipient, any  
> unauthorized review, use, or disclosure or distribution is strictly  
> prohibited. If you have received this message in error, please  
> notify the sender by replying to this e-mail message or by telephone  
> at (800) 999-3456 and permanently destroy all copies of the original  
> message (including any attachments), along with any reply, and  
> delete them from your system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> EDCO Group, Inc.
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
> Confidentiality Notice: The information contained in this e-mail  
> message (including any attachments) may contain confidential and  
> privileged information, and is for the sole use of the intended  
> recipient(s). If you are not the intended recipient, any  
> unauthorized review, use, or disclosure or distribution is strictly  
> prohibited. If you have received this message in error, please  
> notify the sender by replying to this e-mail message or by telephone  
> at (800) 999-3456 and permanently destroy all copies of the original  
> message (including any attachments), along with any reply, and  
> delete them from your system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> EDCO Group, Inc.
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
> Confidentiality Notice: The information contained in this e-mail  
> message (including any attachments) may contain confidential and  
> privileged information, and is for the sole use of the intended  
> recipient(s). If you are not the intended recipient, any  
> unauthorized review, use, or disclosure or distribution is strictly  
> prohibited. If you have received this message in error, please  
> notify the sender by replying to this e-mail message or by telephone  
> at (800) 999-3456 and permanently destroy all copies of the original  
> message (including any attachments), along with any reply, and  
> delete them from your system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bjbrashier at gmail.com  Tue Aug 25 13:23:53 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Tue, 25 Aug 2009 14:23:53 -0600
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <191c3a030908251005v6657943l5cdd7645446fcbc5@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com> 
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com> 
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com> 
	<15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com> 
	<15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com> 
	<65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com> 
	<191c3a030908251005v6657943l5cdd7645446fcbc5@mail.gmail.com>
Message-ID: <7bcfdd290908251323o65654ba3w663900ec693b81cb@mail.gmail.com>

Well, you'd have another nickel from over here, then.
If I can get this working before I'm tasked with something else I'll write
up something more on the wiki about "Freeswitch and SIPp", but I'm not sure
I'll get that chance.

BB

On Tue, Aug 25, 2009 at 11:05 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I wish I had a nickel for every guy struggling with sipp load testing vs
> real world traffic.
>
>
>
> On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> Hello Takeshi,
>>
>> Thanks for your hint... it worked out... so to be precise:
>>
>> VIA header of both INVITE and ACK messages MUST be identical (IP:PORT +
>> branch)... and you are right... it might not be according to SIP
>> specification. Anyhow, i get FS understand my ACK message.
>>
>>
>> Finally, here is what i used and I'm getting some poor results .. but this
>> is another topic :)
>>
>>
>> Thanks for your help.
>> Tihomir.
>>
>>
>> sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1
>> -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
>>
>>
>> <?xml version="1.0" encoding="ISO-8859-1" ?>
>> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>>
>>
>> <scenario name="Basic Sipstone UAC">
>>   <send retrans="500" start_rtd="1" start_rtd="2">
>>
>>     <![CDATA[
>>
>>       INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>       Max-Forwards: 70
>>       Contact: <sip:[field1]@[local_ip]>
>>       From: [field1]
>> <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
>>       To: [service] <sip:[service]@[remote_ip]:[remote_port]>
>>       Call-ID: [call_id]
>>       CSeq: 1 INVITE
>>       Content-Type: application/sdp
>>       Content-Length: [len]
>>
>>       v=0
>>       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>>       s=-
>>       c=IN IP[media_ip_type] [media_ip]
>>       t=0 0
>>       m=audio [media_port] RTP/AVP 0
>>       a=rtpmap:0 PCMU/8000
>>
>>     ]]>
>>   </send>
>>
>>   <recv response="100"
>>         optional="true" rtd="1">
>>   </recv>
>>
>>
>>   <recv response="302" rtd="2">
>>   </recv>
>>
>>   <send>
>>     <![CDATA[
>>
>>       ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>>       From: [field1]
>> <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]
>>       To: [service]
>> <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
>>       Call-ID: [call_id]
>>       CSeq: 1 ACK
>>       Max-Forwards: 70
>>       Content-Length: 0
>>
>>     ]]>
>>   </send>
>>
>>   <!-- definition of the response time repartition table (unit is ms)
>> -->
>>   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
>>
>>   <!-- definition of the call length repartition table (unit is ms)
>> -->
>>   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
>>
>> </scenario>
>>
>>
>>
>> On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>>
>>> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com>
>>> wrote:
>>> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com>
>>> wrote:
>>> >>
>>> >> sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
>>> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m
>>> 1 -l
>>> >> 4000
>>> >> scenario file:      uac_redirect.xml
>>> >> FS dialplan:       public.xml
>>> >> SIP trace:          trace.log
>>> >
>>> > The Via definition in your SIPp scenario differs between the INVITE and
>>> the ACK:
>>> >
>>> > INVITE:
>>> > Via: SIP/2.0/[transport] [local_ip];branch=[branch]
>>> >
>>> > ACK:
>>> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>>> >
>>> >
>>> > In the INVITE, you are not adding the [local_port] as you do in the
>>> ACK.
>>> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept the
>>> ACK.
>>> > So it seems FS is not checking just the ACK's branch against the
>>> > INVITE's; it seems it is checking the whole Via header.
>>> > I don't know if this is in accordance to SIP specs.
>>> > Another thing, about the way you are calling SIPp: do no use "-sn uac"
>>> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
>>> > means "use the internal (embedded) scenario named xxx". So this
>>> > conflicts with the other parameter "-sf" which specifies an external
>>> > profile.
>>>
>>> I mean, an external scenario (file).
>>>
>>>  It seems this doesn't cause any problem (probably because in
>>> > the sipp startup, -sf overrides -sn), but it is misleading.
>>> >
>>> > regards,
>>> > takeshi
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Aug 25 13:39:21 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Aug 2009 15:39:21 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <7bcfdd290908251323o65654ba3w663900ec693b81cb@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
	<15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>
	<15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com>
	<65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com>
	<191c3a030908251005v6657943l5cdd7645446fcbc5@mail.gmail.com>
	<7bcfdd290908251323o65654ba3w663900ec693b81cb@mail.gmail.com>
Message-ID: <191c3a030908251339n57370a92nb6c404903e6f0353@mail.gmail.com>

here's the one i use for making a call waiting x seconds and hanging up
http://www.freeswitch.org/eg/load_test/dft_cap.xml

This requires that the sipp terminate all the calls.

careful with sipp, it's like a roach motel, you can get stuck trying to make
it work and never get it to produce real-life situations.



On Tue, Aug 25, 2009 at 3:23 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> Well, you'd have another nickel from over here, then.
> If I can get this working before I'm tasked with something else I'll write
> up something more on the wiki about "Freeswitch and SIPp", but I'm not sure
> I'll get that chance.
>
> BB
>
> On Tue, Aug 25, 2009 at 11:05 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I wish I had a nickel for every guy struggling with sipp load testing vs
>> real world traffic.
>>
>>
>>
>> On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> Hello Takeshi,
>>>
>>> Thanks for your hint... it worked out... so to be precise:
>>>
>>> VIA header of both INVITE and ACK messages MUST be identical (IP:PORT +
>>> branch)... and you are right... it might not be according to SIP
>>> specification. Anyhow, i get FS understand my ACK message.
>>>
>>>
>>> Finally, here is what i used and I'm getting some poor results .. but
>>> this is another topic :)
>>>
>>>
>>> Thanks for your help.
>>> Tihomir.
>>>
>>>
>>> sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1
>>> -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
>>>
>>>
>>> <?xml version="1.0" encoding="ISO-8859-1" ?>
>>> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>>>
>>>
>>> <scenario name="Basic Sipstone UAC">
>>>   <send retrans="500" start_rtd="1" start_rtd="2">
>>>
>>>     <![CDATA[
>>>
>>>       INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>>>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>>       Max-Forwards: 70
>>>       Contact: <sip:[field1]@[local_ip]>
>>>       From: [field1]
>>> <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
>>>       To: [service] <sip:[service]@[remote_ip]:[remote_port]>
>>>       Call-ID: [call_id]
>>>       CSeq: 1 INVITE
>>>       Content-Type: application/sdp
>>>       Content-Length: [len]
>>>
>>>       v=0
>>>       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>>>       s=-
>>>       c=IN IP[media_ip_type] [media_ip]
>>>       t=0 0
>>>       m=audio [media_port] RTP/AVP 0
>>>       a=rtpmap:0 PCMU/8000
>>>
>>>     ]]>
>>>   </send>
>>>
>>>   <recv response="100"
>>>         optional="true" rtd="1">
>>>   </recv>
>>>
>>>
>>>   <recv response="302" rtd="2">
>>>   </recv>
>>>
>>>   <send>
>>>     <![CDATA[
>>>
>>>       ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>>>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>>>       From: [field1]
>>> <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]
>>>       To: [service]
>>> <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
>>>       Call-ID: [call_id]
>>>       CSeq: 1 ACK
>>>       Max-Forwards: 70
>>>       Content-Length: 0
>>>
>>>     ]]>
>>>   </send>
>>>
>>>   <!-- definition of the response time repartition table (unit is ms)
>>> -->
>>>   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
>>>
>>>   <!-- definition of the call length repartition table (unit is ms)
>>> -->
>>>   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
>>>
>>> </scenario>
>>>
>>>
>>>
>>> On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>>>
>>>> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com>
>>>> wrote:
>>>> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com>
>>>> wrote:
>>>> >>
>>>> >> sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
>>>> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt
>>>> -m 1 -l
>>>> >> 4000
>>>> >> scenario file:      uac_redirect.xml
>>>> >> FS dialplan:       public.xml
>>>> >> SIP trace:          trace.log
>>>> >
>>>> > The Via definition in your SIPp scenario differs between the INVITE
>>>> and the ACK:
>>>> >
>>>> > INVITE:
>>>> > Via: SIP/2.0/[transport] [local_ip];branch=[branch]
>>>> >
>>>> > ACK:
>>>> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>>>> >
>>>> >
>>>> > In the INVITE, you are not adding the [local_port] as you do in the
>>>> ACK.
>>>> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept the
>>>> ACK.
>>>> > So it seems FS is not checking just the ACK's branch against the
>>>> > INVITE's; it seems it is checking the whole Via header.
>>>> > I don't know if this is in accordance to SIP specs.
>>>> > Another thing, about the way you are calling SIPp: do no use "-sn uac"
>>>> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
>>>> > means "use the internal (embedded) scenario named xxx". So this
>>>> > conflicts with the other parameter "-sf" which specifies an external
>>>> > profile.
>>>>
>>>> I mean, an external scenario (file).
>>>>
>>>>  It seems this doesn't cause any problem (probably because in
>>>> > the sipp startup, -sf overrides -sn), but it is misleading.
>>>> >
>>>> > regards,
>>>> > takeshi
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From aep.lists at it46.se  Tue Aug 25 13:46:50 2009
From: aep.lists at it46.se (Alberto Escudero-Pascual (lists))
Date: Tue, 25 Aug 2009 22:46:50 +0200
Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
In-Reply-To: <5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
	<191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
	<6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>
	<5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org>
Message-ID: <45b2e91d7aa77afcf6430a14dde4b886.squirrel@correo.nodo50.org>

Hi Brian,

>From the CLI>

freeswitch at open46> system /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
/tmp/foo.mp3 -S
2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing command:
/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S
API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
/tmp/foo.mp3 -S)] output:
+OK

open46:/tmp# ls
foo.wav


and running the command from the command line:


open46:/tmp#  /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3
-Sopen46:/tmp# ls
foo.mp3  foo.wav


If I do the same with lame397

freeswitch at open46> system /usr/local/freeswitch/bin/lame397 -V2
/tmp/foo.wav /tmp/foo.mp3 -S
2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing command:
/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S
API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav
/tmp/foo.mp3 -S)] output:
+OK

open46:/tmp# ls
foo.mp3  foo.wav


Highly paranormal! Sorry for hijacking the previous thread.

/aep

-- 
Stopping junk mailers is good for the environment

> Try running it at the CLI and see if you see any errors.  Also please
> do not hijack threads.  The original thread "[Freeswitch-users] XML-
> RPC on different ip than 0.0.0.0" which was hijacked by clicking
> reply, changing the subject and clicking send.  Please in the future
> do not do that as it clutters up the threading and could get your
> query lost in the noise.
>
> Thanks,
> Brian
>
> On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote:
>
>> Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
>> appears, if I use version 3.97 (older version), it does!.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>




From anthony.minessale at gmail.com  Tue Aug 25 13:54:15 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Aug 2009 15:54:15 -0500
Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
In-Reply-To: <45b2e91d7aa77afcf6430a14dde4b886.squirrel@correo.nodo50.org>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
	<191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
	<6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>
	<5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org>
	<45b2e91d7aa77afcf6430a14dde4b886.squirrel@correo.nodo50.org>
Message-ID: <191c3a030908251354l25206d0ch19836c8996f43e84@mail.gmail.com>

maybe it's writing some err to stderr that is being suppressed somehow

On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists) <
aep.lists at it46.se> wrote:

> Hi Brian,
>
> >From the CLI>
>
> freeswitch at open46> system /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
> /tmp/foo.mp3 -S
> 2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing command:
> /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S
> API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
> /tmp/foo.mp3 -S)] output:
> +OK
>
> open46:/tmp# ls
> foo.wav
>
>
> and running the command from the command line:
>
>
> open46:/tmp#  /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3
> -Sopen46:/tmp# ls
> foo.mp3  foo.wav
>
>
> If I do the same with lame397
>
> freeswitch at open46> system /usr/local/freeswitch/bin/lame397 -V2
> /tmp/foo.wav /tmp/foo.mp3 -S
> 2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing command:
> /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S
> API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav
> /tmp/foo.mp3 -S)] output:
> +OK
>
> open46:/tmp# ls
> foo.mp3  foo.wav
>
>
> Highly paranormal! Sorry for hijacking the previous thread.
>
> /aep
>
> --
> Stopping junk mailers is good for the environment
>
> > Try running it at the CLI and see if you see any errors.  Also please
> > do not hijack threads.  The original thread "[Freeswitch-users] XML-
> > RPC on different ip than 0.0.0.0" which was hijacked by clicking
> > reply, changing the subject and clicking send.  Please in the future
> > do not do that as it clutters up the threading and could get your
> > query lost in the noise.
> >
> > Thanks,
> > Brian
> >
> > On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote:
> >
> >> Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
> >> appears, if I use version 3.97 (older version), it does!.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From d at d-man.org  Tue Aug 25 13:59:03 2009
From: d at d-man.org (Darren Schreiber)
Date: Tue, 25 Aug 2009 13:59:03 -0700
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer -
	great	but I have a little problem
In-Reply-To: <5800526b0908251202o730d2d18nb6948bad3766b705@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
	<7b197bef0908251128y39de0bc8tcf123ee6343bfb33@mail.gmail.com>
	<5800526b0908251202o730d2d18nb6948bad3766b705@mail.gmail.com>
Message-ID: <8A034A3098ED3C4990F7D9DE40F5585F1762B73424@EXVMBX020-3.exch020.serverdata.net>

Hi there...

So a few things on this.

1) We have a module that's still being worked on called Sip Interface that allows you to configure Sip Profiles in FreeSWITCH. Unfortunately we don't have the ability to easily import your existing SIP profiles, and by NOT displaying them in the UI your stock config conflicts with those profiles. In other words, internal.xml and external.xml define sip profiles on ports 5060 and 5080 that the GUI is unaware. So to simplify things, we just delete those files and expect you to recreate them via the UI on install. We'll probably make this a little more obvious in the near future, but for now, that's what we do.

2) The trunk creation system in the ISO does give the impression that trunk groups & trunks are for external. This is a design flaw I have already fixed in trunk (referenced above). We have now split this module into two modules - one for configuring Sip Interface/Sip Profiles (which are for defining your IPs & ports to use for sending/receiving calls and authentication settings) and the other for defining gateways ('trunks") which are generally used for making outbound calls.

So in other words, we're aware of the issues you are detailing. We will be finalizing the fix for this issue hopefully on Sunday and will rebuild the ISO by Monday.

Hope that helps.

- Darren




________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Carlos Talbot
Sent: Tuesday, August 25, 2009 12:02 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem


This would be a question for Darren and the FreePBX group. :) I guess it does not help that the User Documentation link on this page is currently empty: http://www.freepbx.org/v3/wiki/

If you note the message during FreePBX initialization *all* files in the sip_profiles directory are removed (including internal*.xml). This causes 'sofia status' to come back empty.


Incompatible Configuration
WARNING: THE FOLLOWING FILES WILL BE DELETED!


 *   D:/FreeSWITCH/conf/sip_profiles/external.xml
 *   D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
 *   D:/FreeSWITCH/conf/sip_profiles/internal.xml



regards,

Carlos

On Tue, Aug 25, 2009 at 1:28 PM, Giovanni Maruzzelli <gmaruzz at celliax.org<mailto:gmaruzz at celliax.org>> wrote:

>From the front page of FPBX is not clear you *must* create a trunk/trunk group.

I was thinking trunks were for outgoing calls, or for receiving from external.

I was just testing internal phones, trying an IVR, so I was thinking
trunks were not needed.

Can you explain to me?


Thanks again,

-giovanni

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 8:06 PM, Carlos Talbot<carlos.talbot at gmail.com<mailto:carlos.talbot at gmail.com>> wrote:
> Giovanni,
> you mean like this message?
> "Unable to determine location for device. Voicemail password set via FreePBX
> will not be valid."
> This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36
> Let's keep in ming FreePBX v3 is a developer release and as such many
> features are in flux and might not work. That being said there some features
> in the Windows build that still do not work. The biggest one right now is
> the lack of the php ESL library for Windows which affects the voicemail app.
> I'm trying to get this to compile but it's been difficult.
> I do include the .svn files with the FreePBX install so you can freely
> install TortoiseSVN and update FreePBX at your leisure.
> With regards to the sip_profiles, did you create a trunk group and trunk?
> regards,
> Carlos
>
> On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <gmaruzz at celliax.org<mailto:gmaruzz at celliax.org>>
> wrote:
>>
>> Windows installer does not work for me.
>>
>> I've reinstalled various times, same results.
>>
>> I can correctly create a number, but when I try to create a device for
>> that number, it tells me that cannot locate the device, and the
>> password for vicemail will be invalid.
>>
>> After that, it begins to give the php error page, it cannot find the
>> start < tag in directory/default.xml
>>
>> Also for me there are no sofia profiles...
>>
>> So, I cannot start to test it (eg: I would like to add mod_skypiax
>> support to it).
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>
>>
>>
>> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
>> Guidi<raffaele.p.guidi at gmail.com<mailto:raffaele.p.guidi at gmail.com>> wrote:
>> > This is what I was asking! :D When the installer finished it started the
>> > whole thing and everything got loaded fine, but when I restarted my
>> > system
>> > it didn't (and did not anymore). Well, I will try to install everything
>> > from
>> > scratch again and see...
>> >
>> > On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
>> >>
>> >> If you installed FreePBX then it would be that softwares job to manage
>> >> the sofia profiles... wouldn't it?
>> >>
>> >> /b
>> >>
>> >> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>> >>
>> >> > Actually I did that and it worked fine. I had the problem the SECOND
>> >> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
>> >> > sip_profiles were not
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From MPeace at edcogroupinc.com  Tue Aug 25 13:59:48 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 15:59:48 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <9C4244D1-EC2C-4B3C-BD5E-16643BA5D0D4@avgs.ca>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
	<FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A688B@ESNEXS2.edcogroup.net>
	<9C4244D1-EC2C-4B3C-BD5E-16643BA5D0D4@avgs.ca>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A6921@ESNEXS2.edcogroup.net>

That was it! I had been try to setup the conference java script example and I made an error in the conference-conf.xml. Thanks a lot to everyone!

Can somebody point me in the right direction to get my new T-1 taking to the Sangoma A101?

I have it installed and setup the Wanpipe driver off their site.

When I issue a wanrouter status it shows connected, an ifconfig shows RX and TX packets incrementing on the w1g1 interface.

This is the spec on the phone line:

LD T1 with ISDN, ESF, B8ZS, DTMF, ASCENDING HUNTING, PSEUDO ANI 417-862-4351, REAL TIME ANI, COS US 50 CA,
AND TO ADD 10 8XX #S TO POINT AT THE DS1 WITH NO DNIS.

Thanks again!

Mike







From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 2:45 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

It means you have an error in your XML configuration. You can open /usr/local/freeswitch/log/freeswitch.xml.fsxml to figure out where it is (but do not modify this file, as it is generated from all the other xml files).

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>




On 25-Aug-09, at 3:41 PM, Mike Peace wrote:



OK,

Received a different error: Cannot Initialize [[error near line 294]: unclosed <!--]

?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 2:02 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

You can start it without the -nc switch and it'll stay active on the current terminal so you can look for any errors.

Just do:

/usr/local/freeswitch/bin/freeswitch


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>



On 25-Aug-09, at 2:58 PM, Mike Peace wrote:



It was started from the services configuration window first and I still get the error. I set the service not to load rebooted then ran it from the prompt as you suggested. It stated "3585 Backgrounding" I then ran ps -eaf from the prompt and that PID didn't show up. Then when I attempted to run the fs_cli I received the same error as before  [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error]


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 12:55 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

It means freeswitch is not running. You need to start it with /usr/local/freeswitch/bin/freeswitch -nc

@MikeJ: Nice bash scripting :D

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>




On 25-Aug-09, at 1:53 PM, Mike Peace wrote:




Ok, figured out from your post to load the FS-CLI in order to run the Sophia status Michael mentioned.

Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error] when running ./fs_cli from a terminal window in /usr/local/freeswitch/bin where fs_cli is located.

Mike
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, August 25, 2009 12:39 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

if you want to run that at your prompt instead of at the fs_cli you can do this:

function sofia() { fs_cli -x "$(echo "sofia $@")"; }

(thanks ray for the bash foo)

Mike

On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:





I get a bash: sofia: command not found. Is there something I need to add to my config to use these commands?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 25, 2009 10:55 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

Hmm... can you fire up FS and then do "sofia status" and tell us if the internal profile looks okay? Also, do "sofia status profile internal" and capture the output. Something fishy is going on...
-MC

Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367


________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

_______________________________________________
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________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

_______________________________________________
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http://www.freeswitch.org


________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From gmaruzz at celliax.org  Tue Aug 25 14:06:04 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 23:06:04 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <8A034A3098ED3C4990F7D9DE40F5585F1762B73424@EXVMBX020-3.exch020.serverdata.net>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com> 
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com> 
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com> 
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com> 
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com> 
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com> 
	<7b197bef0908251128y39de0bc8tcf123ee6343bfb33@mail.gmail.com> 
	<5800526b0908251202o730d2d18nb6948bad3766b705@mail.gmail.com> 
	<8A034A3098ED3C4990F7D9DE40F5585F1762B73424@EXVMBX020-3.exch020.serverdata.net>
Message-ID: <7b197bef0908251406ub380301p1ea74cc40c2ae358@mail.gmail.com>

Darren,

thanks! it helps a lot.

If we svn up in /var/www/freepbx we got the trunk?


On Tue, Aug 25, 2009 at 10:59 PM, Darren Schreiber<d at d-man.org> wrote:
> Hi there...
>
> So a few things on this.
>
> 1) We have a module that's still being worked on called Sip Interface that
> allows you to configure Sip Profiles in FreeSWITCH. Unfortunately we don't
> have the ability to easily import your existing SIP profiles, and by NOT
> displaying them in the UI your stock config conflicts with those profiles.
> In other words, internal.xml and external.xml define sip profiles on ports
> 5060 and 5080 that the GUI is unaware. So to simplify things, we just delete
> those files and expect you to recreate them via the UI on install. We'll
> probably make this a little more obvious in the near future, but for now,
> that's what we do.
>
> 2) The trunk creation system in the ISO does give the impression that trunk
> groups & trunks are for external. This is a design flaw I have already fixed
> in trunk (referenced above). We have now split this module into two modules
> - one for configuring Sip Interface/Sip Profiles (which are for defining
> your IPs & ports to use for sending/receiving calls and authentication
> settings) and the other for defining gateways ('trunks") which are generally
> used for making outbound calls.
>
> So in other words, we're aware of the issues you are detailing. We will be
> finalizing the fix for this issue hopefully on Sunday and will rebuild the
> ISO by Monday.
>
> Hope that helps.
>
> - Darren
>
>
>
> ________________________________
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Carlos
> Talbot
> Sent: Tuesday, August 25, 2009 12:02 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
> but I have a little problem
>
>
> This would be a question for Darren and the FreePBX group. :) I guess it
> does not help that the?User Documentation?link on this page is currently
> empty:?http://www.freepbx.org/v3/wiki/
> If you note the message during FreePBX initialization *all* files in the
> sip_profiles directory are removed (including internal*.xml). This causes
> 'sofia status' to come back empty.
>
> Incompatible Configuration
>
> WARNING: THE FOLLOWING FILES WILL BE DELETED!
>
> D:/FreeSWITCH/conf/sip_profiles/external.xml
> D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
> D:/FreeSWITCH/conf/sip_profiles/internal.xml
>
>
> regards,
> Carlos
>
> On Tue, Aug 25, 2009 at 1:28 PM, Giovanni
> Maruzzelli?<gmaruzz at celliax.org>?wrote:
>>
>> >From the front page of FPBX is not clear you *must* create a trunk/trunk
>> group.
>>
>> I was thinking trunks were for outgoing calls, or for receiving from
>> external.
>>
>> I was just testing internal phones, trying an IVR, so I was thinking
>> trunks were not needed.
>>
>> Can you explain to me?
>>
>> Thanks again,
>>
>> -giovanni
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>
>>
>>
>> On Tue, Aug 25, 2009 at 8:06 PM, Carlos Talbot<carlos.talbot at gmail.com>
>> wrote:
>> > Giovanni,
>> > you mean like this message?
>> > "Unable to determine location for device. Voicemail password set via
>> > FreePBX
>> > will not be valid."
>> > This is a known FreePBX issue.?http://www.freepbx.org/v3/ticket/36
>> > Let's keep in ming FreePBX v3 is a developer release and as such many
>> > features are in flux and might not work.?That being said there some
>> > features
>> > in the Windows build that still do not work. The biggest one right now
>> > is
>> > the lack of the php ESL library for Windows which affects the voicemail
>> > app.
>> > I'm trying to get this to compile but it's been difficult.
>> > I do include the .svn files with the FreePBX install so you can freely
>> > install TortoiseSVN and update FreePBX at your leisure.
>> > With regards to the sip_profiles, did you create a trunk group and
>> > trunk?
>> > regards,
>> > Carlos
>> >
>> > On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli
>> > <gmaruzz at celliax.org>
>> > wrote:
>> >>
>> >> Windows installer does not work for me.
>> >>
>> >> I've reinstalled various times, same results.
>> >>
>> >> I can correctly create a number, but when I try to create a device for
>> >> that number, it tells me that cannot locate the device, and the
>> >> password for vicemail will be invalid.
>> >>
>> >> After that, it begins to give the php error page, it cannot find the
>> >> start < tag in directory/default.xml
>> >>
>> >> Also for me there are no sofia profiles...
>> >>
>> >> So, I cannot start to test it (eg: I would like to add mod_skypiax
>> >> support to it).
>> >>
>> >>
>> >> Sincerely,
>> >>
>> >> Giovanni Maruzzelli
>> >> Cell : +39-347-2665618
>> >>
>> >>
>> >>
>> >>
>> >> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
>> >> Guidi<raffaele.p.guidi at gmail.com> wrote:
>> >> > This is what I was asking! :D When the installer finished it started
>> >> > the
>> >> > whole thing and everything got loaded fine, but when I restarted my
>> >> > system
>> >> > it didn't (and did not anymore). Well, I will try to install
>> >> > everything
>> >> > from
>> >> > scratch again and see...
>> >> >
>> >> > On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org>
>> >> > wrote:
>> >> >>
>> >> >> If you installed FreePBX then it would be that softwares job to
>> >> >> manage
>> >> >> the sofia profiles... wouldn't it?
>> >> >>
>> >> >> /b
>> >> >>
>> >> >> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>> >> >>
>> >> >> > Actually I did that and it worked fine. I had the problem the
>> >> >> > SECOND
>> >> >> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
>> >> >> > sip_profiles were not
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Tue Aug 25 14:59:56 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 14:59:56 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A6921@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
	<FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A688B@ESNEXS2.edcogroup.net>
	<9C4244D1-EC2C-4B3C-BD5E-16643BA5D0D4@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A6921@ESNEXS2.edcogroup.net>
Message-ID: <87f2f3b90908251459p29f033c6g7cf2ed234f4b4479@mail.gmail.com>

>
> Can somebody point me in the right direction to get my new T-1 taking to
> the Sangoma A101?
>
>
>
Are you using PRI or just a 24 channel T1, aka CAS or RBS? Check out this
page for some handy information:
http://wiki.freeswitch.org/wiki/OpenZAP

-MC

P.S. - If you join #openzap and/or #freeswitch on irc.freenode.net you can
usually get real-time help with most of these questions.
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From MPeace at edcogroupinc.com  Tue Aug 25 17:14:40 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 19:14:40 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <87f2f3b90908251459p29f033c6g7cf2ed234f4b4479@mail.gmail.com>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
	<FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A688B@ESNEXS2.edcogroup.net>
	<9C4244D1-EC2C-4B3C-BD5E-16643BA5D0D4@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A6921@ESNEXS2.edcogroup.net>
	<87f2f3b90908251459p29f033c6g7cf2ed234f4b4479@mail.gmail.com>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A622D21B4@ESNEXS2.edcogroup.net>

I have looked at that but I am confused on which files need to be edited. Since I have already installed in Wanpipe mode with the Sangoma card I skipped straight to the Wanpipe section. It mentions setting the [span wanpie PRI_1] etc in the openzap.conf then further down it mentions editing the autoload_configs/openzap.conf.xml with a completely different format of config.

Are these the same file? If not where does the first one mentioned reside?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 25, 2009 5:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


Can somebody point me in the right direction to get my new T-1 taking to the Sangoma A101?


Are you using PRI or just a 24 channel T1, aka CAS or RBS? Check out this page for some handy information:
http://wiki.freeswitch.org/wiki/OpenZAP

-MC

P.S. - If you join #openzap and/or #freeswitch on irc.freenode.net<http://irc.freenode.net> you can usually get real-time help with most of these questions.

=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From mitul at enterux.com  Tue Aug 25 18:53:45 2009
From: mitul at enterux.com (Mitul Limbani)
Date: Wed, 26 Aug 2009 07:23:45 +0530
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer -
	great	but I have a little problem
In-Reply-To: <8A034A3098ED3C4990F7D9DE40F5585F1762B73424@EXVMBX020-3.exch020.serverdata.net>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
	<7b197bef0908251128y39de0bc8tcf123ee6343bfb33@mail.gmail.com>
	<5800526b0908251202o730d2d18nb6948bad3766b705@mail.gmail.com>
	<8A034A3098ED3C4990F7D9DE40F5585F1762B73424@EXVMBX020-3.exch020.serverdata.net>
Message-ID: <596902CF-DC31-4E73-81CB-AA41B4CADE12@enterux.com>

Darren,

I shall look forward to check version 3 of FreePBX for FS.
I hope it not trixswitch anyone aware of status on trixswitch (not  
that I m a big fan of Trixbox, but quite a while back I had seen  
trixswitch iso so wondering if anything is progresing on that end.

Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com

On 26-Aug-2009, at 2:29 AM, Darren Schreiber <d at d-man.org> wrote:

> Hi there...
>
> So a few things on this.
>
> 1) We have a module that's still being worked on called Sip  
> Interface that allows you to configure Sip Profiles in FreeSWITCH.  
> Unfortunately we don't have the ability to easily import your  
> existing SIP profiles, and by NOT displaying them in the UI your  
> stock config conflicts with those profiles. In other words,  
> internal.xml and external.xml define sip profiles on ports 5060 and  
> 5080 that the GUI is unaware. So to simplify things, we just delete  
> those files and expect you to recreate them via the UI on install.  
> We'll probably make this a little more obvious in the near future,  
> but for now, that's what we do.
>
> 2) The trunk creation system in the ISO does give the impression  
> that trunk groups & trunks are for external. This is a design flaw I  
> have already fixed in trunk (referenced above). We have now split  
> this module into two modules - one for configuring Sip Interface/Sip  
> Profiles (which are for defining your IPs & ports to use for sending/ 
> receiving calls and authentication settings) and the other for  
> defining gateways ('trunks") which are generally used for making  
> outbound calls.
>
> So in other words, we're aware of the issues you are detailing. We  
> will be finalizing the fix for this issue hopefully on Sunday and  
> will rebuild the ISO by Monday.
>
> Hope that helps.
>
> - Darren
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
> Carlos Talbot
> Sent: Tuesday, August 25, 2009 12:02 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer -  
> great but I have a little problem
>
>
> This would be a question for Darren and the FreePBX group. :) I  
> guess it does not help that the User Documentation link on this page  
> is currently empty: http://www.freepbx.org/v3/wiki/
>
> If you note the message during FreePBX initialization *all* files in  
> the sip_profiles directory are removed (including internal*.xml).  
> This causes 'sofia status' to come back empty.
>
>
> Incompatible Configuration
> WARNING: THE FOLLOWING FILES WILL BE DELETED!
>
>
> D:/FreeSWITCH/conf/sip_profiles/external.xml
> D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
> D:/FreeSWITCH/conf/sip_profiles/internal.xml
>
> regards,
>
> Carlos
>
> On Tue, Aug 25, 2009 at 1:28 PM, Giovanni Maruzzelli <gmaruzz at celliax.org 
> > wrote:
>
> >From the front page of FPBX is not clear you *must* create a trunk/ 
> trunk group.
>
> I was thinking trunks were for outgoing calls, or for receiving from  
> external.
>
> I was just testing internal phones, trying an IVR, so I was thinking
> trunks were not needed.
>
> Can you explain to me?
>
>
> Thanks again,
>
> -giovanni
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
>
> On Tue, Aug 25, 2009 at 8:06 PM, Carlos  
> Talbot<carlos.talbot at gmail.com> wrote:
> > Giovanni,
> > you mean like this message?
> > "Unable to determine location for device. Voicemail password set  
> via FreePBX
> > will not be valid."
> > This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36
> > Let's keep in ming FreePBX v3 is a developer release and as such  
> many
> > features are in flux and might not work. That being said there  
> some features
> > in the Windows build that still do not work. The biggest one right  
> now is
> > the lack of the php ESL library for Windows which affects the  
> voicemail app.
> > I'm trying to get this to compile but it's been difficult.
> > I do include the .svn files with the FreePBX install so you can  
> freely
> > install TortoiseSVN and update FreePBX at your leisure.
> > With regards to the sip_profiles, did you create a trunk group and  
> trunk?
> > regards,
> > Carlos
> >
> > On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <gmaruzz at celliax.org 
> >
> > wrote:
> >>
> >> Windows installer does not work for me.
> >>
> >> I've reinstalled various times, same results.
> >>
> >> I can correctly create a number, but when I try to create a  
> device for
> >> that number, it tells me that cannot locate the device, and the
> >> password for vicemail will be invalid.
> >>
> >> After that, it begins to give the php error page, it cannot find  
> the
> >> start < tag in directory/default.xml
> >>
> >> Also for me there are no sofia profiles...
> >>
> >> So, I cannot start to test it (eg: I would like to add mod_skypiax
> >> support to it).
> >>
> >>
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >>
> >>
> >>
> >> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
> >> Guidi<raffaele.p.guidi at gmail.com> wrote:
> >> > This is what I was asking! :D When the installer finished it  
> started the
> >> > whole thing and everything got loaded fine, but when I  
> restarted my
> >> > system
> >> > it didn't (and did not anymore). Well, I will try to install  
> everything
> >> > from
> >> > scratch again and see...
> >> >
> >> > On Mon, Aug 24, 2009 at 20:30, Brian West  
> <brian at freeswitch.org> wrote:
> >> >>
> >> >> If you installed FreePBX then it would be that softwares job  
> to manage
> >> >> the sofia profiles... wouldn't it?
> >> >>
> >> >> /b
> >> >>
> >> >> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
> >> >>
> >> >> > Actually I did that and it worked fine. I had the problem  
> the SECOND
> >> >> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded  
> but
> >> >> > sip_profiles were not
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From csa at nowthor.com  Tue Aug 25 19:50:08 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Tue, 25 Aug 2009 22:50:08 -0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>	<4A93EFE1.6010007@nowthor.com>
	<191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>
Message-ID: <4A94A2E0.6040306@nowthor.com>

Anthony,

Yes, you are right, I was thinking strictly in terms of SIP gateways. I 
guess that instead on the tag "gateway", one could use "channel"? For 
example:

<action application="bridge">
   <!-- set some kind of *bridge-specific* variable -->
   <channel order=1 data="sofia/gateway/gw-1/1$1"/>
   <channel order=1 data="sofia/gateway/gw-2/1$1"/>
   <channel order=2 data="sofia/gateway/gw-3/1$1">
       <!-- set some kind of *channel-specific* variable -->
   </channel >
   <channel order=3 data="sofia/gateway/gw-4/1$1"/>
</action>

Would this be acceptable to you?

Regarding the combinations of continue_on_fail and hangup_after bridge, 
I'll give that a try although I suspect it will result in less 
structured and harder to understand markup...

Thanks!

Carlos

Anthony Minessale wrote:
> This suggestion violates the scope boundaries.
>
> gateways are specific concept to mod_sofia so a <gateway> tag in 
> <action> (part of agnostic xml dialplan)
> does not flow properly.
>
> you can also use combinations of continue_on_fail and hangup_after 
> bridge so you can
> just put each bridge statement in it's own action.
>
>
> On Tue, Aug 25, 2009 at 9:06 AM, Carlos S. Antunes <csa at nowthor.com 
> <mailto:csa at nowthor.com>> wrote:
>
>     Max,
>
>     I would like to see something similar too. For example, it would be
>     wonderful if one could specify multiple gateways to try like this or
>     something similar:
>
>     <action application="bridge">
>        <!-- set some kind of *bridge-specific* parameter/variable -->
>        <gateway order=1 data="sofia/gateway/gw-1/1$1"/>
>        <gateway order=1 data="sofia/gateway/gw-2/1$1"/>
>        <gateway order=2 data="sofia/gateway/gw-3/1$1">
>            <!-- set some kind of *gw-specific* parameter/variable -->
>        </gateway>
>        <gateway order=3 data="sofia/gateway/gw-4/1$1"/>
>     </action>
>
>     One would be able to avoid the "[]" and "{}" hacks and combine
>     sequential and simultaneous trying of gateways.
>
>     What do the developers think of this?
>
>     Carlos
>
>     Max Ivanov wrote:
>     > Nowdays I 'm forced to put multiple "|" to find first free
>     gateway, ie
>     >
>     sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
>     > ,
>     > the whole sting is tooo long, is there any shorter way to write
>     same thing? Like
>     > "sofia/gateway/panas*/1000" will try all gateways matching the
>     pattern.
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>     >
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
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From gshfreesw at gmail.com  Tue Aug 25 19:51:15 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Tue, 25 Aug 2009 22:51:15 -0400
Subject: [Freeswitch-users] caller id on origination does not work
Message-ID: <5070fcbd0908251951j5a7163b1se346abcdf0664c94@mail.gmail.com>

I am setting the caller id like this in my ESL script:

@con.sendRecv("api originate
{origination_caller_id_number=15103245678}sofia/gateway/junctionnetworks/1#{@number}
&park()")

And the caller id comes out as all zeroes.  The sip trace shows the "from"
as shown in the sofia status command.  This is output of sofia gateway
status and my gateway is under
/usr/local/freeswitch/conf/sip_profiles/external/junctionnetworks.xml .
Also, adding the gateway under /conf/directory/default did not work as the
wiki suugests. What I need to do for the caller_id to come through?


sofia status gateway junctionnetworks
=================================================================================================
Name        junctionnetworks
Scheme      Digest
Realm       jnctn.net
Username    username
Password    yes
>From        <sip:username at jnctn.net <sip%3Ausername at jnctn.net>
;transport=udp>
Contact     <sip:gw+junctionnetworks at 24.87.80.12:5080;transport=udp>
Exten       899xxxxxxx
To          sip:username at sip.jnctn.net <sip%3Ausername at sip.jnctn.net>
Proxy       sip:sip.jnctn.net
Context     public
Expires     3600
Freq        3600
Ping        0
PingFreq    0
State       REGED
Status      UP
CallsIN     0
CallsOUT    1
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From krice at freeswitch.org  Tue Aug 25 20:00:30 2009
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 25 Aug 2009 22:00:30 -0500
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <4A94A2E0.6040306@nowthor.com>
Message-ID: <C6BA0F7E.5D757%krice@freeswitch.org>

That still wouldn?t work... An action has 2 parameters application and
data... And deeper then that and you have to start re-arranging all sorts of
things...

Continue_on_fail and hangup_after_bridge like tony pointed out are what you
want if you don?t want to use the | delimiting ... I use these all the time
with gateway counts > 10 just stacking additional actions for each bridge
line



From: "Carlos S. Antunes" <csa at nowthor.com>
Organization: Nowthor Corporation
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Tue, 25 Aug 2009 22:50:08 -0400
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] how to avoid many "|" in bridge application?

Anthony,

Yes, you are right, I was thinking strictly in terms of SIP gateways. I
guess that instead on the tag "gateway", one could use "channel"? For
example:

<action application="bridge">
   <!-- set some kind of *bridge-specific* variable -->
   <channel order=1 data="sofia/gateway/gw-1/1$1"/>
   <channel order=1 data="sofia/gateway/gw-2/1$1"/>
   <channel order=2 data="sofia/gateway/gw-3/1$1">
       <!-- set some kind of *channel-specific* variable -->
   </channel >
   <channel order=3 data="sofia/gateway/gw-4/1$1"/>
</action>

Would this be acceptable to you?

Regarding the combinations of continue_on_fail and hangup_after bridge, I'll
give that a try although I suspect it will result in less structured and
harder to understand markup...

Thanks!

Carlos

Anthony Minessale wrote:
> This suggestion violates the scope boundaries.
>  
> gateways are specific concept to mod_sofia so a <gateway> tag in <action>
> (part of agnostic xml dialplan)
> does not flow properly.
>  
> you can also use combinations of continue_on_fail and hangup_after bridge so
> you can
> just put each bridge statement in it's own action.
>  
>  
>  
> On Tue, Aug 25, 2009 at 9:06 AM, Carlos S. Antunes <csa at nowthor.com> wrote:
>  
>> Max,
>>  
>> I would like to see something similar too. For example, it would be
>> wonderful if one could specify multiple gateways to try like this or
>> something similar:
>>  
>> <action application="bridge">
>>    <!-- set some kind of *bridge-specific* parameter/variable -->
>>    <gateway order=1 data="sofia/gateway/gw-1/1$1"/>
>>    <gateway order=1 data="sofia/gateway/gw-2/1$1"/>
>>    <gateway order=2 data="sofia/gateway/gw-3/1$1">
>>        <!-- set some kind of *gw-specific* parameter/variable -->
>>    </gateway>
>>    <gateway order=3 data="sofia/gateway/gw-4/1$1"/>
>> </action>
>>  
>> One would be able to avoid the "[]" and "{}" hacks and combine
>> sequential and simultaneous trying of gateways.
>>  
>> What do the developers think of this?
>>  
>> Carlos
>>   
>>  
>> 
>> Max Ivanov wrote:
>>> > Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
>>> > 
>>> sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas1
>>> 13/1000
>>> > ,
>>> > the whole sting is tooo long, is there any shorter way to write same
>>> thing? Like
>>> > "sofia/gateway/panas*/1000" will try all gateways matching the pattern.
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>  
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  FreeSWITCH-users at lists.freeswitch.org
>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  http://www.freeswitch.org
>>  
>>  
>>  
>>  
>>  
>>  
>>  
>> -- 
>> Anthony Minessale II
>>  
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>  
>> AIM: anthm
>>  MSN:anthony_minessale at hotmail.com
>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>  
>> FreeSWITCH Developer Conference
>>  sip:888 at conference.freeswitch.org
>> <mailto:sip%3A888 at conference.freeswitch.org>
>>  iax:guest at conference.freeswitch.org/888
>> <http://iax:guest at conference.freeswitch.org/888>
>>  googletalk:conf+888 at conference.freeswitch.org
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>  
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org

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From sprice at gmail.com  Tue Aug 25 20:07:33 2009
From: sprice at gmail.com (SP)
Date: Tue, 25 Aug 2009 22:07:33 -0500
Subject: [Freeswitch-users] caller id on origination does not work
In-Reply-To: <5070fcbd0908251951j5a7163b1se346abcdf0664c94@mail.gmail.com>
References: <5070fcbd0908251951j5a7163b1se346abcdf0664c94@mail.gmail.com>
Message-ID: <7e2ac3270908252007g5dfb9390w806267550726b99a@mail.gmail.com>

You can try playing with this in your gateway profile
  <param name="caller-id-in-from" value="true"/>

Not sure what it'll do to your registration, give it a try

On Tue, Aug 25, 2009 at 21:51, Shameem Shiek <gshfreesw at gmail.com> wrote:

> I am setting the caller id like this in my ESL script:
>
> @con.sendRecv("api originate
> {origination_caller_id_number=15103245678}sofia/gateway/junctionnetworks/1#{@number}
> &park()")
>
> And the caller id comes out as all zeroes.  The sip trace shows the "from"
> as shown in the sofia status command.  This is output of sofia gateway
> status and my gateway is under
> /usr/local/freeswitch/conf/sip_profiles/external/junctionnetworks.xml .
> Also, adding the gateway under /conf/directory/default did not work as the
> wiki suugests. What I need to do for the caller_id to come through?
>
>
> sofia status gateway junctionnetworks
>
> =================================================================================================
> Name        junctionnetworks
> Scheme      Digest
> Realm       jnctn.net
> Username    username
> Password    yes
> From        <sip:username at jnctn.net <sip%3Ausername at jnctn.net>
> ;transport=udp>
> Contact     <sip:gw+junctionnetworks at 24.87.80.12:5080;transport=udp>
> Exten       899xxxxxxx
> To          sip:username at sip.jnctn.net <sip%3Ausername at sip.jnctn.net>
> Proxy       sip:sip.jnctn.net
> Context     public
> Expires     3600
> Freq        3600
> Ping        0
> PingFreq    0
> State       REGED
> Status      UP
> CallsIN     0
> CallsOUT    1
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Shannon
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From ivanov.maxim at gmail.com  Tue Aug 25 22:44:48 2009
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Wed, 26 Aug 2009 09:44:48 +0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <C6BA0F7E.5D757%krice@freeswitch.org>
References: <4A94A2E0.6040306@nowthor.com>
	<C6BA0F7E.5D757%krice@freeswitch.org>
Message-ID: <fbfa7b0c0908252244x65d19808r967a23332de6a0a0@mail.gmail.com>

>
> Continue_on_fail and hangup_after_bridge like tony pointed out are what you
> want if you don?t want to use the | delimiting ... I use these all the time
> with gateway counts > 10 just stacking additional actions for each bridge
> line

Let's imagine that I need to call 1000,1001,1002 via  gw1,gw2,gw3 by
choosing first available. How dialplan would look like?

continue_on_fail=True
hangup_after_bridge=True
bridge gw1/1000
bridge gw1/1001
bridge gw1/1002
bridge gw2/1000
bridge gw2/1001
bridge gw2/1002
bridge gw3/1000
bridge gw3/1001
bridge gw3/1002

Is it easy to understand? From my point of view it's not. Compare to this:
continue_on_fail=True
hangup_after_bridge=True
bridge ${get_avail_gw(gw1,gw2,gw3)}/1000
bridge ${get_avail_gw(gw1,gw2,gw3)}/1001
bridge ${get_avail_gw(gw1,gw2,gw3)}/1002


From mike at yes.net.ua  Wed Aug 26 00:51:27 2009
From: mike at yes.net.ua (Mike Tkachuk)
Date: Wed, 26 Aug 2009 10:51:27 +0300
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <fbfa7b0c0908252244x65d19808r967a23332de6a0a0@mail.gmail.com>
References: <4A94A2E0.6040306@nowthor.com>
	<C6BA0F7E.5D757%krice@freeswitch.org>
	<fbfa7b0c0908252244x65d19808r967a23332de6a0a0@mail.gmail.com>
Message-ID: <1236433376.20090826105127@yes.net.ua>

Hello Max,

 Don't improve that what's working well. ;)

Wednesday, August 26, 2009 8:44:48 AM, you wrote:

>>
>> Continue_on_fail and hangup_after_bridge like tony pointed out are what you
>> want if you don?t want to use the | delimiting ... I use these all the time
>> with gateway counts > 10 just stacking additional actions for each bridge
>> line

MI> Let's imagine that I need to call 1000,1001,1002 via  gw1,gw2,gw3 by
MI> choosing first available. How dialplan would look like?

MI> continue_on_fail=True
MI> hangup_after_bridge=True
MI> bridge gw1/1000
MI> bridge gw1/1001
MI> bridge gw1/1002
MI> bridge gw2/1000
MI> bridge gw2/1001
MI> bridge gw2/1002
MI> bridge gw3/1000
MI> bridge gw3/1001
MI> bridge gw3/1002

MI> Is it easy to understand? From my point of view it's not. Compare to this:
MI> continue_on_fail=True
MI> hangup_after_bridge=True
MI> bridge ${get_avail_gw(gw1,gw2,gw3)}/1000
MI> bridge ${get_avail_gw(gw1,gw2,gw3)}/1001
MI> bridge ${get_avail_gw(gw1,gw2,gw3)}/1002



--
Mike Tkachuk



From msc at freeswitch.org  Wed Aug 26 01:01:34 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 26 Aug 2009 01:01:34 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
Message-ID: <5DF610F9-6EB5-43CC-BB4F-3513C78F608F@freeswitch.org>


> I have looked at that but I am confused on which files need to be  
> edited. Since I have already installed in Wanpipe mode with the  
> Sangoma card I skipped straight to the Wanpipe section. It mentions  
> setting the [span wanpie PRI_1] etc in the openzap.conf then further  
> down it mentions editing the autoload_configs/openzap.conf.xml with  
> a completely different format of config.
>
>
>
> Are these the same file? If not where does the first one mentioned  
> reside?
>
>
These are two different files. The openzap.conf file is in the  
freeswitch/conf directory. It is for configuring OZ to talk to the  
Sangoma card. The openzap.conf.xml file is for configuring the actual  
openzap module in FreeSWITCH.

The specs you listed on the T1 you have indicate that it's a PRI.  
However, you need to know which protocol, like NI2 or DMS. Have you  
ever done a PRI turn up? Who is the carrier?

-MC
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From aep.lists at it46.se  Wed Aug 26 01:04:00 2009
From: aep.lists at it46.se (Alberto Escudero-Pascual (lists))
Date: Wed, 26 Aug 2009 10:04:00 +0200
Subject: [Freeswitch-users] Internal Profile sends RTP to external IP when
 using two network interfaces
Message-ID: <46b96649f81b0bfcb111928a55450211.squirrel@correo.nodo50.org>

Hi,

I have a FS box with two physical network interfaces. The internal
interface is hosting several internal phones. I have binded the internal
profile SIP/RTP/IP to the private interface. Phones registered correctly
but with User: 1000 at external.ip.address in sofia status profile internal

When I place a call between two internal phones, RTP traffic is send to
the external IP address of the FS box instead of the internal.

The SIP/SDP messages send from FS carry the external IP instead of the
internal.

The result is that no RTP media arrives to any of the phones.

/aep


-- 
Stopping junk mailers is good for the environment





From tculjaga at gmail.com  Wed Aug 26 01:08:57 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 26 Aug 2009 10:08:57 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com>
Message-ID: <65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com>

Hi Giovanny,

regarding ubuntu, did you mean 8.04 server or desktop ?


On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Definitely go for 64 bit OS.
>
> If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
> used both for development and for heavy duty production.
>
> Also Ubuntu 8.04 is good.
>
> Other versions/distros are less used by the community.
>
> Adding RAM and CPUs helps to scale up.
>
> -gm
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
>
> On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com>
> wrote:
> > Hey Giovanni,
> >
> > thanks for the tip... indeed the db files were heavily used regardless if
> i
> > started freeswitch with nosql option (freeswitch -nosql)... FS was not
> > writing anything into that files ... instead it was just accessing it....
> > This behaviour leads to a waste of 40% CPU time... waiting for other
> > processes (mainly disk access) to finish!!!
> >
> > I moved freeswitch/db/ to a ramdisk and the performance got a boost to
> 140
> > CPS with a CPU load of 80%. I was keeping the machine for a while (20 -
> 30
> > minutes) on that rate when i sow CPU suddenly went to 100% and FS
> becoming
> > irresponsive :).
> >
> >
> > What can be wrong?
> > What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
> > cross?
> > What fine tuning do we need in order to asure a long high load run?
> >
> >
> >
> > Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
> sense
> > to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
> will
> > FS perofomr drastically better 20%+ ?
> >
> >
> > Tihomir.
> >
> >
> > On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli <
> gmaruzz at celliax.org>
> > wrote:
> >>
> >> Maybe your load comes from disk access?
> >>
> >> Try putting the sql and log directories on a ramdisk.
> >>
> >> OTH,
> >>
> >> -giovanni
> >>
> >> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com>
> >> wrote:
> >> > Hello,
> >> >
> >> > i'm trying to use freeswitch as a redirecting server meaning FS has to
> >> > receive an INVITE and according to some rules it will redirect calls
> to
> >> > other destinations.
> >> >
> >> >
> >> > CALLING_USER                FREESWITCH
> SOMEWHERE
> >> >
> >> > INVITE ------------------------------->
> >> >            <------------------------------ 100 Trying
> >> >            <------------------------------ 302 Moved Temporary
> >> > ACK    ------------------------------->
> >> >
> >> >
> INVITE--------------------------------------------------------------------------------->
> >> >
> >> >
> >> >
> >> > Well, wverything works well except i have perfromance issues .... on
> my
> >> > HW
> >> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
> Temporary).
> >> > When
> >> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS
> i
> >> > see
> >> > "calls" being build up in FS and the delay begining to grow.
> >> >
> >> > When i observe the machine, load average is almost nothing (load
> >> > average:
> >> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
> >> > taking
> >> > most load... all others are just sitting there with 1-5 % CPU time.
> >> > This looks to me as FS handles 302 messages in a single thread?!?!
> >> >
> >> >
> >> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
> >> >
> >> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41,
> 0.61,
> >> > 0.60
> >> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
> >> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,  0.2%si,
> >> > 0.0%st
> >> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k
> buffers
> >> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k
> cached
> >> >
> >> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
> >> > COMMAND
> >> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
> >> > freeswitch
> >> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
> >> > freeswitch
> >> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
> >> > freeswitch
> >> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
> >> > freeswitch
> >> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
> >> > freeswitch
> >> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
> >> > freeswitch
> >> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
> >> > freeswitch
> >> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
> >> > freeswitch
> >> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
> >> > freeswitch
> >> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50
> freeswitch
> >> >
> >> >
> >> >
> >> > cat /proc/cpuinfo
> >> > processor       : 0
> >> > vendor_id       : GenuineIntel
> >> > cpu family      : 6
> >> > model           : 15
> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> >> > stepping        : 6
> >> > cpu MHz         : 2333.560
> >> > cache size      : 4096 KB
> >> > physical id     : 0
> >> > siblings        : 2
> >> > core id         : 0
> >> > cpu cores       : 2
> >> > apicid          : 0
> >> > initial apicid  : 0
> >> > fdiv_bug        : no
> >> > hlt_bug         : no
> >> > f00f_bug        : no
> >> > coma_bug        : no
> >> > fpu             : yes
> >> > fpu_exception   : yes
> >> > cpuid level     : 10
> >> > wp              : yes
> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
> >> > mca
> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
> ssse3
> >> > cx16
> >> > xtpr dca lahf_lm
> >> > bogomips        : 4670.78
> >> > clflush size    : 64
> >> > power management:
> >> >
> >> > processor       : 1
> >> > vendor_id       : GenuineIntel
> >> > cpu family      : 6
> >> > model           : 15
> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> >> > stepping        : 6
> >> > cpu MHz         : 2333.560
> >> > cache size      : 4096 KB
> >> > physical id     : 0
> >> > siblings        : 2
> >> > core id         : 1
> >> > cpu cores       : 2
> >> > apicid          : 1
> >> > initial apicid  : 1
> >> > fdiv_bug        : no
> >> > hlt_bug         : no
> >> > f00f_bug        : no
> >> > coma_bug        : no
> >> > fpu             : yes
> >> > fpu_exception   : yes
> >> > cpuid level     : 10
> >> > wp              : yes
> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
> >> > mca
> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
> ssse3
> >> > cx16
> >> > xtpr dca lahf_lm
> >> > bogomips        : 4666.82
> >> > clflush size    : 64
> >> > power management:
> >> >
> >> >
> >> >
> >> > uname -a
> >> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009
> i686
> >> > GNU/Linux
> >> >
> >> >
> >> >
> >> > Of course, i've tuned the machine up
> >> >
> >> > ulimit -c unlimited
> >> > ulimit -d unlimited
> >> > ulimit -f unlimited
> >> > ulimit -i unlimited
> >> > ulimit -n 999999
> >> > ulimit -q unlimited
> >> > ulimit -u unlimited
> >> > ulimit -v unlimited
> >> > ulimit -x unlimited
> >> > ulimit -s 240
> >> > ulimit -l unlimited
> >> > ulimit -a
> >> >
> >> >
> >> > Started FS with minimum modules but still 40 CPS seems to be the
> limit.
> >> >
> >> >
> >> > So, is there any way to improve performance?
> >> >
> >> >
> >> > Tihomir.
> >> >
> >> >
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at celliax.org  Wed Aug 26 01:25:19 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 26 Aug 2009 10:25:19 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com> 
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com> 
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com> 
	<7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com> 
	<65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com>
Message-ID: <7b197bef0908260125m56c3d681l26cb18a5e34aaa33@mail.gmail.com>

netbook remix


joking! Server 64bit :-)

-gm



On Wed, Aug 26, 2009 at 10:08 AM, Tihomir Culjaga<tculjaga at gmail.com> wrote:
> Hi Giovanny,
>
> regarding ubuntu, did you mean 8.04 server or desktop ?
>
>
> On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> Definitely go for 64 bit OS.
>>
>> If you want to be safe and sure, ?go for CentOS 5.2 64bit. Is the one
>> used both for development and for heavy duty production.
>>
>> Also Ubuntu 8.04 is good.
>>
>> Other versions/distros are less used by the community.
>>
>> Adding RAM and CPUs helps to scale up.
>>
>> -gm
>>
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>
>>
>>
>> On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com>
>> wrote:
>> > Hey Giovanni,
>> >
>> > thanks for the tip... indeed the db files were heavily used regardless
>> > if i
>> > started freeswitch with nosql option (freeswitch -nosql)... FS was not
>> > writing anything into that files ... instead it was just accessing
>> > it....
>> > This behaviour leads to a waste of 40% CPU time... waiting for other
>> > processes (mainly disk access) to finish!!!
>> >
>> > I moved freeswitch/db/ to a ramdisk and the performance got a boost to
>> > 140
>> > CPS with a CPU load of 80%. I was keeping the machine for a while (20 -
>> > 30
>> > minutes) on that rate when i sow CPU suddenly went to 100% and FS
>> > becoming
>> > irresponsive :).
>> >
>> >
>> > What can be wrong?
>> > What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
>> > cross?
>> > What fine tuning do we need in order to asure a long high load run?
>> >
>> >
>> >
>> > Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
>> > sense
>> > to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
>> > will
>> > FS perofomr drastically better 20%+ ?
>> >
>> >
>> > Tihomir.
>> >
>> >
>> > On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
>> > <gmaruzz at celliax.org>
>> > wrote:
>> >>
>> >> Maybe your load comes from disk access?
>> >>
>> >> Try putting the sql and log directories on a ramdisk.
>> >>
>> >> OTH,
>> >>
>> >> -giovanni
>> >>
>> >> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com>
>> >> wrote:
>> >> > Hello,
>> >> >
>> >> > i'm trying to use freeswitch as a redirecting server meaning FS has
>> >> > to
>> >> > receive an INVITE and according to some rules it will redirect calls
>> >> > to
>> >> > other destinations.
>> >> >
>> >> >
>> >> > CALLING_USER??????????????? FREESWITCH
>> >> > SOMEWHERE
>> >> >
>> >> > INVITE ------------------------------->
>> >> > ?????????? <------------------------------ 100 Trying
>> >> > ?????????? <------------------------------ 302 Moved Temporary
>> >> > ACK ?? ------------------------------->
>> >> >
>> >> >
>> >> > INVITE--------------------------------------------------------------------------------->
>> >> >
>> >> >
>> >> >
>> >> > Well, wverything works well except i have perfromance issues .... on
>> >> > my
>> >> > HW
>> >> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
>> >> > Temporary).
>> >> > When
>> >> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS
>> >> > i
>> >> > see
>> >> > "calls" being build up in FS and the delay begining to grow.
>> >> >
>> >> > When i observe the machine, load average is almost nothing (load
>> >> > average:
>> >> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
>> >> > taking
>> >> > most load... all others are just sitting there with 1-5 % CPU time.
>> >> > This looks to me as FS handles 302 messages in a single thread?!?!
>> >> >
>> >> >
>> >> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>> >> >
>> >> > top - 10:41:37 up 167 days, 20:42,? 3 users,? load average: 1.41,
>> >> > 0.61,
>> >> > 0.60
>> >> > Tasks:? 83 total,?? 2 running,? 81 sleeping,?? 0 stopped,?? 0 zombie
>> >> > Cpu(s): 25.3%us,? 1.5%sy,? 0.0%ni, 30.3%id, 42.7%wa,? 0.0%hi,
>> >> > 0.2%si,
>> >> > 0.0%st
>> >> > Mem:?? 2074520k total,?? 571244k used,? 1503276k free,?? 259604k
>> >> > buffers
>> >> > Swap:? 2650684k total,???? 3020k used,? 2647664k free,?? 153868k
>> >> > cached
>> >> >
>> >> > ? PID USER????? PR? NI? VIRT? RES? SHR S %CPU %MEM??? TIME+
>> >> > COMMAND
>> >> > ?4814 root????? 20?? 0 34188? 20m 3780 S?? 38? 1.0?? 3:10.29
>> >> > freeswitch
>> >> > ?4800 root????? 20?? 0 34188? 20m 3780 S??? 6? 1.0?? 0:08.26
>> >> > freeswitch
>> >> > ?4798 root????? 20?? 0 34188? 20m 3780 R??? 5? 1.0?? 0:24.46
>> >> > freeswitch
>> >> > ?4787 root????? 20?? 0 34188? 20m 3780 S??? 2? 1.0?? 0:11.24
>> >> > freeswitch
>> >> > ?4794 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.42
>> >> > freeswitch
>> >> > ?4803 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.74
>> >> > freeswitch
>> >> > ?4788 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.96
>> >> > freeswitch
>> >> > ?4804 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.64
>> >> > freeswitch
>> >> > ?4807 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.68
>> >> > freeswitch
>> >> > ?4811 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.50
>> >> > freeswitch
>> >> >
>> >> >
>> >> >
>> >> > cat /proc/cpuinfo
>> >> > processor?????? : 0
>> >> > vendor_id?????? : GenuineIntel
>> >> > cpu family????? : 6
>> >> > model?????????? : 15
>> >> > model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
>> >> > stepping??????? : 6
>> >> > cpu MHz???????? : 2333.560
>> >> > cache size????? : 4096 KB
>> >> > physical id???? : 0
>> >> > siblings??????? : 2
>> >> > core id???????? : 0
>> >> > cpu cores?????? : 2
>> >> > apicid????????? : 0
>> >> > initial apicid? : 0
>> >> > fdiv_bug??????? : no
>> >> > hlt_bug???????? : no
>> >> > f00f_bug??????? : no
>> >> > coma_bug??????? : no
>> >> > fpu???????????? : yes
>> >> > fpu_exception?? : yes
>> >> > cpuid level???? : 10
>> >> > wp????????????? : yes
>> >> > flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
>> >> > pge
>> >> > mca
>> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
>> >> > ssse3
>> >> > cx16
>> >> > xtpr dca lahf_lm
>> >> > bogomips??????? : 4670.78
>> >> > clflush size??? : 64
>> >> > power management:
>> >> >
>> >> > processor?????? : 1
>> >> > vendor_id?????? : GenuineIntel
>> >> > cpu family????? : 6
>> >> > model?????????? : 15
>> >> > model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
>> >> > stepping??????? : 6
>> >> > cpu MHz???????? : 2333.560
>> >> > cache size????? : 4096 KB
>> >> > physical id???? : 0
>> >> > siblings??????? : 2
>> >> > core id???????? : 1
>> >> > cpu cores?????? : 2
>> >> > apicid????????? : 1
>> >> > initial apicid? : 1
>> >> > fdiv_bug??????? : no
>> >> > hlt_bug???????? : no
>> >> > f00f_bug??????? : no
>> >> > coma_bug??????? : no
>> >> > fpu???????????? : yes
>> >> > fpu_exception?? : yes
>> >> > cpuid level???? : 10
>> >> > wp????????????? : yes
>> >> > flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
>> >> > pge
>> >> > mca
>> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
>> >> > ssse3
>> >> > cx16
>> >> > xtpr dca lahf_lm
>> >> > bogomips??????? : 4666.82
>> >> > clflush size??? : 64
>> >> > power management:
>> >> >
>> >> >
>> >> >
>> >> > uname -a
>> >> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009
>> >> > i686
>> >> > GNU/Linux
>> >> >
>> >> >
>> >> >
>> >> > Of course, i've tuned the machine up
>> >> >
>> >> > ulimit -c unlimited
>> >> > ulimit -d unlimited
>> >> > ulimit -f unlimited
>> >> > ulimit -i unlimited
>> >> > ulimit -n 999999
>> >> > ulimit -q unlimited
>> >> > ulimit -u unlimited
>> >> > ulimit -v unlimited
>> >> > ulimit -x unlimited
>> >> > ulimit -s 240
>> >> > ulimit -l unlimited
>> >> > ulimit -a
>> >> >
>> >> >
>> >> > Started FS with minimum modules but still 40 CPS seems to be the
>> >> > limit.
>> >> >
>> >> >
>> >> > So, is there any way to improve performance?
>> >> >
>> >> >
>> >> > Tihomir.
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From aep.lists at it46.se  Wed Aug 26 01:28:36 2009
From: aep.lists at it46.se (Alberto Escudero-Pascual (lists))
Date: Wed, 26 Aug 2009 10:28:36 +0200
Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
In-Reply-To: <191c3a030908251354l25206d0ch19836c8996f43e84@mail.gmail.com>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
	<191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
	<6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>
	<5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org>
	<45b2e91d7aa77afcf6430a14dde4b886.squirrel@correo.nodo50.org>
	<191c3a030908251354l25206d0ch19836c8996f43e84@mail.gmail.com>
Message-ID: <a41cc07ca7c2a0f3c51ceb9b37ea6bc6.squirrel@correo.nodo50.org>

I ran strace from freeswitch and from the command line. lame segfaults
when run from system FS.

The only obvious different i see is in the execve() /* XX vars */ apart
from the final Segfault

From
execve("/usr/local/freeswitch/bin/lame",
["/usr/local/freeswitch/bin/lame", "/tmp/foo.wav", "/tmp/foo.mp3", "-S"],
[/* 16 vars */]) = 0


>From FS
execve("/usr/local/freeswitch/bin/lame",
["/usr/local/freeswitch/bin/lame", "/tmp/foo.wav",
"/tmp/fooooooooooooooo.mp3", "-S"], [/* 14 vars */]) = 0

I am attaching the full straces in case they are of any help. Not sure if
this deserves a jira

/aep
-- 
Stopping junk mailers is good for the environment

> maybe it's writing some err to stderr that is being suppressed somehow
>
> On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists) <
> aep.lists at it46.se> wrote:
>
>> Hi Brian,
>>
>> >From the CLI>
>>
>> freeswitch at open46> system /usr/local/freeswitch/bin/lame -V2
>> /tmp/foo.wav
>> /tmp/foo.mp3 -S
>> 2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing
>> command:
>> /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S
>> API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
>> /tmp/foo.mp3 -S)] output:
>> +OK
>>
>> open46:/tmp# ls
>> foo.wav
>>
>>
>> and running the command from the command line:
>>
>>
>> open46:/tmp#  /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
>> /tmp/foo.mp3
>> -Sopen46:/tmp# ls
>> foo.mp3  foo.wav
>>
>>
>> If I do the same with lame397
>>
>> freeswitch at open46> system /usr/local/freeswitch/bin/lame397 -V2
>> /tmp/foo.wav /tmp/foo.mp3 -S
>> 2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing
>> command:
>> /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S
>> API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav
>> /tmp/foo.mp3 -S)] output:
>> +OK
>>
>> open46:/tmp# ls
>> foo.mp3  foo.wav
>>
>>
>> Highly paranormal! Sorry for hijacking the previous thread.
>>
>> /aep
>>
>> --
>> Stopping junk mailers is good for the environment
>>
>> > Try running it at the CLI and see if you see any errors.  Also please
>> > do not hijack threads.  The original thread "[Freeswitch-users] XML-
>> > RPC on different ip than 0.0.0.0" which was hijacked by clicking
>> > reply, changing the subject and clicking send.  Please in the future
>> > do not do that as it clutters up the threading and could get your
>> > query lost in the noise.
>> >
>> > Thanks,
>> > Brian
>> >
>> > On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote:
>> >
>> >> Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
>> >> appears, if I use version 3.97 (older version), it does!.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Wed Aug 26 01:29:07 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 26 Aug 2009 10:29:07 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <7b197bef0908260125m56c3d681l26cb18a5e34aaa33@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com>
	<65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com>
	<7b197bef0908260125m56c3d681l26cb18a5e34aaa33@mail.gmail.com>
Message-ID: <65d96fc80908260129xebbf460raffaa38c72c3281a@mail.gmail.com>

intanto e il centos che si sta installando :)

grazie.

T.

On Wed, Aug 26, 2009 at 10:25 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> netbook remix
>
>
> joking! Server 64bit :-)
>
> -gm
>
>
>
> On Wed, Aug 26, 2009 at 10:08 AM, Tihomir Culjaga<tculjaga at gmail.com>
> wrote:
> > Hi Giovanny,
> >
> > regarding ubuntu, did you mean 8.04 server or desktop ?
> >
> >
> > On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli <
> gmaruzz at celliax.org>
> > wrote:
> >>
> >> Definitely go for 64 bit OS.
> >>
> >> If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
> >> used both for development and for heavy duty production.
> >>
> >> Also Ubuntu 8.04 is good.
> >>
> >> Other versions/distros are less used by the community.
> >>
> >> Adding RAM and CPUs helps to scale up.
> >>
> >> -gm
> >>
> >>
> >>
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >>
> >>
> >>
> >> On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com>
> >> wrote:
> >> > Hey Giovanni,
> >> >
> >> > thanks for the tip... indeed the db files were heavily used regardless
> >> > if i
> >> > started freeswitch with nosql option (freeswitch -nosql)... FS was not
> >> > writing anything into that files ... instead it was just accessing
> >> > it....
> >> > This behaviour leads to a waste of 40% CPU time... waiting for other
> >> > processes (mainly disk access) to finish!!!
> >> >
> >> > I moved freeswitch/db/ to a ramdisk and the performance got a boost to
> >> > 140
> >> > CPS with a CPU load of 80%. I was keeping the machine for a while (20
> -
> >> > 30
> >> > minutes) on that rate when i sow CPU suddenly went to 100% and FS
> >> > becoming
> >> > irresponsive :).
> >> >
> >> >
> >> > What can be wrong?
> >> > What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
> >> > cross?
> >> > What fine tuning do we need in order to asure a long high load run?
> >> >
> >> >
> >> >
> >> > Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
> >> > sense
> >> > to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
> >> > will
> >> > FS perofomr drastically better 20%+ ?
> >> >
> >> >
> >> > Tihomir.
> >> >
> >> >
> >> > On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
> >> > <gmaruzz at celliax.org>
> >> > wrote:
> >> >>
> >> >> Maybe your load comes from disk access?
> >> >>
> >> >> Try putting the sql and log directories on a ramdisk.
> >> >>
> >> >> OTH,
> >> >>
> >> >> -giovanni
> >> >>
> >> >> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com
> >
> >> >> wrote:
> >> >> > Hello,
> >> >> >
> >> >> > i'm trying to use freeswitch as a redirecting server meaning FS has
> >> >> > to
> >> >> > receive an INVITE and according to some rules it will redirect
> calls
> >> >> > to
> >> >> > other destinations.
> >> >> >
> >> >> >
> >> >> > CALLING_USER                FREESWITCH
> >> >> > SOMEWHERE
> >> >> >
> >> >> > INVITE ------------------------------->
> >> >> >            <------------------------------ 100 Trying
> >> >> >            <------------------------------ 302 Moved Temporary
> >> >> > ACK    ------------------------------->
> >> >> >
> >> >> >
> >> >> >
> INVITE--------------------------------------------------------------------------------->
> >> >> >
> >> >> >
> >> >> >
> >> >> > Well, wverything works well except i have perfromance issues ....
> on
> >> >> > my
> >> >> > HW
> >> >> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
> >> >> > Temporary).
> >> >> > When
> >> >> > i increase the rate, FS starts delaying 302 response. Right at 50
> CPS
> >> >> > i
> >> >> > see
> >> >> > "calls" being build up in FS and the delay begining to grow.
> >> >> >
> >> >> > When i observe the machine, load average is almost nothing (load
> >> >> > average:
> >> >> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
> >> >> > taking
> >> >> > most load... all others are just sitting there with 1-5 % CPU time.
> >> >> > This looks to me as FS handles 302 messages in a single thread?!?!
> >> >> >
> >> >> >
> >> >> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
> >> >> >
> >> >> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41,
> >> >> > 0.61,
> >> >> > 0.60
> >> >> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0
> zombie
> >> >> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,
> >> >> > 0.2%si,
> >> >> > 0.0%st
> >> >> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k
> >> >> > buffers
> >> >> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k
> >> >> > cached
> >> >> >
> >> >> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
> >> >> > COMMAND
> >> >> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
> >> >> > freeswitch
> >> >> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
> >> >> > freeswitch
> >> >> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
> >> >> > freeswitch
> >> >> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
> >> >> > freeswitch
> >> >> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
> >> >> > freeswitch
> >> >> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
> >> >> > freeswitch
> >> >> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
> >> >> > freeswitch
> >> >> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
> >> >> > freeswitch
> >> >> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
> >> >> > freeswitch
> >> >> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50
> >> >> > freeswitch
> >> >> >
> >> >> >
> >> >> >
> >> >> > cat /proc/cpuinfo
> >> >> > processor       : 0
> >> >> > vendor_id       : GenuineIntel
> >> >> > cpu family      : 6
> >> >> > model           : 15
> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> >> >> > stepping        : 6
> >> >> > cpu MHz         : 2333.560
> >> >> > cache size      : 4096 KB
> >> >> > physical id     : 0
> >> >> > siblings        : 2
> >> >> > core id         : 0
> >> >> > cpu cores       : 2
> >> >> > apicid          : 0
> >> >> > initial apicid  : 0
> >> >> > fdiv_bug        : no
> >> >> > hlt_bug         : no
> >> >> > f00f_bug        : no
> >> >> > coma_bug        : no
> >> >> > fpu             : yes
> >> >> > fpu_exception   : yes
> >> >> > cpuid level     : 10
> >> >> > wp              : yes
> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
> >> >> > pge
> >> >> > mca
> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
> >> >> > ssse3
> >> >> > cx16
> >> >> > xtpr dca lahf_lm
> >> >> > bogomips        : 4670.78
> >> >> > clflush size    : 64
> >> >> > power management:
> >> >> >
> >> >> > processor       : 1
> >> >> > vendor_id       : GenuineIntel
> >> >> > cpu family      : 6
> >> >> > model           : 15
> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> >> >> > stepping        : 6
> >> >> > cpu MHz         : 2333.560
> >> >> > cache size      : 4096 KB
> >> >> > physical id     : 0
> >> >> > siblings        : 2
> >> >> > core id         : 1
> >> >> > cpu cores       : 2
> >> >> > apicid          : 1
> >> >> > initial apicid  : 1
> >> >> > fdiv_bug        : no
> >> >> > hlt_bug         : no
> >> >> > f00f_bug        : no
> >> >> > coma_bug        : no
> >> >> > fpu             : yes
> >> >> > fpu_exception   : yes
> >> >> > cpuid level     : 10
> >> >> > wp              : yes
> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
> >> >> > pge
> >> >> > mca
> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
> >> >> > ssse3
> >> >> > cx16
> >> >> > xtpr dca lahf_lm
> >> >> > bogomips        : 4666.82
> >> >> > clflush size    : 64
> >> >> > power management:
> >> >> >
> >> >> >
> >> >> >
> >> >> > uname -a
> >> >> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009
> >> >> > i686
> >> >> > GNU/Linux
> >> >> >
> >> >> >
> >> >> >
> >> >> > Of course, i've tuned the machine up
> >> >> >
> >> >> > ulimit -c unlimited
> >> >> > ulimit -d unlimited
> >> >> > ulimit -f unlimited
> >> >> > ulimit -i unlimited
> >> >> > ulimit -n 999999
> >> >> > ulimit -q unlimited
> >> >> > ulimit -u unlimited
> >> >> > ulimit -v unlimited
> >> >> > ulimit -x unlimited
> >> >> > ulimit -s 240
> >> >> > ulimit -l unlimited
> >> >> > ulimit -a
> >> >> >
> >> >> >
> >> >> > Started FS with minimum modules but still 40 CPS seems to be the
> >> >> > limit.
> >> >> >
> >> >> >
> >> >> > So, is there any way to improve performance?
> >> >> >
> >> >> >
> >> >> > Tihomir.
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed Aug 26 01:44:42 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 26 Aug 2009 04:44:42 -0400
Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
In-Reply-To: <a41cc07ca7c2a0f3c51ceb9b37ea6bc6.squirrel@correo.nodo50.org>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
	<191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
	<6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>
	<5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org>
	<45b2e91d7aa77afcf6430a14dde4b886.squirrel@correo.nodo50.org>
	<191c3a030908251354l25206d0ch19836c8996f43e84@mail.gmail.com>
	<a41cc07ca7c2a0f3c51ceb9b37ea6bc6.squirrel@correo.nodo50.org>
Message-ID: <5896508D-3BA9-4B9A-A0DF-E8BE1EA69CB4@jerris.com>

Running out of stack space?  The stack space we run freeswitch in is  
fairly small.  Programs launched from the freeswitch process inherit  
this.

Mike

On Aug 26, 2009, at 4:28 AM, Alberto Escudero-Pascual (lists) wrote:

> I ran strace from freeswitch and from the command line. lame segfaults
> when run from system FS.
>
> The only obvious different i see is in the execve() /* XX vars */  
> apart
> from the final Segfault
>
> From
> execve("/usr/local/freeswitch/bin/lame",
> ["/usr/local/freeswitch/bin/lame", "/tmp/foo.wav", "/tmp/foo.mp3", "- 
> S"],
> [/* 16 vars */]) = 0
>
>
>> From FS
> execve("/usr/local/freeswitch/bin/lame",
> ["/usr/local/freeswitch/bin/lame", "/tmp/foo.wav",
> "/tmp/fooooooooooooooo.mp3", "-S"], [/* 14 vars */]) = 0
>
> I am attaching the full straces in case they are of any help. Not  
> sure if
> this deserves a jira
>
> /aep
> -- 
> Stopping junk mailers is good for the environment
>
>> maybe it's writing some err to stderr that is being suppressed  
>> somehow
>>
>> On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists) <
>> aep.lists at it46.se> wrote:
>>
>>> Hi Brian,
>>>
>>>> From the CLI>
>>>
>>> freeswitch at open46> system /usr/local/freeswitch/bin/lame -V2
>>> /tmp/foo.wav
>>> /tmp/foo.mp3 -S
>>> 2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing
>>> command:
>>> /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S
>>> API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
>>> /tmp/foo.mp3 -S)] output:
>>> +OK
>>>
>>> open46:/tmp# ls
>>> foo.wav
>>>
>>>
>>> and running the command from the command line:
>>>
>>>
>>> open46:/tmp#  /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
>>> /tmp/foo.mp3
>>> -Sopen46:/tmp# ls
>>> foo.mp3  foo.wav
>>>
>>>
>>> If I do the same with lame397
>>>
>>> freeswitch at open46> system /usr/local/freeswitch/bin/lame397 -V2
>>> /tmp/foo.wav /tmp/foo.mp3 -S
>>> 2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing
>>> command:
>>> /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S
>>> API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav
>>> /tmp/foo.mp3 -S)] output:
>>> +OK
>>>
>>> open46:/tmp# ls
>>> foo.mp3  foo.wav
>>>
>>>
>>> Highly paranormal! Sorry for hijacking the previous thread.
>>>
>>> /aep
>>>
>>> --
>>> Stopping junk mailers is good for the environment
>>>
>>>> Try running it at the CLI and see if you see any errors.  Also  
>>>> please
>>>> do not hijack threads.  The original thread "[Freeswitch-users]  
>>>> XML-
>>>> RPC on different ip than 0.0.0.0" which was hijacked by clicking
>>>> reply, changing the subject and clicking send.  Please in the  
>>>> future
>>>> do not do that as it clutters up the threading and could get your
>>>> query lost in the noise.
>>>>
>>>> Thanks,
>>>> Brian
>>>>
>>>> On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists)  
>>>> wrote:
>>>>
>>>>> Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
>>>>> appears, if I use version 3.97 (older version), it does!.
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN 
>> %3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL 
>> %3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip 
>> %3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf 
>> %2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
> <lame_strace.txt>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From raffaele.p.guidi at gmail.com  Wed Aug 26 03:12:31 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Wed, 26 Aug 2009 12:12:31 +0200
Subject: [Freeswitch-users] caller id on origination does not work
In-Reply-To: <5070fcbd0908251951j5a7163b1se346abcdf0664c94@mail.gmail.com>
References: <5070fcbd0908251951j5a7163b1se346abcdf0664c94@mail.gmail.com>
Message-ID: <b0a8792f0908260312v6a84d66bh3d30e582b347e6ab@mail.gmail.com>

I had the same problem with the Sipek2 Softphone (based on the pjsip stack)
and I solved using the sip_contact_user variable like this:

api originate
{sip_contact_user=15103245678}sofia/gateway/junctionnetworks/1#{@number}
&park()


Hope this helps.
Ciao,
   Raffaele

On Wed, Aug 26, 2009 at 04:51, Shameem Shiek <gshfreesw at gmail.com> wrote:

> I am setting the caller id like this in my ESL script:
>
> @con.sendRecv("api originate
> {origination_caller_id_number=15103245678}sofia/gateway/junctionnetworks/1#{@number}
> &park()")
>
> And the caller id comes out as all zeroes.  The sip trace shows the "from"
> as shown in the sofia status command.  This is output of sofia gateway
> status and my gateway is under
> /usr/local/freeswitch/conf/sip_profiles/external/junctionnetworks.xml .
> Also, adding the gateway under /conf/directory/default did not work as the
> wiki suugests. What I need to do for the caller_id to come through?
>
>
> sofia status gateway junctionnetworks
>
> =================================================================================================
> Name        junctionnetworks
> Scheme      Digest
> Realm       jnctn.net
> Username    username
> Password    yes
> From        <sip:username at jnctn.net <sip%3Ausername at jnctn.net>
> ;transport=udp>
> Contact     <sip:gw+junctionnetworks at 24.87.80.12:5080;transport=udp>
> Exten       899xxxxxxx
> To          sip:username at sip.jnctn.net <sip%3Ausername at sip.jnctn.net>
> Proxy       sip:sip.jnctn.net
> Context     public
> Expires     3600
> Freq        3600
> Ping        0
> PingFreq    0
> State       REGED
> Status      UP
> CallsIN     0
> CallsOUT    1
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From odermann at googlemail.com  Wed Aug 26 05:59:33 2009
From: odermann at googlemail.com (Dennis)
Date: Wed, 26 Aug 2009 14:59:33 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
Message-ID: <5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>

> You have a NAT issue.

as i wrote, we are quite sure, that this is a nat problem. but we have
no idea how we could fix this. there must be a reason, why the hangup
always comes after 120 seconds.


>    <!--RTP port range -->
>    <param name="rtp-start-port" value="10000"/>
>    <param name="rtp-end-port" value="32767"/>

we tested that, but this does not change anything (nothing becomes
worse or better).


> param name="session-timeout" value="120"

this line is commented in our profile and we believe, that this has
nothing to do with our problem.


to tell a little bit more about our problem:

under http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios#Scenario_2
you can find the scenario, which is exactly like ours. we played with
all the mentioned settings, but it does not work.

to mention one thing: if we connect with xlite (softphone), we do not
have any problems. but we have snom-voip-phones and with them we have
the problems.

the big question is: wo or what might cause/trigger the hangup? is it
freeswitch or something else? we have a firewall (IPCop) - might there
be a setting, which needs to be set, to avoid theses problems?


kind regards
dennis


From gmaruzz at celliax.org  Wed Aug 26 06:08:38 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 26 Aug 2009 15:08:38 +0200
Subject: [Freeswitch-users] HOW-TO: being on "trunk" of FreePBX,
	starting from the ISO
Message-ID: <7b197bef0908260608l2c3f454cn8c2b8a12bc3d30ce@mail.gmail.com>

Instructions for being on "trunk" of FreePBX, starting from the ISO
(cut and paste to the ssh console after ISO install):

/etc/init.d/httpd stop
cd /var/www/html
mv freepbx freepbx-originale
svn co http://www.freepbx.org/v3/svn/trunk/ freepbx
chown -R apache.apache freepbx
ln -s freepbx  freepbx-v3
cd freepbx-v3/
ln -s freepbx  freepbx-v3
/etc/init.d/httpd start

then browse to:

http://192.168.0.12/freepbx-v3/index.php/installer

it will work! :-)


Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From odermann at googlemail.com  Wed Aug 26 06:18:34 2009
From: odermann at googlemail.com (Dennis)
Date: Wed, 26 Aug 2009 15:18:34 +0200
Subject: [Freeswitch-users] Question about presence
Message-ID: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>

hi,

we have set manage-presence = true to see, who is talking on the phone
and who is free. everyone here has his own snom-voip phone with 12
led-lights.

incoming calls are handled in groups. this means, if someone is
ringing, all voip-phones of one group are ringing and one can see all
led-lights flashing. if someone answers the phonecall, the led of this
person (p1) is on on everybodys phone, so everybody can see, that the
person is talking.
now, if another phonecall comes in, all led-lights are flashing -
including the led of the person who is talking. if another person (p2)
answers the second phonecall, the led of p2 is on, but the led of p1
is off, although p1 is still talking.

is there something we can do about this?


thanks & kind regards
dennis


From brian at freeswitch.org  Wed Aug 26 06:54:40 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 Aug 2009 08:54:40 -0500
Subject: [Freeswitch-users] Internal Profile sends RTP to external IP
	when using two network interfaces
In-Reply-To: <46b96649f81b0bfcb111928a55450211.squirrel@correo.nodo50.org>
References: <46b96649f81b0bfcb111928a55450211.squirrel@correo.nodo50.org>
Message-ID: <3694FC90-3FD1-4F36-8CA9-A3C18EBDEF34@freeswitch.org>

First off you can't bind one profile to two interfaces you have to  
launch two sofia profiles, one for each IP.  Secondly if you're doing  
things like this you'll have to refer to the in tree internal.xml.   
Third can you outline the network topology a little bit more?  Is nat  
involved?

/b

On Aug 26, 2009, at 3:04 AM, Alberto Escudero-Pascual (lists) wrote:

> Hi,
>
> I have a FS box with two physical network interfaces. The internal
> interface is hosting several internal phones. I have binded the  
> internal
> profile SIP/RTP/IP to the private interface. Phones registered  
> correctly
> but with User: 1000 at external.ip.address in sofia status profile  
> internal
>
> When I place a call between two internal phones, RTP traffic is send  
> to
> the external IP address of the FS box instead of the internal.
>
> The SIP/SDP messages send from FS carry the external IP instead of the
> internal.
>
> The result is that no RTP media arrives to any of the phones.
>
> /aep

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From kadantsev.d at gmail.com  Wed Aug 26 04:53:45 2009
From: kadantsev.d at gmail.com (Dmitry Kadantsev)
Date: Wed, 26 Aug 2009 13:53:45 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908260129xebbf460raffaa38c72c3281a@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com> 
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com> 
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com> 
	<7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com> 
	<65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com> 
	<7b197bef0908260125m56c3d681l26cb18a5e34aaa33@mail.gmail.com> 
	<65d96fc80908260129xebbf460raffaa38c72c3281a@mail.gmail.com>
Message-ID: <681a20520908260453q57877b1eg8f0949061cd129ee@mail.gmail.com>

Hi all,

is there same situation with FS for Windows? I mean 64bit is more preferable
than 32bit, isn't it?

Any performance test on Win 32/64 were done?

--
Best regards,
Dmitry Kadantsev


On Wed, Aug 26, 2009 at 10:29 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> intanto e il centos che si sta installando :)
>
> grazie.
>
> T.
>
>
> On Wed, Aug 26, 2009 at 10:25 AM, Giovanni Maruzzelli <gmaruzz at celliax.org
> > wrote:
>
>> netbook remix
>>
>>
>> joking! Server 64bit :-)
>>
>> -gm
>>
>>
>>
>> On Wed, Aug 26, 2009 at 10:08 AM, Tihomir Culjaga<tculjaga at gmail.com>
>> wrote:
>> > Hi Giovanny,
>> >
>> > regarding ubuntu, did you mean 8.04 server or desktop ?
>> >
>> >
>> > On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli <
>> gmaruzz at celliax.org>
>> > wrote:
>> >>
>> >> Definitely go for 64 bit OS.
>> >>
>> >> If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
>> >> used both for development and for heavy duty production.
>> >>
>> >> Also Ubuntu 8.04 is good.
>> >>
>> >> Other versions/distros are less used by the community.
>> >>
>> >> Adding RAM and CPUs helps to scale up.
>> >>
>> >> -gm
>> >>
>> >>
>> >>
>> >> Sincerely,
>> >>
>> >> Giovanni Maruzzelli
>> >> Cell : +39-347-2665618
>> >>
>> >>
>> >>
>> >>
>> >> On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com>
>> >> wrote:
>> >> > Hey Giovanni,
>> >> >
>> >> > thanks for the tip... indeed the db files were heavily used
>> regardless
>> >> > if i
>> >> > started freeswitch with nosql option (freeswitch -nosql)... FS was
>> not
>> >> > writing anything into that files ... instead it was just accessing
>> >> > it....
>> >> > This behaviour leads to a waste of 40% CPU time... waiting for other
>> >> > processes (mainly disk access) to finish!!!
>> >> >
>> >> > I moved freeswitch/db/ to a ramdisk and the performance got a boost
>> to
>> >> > 140
>> >> > CPS with a CPU load of 80%. I was keeping the machine for a while (20
>> -
>> >> > 30
>> >> > minutes) on that rate when i sow CPU suddenly went to 100% and FS
>> >> > becoming
>> >> > irresponsive :).
>> >> >
>> >> >
>> >> > What can be wrong?
>> >> > What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should
>> not
>> >> > cross?
>> >> > What fine tuning do we need in order to asure a long high load run?
>> >> >
>> >> >
>> >> >
>> >> > Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it
>> have
>> >> > sense
>> >> > to move my OS to 64 bit? ... will FS gain more preformance ?... I
>> mean
>> >> > will
>> >> > FS perofomr drastically better 20%+ ?
>> >> >
>> >> >
>> >> > Tihomir.
>> >> >
>> >> >
>> >> > On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
>> >> > <gmaruzz at celliax.org>
>> >> > wrote:
>> >> >>
>> >> >> Maybe your load comes from disk access?
>> >> >>
>> >> >> Try putting the sql and log directories on a ramdisk.
>> >> >>
>> >> >> OTH,
>> >> >>
>> >> >> -giovanni
>> >> >>
>> >> >> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<
>> tculjaga at gmail.com>
>> >> >> wrote:
>> >> >> > Hello,
>> >> >> >
>> >> >> > i'm trying to use freeswitch as a redirecting server meaning FS
>> has
>> >> >> > to
>> >> >> > receive an INVITE and according to some rules it will redirect
>> calls
>> >> >> > to
>> >> >> > other destinations.
>> >> >> >
>> >> >> >
>> >> >> > CALLING_USER                FREESWITCH
>> >> >> > SOMEWHERE
>> >> >> >
>> >> >> > INVITE ------------------------------->
>> >> >> >            <------------------------------ 100 Trying
>> >> >> >            <------------------------------ 302 Moved Temporary
>> >> >> > ACK    ------------------------------->
>> >> >> >
>> >> >> >
>> >> >> >
>> INVITE--------------------------------------------------------------------------------->
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > Well, wverything works well except i have perfromance issues ....
>> on
>> >> >> > my
>> >> >> > HW
>> >> >> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
>> >> >> > Temporary).
>> >> >> > When
>> >> >> > i increase the rate, FS starts delaying 302 response. Right at 50
>> CPS
>> >> >> > i
>> >> >> > see
>> >> >> > "calls" being build up in FS and the delay begining to grow.
>> >> >> >
>> >> >> > When i observe the machine, load average is almost nothing (load
>> >> >> > average:
>> >> >> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one
>> thread
>> >> >> > taking
>> >> >> > most load... all others are just sitting there with 1-5 % CPU
>> time.
>> >> >> > This looks to me as FS handles 302 messages in a single thread?!?!
>> >> >> >
>> >> >> >
>> >> >> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>> >> >> >
>> >> >> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41,
>> >> >> > 0.61,
>> >> >> > 0.60
>> >> >> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0
>> zombie
>> >> >> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,
>> >> >> > 0.2%si,
>> >> >> > 0.0%st
>> >> >> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k
>> >> >> > buffers
>> >> >> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k
>> >> >> > cached
>> >> >> >
>> >> >> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
>> >> >> > COMMAND
>> >> >> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
>> >> >> > freeswitch
>> >> >> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
>> >> >> > freeswitch
>> >> >> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
>> >> >> > freeswitch
>> >> >> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
>> >> >> > freeswitch
>> >> >> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
>> >> >> > freeswitch
>> >> >> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
>> >> >> > freeswitch
>> >> >> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
>> >> >> > freeswitch
>> >> >> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
>> >> >> > freeswitch
>> >> >> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
>> >> >> > freeswitch
>> >> >> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50
>> >> >> > freeswitch
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > cat /proc/cpuinfo
>> >> >> > processor       : 0
>> >> >> > vendor_id       : GenuineIntel
>> >> >> > cpu family      : 6
>> >> >> > model           : 15
>> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
>> >> >> > stepping        : 6
>> >> >> > cpu MHz         : 2333.560
>> >> >> > cache size      : 4096 KB
>> >> >> > physical id     : 0
>> >> >> > siblings        : 2
>> >> >> > core id         : 0
>> >> >> > cpu cores       : 2
>> >> >> > apicid          : 0
>> >> >> > initial apicid  : 0
>> >> >> > fdiv_bug        : no
>> >> >> > hlt_bug         : no
>> >> >> > f00f_bug        : no
>> >> >> > coma_bug        : no
>> >> >> > fpu             : yes
>> >> >> > fpu_exception   : yes
>> >> >> > cpuid level     : 10
>> >> >> > wp              : yes
>> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
>> >> >> > pge
>> >> >> > mca
>> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
>> >> >> > ssse3
>> >> >> > cx16
>> >> >> > xtpr dca lahf_lm
>> >> >> > bogomips        : 4670.78
>> >> >> > clflush size    : 64
>> >> >> > power management:
>> >> >> >
>> >> >> > processor       : 1
>> >> >> > vendor_id       : GenuineIntel
>> >> >> > cpu family      : 6
>> >> >> > model           : 15
>> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
>> >> >> > stepping        : 6
>> >> >> > cpu MHz         : 2333.560
>> >> >> > cache size      : 4096 KB
>> >> >> > physical id     : 0
>> >> >> > siblings        : 2
>> >> >> > core id         : 1
>> >> >> > cpu cores       : 2
>> >> >> > apicid          : 1
>> >> >> > initial apicid  : 1
>> >> >> > fdiv_bug        : no
>> >> >> > hlt_bug         : no
>> >> >> > f00f_bug        : no
>> >> >> > coma_bug        : no
>> >> >> > fpu             : yes
>> >> >> > fpu_exception   : yes
>> >> >> > cpuid level     : 10
>> >> >> > wp              : yes
>> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
>> >> >> > pge
>> >> >> > mca
>> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
>> >> >> > ssse3
>> >> >> > cx16
>> >> >> > xtpr dca lahf_lm
>> >> >> > bogomips        : 4666.82
>> >> >> > clflush size    : 64
>> >> >> > power management:
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > uname -a
>> >> >> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC
>> 2009
>> >> >> > i686
>> >> >> > GNU/Linux
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > Of course, i've tuned the machine up
>> >> >> >
>> >> >> > ulimit -c unlimited
>> >> >> > ulimit -d unlimited
>> >> >> > ulimit -f unlimited
>> >> >> > ulimit -i unlimited
>> >> >> > ulimit -n 999999
>> >> >> > ulimit -q unlimited
>> >> >> > ulimit -u unlimited
>> >> >> > ulimit -v unlimited
>> >> >> > ulimit -x unlimited
>> >> >> > ulimit -s 240
>> >> >> > ulimit -l unlimited
>> >> >> > ulimit -a
>> >> >> >
>> >> >> >
>> >> >> > Started FS with minimum modules but still 40 CPS seems to be the
>> >> >> > limit.
>> >> >> >
>> >> >> >
>> >> >> > So, is there any way to improve performance?
>> >> >> >
>> >> >> >
>> >> >> > Tihomir.
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From se02005-mec at yahoo.com  Wed Aug 26 06:25:51 2009
From: se02005-mec at yahoo.com (Merle J. Ebbert)
Date: Wed, 26 Aug 2009 06:25:51 -0700
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
Message-ID: <4A9537DF.8070609@yahoo.com>


Hi,

I'm trying to avoid taking up a lot of peoples valuable time.

SIP & FS have  brought some ideas for some commercial products but I 
need to know where to start.

Having once written a proprietary DOS & helped with writing a RTOS, I 
consider myself capable
of learning.  I (we) just need to know where to start to come up to 
speed rapidly.

Is there a FreeSWITCH tutorial available?
Should someone new start with Asterisk and then possibly move to FS?

Thanks,
Merle



From harry at vangberg.name  Wed Aug 26 06:54:41 2009
From: harry at vangberg.name (Harry Vangberg)
Date: Wed, 26 Aug 2009 15:54:41 +0200
Subject: [Freeswitch-users] bind_meta_app: "already broadcasting..broadcast
	aborted" if bound via event socket
Message-ID: <74d41a3d0908260654x52bf9d94g124ceaa27f6df6bb@mail.gmail.com>

Hello.

After pestering the IRC channel for a few days with useless
information, I think I've finally made a breakthrough, but I have no
idea why it works this way.

Basically, what I want to achieve is: A calls in and is bridged to B
(1234). If B presses *1, he should be taken out, and A be bridged to C
(8888) instead. Pretty simple.

With the following dialplan it works very nicely:

    <extension name="ivr">
      <condition field="destination_number" expression="^(\d+)$">
        <action application="set" data="bypass_media=false"/>
        <action application="answer"/>
        <action application="bind_meta_app" data="1 b a
bridge::sofia/gateway/secretgw/8888"/>
        <action application="bridge" data="sofia/gateway/secretgw/1234" />
      </condition>
    </extension>

On the other hand, if I try to replicate the above via an outbound
event socket it fails with a: "already broadcasting...broadcast
aborted". In this case my dialplan XML looks like this:

    <extension name="ff-ivr">
      <condition field="destination_number" expression="^(.*)$">
        <action application="set" data="bypass_media=false"/>
        <action application="socket" data="127.0.0.1:8084 full"/>
      </condition>
    </extension>

And this is the event socket communication (in reality I use a Ruby
framework, do a bunch of database calls etc., but for the sake of
clarity I'm mimicking it with a netcat session):

--------------------------------
$ nc -v -l 127.0.0.1 8084
connect

Event-Name: CHANNEL_DATA
Core-UUID: ba78636a-9241-11de-ac94-b736da546252
FreeSWITCH-Hostname: ip-10-226-231-225
FreeSWITCH-IPv4: 10.226.231.225
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-26%2013%3A43%3A20
Event-Date-GMT: Wed,%2026%20Aug%202009%2013%3A43%3A20%20GMT
Event-Date-Timestamp: 1251294200244673
Event-Calling-File: mod_event_socket.c
Event-Calling-Function: parse_command
Event-Calling-Line-Number: 1482
Channel-Username: hemmeligt
Channel-Dialplan: XML
Channel-Caller-ID-Name: hemmeligt
Channel-Caller-ID-Number: hemmeligt
Channel-Network-Addr: 129.142.224.250
Channel-Destination-Number: 77344541
Channel-Unique-ID: 681102bc-9246-11de-ac94-b736da546252
Channel-Source: mod_sofia
Channel-Context: public
Channel-Channel-Name: sofia/external/hemmeligt%40129.142.224.250
Channel-Profile-Index: 1
Channel-Profile-Created-Time: 1251294195778388
Channel-Channel-Created-Time: 1251294195778388
Channel-Channel-Answered-Time: 0
Channel-Channel-Progress-Time: 0
Channel-Channel-Progress-Media-Time: 0
Channel-Channel-Hangup-Time: 0
Channel-Channel-Transfer-Time: 0
Channel-Screen-Bit: false
Channel-Privacy-Hide-Name: true
Channel-Privacy-Hide-Number: true
Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/external/hemmeligt%40129.142.224.250
Unique-ID: 681102bc-9246-11de-ac94-b736da546252
Call-Direction: inbound
Presence-Call-Direction: inbound
Answer-State: ringing
Channel-Read-Codec-Name: PCMA
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMA
Channel-Write-Codec-Rate: 8000
Caller-Username: hemmeligt
Caller-Dialplan: XML
Caller-Caller-ID-Name: hemmeligt
Caller-Caller-ID-Number: hemmeligt
Caller-Network-Addr: 129.142.224.250
Caller-Destination-Number: 77344541
Caller-Unique-ID: 681102bc-9246-11de-ac94-b736da546252
Caller-Source: mod_sofia
Caller-Context: public
Caller-Channel-Name: sofia/external/hemmeligt%40129.142.224.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1251294195778388
Caller-Channel-Created-Time: 1251294195778388
Caller-Channel-Answered-Time: 0
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 0
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: false
Caller-Privacy-Hide-Name: true
Caller-Privacy-Hide-Number: true
variable_sip_received_ip: 129.142.224.250
variable_sip_received_port: 5060
variable_sip_via_protocol: udp
variable_sip_from_user: hemmeligt
variable_sip_from_uri: hemmeligt%40129.142.224.250
variable_sip_from_host: 129.142.224.250
variable_sip_from_user_stripped: hemmeligt
variable_sip_from_tag: as19641342
variable_sofia_profile_name: external
variable_sip_Remote-Party-ID:
%22hemmeligt%22%20%3Csip%3Ahemmeligt%40129.142.224.250%3E%3Bprivacy%3Dfull%3Bscreen%3Dno
variable_sip_cid_type: rpid
variable_sip_req_user: 77344541
variable_sip_req_port: 5080
variable_sip_req_uri: 77344541%4079.125.42.248%3A5080
variable_sip_req_host: 79.125.42.248
variable_sip_to_user: 77344541
variable_sip_to_port: 5080
variable_sip_to_uri: 77344541%4079.125.42.248%3A5080
variable_sip_to_host: 79.125.42.248
variable_sip_contact_user: hemmeligt
variable_sip_contact_uri: hemmeligt%40129.142.224.250
variable_sip_contact_host: 129.142.224.250
variable_channel_name: sofia/external/hemmeligt%40129.142.224.250
variable_sip_call_id: 1aa24dab25cf03f07933f7136822ceff%40129.142.224.250
variable_sip_user_agent: Condor_gw1
variable_sip_via_host: 129.142.224.250
variable_sip_via_port: 5060
variable_sip_via_rport: 5060
variable_max_forwards: 70
variable_switch_r_sdp:
v%3D0%0D%0Ao%3Droot%2018982%2018982%20IN%20IP4%20129.142.224.250%0D%0As%3Dsession%0D%0Ac%3DIN%20IP4%20129.142.224.250%0D%0At%3D0%200%0D%0Am%3Daudio%2015430%20RTP/AVP%208%200%203%2097%20101%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A97%20iLBC/8000%0D%0Aa%3Dfmtp%3A97%20mode%3D30%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0D%0Aa%3Dptime%3A20%0D%0A
variable_remote_media_ip: 129.142.224.250
variable_remote_media_port: 15430
variable_read_codec: PCMA
variable_read_rate: 8000
variable_write_codec: PCMA
variable_write_rate: 8000
variable_endpoint_disposition: RECEIVED
variable_bypass_media: false
variable_current_application_data: 127.0.0.1%3A8084%20full
variable_current_application: socket
variable_socket_host: 127.0.0.1
Content-Type: command/reply
Reply-Text: %2BOK%0A
Socket-Mode: static
Control: full

sendmsg
call-command: execute
execute-app-name: answer

Content-Type: command/reply
Reply-Text: +OK

sendmsg
call-command: execute
execute-app-name: bind_meta_app
execute-app-arg: 1 b a bridge::sofia/gateway/secretgw/8888

Content-Type: command/reply
Reply-Text: +OK

sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg: sofia/gateway/secretgw/1234

Content-Type: command/reply
Reply-Text: +OK
--------------------------------

And on the FreeSWITCH side:

2009-08-26 13:43:15.778388 [NOTICE] switch_channel.c:602 New Channel
sofia/external/hemmeligt at 129.142.224.250
[681102bc-9246-11de-ac94-b736da546252]
2009-08-26 13:43:15.778388 [DEBUG] sofia.c:3302 Channel
sofia/external/hemmeligt at 129.142.224.250 entering state
[received][100]
2009-08-26 13:43:15.778388 [DEBUG] sofia.c:3309 Remote SDP:
v=0
o=root 18982 18982 IN IP4 129.142.224.250
s=session
c=IN IP4 129.142.224.250
t=0 0
m=audio 15430 RTP/AVP 8 0 3 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

2009-08-26 13:43:15.778388 [DEBUG] sofia_glue.c:3132 Audio Codec
Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2009-08-26 13:43:15.778388 [DEBUG] sofia_glue.c:2090 Set Codec
sofia/external/hemmeligt at 129.142.224.250 PCMA/8000 20 ms 160 samples
2009-08-26 13:43:15.778388 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
payload to 101
2009-08-26 13:43:15.778388 [DEBUG] sofia.c:3468
(sofia/external/hemmeligt at 129.142.224.250) State Change CS_NEW ->
CS_INIT
2009-08-26 13:43:15.778388 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:398
(sofia/external/hemmeligt at 129.142.224.250) Running State Change
CS_INIT
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:481
(sofia/external/hemmeligt at 129.142.224.250) State INIT
2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:83
sofia/external/hemmeligt at 129.142.224.250 SOFIA INIT
2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:111
(sofia/external/hemmeligt at 129.142.224.250) State Change CS_INIT ->
CS_ROUTING
2009-08-26 13:43:15.778388 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:481
(sofia/external/hemmeligt at 129.142.224.250) State INIT going to sleep
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:398
(sofia/external/hemmeligt at 129.142.224.250) Running State Change
CS_ROUTING
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:484
(sofia/external/hemmeligt at 129.142.224.250) State ROUTING
2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:130
sofia/external/hemmeligt at 129.142.224.250 SOFIA ROUTING
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:78
sofia/external/hemmeligt at 129.142.224.250 Standard ROUTING
2009-08-26 13:43:15.778388 [INFO] mod_dialplan_xml.c:315 Processing
hemmeligt->77344541 in context public
Dialplan: sofia/external/hemmeligt at 129.142.224.250 parsing
[public->ff-ivr] continue=false
Dialplan: sofia/external/hemmeligt at 129.142.224.250 Regex (PASS)
[ff-ivr] destination_number(77344541) =~ /^(.*)$/ break=on-false
Dialplan: sofia/external/hemmeligt at 129.142.224.250 Action
set(bypass_media=false)
Dialplan: sofia/external/hemmeligt at 129.142.224.250 Action
socket(127.0.0.1:8084 full)
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:114
(sofia/external/hemmeligt at 129.142.224.250) State Change CS_ROUTING ->
CS_EXECUTE
2009-08-26 13:43:15.778388 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:484
(sofia/external/hemmeligt at 129.142.224.250) State ROUTING going to
sleep
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:398
(sofia/external/hemmeligt at 129.142.224.250) Running State Change
CS_EXECUTE
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:491
(sofia/external/hemmeligt at 129.142.224.250) State EXECUTE
2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:173
sofia/external/hemmeligt at 129.142.224.250 SOFIA EXECUTE
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:151
sofia/external/hemmeligt at 129.142.224.250 Standard EXECUTE
EXECUTE sofia/external/hemmeligt at 129.142.224.250 set(bypass_media=false)
2009-08-26 13:43:15.778388 [DEBUG] mod_dptools.c:748
sofia/external/hemmeligt at 129.142.224.250 SET [bypass_media]=[false]
EXECUTE sofia/external/hemmeligt at 129.142.224.250 socket(127.0.0.1:8084 full)
2009-08-26 13:43:21.701447 [DEBUG] switch_ivr.c:540
sofia/external/hemmeligt at 129.142.224.250 Command Execute answer()
EXECUTE sofia/external/hemmeligt at 129.142.224.250 answer()
2009-08-26 13:43:21.701447 [DEBUG] mod_dptools.c:649
sofia/external/hemmeligt at 129.142.224.250 receive message [ANSWER]
2009-08-26 13:43:21.701447 [DEBUG] sofia_glue.c:2324 AUDIO RTP
[sofia/external/hemmeligt at 129.142.224.250] 10.226.231.225 port 19016
-> 129.142.224.250 port 15430 codec: 8 ms: 20
2009-08-26 13:43:21.701447 [DEBUG] switch_rtp.c:1138 Starting timer
[soft] 160 bytes per 20ms
2009-08-26 13:43:21.702447 [DEBUG] mod_sofia.c:536 Local SDP
sofia/external/hemmeligt at 129.142.224.250:
v=0
o=FreeSWITCH 1251275185 1251275186 IN IP4 79.125.42.248
s=FreeSWITCH
c=IN IP4 79.125.42.248
t=0 0
m=audio 19016 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-08-26 13:43:21.702447 [DEBUG] switch_core_session.c:630 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:21.702447 [NOTICE] mod_dptools.c:649 Channel
[sofia/external/hemmeligt at 129.142.224.250] has been answered
2009-08-26 13:43:21.702447 [DEBUG] switch_channel.c:182
sofia/external/hemmeligt at 129.142.224.250 receive message [AUDIO_SYNC]
2009-08-26 13:43:21.702447 [DEBUG] sofia.c:3302 Channel
sofia/external/hemmeligt at 129.142.224.250 entering state
[completed][200]
2009-08-26 13:43:21.782616 [DEBUG] sofia.c:3302 Channel
sofia/external/hemmeligt at 129.142.224.250 entering state [ready][200]
2009-08-26 13:43:26.388737 [DEBUG] switch_ivr.c:540
sofia/external/hemmeligt at 129.142.224.250 Command Execute
bind_meta_app(1 b a bridge::sofia/gateway/secretgw/8888)
EXECUTE sofia/external/hemmeligt at 129.142.224.250 bind_meta_app(1 b a
bridge::sofia/gateway/secretgw/8888)
2009-08-26 13:43:26.388737 [INFO] switch_ivr_async.c:1795 Bound B-Leg:
1 bridge::sofia/gateway/secretgw/8888
2009-08-26 13:43:29.684236 [DEBUG] switch_ivr.c:540
sofia/external/hemmeligt at 129.142.224.250 Command Execute
bridge(sofia/gateway/secretgw/1234)
EXECUTE sofia/external/hemmeligt at 129.142.224.250
bridge(sofia/gateway/secretgw/1234)
2009-08-26 13:43:29.684236 [NOTICE] switch_channel.c:602 New Channel
sofia/external/1234 [705ad5d8-9246-11de-ac94-b736da546252]
2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:2809
(sofia/external/1234) State Change CS_NEW -> CS_INIT
2009-08-26 13:43:29.684236 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:398
(sofia/external/1234) Running State Change CS_INIT
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:481
(sofia/external/1234) State INIT
2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:83 sofia/external/1234 SOFIA INIT
2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:111
(sofia/external/1234) State Change CS_INIT -> CS_ROUTING
2009-08-26 13:43:29.684236 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:29.684236 [DEBUG] sofia.c:3302 Channel
sofia/external/1234 entering state [calling][0]
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:481
(sofia/external/1234) State INIT going to sleep
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:398
(sofia/external/1234) Running State Change CS_ROUTING
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:484
(sofia/external/1234) State ROUTING
2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:130 sofia/external/1234
SOFIA ROUTING
2009-08-26 13:43:29.684236 [DEBUG] switch_ivr_originate.c:63
(sofia/external/1234) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-08-26 13:43:29.684236 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:484
(sofia/external/1234) State ROUTING going to sleep
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:398
(sofia/external/1234) Running State Change CS_CONSUME_MEDIA
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:503
(sofia/external/1234) State CONSUME_MEDIA
2009-08-26 13:43:29.731221 [DEBUG] sofia.c:3302 Channel
sofia/external/1234 entering state [calling][0]
2009-08-26 13:43:30.660221 [DEBUG] sofia.c:3302 Channel
sofia/external/1234 entering state [proceeding][183]
2009-08-26 13:43:30.660221 [DEBUG] sofia.c:3309 Remote SDP:
v=0
o=root 18982 18982 IN IP4 129.142.224.250
s=session
c=IN IP4 129.142.224.250
t=0 0
m=audio 12480 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:3132 Audio Codec
Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:2090 Set Codec
sofia/external/1234 PCMA/8000 20 ms 160 samples
2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
payload to 101
2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:2324 AUDIO RTP
[sofia/external/1234] 10.226.231.225 port 25540 -> 129.142.224.250
port 12480 codec: 8 ms: 20
2009-08-26 13:43:30.660221 [DEBUG] switch_rtp.c:1138 Starting timer
[soft] 160 bytes per 20ms
2009-08-26 13:43:30.661084 [NOTICE] sofia_glue.c:2759 Pre-Answer
sofia/external/1234!
2009-08-26 13:43:30.661084 [DEBUG] switch_channel.c:1778 Send signal
sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_originate.c:2061
Originate Resulted in Success: [sofia/external/1234]
2009-08-26 13:43:30.665170 [DEBUG] switch_channel.c:182
sofia/external/1234 receive message [AUDIO_SYNC]
2009-08-26 13:43:30.665170 [DEBUG] switch_channel.c:182
sofia/external/hemmeligt at 129.142.224.250 receive message [AUDIO_SYNC]
2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_bridge.c:889
sofia/external/1234 receive message [BRIDGE]
2009-08-26 13:43:30.665170 [DEBUG] switch_core_session.c:630 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_bridge.c:896
sofia/external/hemmeligt at 129.142.224.250 receive message [BRIDGE]
2009-08-26 13:43:30.665170 [DEBUG] switch_core_session.c:630 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_bridge.c:940
(sofia/external/1234) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2009-08-26 13:43:30.665170 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:30.705074 [DEBUG] switch_core_state_machine.c:503
(sofia/external/1234) State CONSUME_MEDIA going to sleep
2009-08-26 13:43:30.705074 [DEBUG] switch_core_state_machine.c:398
(sofia/external/1234) Running State Change CS_EXCHANGE_MEDIA
2009-08-26 13:43:30.705074 [DEBUG] switch_core_state_machine.c:494
(sofia/external/1234) State EXCHANGE_MEDIA
2009-08-26 13:43:30.705074 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK
2009-08-26 13:43:36.642173 [DEBUG] sofia.c:3302 Channel
sofia/external/1234 entering state [ready][200]
2009-08-26 13:43:36.642173 [DEBUG] switch_channel.c:1891 Send signal
sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:36.642173 [NOTICE] sofia.c:3752 Channel
[sofia/external/1234] has been answered
2009-08-26 13:43:36.642173 [DEBUG] switch_channel.c:182
sofia/external/1234 receive message [AUDIO_SYNC]
2009-08-26 13:43:40.309616 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF *:2000
2009-08-26 13:43:40.970513 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF 1:2000
2009-08-26 13:43:40.970513 [DEBUG] switch_ivr_async.c:1711
sofia/external/hemmeligt at 129.142.224.250 Processing meta digit '1'
[bridge::sofia/gateway/secretgw/8888]
2009-08-26 13:43:40.970513 [WARNING] switch_ivr_async.c:2310 Channel
[sofia/external/hemmeligt at 129.142.224.250][bridge::sofia/gateway/secretgw/8888]
already broadcasting...broadcast aborted
2009-08-26 13:43:46.272706 [NOTICE] sofia.c:327 Hangup
sofia/external/1234 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2009-08-26 13:43:46.272706 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/1234 [KILL]
2009-08-26 13:43:46.272706 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:371
sofia/external/1234 ending bridge by request from write function
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:426
sofia/external/hemmeligt at 129.142.224.250 receive message [UNBRIDGE]
2009-08-26 13:43:46.284704 [DEBUG] switch_core_session.c:630 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:452 BRIDGE
THREAD DONE [sofia/external/hemmeligt at 129.142.224.250]
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:454 Send signal
sofia/external/1234 [BREAK]
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:377
sofia/external/1234 ending bridge by request from read function
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:452 BRIDGE
THREAD DONE [sofia/external/1234]
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:454 Send signal
sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:494
(sofia/external/1234) State EXCHANGE_MEDIA going to sleep
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:398
(sofia/external/1234) Running State Change CS_HANGUP
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:434
(sofia/external/1234) State HANGUP
2009-08-26 13:43:46.284704 [DEBUG] mod_sofia.c:338 Channel
sofia/external/1234 hanging up, cause: NORMAL_CLEARING
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:46
sofia/external/1234 Standard HANGUP, cause: NORMAL_CLEARING
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:434
(sofia/external/1234) State HANGUP going to sleep
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:476
(sofia/external/1234) State Change CS_HANGUP -> CS_REPORTING
2009-08-26 13:43:46.284704 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:398
(sofia/external/1234) Running State Change CS_REPORTING
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:612
(sofia/external/1234) State REPORTING
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:53
sofia/external/1234 Standard REPORTING, cause: NORMAL_CLEARING
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:612
(sofia/external/1234) State REPORTING going to sleep
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:411
(sofia/external/1234) State Change CS_REPORTING -> CS_DESTROY
2009-08-26 13:43:46.284704 [DEBUG] switch_core_session.c:1068 Session
5 (sofia/external/1234) Locked, Waiting on external entities
2009-08-26 13:43:46.284704 [NOTICE] switch_core_session.c:1086 Session
5 (sofia/external/1234) Ended
2009-08-26 13:43:46.284704 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/external/1234 [CS_DESTROY]
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:564
(sofia/external/1234) State DESTROY
2009-08-26 13:43:46.284704 [DEBUG] mod_sofia.c:255 sofia/external/1234
SOFIA DESTROY
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:60
sofia/external/1234 Standard DESTROY
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:564
(sofia/external/1234) State DESTROY going to sleep
2009-08-26 13:44:46.746524 [NOTICE] sofia.c:327 Hangup
sofia/external/hemmeligt at 129.142.224.250 [CS_EXECUTE]
[NORMAL_CLEARING]
2009-08-26 13:44:46.746524 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/hemmeligt at 129.142.224.250 [KILL]
2009-08-26 13:44:46.746524 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:491
(sofia/external/hemmeligt at 129.142.224.250) State EXECUTE going to
sleep
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:398
(sofia/external/hemmeligt at 129.142.224.250) Running State Change
CS_HANGUP
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:434
(sofia/external/hemmeligt at 129.142.224.250) State HANGUP
2009-08-26 13:44:46.747796 [DEBUG] mod_sofia.c:338 Channel
sofia/external/hemmeligt at 129.142.224.250 hanging up, cause:
NORMAL_CLEARING
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:46
sofia/external/hemmeligt at 129.142.224.250 Standard HANGUP, cause:
NORMAL_CLEARING
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:434
(sofia/external/hemmeligt at 129.142.224.250) State HANGUP going to sleep
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:476
(sofia/external/hemmeligt at 129.142.224.250) State Change CS_HANGUP ->
CS_REPORTING
2009-08-26 13:44:46.747796 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:398
(sofia/external/hemmeligt at 129.142.224.250) Running State Change
CS_REPORTING
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:612
(sofia/external/hemmeligt at 129.142.224.250) State REPORTING
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:53
sofia/external/hemmeligt at 129.142.224.250 Standard REPORTING, cause:
NORMAL_CLEARING
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:612
(sofia/external/hemmeligt at 129.142.224.250) State REPORTING going to
sleep
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:411
(sofia/external/hemmeligt at 129.142.224.250) State Change CS_REPORTING
-> CS_DESTROY
2009-08-26 13:44:46.747796 [DEBUG] switch_core_session.c:1068 Session
4 (sofia/external/hemmeligt at 129.142.224.250) Locked, Waiting on
external entities
2009-08-26 13:44:46.747796 [NOTICE] switch_core_session.c:1086 Session
4 (sofia/external/hemmeligt at 129.142.224.250) Ended
2009-08-26 13:44:46.747796 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/external/hemmeligt at 129.142.224.250 [CS_DESTROY]
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:564
(sofia/external/hemmeligt at 129.142.224.250) State DESTROY
2009-08-26 13:44:46.747796 [DEBUG] mod_sofia.c:255
sofia/external/hemmeligt at 129.142.224.250 SOFIA DESTROY
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:60
sofia/external/hemmeligt at 129.142.224.250 Standard DESTROY
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:564
(sofia/external/hemmeligt at 129.142.224.250) State DESTROY going to
sleep

Of which I mainly find these interesting:

EXECUTE sofia/external/hemmeligt at 129.142.224.250 bind_meta_app(1 b a
bridge::sofia/gateway/secretgw/8888)
2009-08-26 13:43:26.388737 [INFO] switch_ivr_async.c:1795 Bound B-Leg:
1 bridge::sofia/gateway/secretgw/8888
[...]
2009-08-26 13:43:40.309616 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF *:2000
2009-08-26 13:43:40.970513 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF 1:2000
2009-08-26 13:43:40.970513 [DEBUG] switch_ivr_async.c:1711
sofia/external/hemmeligt at 129.142.224.250 Processing meta digit '1'
[bridge::sofia/gateway/secretgw/8888]
2009-08-26 13:43:40.970513 [WARNING] switch_ivr_async.c:2310 Channel
[sofia/external/hemmeligt at 129.142.224.250][bridge::sofia/gateway/secretgw/8888]
already broadcasting...broadcast aborted

Can anybody explain what is going on here? It seems like the
bind_meta_app is bound perfectly via event socket, but when it comes
down, it doesn't work.

Best regards,
Harry


From MPeace at edcogroupinc.com  Wed Aug 26 06:58:02 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Wed, 26 Aug 2009 08:58:02 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <5DF610F9-6EB5-43CC-BB4F-3513C78F608F@freeswitch.org>
References: <5DF610F9-6EB5-43CC-BB4F-3513C78F608F@freeswitch.org>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A622D22BE@ESNEXS2.edcogroup.net>

I found and edited openzap.cong file, the openzap.conf.xml file is in the correct dir but the only example I can find is for a Euro E1 setup not a US T1. I

I have turned up data T1 circuits for WAN links before but not telco stuff. The provider is Qwest, I?ll have to ask about the NI2 , DMS question.

Mike

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins
Sent: Wednesday, August 26, 2009 3:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


I have looked at that but I am confused on which files need to be edited. Since I have already installed in Wanpipe mode with the Sangoma card I skipped straight to the Wanpipe section. It mentions setting the [span wanpie PRI_1] etc in the openzap.conf then further down it mentions editing the autoload_configs/openzap.conf.xml with a completely different format of config.

Are these the same file? If not where does the first one mentioned reside?

These are two different files. The openzap.conf file is in the freeswitch/conf directory. It is for configuring OZ to talk to the Sangoma card. The openzap.conf.xml file is for configuring the actual openzap module in FreeSWITCH.

The specs you listed on the T1 you have indicate that it's a PRI. However, you need to know which protocol, like NI2 or DMS. Have you ever done a PRI turn up? Who is the carrier?

-MC
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From gmaruzz at celliax.org  Wed Aug 26 07:01:44 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 26 Aug 2009 16:01:44 +0200
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
In-Reply-To: <4A9537DF.8070609@yahoo.com>
References: <4A9537DF.8070609@yahoo.com>
Message-ID: <7b197bef0908260701n60a671dct26394947e4db65c4@mail.gmail.com>

You will find all the informations, and then some, here:
http://wiki.freeswitch.org

Then, after reading and testing and experimenting, come to the IRC
channel for direct help with esoteric (or not so esoteric) problems.

You'll find a nice and friendly community.

-gm




Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Wed, Aug 26, 2009 at 3:25 PM, Merle J. Ebbert<se02005-mec at yahoo.com> wrote:
>
> Hi,
>
> I'm trying to avoid taking up a lot of peoples valuable time.
>
> SIP & FS have ?brought some ideas for some commercial products but I
> need to know where to start.
>
> Having once written a proprietary DOS & helped with writing a RTOS, I
> consider myself capable
> of learning. ?I (we) just need to know where to start to come up to
> speed rapidly.
>
> Is there a FreeSWITCH tutorial available?
> Should someone new start with Asterisk and then possibly move to FS?
>
> Thanks,
> Merle
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From bjbrashier at gmail.com  Wed Aug 26 07:05:05 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Wed, 26 Aug 2009 08:05:05 -0600
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
In-Reply-To: <4A9537DF.8070609@yahoo.com>
References: <4A9537DF.8070609@yahoo.com>
Message-ID: <7bcfdd290908260705t644c42f0g52fd6c10b9bbaed1@mail.gmail.com>

Speaking as someone who went through this recently myself, my
suggestion is to learn by doing. Choose some simple-sounding task you
want to accomplish in FS, and try to carry it out. Use the wiki as a
reference and the mailing list if you can't figure out what you need
from that.

I don't suggest you start with Asterisk or anything else, as I'd guess
that would just confuse the issue. I came from a BIOS background, went
directly into Freeswitch, and am so far glad I did.

BB

On Wed, Aug 26, 2009 at 7:25 AM, Merle J. Ebbert<se02005-mec at yahoo.com> wrote:
>
> Hi,
>
> I'm trying to avoid taking up a lot of peoples valuable time.
>
> SIP & FS have ?brought some ideas for some commercial products but I
> need to know where to start.
>
> Having once written a proprietary DOS & helped with writing a RTOS, I
> consider myself capable
> of learning. ?I (we) just need to know where to start to come up to
> speed rapidly.
>
> Is there a FreeSWITCH tutorial available?
> Should someone new start with Asterisk and then possibly move to FS?
>
> Thanks,
> Merle
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From aep.lists at it46.se  Wed Aug 26 07:18:03 2009
From: aep.lists at it46.se (Alberto Escudero-Pascual (lists))
Date: Wed, 26 Aug 2009 16:18:03 +0200
Subject: [Freeswitch-users] Internal Profile sends RTP to external IP
 when using two network interfaces
In-Reply-To: <3694FC90-3FD1-4F36-8CA9-A3C18EBDEF34@freeswitch.org>
References: <46b96649f81b0bfcb111928a55450211.squirrel@correo.nodo50.org>
	<3694FC90-3FD1-4F36-8CA9-A3C18EBDEF34@freeswitch.org>
Message-ID: <7682ba3b2cbb1cf578dcc2226e7d4787.squirrel@correo.nodo50.org>



ASCII art follows

voip-phones..... 10.0.46.1 [ FS ] 216.82.231.69 ----> Internet


I have two profiles binded to two different IPs

internal is binded to 10.0.46.1 port 5060 /24 network
external is binded to 216.82.231.69 port 5080

My IP phones are in the 10.0.46.X range and they register @ 10.0.46.1 IP
address.

When I place a call between the two phones in the 10.0.46.0/24 network
the phones register but audio RTP is sent to 216.82.231.69 instead of
10.0.46.1 (FS)

In this scenario i have FS without any NAT configuration.

-- 
The way that i solved was to run both profiles (internal and external) in
the public IP (local_ipv4) and force the phones to register with
216.82.231.69 instead of 10.0.46.1

Thanks looong time!


-- 
Stopping junk mailers is good for the environment

> First off you can't bind one profile to two interfaces you have to
> launch two sofia profiles, one for each IP.  Secondly if you're doing
> things like this you'll have to refer to the in tree internal.xml.
> Third can you outline the network topology a little bit more?  Is nat
> involved?
>
> /b
>
> On Aug 26, 2009, at 3:04 AM, Alberto Escudero-Pascual (lists) wrote:
>
>> Hi,
>>
>> I have a FS box with two physical network interfaces. The internal
>> interface is hosting several internal phones. I have binded the
>> internal
>> profile SIP/RTP/IP to the private interface. Phones registered
>> correctly
>> but with User: 1000 at external.ip.address in sofia status profile
>> internal
>>
>> When I place a call between two internal phones, RTP traffic is send
>> to
>> the external IP address of the FS box instead of the internal.
>>
>> The SIP/SDP messages send from FS carry the external IP instead of the
>> internal.
>>
>> The result is that no RTP media arrives to any of the phones.
>>
>> /aep
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From brian at freeswitch.org  Wed Aug 26 07:25:02 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 Aug 2009 09:25:02 -0500
Subject: [Freeswitch-users] Internal Profile sends RTP to external IP
	when using two network interfaces
In-Reply-To: <7682ba3b2cbb1cf578dcc2226e7d4787.squirrel@correo.nodo50.org>
References: <46b96649f81b0bfcb111928a55450211.squirrel@correo.nodo50.org>
	<3694FC90-3FD1-4F36-8CA9-A3C18EBDEF34@freeswitch.org>
	<7682ba3b2cbb1cf578dcc2226e7d4787.squirrel@correo.nodo50.org>
Message-ID: <56E6F9F8-82FB-4A4D-9B7A-44D9BE2BDE43@freeswitch.org>

Well actually you do have issues you're going to have to specify the  
rtp-ip and sip-ip on BOTH profiles yourself because you're config is  
currently putting your external IP into the internal's settings.   
Please correct that and I'm sure it'll work then.

/b

On Aug 26, 2009, at 9:18 AM, Alberto Escudero-Pascual (lists) wrote:

>
>
> ASCII art follows
>
> voip-phones..... 10.0.46.1 [ FS ] 216.82.231.69 ----> Internet
>
>
> I have two profiles binded to two different IPs
>
> internal is binded to 10.0.46.1 port 5060 /24 network
> external is binded to 216.82.231.69 port 5080
>
> My IP phones are in the 10.0.46.X range and they register @  
> 10.0.46.1 IP
> address.
>
> When I place a call between the two phones in the 10.0.46.0/24 network
> the phones register but audio RTP is sent to 216.82.231.69 instead of
> 10.0.46.1 (FS)
>
> In this scenario i have FS without any NAT configuration.
>
> -- 
> The way that i solved was to run both profiles (internal and  
> external) in
> the public IP (local_ipv4) and force the phones to register with
> 216.82.231.69 instead of 10.0.46.1
>
> Thanks looong time!
>



From MPeace at edcogroupinc.com  Wed Aug 26 07:26:45 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Wed, 26 Aug 2009 09:26:45 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A622D22BE@ESNEXS2.edcogroup.net>
References: <5DF610F9-6EB5-43CC-BB4F-3513C78F608F@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A622D22BE@ESNEXS2.edcogroup.net>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A622D2307@ESNEXS2.edcogroup.net>

Ok it is NI2 according to Qwest,

Mike Peace
Network Analyst
EDCO, The Document People
Direct 417-447-3367

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike Peace
Sent: Wednesday, August 26, 2009 8:58 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

I found and edited openzap.cong file, the openzap.conf.xml file is in the correct dir but the only example I can find is for a Euro E1 setup not a US T1. I

I have turned up data T1 circuits for WAN links before but not telco stuff. The provider is Qwest, I?ll have to ask about the NI2 , DMS question.

Mike

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins
Sent: Wednesday, August 26, 2009 3:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


I have looked at that but I am confused on which files need to be edited. Since I have already installed in Wanpipe mode with the Sangoma card I skipped straight to the Wanpipe section. It mentions setting the [span wanpie PRI_1] etc in the openzap.conf then further down it mentions editing the autoload_configs/openzap.conf.xml with a completely different format of config.

Are these the same file? If not where does the first one mentioned reside?

These are two different files. The openzap.conf file is in the freeswitch/conf directory. It is for configuring OZ to talk to the Sangoma card. The openzap.conf.xml file is for configuring the actual openzap module in FreeSWITCH.

The specs you listed on the T1 you have indicate that it's a PRI. However, you need to know which protocol, like NI2 or DMS. Have you ever done a PRI turn up? Who is the carrier?

-MC
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From se02005-mec at yahoo.com  Wed Aug 26 07:45:11 2009
From: se02005-mec at yahoo.com (Merle J. Ebbert)
Date: Wed, 26 Aug 2009 07:45:11 -0700
Subject: [Freeswitch-users] I would like to send an IM. From a Windows
 PC. To a BlackBerry @vzw.blackberry.com
Message-ID: <4A954A77.7050508@yahoo.com>

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From anatoliy at kounitskiy.com  Wed Aug 26 07:08:55 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Wed, 26 Aug 2009 17:08:55 +0300
Subject: [Freeswitch-users] Questions about att_xfer (freeswitch version 1.0
	trunk 14633M)
Message-ID: <4A9541F7.4070901@kounitskiy.com>

Hello everybody!
I have few questions about the att_xfer application. First, what i want 
to accomplish is: user A calls user B, after that user B makes attended 
transfer to user C.
In the dialplan i have:

<context name="vpbx">
  <extension name="local_number">
...
      <action application="bind_meta_app" data="1 b s 
execute_extension::dx XML features"/>
      <action application="bind_meta_app" data="2 b s 
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
      <action application="bind_meta_app" data="3 b s 
execute_extension::cf XML features"/>
      <action application="bind_meta_app" data="4 b s 
execute_extension::attented_xfer XML features"/>
 ....
    </condition>
  </extension>

So when user B answers the call, he sends *4 and the extensions for the 
attended transfer is started - the usual - plays message and read the 
input dtmf:

features.xml
...
    <extension name="attented_xfer">
      <condition field="${toll_allow}" expression="local"/>
      <condition field="destination_number" expression="^attented_xfer$">
        <action application="info"/>
        <action application="read" data="3 4 ivr/ivr-enter_ext.wav 
attxfer_callthis 30000 #"/>
        <action application="set" data="call_timeout=15"/>
        <action application="att_xfer" 
data="user/${attxfer_callthis}@${domain_name}"/>
      </condition>
    </extension>
...

To this problems everything is perfect. But here comes the questions, so 
if you can give some tips would be great.

1) when user B enters the extension number of C - the C's phone starts 
ringing in the tcpdump i can see that the phone is sending 180 ringing, 
BUT user B does not hear the ringing.
2) as mentioned in the 
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
quote: "If the other leg is a voicemail or doesn't answered you can 
hangup that leg by pressing dtmf # (fixed in r14438) "
It doesn't work. The option 0 is working even before C answering the 
phone - after he answers it's a threeway conference :) - i like this 
feature.

I'm using FreeSWITCH Version 1.0.trunk (14633M)

Also I tried to set call timeout to see if I can go back the user A, who 
is listening to MOH - no luck here.

Probably I'm missing something. Tried to look in the source of att_xfer 
to understand why the feature i want is not working - but it seems my 
C/C++ skills are not so good, as i want :( .

Thank you in advance,
Anatoliy Kounitskiy


From anatoliy at kounitskiy.com  Wed Aug 26 07:51:49 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Wed, 26 Aug 2009 17:51:49 +0300
Subject: [Freeswitch-users] Questions about att_xfer
Message-ID: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>

Hello everybody!
I have few questions about the att_xfer application. First, what i want
to accomplish is: user A calls user B, after that user B makes attended
transfer to user C.
In the dialplan i have:

<context name="vpbx">
  <extension name="local_number">
...
      <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
      <action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
      <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
      <action application="bind_meta_app" data="4 b s
execute_extension::attented_xfer XML features"/>
 ....
    </condition>
  </extension>

So when user B answers the call, he sends *4 and the extensions for the
attended transfer is started - the usual - plays message and read the
input dtmf:

features.xml
...
    <extension name="attented_xfer">
      <condition field="${toll_allow}" expression="local"/>
      <condition field="destination_number" expression="^attented_xfer$">
        <action application="info"/>
        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 30000 #"/>
        <action application="set" data="call_timeout=15"/>
        <action application="att_xfer"
data="user/${attxfer_callthis}@${domain_name}"/>
      </condition>
    </extension>
...

To this problems everything is perfect. But here comes the questions, so
if you can give some tips would be great.

1) when user B enters the extension number of C - the C's phone starts
ringing in the tcpdump i can see that the phone is sending 180 ringing,
BUT user B does not hear the ringing.
2) as mentioned in the
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
quote: "If the other leg is a voicemail or doesn't answered you can
hangup that leg by pressing dtmf # (fixed in r14438) "
It doesn't work. The option 0 is working even before C answering the
phone - after he answers it's a threeway conference :) - i like this
feature.

I'm using FreeSWITCH Version 1.0.trunk (14633M)

Also I tried to set call timeout to see if I can go back the user A, who
is listening to MOH - no luck here.

Probably I'm missing something. Tried to look in the source of att_xfer
to understand why the feature i want is not working - but it seems my
C/C++ skills are not so good, as i want :( .

Thank you in advance,
Anatoliy Kounitskiy


From delianspam at gmail.com  Wed Aug 26 07:59:08 2009
From: delianspam at gmail.com (delianSPAM)
Date: Wed, 26 Aug 2009 17:59:08 +0300
Subject: [Freeswitch-users] Timers/DTMFs During a Call
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>

Hello Everybody!

 

1.       Scenario.

I am writing an IVR in Python that gets a destination from the calling party
(party A) and then connects to the destination (party B).

When the call is CONNECTED, I want to:

-          Receive DTMFs

-          Have a timer that can call a certain function in my script. The
script will have to play a message to party A.

-          Have a timer that can call a certain function in my script. The
script will have to drop the call.

Please notice that I want to do the things after the two parties are
connected, and not after I send the Invite to party B.

 

2.       Problem.

I will be happy to receive help on:

-          Which methods should I look for to implement this.

 

3.       Details

Here is how I connect the call currently:

session.execute("bridge","sofia/internal/" + destination_number +
"@domain.com")

 

I have tried to create a timer callback function "my_method()" using:

ivr_timer =threading.Timer(30,my_method)

This never called the function "my_method()".

 

Maybe I am wrong in using threading.Timer and the "bridge" application?
Maybe I need to create a new thread and a new timer using the API of
freeswitch, plus to use the "session.setInputCallback", plus use a
conference rather than a bridge? Can you please provide any suggestions or
examples? 

 

Thank you!

Best Regards, Delian Tashev

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From mike at jerris.com  Wed Aug 26 08:18:38 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 26 Aug 2009 11:18:38 -0400
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
Message-ID: <0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>


On Aug 26, 2009, at 9:18 AM, Dennis wrote:

> hi,
>
> we have set manage-presence = true to see, who is talking on the phone
> and who is free. everyone here has his own snom-voip phone with 12
> led-lights.
>
> incoming calls are handled in groups. this means, if someone is
> ringing, all voip-phones of one group are ringing and one can see all
> led-lights flashing. if someone answers the phonecall, the led of this
> person (p1) is on on everybodys phone, so everybody can see, that the
> person is talking.
> now, if another phonecall comes in, all led-lights are flashing -
> including the led of the person who is talking. if another person (p2)
> answers the second phonecall, the led of p2 is on, but the led of p1
> is off, although p1 is still talking.
>
> is there something we can do about this?
>

Do about what?  Your description sounds fine, what is the problem

Mike



From anthony.minessale at gmail.com  Wed Aug 26 08:25:05 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 10:25:05 -0500
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <4A94A2E0.6040306@nowthor.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
	<4A93EFE1.6010007@nowthor.com>
	<191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>
	<4A94A2E0.6040306@nowthor.com>
Message-ID: <191c3a030908260825n73d6ffc2kbc87296197e72c90@mail.gmail.com>

Keep in mind I am not trying to just arrogantly shoot down your idea, I
enjoy the brainstorming.

Unfortunately it's still a problem because the core bridge/originate
mechanism does not parse XML it's only parses the
dial string markup.

The XML dialplan is completely abstract from the originate mechanism because
there can be many
dialplans not just XML.

In the end you must end up with one string of application and one string of
data.

I am very strict about maintaining scope boundaries between abstract
concepts.  I think it's the number 1 breakdown that
led to asterisk getting so messed up over the years and I want to avoid that
same plague.

I hope you understand and if you want to continue to ponder some more
functionality I welcome it.



On Tue, Aug 25, 2009 at 9:50 PM, Carlos S. Antunes <csa at nowthor.com> wrote:

>  Anthony,
>
> Yes, you are right, I was thinking strictly in terms of SIP gateways. I
> guess that instead on the tag "gateway", one could use "channel"? For
> example:
>
> <action application="bridge">
>    <!-- set some kind of *bridge-specific* variable -->
>    <channel order=1 data="sofia/gateway/gw-1/1$1"/>
>    <channel order=1 data="sofia/gateway/gw-2/1$1"/>
>    <channel order=2 data="sofia/gateway/gw-3/1$1">
>        <!-- set some kind of *channel-specific* variable -->
>    </channel >
>    <channel order=3 data="sofia/gateway/gw-4/1$1"/>
> </action>
>
> Would this be acceptable to you?
>
> Regarding the combinations of continue_on_fail and hangup_after bridge,
> I'll give that a try although I suspect it will result in less structured
> and harder to understand markup...
>
> Thanks!
>
> Carlos
>
> Anthony Minessale wrote:
>
> This suggestion violates the scope boundaries.
>
> gateways are specific concept to mod_sofia so a <gateway> tag in <action>
> (part of agnostic xml dialplan)
> does not flow properly.
>
> you can also use combinations of continue_on_fail and hangup_after bridge
> so you can
> just put each bridge statement in it's own action.
>
>
> On Tue, Aug 25, 2009 at 9:06 AM, Carlos S. Antunes <csa at nowthor.com>wrote:
>
>> Max,
>>
>> I would like to see something similar too. For example, it would be
>> wonderful if one could specify multiple gateways to try like this or
>> something similar:
>>
>> <action application="bridge">
>>    <!-- set some kind of *bridge-specific* parameter/variable -->
>>    <gateway order=1 data="sofia/gateway/gw-1/1$1"/>
>>    <gateway order=1 data="sofia/gateway/gw-2/1$1"/>
>>    <gateway order=2 data="sofia/gateway/gw-3/1$1">
>>        <!-- set some kind of *gw-specific* parameter/variable -->
>>    </gateway>
>>    <gateway order=3 data="sofia/gateway/gw-4/1$1"/>
>> </action>
>>
>> One would be able to avoid the "[]" and "{}" hacks and combine
>> sequential and simultaneous trying of gateways.
>>
>> What do the developers think of this?
>>
>> Carlos
>>
>> Max Ivanov wrote:
>> > Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
>> >
>> sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
>> > ,
>> > the whole sting is tooo long, is there any shorter way to write same
>> thing? Like
>> > "sofia/gateway/panas*/1000" will try all gateways matching the pattern.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From mitul at enterux.com  Wed Aug 26 08:31:09 2009
From: mitul at enterux.com (Mitul Limbani)
Date: Wed, 26 Aug 2009 11:31:09 -0400
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
Message-ID: <2918.1251300669@enterux.com>

  BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px;
}Infact coming from Asterisk world is not too hard to figure out FS,
but yeah i wont recommend a newbie that route :)
 Although an Asterisk dude /dudette can check out the Rosetta_Stone
page on wiki
 Thanks & Regards, 
 Mitul Limbani, 
 Founder & CEO, 
 Enterux Solutions, 
 The Enterprise Linux Company (TM), 
 www.enterux.com 
 On Wed 26/08/09 19:35 , Bradley Brashier bjbrashier at gmail.com sent:
 Speaking as someone who went through this recently myself, my
 suggestion is to learn by doing. Choose some simple-sounding task
you
 want to accomplish in FS, and try to carry it out. Use the wiki as a
 reference and the mailing list if you can't figure out what you need
 from that.
 I don't suggest you start with Asterisk or anything else, as I'd
guess
 that would just confuse the issue. I came from a BIOS background,
went
 directly into Freeswitch, and am so far glad I did.
 BB
 On Wed, Aug 26, 2009 at 7:25 AM, Merle J. Ebbert wrote:
 >
 > Hi,
 >
 > I'm trying to avoid taking up a lot of peoples valuable time.
 >
 > SIP & FS have  brought some ideas for some commercial products but
I
 > need to know where to start.
 >
 > Having once written a proprietary DOS & helped with writing a
RTOS, I
 > consider myself capable
 > of learning.  I (we) just need to know where to start to come up
to
 > speed rapidly.
 >
 > Is there a FreeSWITCH tutorial available?
 > Should someone new start with Asterisk and then possibly move to
FS?
 >
 > Thanks,
 > Merle
 >
 >
 > _______________________________________________
 > FreeSWITCH-users mailing list
 > FreeSWITCH-users at lists.freeswitch.org [2]
 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > http://www.freeswitch.org
 >
 _______________________________________________
 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org [3]
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
-------------------------
Msg sent via Enterux Enterprise Email Server :
http://www.enterux.com/

Links:
------
[1] mailto:se02005-mec at yahoo.com
[2] mailto:FreeSWITCH-users at lists.freeswitch.org
[3] mailto:FreeSWITCH-users at lists.freeswitch.org
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From odermann at googlemail.com  Wed Aug 26 08:34:13 2009
From: odermann at googlemail.com (Dennis)
Date: Wed, 26 Aug 2009 17:34:13 +0200
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
Message-ID: <5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>

> Do about what? ?Your description sounds fine, what is the problem

2 people are talking and only ONE led is on? for me it sounds wrong.

the led of p1 AND p2 should be on! the led should indicate, who in the
company/group is busy/talking and who is available - i think that's
the joke about the leds.


kind regards
dennis


From tparikh at gmail.com  Wed Aug 26 08:38:53 2009
From: tparikh at gmail.com (Tapan Parikh)
Date: Wed, 26 Aug 2009 08:38:53 -0700
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
In-Reply-To: <2918.1251300669@enterux.com>
References: <2918.1251300669@enterux.com>
Message-ID: <1ecdcb6a0908260838r405dfcf5j44485101b3447505@mail.gmail.com>

I went through this a couple of weeks ago, and found that this
particular example is the most straightforward way to get things
working end-to-end.  I just chose to implement the Hello World Lua
script instead.

http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts


On Wed, Aug 26, 2009 at 8:31 AM, Mitul Limbani<mitul at enterux.com> wrote:
> Infact coming from Asterisk world is not too hard to figure out FS, but yeah
> i wont recommend a newbie that route :)
>
> Although an Asterisk dude /dudette can check out the Rosetta_Stone page on
> wiki
>
> Thanks & Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions,
> The Enterprise Linux Company (TM),
> www.enterux.com
>
>
> On Wed 26/08/09 19:35 , Bradley Brashier bjbrashier at gmail.com sent:
>
> Speaking as someone who went through this recently myself, my
> suggestion is to learn by doing. Choose some simple-sounding task you
> want to accomplish in FS, and try to carry it out. Use the wiki as a
> reference and the mailing list if you can't figure out what you need
> from that.
>
> I don't suggest you start with Asterisk or anything else, as I'd guess
> that would just confuse the issue. I came from a BIOS background, went
> directly into Freeswitch, and am so far glad I did.
>
> BB
>
> On Wed, Aug 26, 2009 at 7:25 AM, Merle J. Ebbert<se02005-mec at yahoo.com>
> wrote:
>>
>> Hi,
>>
>> I'm trying to avoid taking up a lot of peoples valuable time.
>>
>> SIP & FS have ?brought some ideas for some commercial products but I
>> need to know where to start.
>>
>> Having once written a proprietary DOS & helped with writing a RTOS, I
>> consider myself capable
>> of learning. ?I (we) just need to know where to start to come up to
>> speed rapidly.
>>
>> Is there a FreeSWITCH tutorial available?
>> Should someone new start with Asterisk and then possibly move to FS?
>>
>> Thanks,
>> Merle
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ________________________________
> Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From brian at freeswitch.org  Wed Aug 26 08:41:40 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 Aug 2009 10:41:40 -0500
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
In-Reply-To: <1ecdcb6a0908260838r405dfcf5j44485101b3447505@mail.gmail.com>
References: <2918.1251300669@enterux.com>
	<1ecdcb6a0908260838r405dfcf5j44485101b3447505@mail.gmail.com>
Message-ID: <F7D32744-B3A0-4769-BC98-02F866A77607@freeswitch.org>

This example need to be updated as the grammar format was changed to  
JSGF which is a standard grammar format.

/b

On Aug 26, 2009, at 10:38 AM, Tapan Parikh wrote:

> I went through this a couple of weeks ago, and found that this
> particular example is the most straightforward way to get things
> working end-to-end.  I just chose to implement the Hello World Lua
> script instead.
>
> http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts

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From anthony.minessale at gmail.com  Wed Aug 26 08:42:05 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 10:42:05 -0500
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <fbfa7b0c0908252244x65d19808r967a23332de6a0a0@mail.gmail.com>
References: <4A94A2E0.6040306@nowthor.com>
	<C6BA0F7E.5D757%krice@freeswitch.org>
	<fbfa7b0c0908252244x65d19808r967a23332de6a0a0@mail.gmail.com>
Message-ID: <191c3a030908260842s573de3b8ncf1ea4b76ee86232@mail.gmail.com>

But how does get_avail_gw find out it's avail?
It would have to try making the call so you would call over it to find out
it's ok to call then pass that back up to bridge?

You can try to hide the complexity but you still have to stash it somewhere.

So if you don't like looking at it consider making a script or a C mod to
hide it.

Here's one cool way you can do with just regular dialplan.


    <extension name="gateways">
      <condition field="destination_number" expression="gateway_macro">
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="bridge"
data="sofia/gateway/gw1/${dialed_number}"/>
        <action application="bridge"
data="sofia/gateway/gw2/${dialed_number}"/>
        <action application="bridge"
data="sofia/gateway/gw3/${dialed_number}"/>
        <action application="bridge"
data="sofia/gateway/gw4/${dialed_number}"/>
        <action application="bridge"
data="sofia/gateway/gw5/${dialed_number}"/>
      </condition>
    </extension>

    <extension name="myext">
      <condition field="destination_number" expression="^(whatever)$">
        <action application="set" data="dialed_number=$1"/>
        <action application="transfer" data="gateway_macro"/>
      </condition>
    </extension>

change whatever to your favorite regex
so now all you have to do is set dialed_number and transfer to gateway_macro
(you could have many of these with different names)


One way or another all your real goal seems to be is hiding the complexity
so *shrug* here's and easy way with existing tools.



On Wed, Aug 26, 2009 at 12:44 AM, Max Ivanov <ivanov.maxim at gmail.com> wrote:

> >
> > Continue_on_fail and hangup_after_bridge like tony pointed out are what
> you
> > want if you don?t want to use the | delimiting ... I use these all the
> time
> > with gateway counts > 10 just stacking additional actions for each bridge
> > line
>
> Let's imagine that I need to call 1000,1001,1002 via  gw1,gw2,gw3 by
> choosing first available. How dialplan would look like?
>
> continue_on_fail=True
> hangup_after_bridge=True
> bridge gw1/1000
> bridge gw1/1001
> bridge gw1/1002
> bridge gw2/1000
> bridge gw2/1001
> bridge gw2/1002
> bridge gw3/1000
> bridge gw3/1001
> bridge gw3/1002
>
> Is it easy to understand? From my point of view it's not. Compare to this:
> continue_on_fail=True
> hangup_after_bridge=True
> bridge ${get_avail_gw(gw1,gw2,gw3)}/1000
> bridge ${get_avail_gw(gw1,gw2,gw3)}/1001
> bridge ${get_avail_gw(gw1,gw2,gw3)}/1002
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Wed Aug 26 08:44:40 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 10:44:40 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
Message-ID: <191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>

if you are talking to a device that pretends to do session timers but really
does not, the re-invite can cause a hangup.
We have seen this with sonus many times, do you have sonus involved in your
call per chance?


On Wed, Aug 26, 2009 at 7:59 AM, Dennis <odermann at googlemail.com> wrote:

> > You have a NAT issue.
>
> as i wrote, we are quite sure, that this is a nat problem. but we have
> no idea how we could fix this. there must be a reason, why the hangup
> always comes after 120 seconds.
>
>
> >    <!--RTP port range -->
> >    <param name="rtp-start-port" value="10000"/>
> >    <param name="rtp-end-port" value="32767"/>
>
> we tested that, but this does not change anything (nothing becomes
> worse or better).
>
>
> > param name="session-timeout" value="120"
>
> this line is commented in our profile and we believe, that this has
> nothing to do with our problem.
>
>
> to tell a little bit more about our problem:
>
> under
> http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios#Scenario_2
> you can find the scenario, which is exactly like ours. we played with
> all the mentioned settings, but it does not work.
>
> to mention one thing: if we connect with xlite (softphone), we do not
> have any problems. but we have snom-voip-phones and with them we have
> the problems.
>
> the big question is: wo or what might cause/trigger the hangup? is it
> freeswitch or something else? we have a firewall (IPCop) - might there
> be a setting, which needs to be set, to avoid theses problems?
>
>
> kind regards
> dennis
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Wed Aug 26 08:52:57 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 26 Aug 2009 11:52:57 -0400
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
	<5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
Message-ID: <274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>


On Aug 26, 2009, at 11:34 AM, Dennis wrote:

>> Do about what?  Your description sounds fine, what is the problem
>
> 2 people are talking and only ONE led is on? for me it sounds wrong.
>
> the led of p1 AND p2 should be on! the led should indicate, who in the
> company/group is busy/talking and who is available - i think that's
> the joke about the leds.
>
ahh, I understand the issue now.  Please open a jira on jira.freeswitch.org 
  for this issue.

Mike



From odermann at googlemail.com  Wed Aug 26 09:10:06 2009
From: odermann at googlemail.com (Dennis)
Date: Wed, 26 Aug 2009 18:10:06 +0200
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
	<5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
	<274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>
Message-ID: <5e414ed0908260910h55122d77o5013ef1795d0c75@mail.gmail.com>

> ahh, I understand the issue now. ?Please open a jira on jira.freeswitch.org
> ?for this issue.

ok, we could not imagine, that this behavior is meant to be.

we will try to open a jira with this issue (never opened a jira before).


kind regards
dennis


From odermann at googlemail.com  Wed Aug 26 09:23:00 2009
From: odermann at googlemail.com (Dennis)
Date: Wed, 26 Aug 2009 18:23:00 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
Message-ID: <5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>

> if you are talking to a device that pretends to do session timers but really
> does not, the re-invite can cause a hangup.
> We have seen this with sonus many times, do you have sonus involved in your
> call per chance?

hi anthony,

this is what we found on the snom website about the session timer
settings: "If SIP Session Timer Support is enabled, this option
specifies the SIP session timer in seconds. For instance, a Re-INVITE
will be sent after 50% of its value has elapsed." our setting is 3600.

is session timer and re-invite helpful or can we disable it? behaps
disabling works?

is it possible, that snom does not support a REAL session timer?

sonus is not involved.


kind regards
dennis


From MPeace at edcogroupinc.com  Wed Aug 26 09:20:49 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Wed, 26 Aug 2009 11:20:49 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A622D22BE@ESNEXS2.edcogroup.net>
References: <5DF610F9-6EB5-43CC-BB4F-3513C78F608F@freeswitch.org>,
	<69D652F49932C34199968DFB8AEAABA33A622D22BE@ESNEXS2.edcogroup.net>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A622F13DB@ESNEXS2.edcogroup.net>

OK I edited the openzap.conf and the open.conf.xml and added the gateway info from the Wiki for dilaing out through a gateway. Still no love but I also noticed this when starting Freeswitch:

2009-08-26 11:04:53 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_openzap.so
**/usr/local/freeswitch/mod/mod_openzap.so: cannot open shared object file: No such file or directory**


Sure enough the openzap.so is not there, does the fact I used the "wanpipe mode" have anything to do with this ? I have searched the drive and it is not there. 
________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike Peace [MPeace at edcogroupinc.com]
Sent: Wednesday, August 26, 2009 8:58 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

I found and edited openzap.cong file, the openzap.conf.xml file is in the correct dir but the only example I can find is for a Euro E1 setup not a US T1. I

I have turned up data T1 circuits for WAN links before but not telco stuff. The provider is Qwest, I?ll have to ask about the NI2 , DMS question.

Mike

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins
Sent: Wednesday, August 26, 2009 3:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


I have looked at that but I am confused on which files need to be edited. Since I have already installed in Wanpipe mode with the Sangoma card I skipped straight to the Wanpipe section. It mentions setting the [span wanpie PRI_1] etc in the openzap.conf then further down it mentions editing the autoload_configs/openzap.conf.xml with a completely different format of config.

Are these the same file? If not where does the first one mentioned reside?

These are two different files. The openzap.conf file is in the freeswitch/conf directory. It is for configuring OZ to talk to the Sangoma card. The openzap.conf.xml file is for configuring the actual openzap module in FreeSWITCH.

The specs you listed on the T1 you have indicate that it's a PRI. However, you need to know which protocol, like NI2 or DMS. Have you ever done a PRI turn up? Who is the carrier?

-MC
=-=-=-=-=-=-=-=-=-=-=-=-=-=-EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.


From foxb at abv.bg  Wed Aug 26 09:37:16 2009
From: foxb at abv.bg (Hristo Benev)
Date: Wed, 26 Aug 2009 19:37:16 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <2089188486.120014.1251304644860.JavaMail.apache@mail22.abv.bg>

 Hello

I followed the tutorial
http://wiki.freeswitch.org/wiki/SBC_Setup

I have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not route

Where to look for problems?

Here is the debug:

2009-08-26 20:21:42.332154 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context default
Dialplan: sofia/internal/1001 at 10.10.10.10 parsing [default->LOOKUP_ROUTE] continue=false
Dialplan: sofia/internal/1001 at 10.10.10.10 Regex (PASS) [LOOKUP_ROUTE] destination_number(1000) =~ /(\d+)$/ break=on-false
Dialplan: sofia/internal/1001 at 10.10.10.10 Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1001 at 10.10.10.10 Action set(continue_on_fail=true)
Dialplan: sofia/internal/1001 at 10.10.10.10 Action export(sip_h_X-ROUTE=LOOKUP)
Dialplan: sofia/internal/1001 at 10.10.10.10 Action bridge(sofia/internal/${destination_number}@127.0.0.1:5062)
Dialplan: sofia/internal/1001 at 10.10.10.10 Action set(ROUTE_GW=${sip_redirect_contact_user_0})
Dialplan: sofia/internal/1001 at 10.10.10.10 Action set(AREA=${sip_redirect_contact_user_0})
Dialplan: sofia/internal/1001 at 10.10.10.10 Action transfer(${destination_number} XML ROUTING)
2009-08-26 20:21:42.334579 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/1001 at 10.10.10.10) State Change CS_ROUTING -> CS_EXECUTE
2009-08-26 20:21:42.334579 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1001 at 10.10.10.10 [BREAK]
2009-08-26 20:21:42.334579 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/1001 at 10.10.10.10) State ROUTING going to sleep
2009-08-26 20:21:42.335292 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1001 at 10.10.10.10) Running State Change CS_EXECUTE
2009-08-26 20:21:42.335292 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1001 at 10.10.10.10) State EXECUTE
2009-08-26 20:21:42.335292 [DEBUG] mod_sofia.c:173 sofia/internal/1001 at 10.10.10.10 SOFIA EXECUTE
2009-08-26 20:21:42.335292 [DEBUG] switch_core_state_machine.c:151 sofia/internal/1001 at 10.10.10.10 Standard EXECUTE
EXECUTE sofia/internal/1001 at 10.10.10.10 set(hangup_after_bridge=true)
2009-08-26 20:21:42.336509 [DEBUG] mod_dptools.c:748 sofia/internal/1001 at 10.10.10.10 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1001 at 10.10.10.10 set(continue_on_fail=true)
2009-08-26 20:21:42.337353 [DEBUG] mod_dptools.c:748 sofia/internal/1001 at 10.10.10.10 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1001 at 10.10.10.10 export(sip_h_X-ROUTE=LOOKUP)
2009-08-26 20:21:42.338352 [DEBUG] mod_dptools.c:886 EXPORT [sip_h_X-ROUTE]=[LOOKUP]
EXECUTE sofia/internal/1001 at 10.10.10.10 bridge(sofia/internal/1000 at 127.0.0.1:5062)
2009-08-26 20:21:42.339309 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1000 at 127.0.0.1:5062 [8a588e6a-925c-11de-85dd-15dc0a06983f]
2009-08-26 20:21:42.339309 [DEBUG] mod_sofia.c:2811 (sofia/internal/1000 at 127.0.0.1:5062) State Change CS_NEW -> CS_INIT
2009-08-26 20:21:42.340376 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1000 at 127.0.0.1:5062 [BREAK]
2009-08-26 20:21:42.341175 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1000 at 127.0.0.1:5062) Running State Change CS_INIT
2009-08-26 20:21:42.341175 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/1000 at 127.0.0.1:5062) State INIT
2009-08-26 20:21:42.341175 [DEBUG] mod_sofia.c:83 sofia/internal/1000 at 127.0.0.1:5062 SOFIA INIT
2009-08-26 20:21:42.344378 [DEBUG] mod_sofia.c:111 (sofia/internal/1000 at 127.0.0.1:5062) State Change CS_INIT -> CS_ROUTING
2009-08-26 20:21:42.344378 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1000 at 127.0.0.1:5062 [BREAK]
2009-08-26 20:21:42.344378 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/1000 at 127.0.0.1:5062) State INIT going to sleep
2009-08-26 20:21:42.344378 [DEBUG] sofia.c:3289 Channel sofia/internal/1000 at 127.0.0.1:5062 entering state [calling][0]
2009-08-26 20:21:42.344378 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1000 at 127.0.0.1:5062) Running State Change CS_ROUTING
2009-08-26 20:21:42.345314 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/1000 at 127.0.0.1:5062) State ROUTING
2009-08-26 20:21:42.345314 [DEBUG] mod_sofia.c:130 sofia/internal/1000 at 127.0.0.1:5062 SOFIA ROUTING
2009-08-26 20:21:42.345314 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/1000 at 127.0.0.1:5062) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-08-26 20:21:42.345314 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1000 at 127.0.0.1:5062 [BREAK]
2009-08-26 20:21:42.345314 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/1000 at 127.0.0.1:5062) State ROUTING going to sleep
2009-08-26 20:21:42.346255 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1000 at 127.0.0.1:5062) Running State Change CS_CONSUME_MEDIA
2009-08-26 20:21:42.346255 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/1000 at 127.0.0.1:5062) State CONSUME_MEDIA
2009-08-26 20:21:42.349173 [DEBUG] sofia.c:3289 Channel sofia/internal/1000 at 127.0.0.1:5062 entering state [calling][0]
2009-08-26 20:21:42.350242 [DEBUG] sofia.c:3289 Channel sofia/internal/1000 at 127.0.0.1:5062 entering state [terminated][503]
2009-08-26 20:21:42.350242 [NOTICE] sofia.c:3849 Hangup sofia/internal/1000 at 127.0.0.1:5062 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2009-08-26 20:21:42.351172 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]
2009-08-26 20:21:42.351172 [INFO] mod_dptools.c:2093 Originate Failed.  Cause: NORMAL_TEMPORARY_FAILURE
2009-08-26 20:21:42.351172 [DEBUG] mod_dptools.c:2120 Continue on fail [true]:  Cause: NORMAL_TEMPORARY_FAILURE
EXECUTE sofia/internal/1001 at 10.10.10.10 set(ROUTE_GW=PEER-01)
2009-08-26 20:21:42.352234 [DEBUG] mod_dptools.c:748 sofia/internal/1001 at 10.10.10.10 SET [ROUTE_GW]=[PEER-01]
2009-08-26 20:21:42.352234 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/1000 at 127.0.0.1:5062 [KILL]
2009-08-26 20:21:42.352234 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1000 at 127.0.0.1:5062 [BREAK]
EXECUTE sofia/internal/1001 at 10.10.10.10 set(AREA=PEER-01)
2009-08-26 20:21:42.353179 [DEBUG] mod_dptools.c:748 sofia/internal/1001 at 10.10.10.10 SET [AREA]=[PEER-01]
EXECUTE sofia/internal/1001 at 10.10.10.10 transfer(1000 XML ROUTING)
2009-08-26 20:21:42.355174 [DEBUG] switch_ivr.c:1343 (sofia/internal/1001 at 10.10.10.10) State Change CS_EXECUTE -> CS_ROUTING
2009-08-26 20:21:42.355174 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1001 at 10.10.10.10 [BREAK]
2009-08-26 20:21:42.355174 [DEBUG] switch_ivr.c:1347 sofia/internal/1001 at 10.10.10.10 receive message [TRANSFER]
2009-08-26 20:21:42.355174 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/1001 at 10.10.10.10 [BREAK]
2009-08-26 20:21:42.355174 [NOTICE] switch_ivr.c:1349 Transfer sofia/internal/1001 at 10.10.10.10 to XML[1000 at ROUTING]
2009-08-26 20:21:42.356405 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1001 at 10.10.10.10) State EXECUTE going to sleep
2009-08-26 20:21:42.356405 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1001 at 10.10.10.10) Running State Change CS_ROUTING
2009-08-26 20:21:42.357172 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/1000 at 127.0.0.1:5062) State CONSUME_MEDIA going to sleep
2009-08-26 20:21:42.357172 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1000 at 127.0.0.1:5062) Running State Change CS_HANGUP
2009-08-26 20:21:42.357172 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/1001 at 10.10.10.10) State ROUTING
2009-08-26 20:21:42.357172 [DEBUG] mod_sofia.c:130 sofia/internal/1001 at 10.10.10.10 SOFIA ROUTING
2009-08-26 20:21:42.357172 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001 at 10.10.10.10 Standard ROUTING
2009-08-26 20:21:42.357172 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
Dialplan: sofia/internal/1001 at 10.10.10.10 parsing [ROUTING->PEER_01] continue=false
2009-08-26 20:21:42.358179 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/1000 at 127.0.0.1:5062) State HANGUP
Dialplan: sofia/internal/1001 at 10.10.10.10 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
2009-08-26 20:21:42.358179 [INFO] switch_core_state_machine.c:136 No Route, Aborting
2009-08-26 20:21:42.358179 [DEBUG] mod_sofia.c:306 sofia/internal/1000 at 127.0.0.1:5062 Overriding SIP cause 503 with 503 from the other leg
2009-08-26 20:21:42.359202 [NOTICE] switch_core_state_machine.c:137 Hangup sofia/internal/1001 at 10.10.10.10 [CS_ROUTING] [NO_ROUTE_DESTINATION]



From brian at freeswitch.org  Wed Aug 26 09:47:37 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 Aug 2009 11:47:37 -0500
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
In-Reply-To: <2089188486.120014.1251304644860.JavaMail.apache@mail22.abv.bg>
References: <2089188486.120014.1251304644860.JavaMail.apache@mail22.abv.bg>
Message-ID: <94B8E08E-2537-4FA4-9047-8904975D7BFB@freeswitch.org>

We do not blindly follow 302's as that is a dangerous thing to do.   
You have to process all 302's in the dialplan.

Set this on your sofia profile <param name="manual-redirect"  
value="true"/>

You can set these variables sip_redirect_profile,  
sip_redirect_context, sip_redirect_dialplan,

When a redirect happens you get these variables - sip_redirect_contact_ 
%d, sip_redirected_to, sip_redirect_contact_user_%d,  
sip_redirect_contact_host_%d, sip_redirect_contact_params_%d,  
sip_redirect_dialstring_%d, sip_redirect_dialstring, sip_redirected_by

Then its up to you to process the redirect in your dialplan, If you  
don't set the sip_redirect_context then it'll default to redirected  
context and XML as the dialplan.

/b


On Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:

> Hello
>
> I followed the tutorial
> http://wiki.freeswitch.org/wiki/SBC_Setup
>
> I have following problem when I dial 1000 Kamalio reports 302, but  
> freeswitch does not route
>
> Where to look for problems?

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From anthony.minessale at gmail.com  Wed Aug 26 09:53:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 11:53:03 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
Message-ID: <191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>

Is the snom firmware up to the latest?
I believe session timers should work properly with snom?
you can try disabling it but make sure you have your firmware to latest.


On Wed, Aug 26, 2009 at 11:23 AM, Dennis <odermann at googlemail.com> wrote:

> > if you are talking to a device that pretends to do session timers but
> really
> > does not, the re-invite can cause a hangup.
> > We have seen this with sonus many times, do you have sonus involved in
> your
> > call per chance?
>
> hi anthony,
>
> this is what we found on the snom website about the session timer
> settings: "If SIP Session Timer Support is enabled, this option
> specifies the SIP session timer in seconds. For instance, a Re-INVITE
> will be sent after 50% of its value has elapsed." our setting is 3600.
>
> is session timer and re-invite helpful or can we disable it? behaps
> disabling works?
>
> is it possible, that snom does not support a REAL session timer?
>
> sonus is not involved.
>
>
> kind regards
> dennis
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From pbd at suspiria.net  Wed Aug 26 10:28:50 2009
From: pbd at suspiria.net (Public Dump)
Date: Wed, 26 Aug 2009 19:28:50 +0200
Subject: [Freeswitch-users] delay buildup in conference
Message-ID: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>

When running conferences with users dialed in from a PSTN gateway (SIP) and directly from remote SIP endpoints there is an ever longer buildup in delay, reaching up to multiple seconds. Is there any way to limit the delay ?



I am not 100% sure whether the delays is caused by the SIP jitter buffer of freeswitch or directly by the conference module.



Any advice?



Thanks

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From tculjaga at gmail.com  Wed Aug 26 10:42:19 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 26 Aug 2009 19:42:19 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <681a20520908260453q57877b1eg8f0949061cd129ee@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com>
	<65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com>
	<7b197bef0908260125m56c3d681l26cb18a5e34aaa33@mail.gmail.com>
	<65d96fc80908260129xebbf460raffaa38c72c3281a@mail.gmail.com>
	<681a20520908260453q57877b1eg8f0949061cd129ee@mail.gmail.com>
Message-ID: <65d96fc80908261042l42437ed9xc4fab87b757adc80@mail.gmail.com>

Guys you made a monster!!


so, i moved the machine to 64bit CentOS 5.3... recompiled the latest trunk
and did my tests again.

The old tests on 32bit debian 5 on the same hardware shown a CPS rate of 120
as 75 - 80% CPU.... and after some time on that 120 CPS rate the CPU goes to
100% without any chance FS recovers at all.
New tests on 64bit CentOS shown a monster.... 400 CPS rate at 75% CPU....
during the tests FS was really stable and responsive... i placed few calls
that went through as a charm :).


Also, i didn't optimize the machine at all ... as it is after CentOS
install!.... not even db files are on ramdisk.

What did it really happen? .. did you guys change something in the trunk
overnight or it is just moving to CentOS 64bit that boosted drastically?




Here are some details:

?nmon?12a??????[H for help]???Hostname=l01sipindir2?Refresh= 2secs
???19:17.48??????????????????????????????????????????????????????
? CPU
+-------------------------------------------------------------------------+
?
?100%-|
|
?
? 95%-|
|
?
? 90%-|
|
?
? 85%-|
|
?
? 80%-|  | w              s  w      www  w        s   sw    s    s
s                                                     ?
? 75%-|
|ssssssssssssssssssssswssssssssssssssssssssssssssssssssssssssssssssss
?
? 70%-|
+sssssssssssssssssssssssssssssssUssssssssUsssUssssssssssUssssssssssss
?
? 65%-|
|UUUUUsUsUUUUUUUUUUUUUsUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUsUUU
?
? 60%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 55%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 50%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 45%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 40%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUw
?
? 35%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUs
?
? 30%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 25%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 20%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 15%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 10%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
?  5%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
?
+-------------------------------------------------------------------------+
?
? CPU Utilisation
??????????????????????????????????????????????????????????????????????????????????????????????????????????????????
?
+-------------------------------------------------+
?
?CPU  User%  Sys% Wait% Idle|0          |25         |50          |75
100|                                                    ?
? 1   1.0   0.5   0.0   98.5|
>          |                                                    ?
? 2   1.5   1.0   0.0   97.5|
>  |                                                    ?
?
+-------------------------------------------------+
?
?Avg  1.2   0.5   0.0   98.3|
>         |                                                    ?
?
+-------------------------------------------------+
?
? Disk I/O ?????(/proc/diskstats)????????all data is Kbytes per
second??????????????????????????????????????????????????????????????
?DiskName Busy  Read WriteKB|0          |25         |50          |75
100|                                                    ?
?iss/c0d0   0%    0.0    0.0|
>   |                                                    ?
?s/c0d0p1   0%    0.0
0.0|>
|                                                    ?
?s/c0d0p2   0%    0.0    0.0|
>  |                                                    ?
?dm-0       0%    0.0    0.0|
>  |                                                    ?
?dm-1       0%    0.0
0.0|>
|                                                    ?
????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????







------------------------------ Scenario Screen -------- [1-9]: Change Screen
--
  Call-rate(length)   Port   Total-time  Total-calls  Remote-host
  40.0(0 ms)/0.100s   5060     488.20 s       193555  10.4.4.252:5060(UDP)

  402 new calls during 1.002 s period    0 ms scheduler resolution
  3 calls (limit 4000)                   Peak was 53 calls, after 351 s
  0 Running, 13216 Paused, 670 Woken up
  0 dead call msg (discarded)            0 out-of-call msg
(discarded)
  3 open sockets

                                 Messages  Retrans   Timeout
Unexpected-Msg
      INVITE ---------->  B-RTD1 193553    0         0
         100 <----------  E-RTD1 193553    0         0         0
         302 <----------  E-RTD2 193552    0         0         0
         ACK ---------->         193552    0
------ [+|-|*|/]: Adjust rate ---- [q]: Soft exit ---- [p]: Pause traffic
-----

------------------------------ Scenario Screen -------- [1-9]: Change Screen
--
  Call-rate(length)   Port   Total-time  Total-calls  Remote-host
  40.0(0 ms)/0.100s   5060     489.10 s       193917  10.4.4.252:5060(UDP)

  362 new calls during 0.906 s period    1 ms scheduler resolution
  2 calls (limit 4000)                   Peak was 53 calls, after 351 s
  0 Running, 13215 Paused, 623 Woken up
  0 dead call msg (discarded)            0 out-of-call msg
(discarded)
  3 open sockets

                                 Messages  Retrans   Timeout
Unexpected-Msg
      INVITE ---------->  B-RTD1 193917    0         0
         100 <----------  E-RTD1 193917    0         0         0
         302 <----------  E-RTD2 193915    0         0         0
         ACK ---------->         193915    0
------------------------------ Test Terminated
--------------------------------


----------------------------- Statistics Screen ------- [1-9]: Change Screen
--
  Start Time             | 2009-08-26   19:09:34:575
1251306574.575684
  Last Reset Time        | 2009-08-26   19:17:42:779
1251307062.779468
  Current Time           | 2009-08-26   19:17:43:685
1251307063.685281
-------------------------+---------------------------+--------------------------
  Counter Name           | Periodic value            | Cumulative value
-------------------------+---------------------------+--------------------------
  Elapsed Time           | 00:00:00:905              |
00:08:09:109
  Call Rate              |  400.000 cps              |  396.470
cps
-------------------------+---------------------------+--------------------------
  Incoming call created  |        0                  |
0
  OutGoing call created  |      362                  |
193917
  Total Call created     |                           |
193917
  Current Call           |        2
|
-------------------------+---------------------------+--------------------------
  Successful call        |      363                  |
193915
  Failed call            |        0                  |
0
-------------------------+---------------------------+--------------------------
  Response Time 1        | 00:00:00:001              |
00:00:00:000
  Response Time 2        | 00:00:00:010              |
00:00:00:008
  Call Length            | 00:00:00:010              |
00:00:00:008
------------------------------ Test Terminated
--------------------------------



...i didn't beleive to SIPp and i went to FS console issuing status command
to conferm the results.


freeswitch at l01sipindir2.ot.hr> status
API CALL [status()] output:
UP 0 years, 0 days, 0 hours, 8 minutes, 13 seconds, 703 milliseconds, 971
microseconds
183382 session(s) since startup
1 session(s) 410/800
8000 session(s) max

freeswitch at l01sipindir2.ot.hr> status
API CALL [status()] output:
UP 0 years, 0 days, 0 hours, 8 minutes, 15 seconds, 109 milliseconds, 891
microseconds
183944 session(s) since startup
1 session(s) 401/800
8000 session(s) max

freeswitch at l01sipindir2.ot.hr> status
API CALL [status()] output:
UP 0 years, 0 days, 0 hours, 8 minutes, 16 seconds, 139 milliseconds, 412
microseconds
184356 session(s) since startup
2 session(s) 389/800
8000 session(s) max

freeswitch at l01sipindir2.ot.hr> status
API CALL [status()] output:
UP 0 years, 0 days, 0 hours, 8 minutes, 17 seconds, 62 milliseconds, 16
microseconds
184717 session(s) since startup
6 session(s) 410/800
8000 session(s) max

freeswitch at l01sipindir2.ot.hr> status
API CALL [status()] output:
UP 0 years, 0 days, 0 hours, 8 minutes, 35 seconds, 150 milliseconds, 253
microseconds
191959 session(s) since startup
1 session(s) 400/800
8000 session(s) max

freeswitch at l01sipindir2.ot.hr> status
API CALL [status()] output:
UP 0 years, 0 days, 0 hours, 8 minutes, 36 seconds, 892 milliseconds, 672
microseconds
192657 session(s) since startup
1 session(s) 393/800
8000 session(s) max



On Wed, Aug 26, 2009 at 1:53 PM, Dmitry Kadantsev <kadantsev.d at gmail.com>wrote:

> Hi all,
>
> is there same situation with FS for Windows? I mean 64bit is more
> preferable than 32bit, isn't it?
>
> Any performance test on Win 32/64 were done?
>
> --
> Best regards,
> Dmitry Kadantsev
>
>
>
> On Wed, Aug 26, 2009 at 10:29 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> intanto e il centos che si sta installando :)
>>
>> grazie.
>>
>> T.
>>
>>
>> On Wed, Aug 26, 2009 at 10:25 AM, Giovanni Maruzzelli <
>> gmaruzz at celliax.org> wrote:
>>
>>> netbook remix
>>>
>>>
>>> joking! Server 64bit :-)
>>>
>>> -gm
>>>
>>>
>>>
>>> On Wed, Aug 26, 2009 at 10:08 AM, Tihomir Culjaga<tculjaga at gmail.com>
>>> wrote:
>>> > Hi Giovanny,
>>> >
>>> > regarding ubuntu, did you mean 8.04 server or desktop ?
>>> >
>>> >
>>> > On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli <
>>> gmaruzz at celliax.org>
>>> > wrote:
>>> >>
>>> >> Definitely go for 64 bit OS.
>>> >>
>>> >> If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
>>> >> used both for development and for heavy duty production.
>>> >>
>>> >> Also Ubuntu 8.04 is good.
>>> >>
>>> >> Other versions/distros are less used by the community.
>>> >>
>>> >> Adding RAM and CPUs helps to scale up.
>>> >>
>>> >> -gm
>>> >>
>>> >>
>>> >>
>>> >> Sincerely,
>>> >>
>>> >> Giovanni Maruzzelli
>>> >> Cell : +39-347-2665618
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com>
>>> >> wrote:
>>> >> > Hey Giovanni,
>>> >> >
>>> >> > thanks for the tip... indeed the db files were heavily used
>>> regardless
>>> >> > if i
>>> >> > started freeswitch with nosql option (freeswitch -nosql)... FS was
>>> not
>>> >> > writing anything into that files ... instead it was just accessing
>>> >> > it....
>>> >> > This behaviour leads to a waste of 40% CPU time... waiting for other
>>> >> > processes (mainly disk access) to finish!!!
>>> >> >
>>> >> > I moved freeswitch/db/ to a ramdisk and the performance got a boost
>>> to
>>> >> > 140
>>> >> > CPS with a CPU load of 80%. I was keeping the machine for a while
>>> (20 -
>>> >> > 30
>>> >> > minutes) on that rate when i sow CPU suddenly went to 100% and FS
>>> >> > becoming
>>> >> > irresponsive :).
>>> >> >
>>> >> >
>>> >> > What can be wrong?
>>> >> > What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should
>>> not
>>> >> > cross?
>>> >> > What fine tuning do we need in order to asure a long high load run?
>>> >> >
>>> >> >
>>> >> >
>>> >> > Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it
>>> have
>>> >> > sense
>>> >> > to move my OS to 64 bit? ... will FS gain more preformance ?... I
>>> mean
>>> >> > will
>>> >> > FS perofomr drastically better 20%+ ?
>>> >> >
>>> >> >
>>> >> > Tihomir.
>>> >> >
>>> >> >
>>> >> > On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
>>> >> > <gmaruzz at celliax.org>
>>> >> > wrote:
>>> >> >>
>>> >> >> Maybe your load comes from disk access?
>>> >> >>
>>> >> >> Try putting the sql and log directories on a ramdisk.
>>> >> >>
>>> >> >> OTH,
>>> >> >>
>>> >> >> -giovanni
>>> >> >>
>>> >> >> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<
>>> tculjaga at gmail.com>
>>> >> >> wrote:
>>> >> >> > Hello,
>>> >> >> >
>>> >> >> > i'm trying to use freeswitch as a redirecting server meaning FS
>>> has
>>> >> >> > to
>>> >> >> > receive an INVITE and according to some rules it will redirect
>>> calls
>>> >> >> > to
>>> >> >> > other destinations.
>>> >> >> >
>>> >> >> >
>>> >> >> > CALLING_USER                FREESWITCH
>>> >> >> > SOMEWHERE
>>> >> >> >
>>> >> >> > INVITE ------------------------------->
>>> >> >> >            <------------------------------ 100 Trying
>>> >> >> >            <------------------------------ 302 Moved Temporary
>>> >> >> > ACK    ------------------------------->
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> INVITE--------------------------------------------------------------------------------->
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > Well, wverything works well except i have perfromance issues ....
>>> on
>>> >> >> > my
>>> >> >> > HW
>>> >> >> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
>>> >> >> > Temporary).
>>> >> >> > When
>>> >> >> > i increase the rate, FS starts delaying 302 response. Right at 50
>>> CPS
>>> >> >> > i
>>> >> >> > see
>>> >> >> > "calls" being build up in FS and the delay begining to grow.
>>> >> >> >
>>> >> >> > When i observe the machine, load average is almost nothing (load
>>> >> >> > average:
>>> >> >> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one
>>> thread
>>> >> >> > taking
>>> >> >> > most load... all others are just sitting there with 1-5 % CPU
>>> time.
>>> >> >> > This looks to me as FS handles 302 messages in a single
>>> thread?!?!
>>> >> >> >
>>> >> >> >
>>> >> >> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>>> >> >> >
>>> >> >> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41,
>>> >> >> > 0.61,
>>> >> >> > 0.60
>>> >> >> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0
>>> zombie
>>> >> >> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,
>>> >> >> > 0.2%si,
>>> >> >> > 0.0%st
>>> >> >> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k
>>> >> >> > buffers
>>> >> >> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k
>>> >> >> > cached
>>> >> >> >
>>> >> >> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
>>> >> >> > COMMAND
>>> >> >> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
>>> >> >> > freeswitch
>>> >> >> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
>>> >> >> > freeswitch
>>> >> >> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
>>> >> >> > freeswitch
>>> >> >> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
>>> >> >> > freeswitch
>>> >> >> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
>>> >> >> > freeswitch
>>> >> >> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
>>> >> >> > freeswitch
>>> >> >> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
>>> >> >> > freeswitch
>>> >> >> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
>>> >> >> > freeswitch
>>> >> >> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
>>> >> >> > freeswitch
>>> >> >> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50
>>> >> >> > freeswitch
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > cat /proc/cpuinfo
>>> >> >> > processor       : 0
>>> >> >> > vendor_id       : GenuineIntel
>>> >> >> > cpu family      : 6
>>> >> >> > model           : 15
>>> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
>>> >> >> > stepping        : 6
>>> >> >> > cpu MHz         : 2333.560
>>> >> >> > cache size      : 4096 KB
>>> >> >> > physical id     : 0
>>> >> >> > siblings        : 2
>>> >> >> > core id         : 0
>>> >> >> > cpu cores       : 2
>>> >> >> > apicid          : 0
>>> >> >> > initial apicid  : 0
>>> >> >> > fdiv_bug        : no
>>> >> >> > hlt_bug         : no
>>> >> >> > f00f_bug        : no
>>> >> >> > coma_bug        : no
>>> >> >> > fpu             : yes
>>> >> >> > fpu_exception   : yes
>>> >> >> > cpuid level     : 10
>>> >> >> > wp              : yes
>>> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep
>>> mtrr
>>> >> >> > pge
>>> >> >> > mca
>>> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>>> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
>>> >> >> > ssse3
>>> >> >> > cx16
>>> >> >> > xtpr dca lahf_lm
>>> >> >> > bogomips        : 4670.78
>>> >> >> > clflush size    : 64
>>> >> >> > power management:
>>> >> >> >
>>> >> >> > processor       : 1
>>> >> >> > vendor_id       : GenuineIntel
>>> >> >> > cpu family      : 6
>>> >> >> > model           : 15
>>> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
>>> >> >> > stepping        : 6
>>> >> >> > cpu MHz         : 2333.560
>>> >> >> > cache size      : 4096 KB
>>> >> >> > physical id     : 0
>>> >> >> > siblings        : 2
>>> >> >> > core id         : 1
>>> >> >> > cpu cores       : 2
>>> >> >> > apicid          : 1
>>> >> >> > initial apicid  : 1
>>> >> >> > fdiv_bug        : no
>>> >> >> > hlt_bug         : no
>>> >> >> > f00f_bug        : no
>>> >> >> > coma_bug        : no
>>> >> >> > fpu             : yes
>>> >> >> > fpu_exception   : yes
>>> >> >> > cpuid level     : 10
>>> >> >> > wp              : yes
>>> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep
>>> mtrr
>>> >> >> > pge
>>> >> >> > mca
>>> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>>> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
>>> >> >> > ssse3
>>> >> >> > cx16
>>> >> >> > xtpr dca lahf_lm
>>> >> >> > bogomips        : 4666.82
>>> >> >> > clflush size    : 64
>>> >> >> > power management:
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > uname -a
>>> >> >> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC
>>> 2009
>>> >> >> > i686
>>> >> >> > GNU/Linux
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > Of course, i've tuned the machine up
>>> >> >> >
>>> >> >> > ulimit -c unlimited
>>> >> >> > ulimit -d unlimited
>>> >> >> > ulimit -f unlimited
>>> >> >> > ulimit -i unlimited
>>> >> >> > ulimit -n 999999
>>> >> >> > ulimit -q unlimited
>>> >> >> > ulimit -u unlimited
>>> >> >> > ulimit -v unlimited
>>> >> >> > ulimit -x unlimited
>>> >> >> > ulimit -s 240
>>> >> >> > ulimit -l unlimited
>>> >> >> > ulimit -a
>>> >> >> >
>>> >> >> >
>>> >> >> > Started FS with minimum modules but still 40 CPS seems to be the
>>> >> >> > limit.
>>> >> >> >
>>> >> >> >
>>> >> >> > So, is there any way to improve performance?
>>> >> >> >
>>> >> >> >
>>> >> >> > Tihomir.
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > _______________________________________________
>>> >> >> > FreeSWITCH-users mailing list
>>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >
>>> >> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> > http://www.freeswitch.org
>>> >> >> >
>>> >> >> >
>>> >> >>
>>> >> >> _______________________________________________
>>> >> >> FreeSWITCH-users mailing list
>>> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >>
>>> >> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From csa at nowthor.com  Wed Aug 26 10:48:11 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Wed, 26 Aug 2009 13:48:11 -0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <191c3a030908260825n73d6ffc2kbc87296197e72c90@mail.gmail.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>	<4A93EFE1.6010007@nowthor.com>	<191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>	<4A94A2E0.6040306@nowthor.com>
	<191c3a030908260825n73d6ffc2kbc87296197e72c90@mail.gmail.com>
Message-ID: <4A95755B.9020201@nowthor.com>

Anthony Minessale wrote:
> Unfortunately it's still a problem because the core bridge/originate 
> mechanism does not parse XML it's only parses the dial string markup. 

Wouldn't it be possible to "translate" the markup to a dial string prior 
to hitting the core mechanism?

> In the end you must end up with one string of application and one 
> string of data.
>

How about something like

<action application="bridge"><data> ... </data></action>

being somewhat* equivalent to

<action application="bridge" data=" ... ">?

(* somewhat to be properly define later)

> I am very strict about maintaining scope boundaries between abstract 
> concepts.  I think it's the number 1 breakdown that led to asterisk 
> getting so messed up over the years and I want to avoid that same plague.

I totally agree with you on this.

> I hope you understand and if you want to continue to ponder some more 
> functionality I welcome it.

Don't worry, I rarely stop having ideas. Feel free to shoot them down at 
will! :)

Carlos



From msc at freeswitch.org  Wed Aug 26 10:50:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Aug 2009 10:50:31 -0700
Subject: [Freeswitch-users] HOW-TO: being on "trunk" of FreePBX,
	starting 	from the ISO
In-Reply-To: <7b197bef0908260608l2c3f454cn8c2b8a12bc3d30ce@mail.gmail.com>
References: <7b197bef0908260608l2c3f454cn8c2b8a12bc3d30ce@mail.gmail.com>
Message-ID: <87f2f3b90908261050i25a6fbeeke30d21145d679616@mail.gmail.com>

On Wed, Aug 26, 2009 at 6:08 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Instructions for being on "trunk" of FreePBX, starting from the ISO
> (cut and paste to the ssh console after ISO install):
>
> /etc/init.d/httpd stop
> cd /var/www/html
> mv freepbx freepbx-originale
> svn co http://www.freepbx.org/v3/svn/trunk/ freepbx
> chown -R apache.apache freepbx
> ln -s freepbx  freepbx-v3
> cd freepbx-v3/
> ln -s freepbx  freepbx-v3
> /etc/init.d/httpd start
>
> then browse to:
>
> http://192.168.0.12/freepbx-v3/index.php/installer
>
> it will work! :-)


Giovanni, you're such a swell guy! :P
-MC
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From aep.lists at it46.se  Wed Aug 26 11:25:22 2009
From: aep.lists at it46.se (Alberto Escudero-Pascual (lists))
Date: Wed, 26 Aug 2009 20:25:22 +0200
Subject: [Freeswitch-users] Internal Profile sends RTP to external IP
 when using two network interfaces
In-Reply-To: <56E6F9F8-82FB-4A4D-9B7A-44D9BE2BDE43@freeswitch.org>
References: <46b96649f81b0bfcb111928a55450211.squirrel@correo.nodo50.org>
	<3694FC90-3FD1-4F36-8CA9-A3C18EBDEF34@freeswitch.org>
	<7682ba3b2cbb1cf578dcc2226e7d4787.squirrel@correo.nodo50.org>
	<56E6F9F8-82FB-4A4D-9B7A-44D9BE2BDE43@freeswitch.org>
Message-ID: <131017dbd9f5c898b164231856682f14.squirrel@correo.nodo50.org>


Yes, I did fix the internal IP in the internal profile overwritting the
local_ipv4 var in sip-ip and rtp-ip. When the internal clients

>sofia status profile internet


User:       	1000 at 216.82.231.69
Contact:    	"user"
<sip:1000 at 10.0.46.51;user=phone;fs_nat=yes;fs_path=sip%3A1000%4010.0.46.51%3A5060%3Buser%3Dphone>
Agent:      	Grandstream BT110 1.0.8.12
Status:     	Registered(UDP-NAT)(unknown) EXP(2009-08-26 22:22:32)
Host:       	open46
IP:         	10.0.46.51
Instead of

User: 1000 at 10.0.46.51


I can see SIP/SDP messages announcing the external IP and port 5060 in the
INVITE.


It is a very old phone... will try with different gadgets tomorrow.

/aep
-- 
Stopping junk mailers is good for the environment

> Well actually you do have issues you're going to have to specify the
> rtp-ip and sip-ip on BOTH profiles yourself because you're config is
> currently putting your external IP into the internal's settings.
> Please correct that and I'm sure it'll work then.
>
> /b
>
> On Aug 26, 2009, at 9:18 AM, Alberto Escudero-Pascual (lists) wrote:
>
>>
>>
>> ASCII art follows
>>
>> voip-phones..... 10.0.46.1 [ FS ] 216.82.231.69 ----> Internet
>>
>>
>> I have two profiles binded to two different IPs
>>
>> internal is binded to 10.0.46.1 port 5060 /24 network
>> external is binded to 216.82.231.69 port 5080
>>
>> My IP phones are in the 10.0.46.X range and they register @
>> 10.0.46.1 IP
>> address.
>>
>> When I place a call between the two phones in the 10.0.46.0/24 network
>> the phones register but audio RTP is sent to 216.82.231.69 instead of
>> 10.0.46.1 (FS)
>>
>> In this scenario i have FS without any NAT configuration.
>>
>> --
>> The way that i solved was to run both profiles (internal and
>> external) in
>> the public IP (local_ipv4) and force the phones to register with
>> 216.82.231.69 instead of 10.0.46.1
>>
>> Thanks looong time!
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>




From anthony.minessale at gmail.com  Wed Aug 26 11:26:40 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 13:26:40 -0500
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <4A95755B.9020201@nowthor.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
	<4A93EFE1.6010007@nowthor.com>
	<191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>
	<4A94A2E0.6040306@nowthor.com>
	<191c3a030908260825n73d6ffc2kbc87296197e72c90@mail.gmail.com>
	<4A95755B.9020201@nowthor.com>
Message-ID: <191c3a030908261126o1d0e4255k6f31a8de84e8fc44@mail.gmail.com>

did you see my other email on this thread with the xml macro idea?
When it comes to displacing the complexity there are many choices.

if you want to make some api call that can look in xml and generate a big |
sep dial string that's entirely possible.
you could do something similar with lua or some other embedded language as
well.


On Wed, Aug 26, 2009 at 12:48 PM, Carlos S. Antunes <csa at nowthor.com> wrote:

> Anthony Minessale wrote:
> > Unfortunately it's still a problem because the core bridge/originate
> > mechanism does not parse XML it's only parses the dial string markup.
>
> Wouldn't it be possible to "translate" the markup to a dial string prior
> to hitting the core mechanism?
>
> > In the end you must end up with one string of application and one
> > string of data.
> >
>
> How about something like
>
> <action application="bridge"><data> ... </data></action>
>
> being somewhat* equivalent to
>
> <action application="bridge" data=" ... ">?
>
> (* somewhat to be properly define later)
>
> > I am very strict about maintaining scope boundaries between abstract
> > concepts.  I think it's the number 1 breakdown that led to asterisk
> > getting so messed up over the years and I want to avoid that same plague.
>
> I totally agree with you on this.
>
> > I hope you understand and if you want to continue to ponder some more
> > functionality I welcome it.
>
> Don't worry, I rarely stop having ideas. Feel free to shoot them down at
> will! :)
>
> Carlos
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From anatoliy at kounitskiy.com  Wed Aug 26 11:27:52 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Wed, 26 Aug 2009 21:27:52 +0300
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
Message-ID: <1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>

After several hours of testing I was able to answer myself the
previous mentioned questions.

It appears that # and the 0 option work _only_ if user C has answered
the call OR voicemail system answers it.

user A ---call---> user B----attended xfer---> user C

At this point I have new question. In example user C does not have a
voicemail and the call timeout is not an option to wait for. How can
user B go back to the user A, who is listening to MOH?
Could someone help me with an advice/tip?

At the moment I have just one idea for accomplishing it:
1) try to use bind_meta_app in the extension with the att_xfer (not
sure if it can be done). To have a key feature that takes the user A
call leg id and bridging it with user B

Thank you in advnace,
Anatoliy Kounitskiy


On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy
Kounitskiy<anatoliy at kounitskiy.com> wrote:
> Hello everybody!
> I have few questions about the att_xfer application. First, what i want
> to accomplish is: user A calls user B, after that user B makes attended
> transfer to user C.
> In the dialplan i have:
>
> <context name="vpbx">
> ?<extension name="local_number">
> ...
> ? ? ?<action application="bind_meta_app" data="1 b s
> execute_extension::dx XML features"/>
> ? ? ?<action application="bind_meta_app" data="2 b s
> record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> ? ? ?<action application="bind_meta_app" data="3 b s
> execute_extension::cf XML features"/>
> ? ? ?<action application="bind_meta_app" data="4 b s
> execute_extension::attented_xfer XML features"/>
> ?....
> ? ?</condition>
> ?</extension>
>
> So when user B answers the call, he sends *4 and the extensions for the
> attended transfer is started - the usual - plays message and read the
> input dtmf:
>
> features.xml
> ...
> ? ?<extension name="attented_xfer">
> ? ? ?<condition field="${toll_allow}" expression="local"/>
> ? ? ?<condition field="destination_number" expression="^attented_xfer$">
> ? ? ? ?<action application="info"/>
> ? ? ? ?<action application="read" data="3 4 ivr/ivr-enter_ext.wav
> attxfer_callthis 30000 #"/>
> ? ? ? ?<action application="set" data="call_timeout=15"/>
> ? ? ? ?<action application="att_xfer"
> data="user/${attxfer_callthis}@${domain_name}"/>
> ? ? ?</condition>
> ? ?</extension>
> ...
>
> To this problems everything is perfect. But here comes the questions, so
> if you can give some tips would be great.
>
> 1) when user B enters the extension number of C - the C's phone starts
> ringing in the tcpdump i can see that the phone is sending 180 ringing,
> BUT user B does not hear the ringing.
> 2) as mentioned in the
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
> quote: "If the other leg is a voicemail or doesn't answered you can
> hangup that leg by pressing dtmf # (fixed in r14438) "
> It doesn't work. The option 0 is working even before C answering the
> phone - after he answers it's a threeway conference :) - i like this
> feature.
>
> I'm using FreeSWITCH Version 1.0.trunk (14633M)
>
> Also I tried to set call timeout to see if I can go back the user A, who
> is listening to MOH - no luck here.
>
> Probably I'm missing something. Tried to look in the source of att_xfer
> to understand why the feature i want is not working - but it seems my
> C/C++ skills are not so good, as i want :( .
>
> Thank you in advance,
> Anatoliy Kounitskiy
>



-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540


From anthony.minessale at gmail.com  Wed Aug 26 11:39:46 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 13:39:46 -0500
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
Message-ID: <191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>

maybe we can make an origination_cancel_key=# you could set on the dial
string to be able to cancel that originate with dtmf


On Wed, Aug 26, 2009 at 1:27 PM, Anatoliy Kounitskiy <
anatoliy at kounitskiy.com> wrote:

> After several hours of testing I was able to answer myself the
> previous mentioned questions.
>
> It appears that # and the 0 option work _only_ if user C has answered
> the call OR voicemail system answers it.
>
> user A ---call---> user B----attended xfer---> user C
>
> At this point I have new question. In example user C does not have a
> voicemail and the call timeout is not an option to wait for. How can
> user B go back to the user A, who is listening to MOH?
> Could someone help me with an advice/tip?
>
> At the moment I have just one idea for accomplishing it:
> 1) try to use bind_meta_app in the extension with the att_xfer (not
> sure if it can be done). To have a key feature that takes the user A
> call leg id and bridging it with user B
>
> Thank you in advnace,
> Anatoliy Kounitskiy
>
>
> On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy
> Kounitskiy<anatoliy at kounitskiy.com> wrote:
> > Hello everybody!
> > I have few questions about the att_xfer application. First, what i want
> > to accomplish is: user A calls user B, after that user B makes attended
> > transfer to user C.
> > In the dialplan i have:
> >
> > <context name="vpbx">
> >  <extension name="local_number">
> > ...
> >      <action application="bind_meta_app" data="1 b s
> > execute_extension::dx XML features"/>
> >      <action application="bind_meta_app" data="2 b s
> >
> record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> >      <action application="bind_meta_app" data="3 b s
> > execute_extension::cf XML features"/>
> >      <action application="bind_meta_app" data="4 b s
> > execute_extension::attented_xfer XML features"/>
> >  ....
> >    </condition>
> >  </extension>
> >
> > So when user B answers the call, he sends *4 and the extensions for the
> > attended transfer is started - the usual - plays message and read the
> > input dtmf:
> >
> > features.xml
> > ...
> >    <extension name="attented_xfer">
> >      <condition field="${toll_allow}" expression="local"/>
> >      <condition field="destination_number" expression="^attented_xfer$">
> >        <action application="info"/>
> >        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
> > attxfer_callthis 30000 #"/>
> >        <action application="set" data="call_timeout=15"/>
> >        <action application="att_xfer"
> > data="user/${attxfer_callthis}@${domain_name}"/>
> >      </condition>
> >    </extension>
> > ...
> >
> > To this problems everything is perfect. But here comes the questions, so
> > if you can give some tips would be great.
> >
> > 1) when user B enters the extension number of C - the C's phone starts
> > ringing in the tcpdump i can see that the phone is sending 180 ringing,
> > BUT user B does not hear the ringing.
> > 2) as mentioned in the
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
> > quote: "If the other leg is a voicemail or doesn't answered you can
> > hangup that leg by pressing dtmf # (fixed in r14438) "
> > It doesn't work. The option 0 is working even before C answering the
> > phone - after he answers it's a threeway conference :) - i like this
> > feature.
> >
> > I'm using FreeSWITCH Version 1.0.trunk (14633M)
> >
> > Also I tried to set call timeout to see if I can go back the user A, who
> > is listening to MOH - no luck here.
> >
> > Probably I'm missing something. Tried to look in the source of att_xfer
> > to understand why the feature i want is not working - but it seems my
> > C/C++ skills are not so good, as i want :( .
> >
> > Thank you in advance,
> > Anatoliy Kounitskiy
> >
>
>
>
> --
> Anatoliy Kounitskiy
> -------------------------
> E-mail: anatoliy at kounitskiy.com
> Mobile: +359898913540
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From csa at nowthor.com  Wed Aug 26 11:54:52 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Wed, 26 Aug 2009 14:54:52 -0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <191c3a030908261126o1d0e4255k6f31a8de84e8fc44@mail.gmail.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>	<4A93EFE1.6010007@nowthor.com>	<191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>	<4A94A2E0.6040306@nowthor.com>	<191c3a030908260825n73d6ffc2kbc87296197e72c90@mail.gmail.com>	<4A95755B.9020201@nowthor.com>
	<191c3a030908261126o1d0e4255k6f31a8de84e8fc44@mail.gmail.com>
Message-ID: <4A9584FC.9030305@nowthor.com>

Anthony Minessale wrote:
> did you see my other email on this thread with the xml macro idea?
> When it comes to displacing the complexity there are many choices.

Yes, I did see that macro idea. It looks indeed very good in the case of 
a failover scenario. But what about something like I proposed initially, 
that is, a mix of simultaneous and sequential dialing in arbitrary 
order? I guess this is also what Phillip Jones is after with his feature 
request (http://jira.freeswitch.org/browse/FSCORE-422).

In any case, I am going to adopt your solution for the failover scenario 
as the most elegant so far! :)

> if you want to make some api call that can look in xml and generate a 
> big | sep dial string that's entirely possible. you could do something 
> similar with lua or some other embedded language as well.

Yes, that is indeed another possibility. However, I am not sure it would 
allow that mix of simultaneous and sequential dialing, would it?

Carlos


From anthony.minessale at gmail.com  Wed Aug 26 11:56:35 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 13:56:35 -0500
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
Message-ID: <191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>

which revision are you on?
The defaults on the latest code and examples should be configured to
minimize delay.
Some of the older revisions built up some delay issues from udp buffering
when timers were not synced.


On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net> wrote:

>  When running conferences with users dialed in from a PSTN gateway (SIP)
> and directly from remote SIP endpoints there is an ever longer buildup in
> delay, reaching up to multiple seconds. Is there any way to limit the delay
> ?
>
>
>
> I am not 100% sure whether the delays is caused by the SIP jitter buffer of
> freeswitch or directly by the conference module.
>
>
>
> Any advice?
>
>
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
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From foxb at abv.bg  Wed Aug 26 12:02:13 2009
From: foxb at abv.bg (Hristo Benev)
Date: Wed, 26 Aug 2009 22:02:13 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <925579051.122056.1251313341898.JavaMail.apache@mail22.abv.bg>

 I think that the problem is here:
-------------------------
2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
--------------------------

Actually Regex FAIL

I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
Here is my  default.xml:
----------------
<?xml version="1.0" encoding="utf-8"?>
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
  <context name="default">

     <extension name="LOOKUP_ROUTE">
      <condition field="destination_number" expression="(\d+)$">
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" data="continue_on_fail=true"/>
         <action application="export" data="sip_h_X-ROUTE=LOOKUP"/>
         <action application="bridge" data="sofia/internal/${destination_number}@127.0.0.1:5062"/>
         <action application="set" data="ROUTE_GW=${sip_redirect_contact_user_0}"/>
         <action application="set" data="AREA=${sip_redirect_contact_user_0}"/>
         <action application="transfer" data="${destination_number} XML ROUTING"/>
      </condition>
    </extension>

   </context>

  <context name="ROUTING">

    <extension name="PEER_01">
      <condition field="${sip_h_X-ROUTE}" expression="PEER_01">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
        <action application="set" data="PEER=1.1.1.1"/>
        <action application="bridge" data="sofia/external/${destination_number}@1.1.1.1"/>
        <action application="set" data="PEER=2.2.2.2"/>
        <action application="bridge" data="sofia/external/${destination_number}@2.2.2.2"/>
        <action application="set" data="PEER=3.3.3.3"/>
        <action application="bridge" data="sofia/external/${destination_number}@3.3.3.3"/>
      </condition>
    </extension>

  </context>

</include>

--------------------------



 >-------- ?????????? ????? --------
 >??:  Brian West 
 >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >??: freeswitch-users at lists.freeswitch.org
 >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST

 >We do not blindly follow 302's as that is a dangerous thing to do. 
You have to process all 302's in the dialplan. 
Set this on your sofia profile
You can set these variables sip_redirect_profile,
sip_redirect_context,
sip_redirect_dialplan,
When a redirect happens you get these variables - sip_redirect_contact_%d,
sip_redirected_to,
sip_redirect_contact_user_%d,
sip_redirect_contact_host_%d,
sip_redirect_contact_params_%d,
sip_redirect_dialstring_%d,
sip_redirect_dialstring,
sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
 >


From anthony.minessale at gmail.com  Wed Aug 26 12:11:22 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 14:11:22 -0500
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <4A9584FC.9030305@nowthor.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
	<4A93EFE1.6010007@nowthor.com>
	<191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>
	<4A94A2E0.6040306@nowthor.com>
	<191c3a030908260825n73d6ffc2kbc87296197e72c90@mail.gmail.com>
	<4A95755B.9020201@nowthor.com>
	<191c3a030908261126o1d0e4255k6f31a8de84e8fc44@mail.gmail.com>
	<4A9584FC.9030305@nowthor.com>
Message-ID: <191c3a030908261211r40698642r37d338c08679b03d@mail.gmail.com>

that simo-sequential dialing thing is highly complicated because like
highlander there can be only one in the end.
The way we do it now is all controlled by the same monitor thread and it's a
ton more complexity added to a system that many
now depend on.  The things originate can do easily outdo the leading brand
and i would hate to break anything.

You already hate the syntax now and the only real solution would be to add
even more delimiters to the originate syntax
to denote sequential within simo as sequential currently has the highest
priority.

most chars are used up already maybe ^ is still free (it has to be valid in
an xml attr too, we got rid of + for simo dial because of that)
sofia/foo1^sofia/foo2,sofia/bar1^sofia/bar2

on top of all that it's a lot of work and we are not exactly swimming with
coders blessed to hack the core volunteering their time =D





On Wed, Aug 26, 2009 at 1:54 PM, Carlos S. Antunes <csa at nowthor.com> wrote:

> Anthony Minessale wrote:
> > did you see my other email on this thread with the xml macro idea?
> > When it comes to displacing the complexity there are many choices.
>
> Yes, I did see that macro idea. It looks indeed very good in the case of
> a failover scenario. But what about something like I proposed initially,
> that is, a mix of simultaneous and sequential dialing in arbitrary
> order? I guess this is also what Phillip Jones is after with his feature
> request (http://jira.freeswitch.org/browse/FSCORE-422).
>
> In any case, I am going to adopt your solution for the failover scenario
> as the most elegant so far! :)
>
> > if you want to make some api call that can look in xml and generate a
> > big | sep dial string that's entirely possible. you could do something
> > similar with lua or some other embedded language as well.
>
> Yes, that is indeed another possibility. However, I am not sure it would
> allow that mix of simultaneous and sequential dialing, would it?
>
> Carlos
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gshfreesw at gmail.com  Wed Aug 26 12:55:38 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Wed, 26 Aug 2009 15:55:38 -0400
Subject: [Freeswitch-users] caller id on origination does not work
In-Reply-To: <7e2ac3270908252007g5dfb9390w806267550726b99a@mail.gmail.com>
References: <5070fcbd0908251951j5a7163b1se346abcdf0664c94@mail.gmail.com>
	<7e2ac3270908252007g5dfb9390w806267550726b99a@mail.gmail.com>
Message-ID: <5070fcbd0908261255o677d1abbseba42569ba9ebaf0@mail.gmail.com>

This is one did the trick.

Thanks.

On Tue, Aug 25, 2009 at 11:07 PM, SP <sprice at gmail.com> wrote:

> You can try playing with this in your gateway profile
>   <param name="caller-id-in-from" value="true"/>
>
> Not sure what it'll do to your registration, give it a try
>
> On Tue, Aug 25, 2009 at 21:51, Shameem Shiek <gshfreesw at gmail.com> wrote:
>
>> I am setting the caller id like this in my ESL script:
>>
>> @con.sendRecv("api originate
>> {origination_caller_id_number=15103245678}sofia/gateway/junctionnetworks/1#{@number}
>> &park()")
>>
>> And the caller id comes out as all zeroes.  The sip trace shows the "from"
>> as shown in the sofia status command.  This is output of sofia gateway
>> status and my gateway is under
>> /usr/local/freeswitch/conf/sip_profiles/external/junctionnetworks.xml .
>> Also, adding the gateway under /conf/directory/default did not work as the
>> wiki suugests. What I need to do for the caller_id to come through?
>>
>>
>> sofia status gateway junctionnetworks
>>
>> =================================================================================================
>> Name        junctionnetworks
>> Scheme      Digest
>> Realm       jnctn.net
>> Username    username
>> Password    yes
>> From        <sip:username at jnctn.net <sip%3Ausername at jnctn.net>
>> ;transport=udp>
>> Contact     <sip:gw+junctionnetworks at 24.87.80.12:5080;transport=udp>
>> Exten       899xxxxxxx
>> To          sip:username at sip.jnctn.net <sip%3Ausername at sip.jnctn.net>
>> Proxy       sip:sip.jnctn.net
>> Context     public
>> Expires     3600
>> Freq        3600
>> Ping        0
>> PingFreq    0
>> State       REGED
>> Status      UP
>> CallsIN     0
>> CallsOUT    1
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Shannon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Aug 26 12:58:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Aug 2009 12:58:10 -0700
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
In-Reply-To: <4A9537DF.8070609@yahoo.com>
References: <4A9537DF.8070609@yahoo.com>
Message-ID: <87f2f3b90908261258r77869c3ara40dbd5a7a4259d6@mail.gmail.com>

On Wed, Aug 26, 2009 at 6:25 AM, Merle J. Ebbert <se02005-mec at yahoo.com>wrote:

>
> Hi,
>
> I'm trying to avoid taking up a lot of peoples valuable time.
>
> SIP & FS have  brought some ideas for some commercial products but I
> need to know where to start.
>
> Having once written a proprietary DOS & helped with writing a RTOS, I
> consider myself capable
> of learning.  I (we) just need to know where to start to come up to
> speed rapidly.
>
> Is there a FreeSWITCH tutorial available?
> Should someone new start with Asterisk and then possibly move to FS?
>
> Thanks,
> Merle
>

BTW, if you're in Europe then you can get a copy of the September issue of
Linux-Pro magazine. (I thinks it's just called Linux Magazine in Europe.) It
has a very gentle introduction to FS. The mag should be available any time
in US and end of August in Australia/NZ.
-MC
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From anthony.minessale at gmail.com  Wed Aug 26 12:59:35 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 14:59:35 -0500
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
Message-ID: <191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>

i added a patch to attempt to do this so try adding

{origination_cancel_key=#} before the dial string
or
<action application="set" data="origination_cancel_key=#"/>
before you bridge.

On Wed, Aug 26, 2009 at 1:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> maybe we can make an origination_cancel_key=# you could set on the dial
> string to be able to cancel that originate with dtmf
>
>
>
> On Wed, Aug 26, 2009 at 1:27 PM, Anatoliy Kounitskiy <
> anatoliy at kounitskiy.com> wrote:
>
>> After several hours of testing I was able to answer myself the
>> previous mentioned questions.
>>
>> It appears that # and the 0 option work _only_ if user C has answered
>> the call OR voicemail system answers it.
>>
>> user A ---call---> user B----attended xfer---> user C
>>
>> At this point I have new question. In example user C does not have a
>> voicemail and the call timeout is not an option to wait for. How can
>> user B go back to the user A, who is listening to MOH?
>> Could someone help me with an advice/tip?
>>
>> At the moment I have just one idea for accomplishing it:
>> 1) try to use bind_meta_app in the extension with the att_xfer (not
>> sure if it can be done). To have a key feature that takes the user A
>> call leg id and bridging it with user B
>>
>> Thank you in advnace,
>> Anatoliy Kounitskiy
>>
>>
>> On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy
>> Kounitskiy<anatoliy at kounitskiy.com> wrote:
>> > Hello everybody!
>> > I have few questions about the att_xfer application. First, what i want
>> > to accomplish is: user A calls user B, after that user B makes attended
>> > transfer to user C.
>> > In the dialplan i have:
>> >
>> > <context name="vpbx">
>> >  <extension name="local_number">
>> > ...
>> >      <action application="bind_meta_app" data="1 b s
>> > execute_extension::dx XML features"/>
>> >      <action application="bind_meta_app" data="2 b s
>> >
>> record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>> >      <action application="bind_meta_app" data="3 b s
>> > execute_extension::cf XML features"/>
>> >      <action application="bind_meta_app" data="4 b s
>> > execute_extension::attented_xfer XML features"/>
>> >  ....
>> >    </condition>
>> >  </extension>
>> >
>> > So when user B answers the call, he sends *4 and the extensions for the
>> > attended transfer is started - the usual - plays message and read the
>> > input dtmf:
>> >
>> > features.xml
>> > ...
>> >    <extension name="attented_xfer">
>> >      <condition field="${toll_allow}" expression="local"/>
>> >      <condition field="destination_number" expression="^attented_xfer$">
>> >        <action application="info"/>
>> >        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
>> > attxfer_callthis 30000 #"/>
>> >        <action application="set" data="call_timeout=15"/>
>> >        <action application="att_xfer"
>> > data="user/${attxfer_callthis}@${domain_name}"/>
>> >      </condition>
>> >    </extension>
>> > ...
>> >
>> > To this problems everything is perfect. But here comes the questions, so
>> > if you can give some tips would be great.
>> >
>> > 1) when user B enters the extension number of C - the C's phone starts
>> > ringing in the tcpdump i can see that the phone is sending 180 ringing,
>> > BUT user B does not hear the ringing.
>> > 2) as mentioned in the
>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
>> > quote: "If the other leg is a voicemail or doesn't answered you can
>> > hangup that leg by pressing dtmf # (fixed in r14438) "
>> > It doesn't work. The option 0 is working even before C answering the
>> > phone - after he answers it's a threeway conference :) - i like this
>> > feature.
>> >
>> > I'm using FreeSWITCH Version 1.0.trunk (14633M)
>> >
>> > Also I tried to set call timeout to see if I can go back the user A, who
>> > is listening to MOH - no luck here.
>> >
>> > Probably I'm missing something. Tried to look in the source of att_xfer
>> > to understand why the feature i want is not working - but it seems my
>> > C/C++ skills are not so good, as i want :( .
>> >
>> > Thank you in advance,
>> > Anatoliy Kounitskiy
>> >
>>
>>
>>
>> --
>> Anatoliy Kounitskiy
>> -------------------------
>> E-mail: anatoliy at kounitskiy.com
>> Mobile: +359898913540
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anatoliy at kounitskiy.com  Wed Aug 26 13:10:59 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Wed, 26 Aug 2009 23:10:59 +0300
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
Message-ID: <1cd828b60908261310m3cca0e56u98c8ccc29a54b21c@mail.gmail.com>

Thank you for the fast patch :)

I'm going to test it right away and report you if there are problems or not.


On Wed, Aug 26, 2009 at 10:59 PM, Anthony
Minessale<anthony.minessale at gmail.com> wrote:
> i added a patch to attempt to do this so try adding
>
> {origination_cancel_key=#} before the dial string
> or
> <action application="set" data="origination_cancel_key=#"/>
> before you bridge.
>
> On Wed, Aug 26, 2009 at 1:39 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> maybe we can make an origination_cancel_key=# you could set on the dial
>> string to be able to cancel that originate with dtmf
>>
>>
>> On Wed, Aug 26, 2009 at 1:27 PM, Anatoliy Kounitskiy
>> <anatoliy at kounitskiy.com> wrote:
>>>
>>> After several hours of testing I was able to answer myself the
>>> previous mentioned questions.
>>>
>>> It appears that # and the 0 option work _only_ if user C has answered
>>> the call OR voicemail system answers it.
>>>
>>> user A ---call---> user B----attended xfer---> user C
>>>
>>> At this point I have new question. In example user C does not have a
>>> voicemail and the call timeout is not an option to wait for. How can
>>> user B go back to the user A, who is listening to MOH?
>>> Could someone help me with an advice/tip?
>>>
>>> At the moment I have just one idea for accomplishing it:
>>> 1) try to use bind_meta_app in the extension with the att_xfer (not
>>> sure if it can be done). To have a key feature that takes the user A
>>> call leg id and bridging it with user B
>>>
>>> Thank you in advnace,
>>> Anatoliy Kounitskiy
>>>
>>>
>>> On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy
>>> Kounitskiy<anatoliy at kounitskiy.com> wrote:
>>> > Hello everybody!
>>> > I have few questions about the att_xfer application. First, what i want
>>> > to accomplish is: user A calls user B, after that user B makes attended
>>> > transfer to user C.
>>> > In the dialplan i have:
>>> >
>>> > <context name="vpbx">
>>> > ?<extension name="local_number">
>>> > ...
>>> > ? ? ?<action application="bind_meta_app" data="1 b s
>>> > execute_extension::dx XML features"/>
>>> > ? ? ?<action application="bind_meta_app" data="2 b s
>>> >
>>> > record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>> > ? ? ?<action application="bind_meta_app" data="3 b s
>>> > execute_extension::cf XML features"/>
>>> > ? ? ?<action application="bind_meta_app" data="4 b s
>>> > execute_extension::attented_xfer XML features"/>
>>> > ?....
>>> > ? ?</condition>
>>> > ?</extension>
>>> >
>>> > So when user B answers the call, he sends *4 and the extensions for the
>>> > attended transfer is started - the usual - plays message and read the
>>> > input dtmf:
>>> >
>>> > features.xml
>>> > ...
>>> > ? ?<extension name="attented_xfer">
>>> > ? ? ?<condition field="${toll_allow}" expression="local"/>
>>> > ? ? ?<condition field="destination_number"
>>> > expression="^attented_xfer$">
>>> > ? ? ? ?<action application="info"/>
>>> > ? ? ? ?<action application="read" data="3 4 ivr/ivr-enter_ext.wav
>>> > attxfer_callthis 30000 #"/>
>>> > ? ? ? ?<action application="set" data="call_timeout=15"/>
>>> > ? ? ? ?<action application="att_xfer"
>>> > data="user/${attxfer_callthis}@${domain_name}"/>
>>> > ? ? ?</condition>
>>> > ? ?</extension>
>>> > ...
>>> >
>>> > To this problems everything is perfect. But here comes the questions,
>>> > so
>>> > if you can give some tips would be great.
>>> >
>>> > 1) when user B enters the extension number of C - the C's phone starts
>>> > ringing in the tcpdump i can see that the phone is sending 180 ringing,
>>> > BUT user B does not hear the ringing.
>>> > 2) as mentioned in the
>>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
>>> > quote: "If the other leg is a voicemail or doesn't answered you can
>>> > hangup that leg by pressing dtmf # (fixed in r14438) "
>>> > It doesn't work. The option 0 is working even before C answering the
>>> > phone - after he answers it's a threeway conference :) - i like this
>>> > feature.
>>> >
>>> > I'm using FreeSWITCH Version 1.0.trunk (14633M)
>>> >
>>> > Also I tried to set call timeout to see if I can go back the user A,
>>> > who
>>> > is listening to MOH - no luck here.
>>> >
>>> > Probably I'm missing something. Tried to look in the source of att_xfer
>>> > to understand why the feature i want is not working - but it seems my
>>> > C/C++ skills are not so good, as i want :( .
>>> >
>>> > Thank you in advance,
>>> > Anatoliy Kounitskiy
>>> >
>>>
>>>
>>>
>>> --
>>> Anatoliy Kounitskiy
>>> -------------------------
>>> E-mail: anatoliy at kounitskiy.com
>>> Mobile: +359898913540
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540


From msc at freeswitch.org  Wed Aug 26 13:11:14 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Aug 2009 13:11:14 -0700
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
Message-ID: <87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>

On Wed, Aug 26, 2009 at 12:59 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> i added a patch to attempt to do this so try adding
>

Nice work!

>
> {origination_cancel_key=#} before the dial string
> or
> <action application="set" data="origination_cancel_key=#"/>
> before you bridge.
>

Anatoliy,

Please try this and let us know if it works for you as expected. If so,
please write up the exact procedure that you used and include dialplan
examples. I'd like to get this on the wiki while it's fresh in my mind. If
you feel comfortable editing the wiki on your own then just let me know so
that we can coordinate efforts.

Thanks,
MC
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From msc at freeswitch.org  Wed Aug 26 13:28:17 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Aug 2009 13:28:17 -0700
Subject: [Freeswitch-users] Timers/DTMFs During a Call
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>
Message-ID: <87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com>

Do you actually need Python for the IVR, or is it that you're comfortable
using a scripting lang for an IVR? I like using XML for IVRs, but using
scripting langs does give you a bit more power & flexibility at the cost of
some resources.

For the record, you can do this in the dialplan using XML and sched_api
without touching a scripting language. Checkout the sched_api channel
variable on the wiki - it may give you the functionality you need.

-MC

On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM <delianspam at gmail.com> wrote:

>  Hello Everybody!
>
>
>
> 1.       Scenario.
>
> I am writing an IVR in Python that gets a destination from the calling
> party (party A) and then connects to the destination (party B).
>
> When the call is CONNECTED, I want to:
>
> -          Receive DTMFs
>
> -          Have a timer that can call a certain function in my script. The
> script will have to play a message to party A.
>
> -          Have a timer that can call a certain function in my script. The
> script will have to drop the call.
>
> Please notice that I want to do the things after the two parties are
> connected, and not after I send the Invite to party B.
>
>
>
> 2.       Problem.
>
> I will be happy to receive help on:
>
> -          Which methods should I look for to implement this.
>
>
>
> 3.       Details
>
> Here is how I connect the call currently:
>
> session.execute("bridge",?sofia/internal/" + destination_number + "@
> domain.com?)
>
>
>
> I have tried to create a timer callback function ?my_method()? using:
>
> ivr_timer =threading.Timer(30,my_method)
>
> This never called the function ?my_method()?.
>
>
>
> Maybe I am wrong in using threading.Timer and the ?bridge? application?
> Maybe I need to create a new thread and a new timer using the API of
> freeswitch, plus to use the ?session.setInputCallback?, plus use a
> conference rather than a bridge? Can you please provide any suggestions or
> examples?
>
>
>
> Thank you!
>
> Best Regards, Delian Tashev
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pbd at suspiria.net  Wed Aug 26 13:30:07 2009
From: pbd at suspiria.net (Public Dump)
Date: Wed, 26 Aug 2009 22:30:07 +0200
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
Message-ID: <13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>

Hi,



I am on a quite recent version (i assume):



FreeSWITCH Version 1.0.trunk (14461)



Should the bug be fixed in this revision ? What config settings would a have to check to limit delay (even at the cost of reduction in quality).



Thanks



Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 20:57
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] delay buildup in conference



which revision are you on?
The defaults on the latest code and examples should be configured to minimize delay.
Some of the older revisions built up some delay issues from udp buffering when timers were not synced.



On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net<mailto:pbd at suspiria.net>> wrote:

When running conferences with users dialed in from a PSTN gateway (SIP) and directly from remote SIP endpoints there is an ever longer buildup in delay, reaching up to multiple seconds. Is there any way to limit the delay ?



I am not 100% sure whether the delays is caused by the SIP jitter buffer of freeswitch or directly by the conference module.



Any advice?



Thanks


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400

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From anthony.minessale at gmail.com  Wed Aug 26 13:45:29 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 15:45:29 -0500
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
Message-ID: <191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>

There is no bug, it's all dependent on your network conditions.
I spend literally 8-12 hours a day on a conference and there is no delay.

The important param is
<param name="rtp-autoflush-during-bridge" value="true"/>

in the sofia profile in question.

if you have delay with that in place, then it's probably not FS


On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd at suspiria.net> wrote:

>  Hi,
>
>
>
> I am on a quite recent version (i assume):
>
>
>
> FreeSWITCH Version 1.0.trunk (14461)
>
>
>
> Should the bug be fixed in this revision ? What config settings would a
> have to check to limit delay (even at the cost of reduction in quality).
>
>
>
> Thanks
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Anthony
> Minessale
> *Gesendet:* Mittwoch, 26. August 2009 20:57
> *An:* freeswitch-users at lists.freeswitch.org
> *Betreff:* Re: [Freeswitch-users] delay buildup in conference
>
>
>
> which revision are you on?
> The defaults on the latest code and examples should be configured to
> minimize delay.
> Some of the older revisions built up some delay issues from udp buffering
> when timers were not synced.
>
>  On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net> wrote:
>
> When running conferences with users dialed in from a PSTN gateway (SIP) and
> directly from remote SIP endpoints there is an ever longer buildup in delay,
> reaching up to multiple seconds. Is there any way to limit the delay ?
>
>
>
> I am not 100% sure whether the delays is caused by the SIP jitter buffer of
> freeswitch or directly by the conference module.
>
>
>
> Any advice?
>
>
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From harry at vangberg.name  Wed Aug 26 13:59:02 2009
From: harry at vangberg.name (Harry Vangberg)
Date: Wed, 26 Aug 2009 22:59:02 +0200
Subject: [Freeswitch-users] bind_meta_app: "already
	broadcasting..broadcast aborted" if bound via event socket
In-Reply-To: <74d41a3d0908260654x52bf9d94g124ceaa27f6df6bb@mail.gmail.com>
References: <74d41a3d0908260654x52bf9d94g124ceaa27f6df6bb@mail.gmail.com>
Message-ID: <74d41a3d0908261359sa78607bx20852471dfc3aa49@mail.gmail.com>

For your information, revision 14644 committed by anthm fixes this.
Again, thanks!

2009/8/26 Harry Vangberg <harry at vangberg.name>:
> Hello.
>
> After pestering the IRC channel for a few days with useless
> information, I think I've finally made a breakthrough, but I have no
> idea why it works this way.
>
> Basically, what I want to achieve is: A calls in and is bridged to B
> (1234). If B presses *1, he should be taken out, and A be bridged to C
> (8888) instead. Pretty simple.
>
> With the following dialplan it works very nicely:
>
> ? ?<extension name="ivr">
> ? ? ?<condition field="destination_number" expression="^(\d+)$">
> ? ? ? ?<action application="set" data="bypass_media=false"/>
> ? ? ? ?<action application="answer"/>
> ? ? ? ?<action application="bind_meta_app" data="1 b a
> bridge::sofia/gateway/secretgw/8888"/>
> ? ? ? ?<action application="bridge" data="sofia/gateway/secretgw/1234" />
> ? ? ?</condition>
> ? ?</extension>
>
> On the other hand, if I try to replicate the above via an outbound
> event socket it fails with a: "already broadcasting...broadcast
> aborted". In this case my dialplan XML looks like this:
>
> ? ?<extension name="ff-ivr">
> ? ? ?<condition field="destination_number" expression="^(.*)$">
> ? ? ? ?<action application="set" data="bypass_media=false"/>
> ? ? ? ?<action application="socket" data="127.0.0.1:8084 full"/>
> ? ? ?</condition>
> ? ?</extension>
>
> And this is the event socket communication (in reality I use a Ruby
> framework, do a bunch of database calls etc., but for the sake of
> clarity I'm mimicking it with a netcat session):
>
> --------------------------------
> $ nc -v -l 127.0.0.1 8084
> connect
>
> Event-Name: CHANNEL_DATA
> Core-UUID: ba78636a-9241-11de-ac94-b736da546252
> FreeSWITCH-Hostname: ip-10-226-231-225
> FreeSWITCH-IPv4: 10.226.231.225
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-08-26%2013%3A43%3A20
> Event-Date-GMT: Wed,%2026%20Aug%202009%2013%3A43%3A20%20GMT
> Event-Date-Timestamp: 1251294200244673
> Event-Calling-File: mod_event_socket.c
> Event-Calling-Function: parse_command
> Event-Calling-Line-Number: 1482
> Channel-Username: hemmeligt
> Channel-Dialplan: XML
> Channel-Caller-ID-Name: hemmeligt
> Channel-Caller-ID-Number: hemmeligt
> Channel-Network-Addr: 129.142.224.250
> Channel-Destination-Number: 77344541
> Channel-Unique-ID: 681102bc-9246-11de-ac94-b736da546252
> Channel-Source: mod_sofia
> Channel-Context: public
> Channel-Channel-Name: sofia/external/hemmeligt%40129.142.224.250
> Channel-Profile-Index: 1
> Channel-Profile-Created-Time: 1251294195778388
> Channel-Channel-Created-Time: 1251294195778388
> Channel-Channel-Answered-Time: 0
> Channel-Channel-Progress-Time: 0
> Channel-Channel-Progress-Media-Time: 0
> Channel-Channel-Hangup-Time: 0
> Channel-Channel-Transfer-Time: 0
> Channel-Screen-Bit: false
> Channel-Privacy-Hide-Name: true
> Channel-Privacy-Hide-Number: true
> Channel-State: CS_EXECUTE
> Channel-State-Number: 4
> Channel-Name: sofia/external/hemmeligt%40129.142.224.250
> Unique-ID: 681102bc-9246-11de-ac94-b736da546252
> Call-Direction: inbound
> Presence-Call-Direction: inbound
> Answer-State: ringing
> Channel-Read-Codec-Name: PCMA
> Channel-Read-Codec-Rate: 8000
> Channel-Write-Codec-Name: PCMA
> Channel-Write-Codec-Rate: 8000
> Caller-Username: hemmeligt
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: hemmeligt
> Caller-Caller-ID-Number: hemmeligt
> Caller-Network-Addr: 129.142.224.250
> Caller-Destination-Number: 77344541
> Caller-Unique-ID: 681102bc-9246-11de-ac94-b736da546252
> Caller-Source: mod_sofia
> Caller-Context: public
> Caller-Channel-Name: sofia/external/hemmeligt%40129.142.224.250
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1251294195778388
> Caller-Channel-Created-Time: 1251294195778388
> Caller-Channel-Answered-Time: 0
> Caller-Channel-Progress-Time: 0
> Caller-Channel-Progress-Media-Time: 0
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: false
> Caller-Privacy-Hide-Name: true
> Caller-Privacy-Hide-Number: true
> variable_sip_received_ip: 129.142.224.250
> variable_sip_received_port: 5060
> variable_sip_via_protocol: udp
> variable_sip_from_user: hemmeligt
> variable_sip_from_uri: hemmeligt%40129.142.224.250
> variable_sip_from_host: 129.142.224.250
> variable_sip_from_user_stripped: hemmeligt
> variable_sip_from_tag: as19641342
> variable_sofia_profile_name: external
> variable_sip_Remote-Party-ID:
> %22hemmeligt%22%20%3Csip%3Ahemmeligt%40129.142.224.250%3E%3Bprivacy%3Dfull%3Bscreen%3Dno
> variable_sip_cid_type: rpid
> variable_sip_req_user: 77344541
> variable_sip_req_port: 5080
> variable_sip_req_uri: 77344541%4079.125.42.248%3A5080
> variable_sip_req_host: 79.125.42.248
> variable_sip_to_user: 77344541
> variable_sip_to_port: 5080
> variable_sip_to_uri: 77344541%4079.125.42.248%3A5080
> variable_sip_to_host: 79.125.42.248
> variable_sip_contact_user: hemmeligt
> variable_sip_contact_uri: hemmeligt%40129.142.224.250
> variable_sip_contact_host: 129.142.224.250
> variable_channel_name: sofia/external/hemmeligt%40129.142.224.250
> variable_sip_call_id: 1aa24dab25cf03f07933f7136822ceff%40129.142.224.250
> variable_sip_user_agent: Condor_gw1
> variable_sip_via_host: 129.142.224.250
> variable_sip_via_port: 5060
> variable_sip_via_rport: 5060
> variable_max_forwards: 70
> variable_switch_r_sdp:
> v%3D0%0D%0Ao%3Droot%2018982%2018982%20IN%20IP4%20129.142.224.250%0D%0As%3Dsession%0D%0Ac%3DIN%20IP4%20129.142.224.250%0D%0At%3D0%200%0D%0Am%3Daudio%2015430%20RTP/AVP%208%200%203%2097%20101%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A97%20iLBC/8000%0D%0Aa%3Dfmtp%3A97%20mode%3D30%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0D%0Aa%3Dptime%3A20%0D%0A
> variable_remote_media_ip: 129.142.224.250
> variable_remote_media_port: 15430
> variable_read_codec: PCMA
> variable_read_rate: 8000
> variable_write_codec: PCMA
> variable_write_rate: 8000
> variable_endpoint_disposition: RECEIVED
> variable_bypass_media: false
> variable_current_application_data: 127.0.0.1%3A8084%20full
> variable_current_application: socket
> variable_socket_host: 127.0.0.1
> Content-Type: command/reply
> Reply-Text: %2BOK%0A
> Socket-Mode: static
> Control: full
>
> sendmsg
> call-command: execute
> execute-app-name: answer
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> sendmsg
> call-command: execute
> execute-app-name: bind_meta_app
> execute-app-arg: 1 b a bridge::sofia/gateway/secretgw/8888
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: sofia/gateway/secretgw/1234
>
> Content-Type: command/reply
> Reply-Text: +OK
> --------------------------------
>
> And on the FreeSWITCH side:
>
> 2009-08-26 13:43:15.778388 [NOTICE] switch_channel.c:602 New Channel
> sofia/external/hemmeligt at 129.142.224.250
> [681102bc-9246-11de-ac94-b736da546252]
> 2009-08-26 13:43:15.778388 [DEBUG] sofia.c:3302 Channel
> sofia/external/hemmeligt at 129.142.224.250 entering state
> [received][100]
> 2009-08-26 13:43:15.778388 [DEBUG] sofia.c:3309 Remote SDP:
> v=0
> o=root 18982 18982 IN IP4 129.142.224.250
> s=session
> c=IN IP4 129.142.224.250
> t=0 0
> m=audio 15430 RTP/AVP 8 0 3 97 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2009-08-26 13:43:15.778388 [DEBUG] sofia_glue.c:3132 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
> 2009-08-26 13:43:15.778388 [DEBUG] sofia_glue.c:2090 Set Codec
> sofia/external/hemmeligt at 129.142.224.250 PCMA/8000 20 ms 160 samples
> 2009-08-26 13:43:15.778388 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
> payload to 101
> 2009-08-26 13:43:15.778388 [DEBUG] sofia.c:3468
> (sofia/external/hemmeligt at 129.142.224.250) State Change CS_NEW ->
> CS_INIT
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/hemmeligt at 129.142.224.250) Running State Change
> CS_INIT
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:481
> (sofia/external/hemmeligt at 129.142.224.250) State INIT
> 2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:83
> sofia/external/hemmeligt at 129.142.224.250 SOFIA INIT
> 2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:111
> (sofia/external/hemmeligt at 129.142.224.250) State Change CS_INIT ->
> CS_ROUTING
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:481
> (sofia/external/hemmeligt at 129.142.224.250) State INIT going to sleep
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/hemmeligt at 129.142.224.250) Running State Change
> CS_ROUTING
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:484
> (sofia/external/hemmeligt at 129.142.224.250) State ROUTING
> 2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:130
> sofia/external/hemmeligt at 129.142.224.250 SOFIA ROUTING
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:78
> sofia/external/hemmeligt at 129.142.224.250 Standard ROUTING
> 2009-08-26 13:43:15.778388 [INFO] mod_dialplan_xml.c:315 Processing
> hemmeligt->77344541 in context public
> Dialplan: sofia/external/hemmeligt at 129.142.224.250 parsing
> [public->ff-ivr] continue=false
> Dialplan: sofia/external/hemmeligt at 129.142.224.250 Regex (PASS)
> [ff-ivr] destination_number(77344541) =~ /^(.*)$/ break=on-false
> Dialplan: sofia/external/hemmeligt at 129.142.224.250 Action
> set(bypass_media=false)
> Dialplan: sofia/external/hemmeligt at 129.142.224.250 Action
> socket(127.0.0.1:8084 full)
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:114
> (sofia/external/hemmeligt at 129.142.224.250) State Change CS_ROUTING ->
> CS_EXECUTE
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:484
> (sofia/external/hemmeligt at 129.142.224.250) State ROUTING going to
> sleep
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/hemmeligt at 129.142.224.250) Running State Change
> CS_EXECUTE
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:491
> (sofia/external/hemmeligt at 129.142.224.250) State EXECUTE
> 2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:173
> sofia/external/hemmeligt at 129.142.224.250 SOFIA EXECUTE
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:151
> sofia/external/hemmeligt at 129.142.224.250 Standard EXECUTE
> EXECUTE sofia/external/hemmeligt at 129.142.224.250 set(bypass_media=false)
> 2009-08-26 13:43:15.778388 [DEBUG] mod_dptools.c:748
> sofia/external/hemmeligt at 129.142.224.250 SET [bypass_media]=[false]
> EXECUTE sofia/external/hemmeligt at 129.142.224.250 socket(127.0.0.1:8084 full)
> 2009-08-26 13:43:21.701447 [DEBUG] switch_ivr.c:540
> sofia/external/hemmeligt at 129.142.224.250 Command Execute answer()
> EXECUTE sofia/external/hemmeligt at 129.142.224.250 answer()
> 2009-08-26 13:43:21.701447 [DEBUG] mod_dptools.c:649
> sofia/external/hemmeligt at 129.142.224.250 receive message [ANSWER]
> 2009-08-26 13:43:21.701447 [DEBUG] sofia_glue.c:2324 AUDIO RTP
> [sofia/external/hemmeligt at 129.142.224.250] 10.226.231.225 port 19016
> -> 129.142.224.250 port 15430 codec: 8 ms: 20
> 2009-08-26 13:43:21.701447 [DEBUG] switch_rtp.c:1138 Starting timer
> [soft] 160 bytes per 20ms
> 2009-08-26 13:43:21.702447 [DEBUG] mod_sofia.c:536 Local SDP
> sofia/external/hemmeligt at 129.142.224.250:
> v=0
> o=FreeSWITCH 1251275185 1251275186 IN IP4 79.125.42.248
> s=FreeSWITCH
> c=IN IP4 79.125.42.248
> t=0 0
> m=audio 19016 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 2009-08-26 13:43:21.702447 [DEBUG] switch_core_session.c:630 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:21.702447 [NOTICE] mod_dptools.c:649 Channel
> [sofia/external/hemmeligt at 129.142.224.250] has been answered
> 2009-08-26 13:43:21.702447 [DEBUG] switch_channel.c:182
> sofia/external/hemmeligt at 129.142.224.250 receive message [AUDIO_SYNC]
> 2009-08-26 13:43:21.702447 [DEBUG] sofia.c:3302 Channel
> sofia/external/hemmeligt at 129.142.224.250 entering state
> [completed][200]
> 2009-08-26 13:43:21.782616 [DEBUG] sofia.c:3302 Channel
> sofia/external/hemmeligt at 129.142.224.250 entering state [ready][200]
> 2009-08-26 13:43:26.388737 [DEBUG] switch_ivr.c:540
> sofia/external/hemmeligt at 129.142.224.250 Command Execute
> bind_meta_app(1 b a bridge::sofia/gateway/secretgw/8888)
> EXECUTE sofia/external/hemmeligt at 129.142.224.250 bind_meta_app(1 b a
> bridge::sofia/gateway/secretgw/8888)
> 2009-08-26 13:43:26.388737 [INFO] switch_ivr_async.c:1795 Bound B-Leg:
> 1 bridge::sofia/gateway/secretgw/8888
> 2009-08-26 13:43:29.684236 [DEBUG] switch_ivr.c:540
> sofia/external/hemmeligt at 129.142.224.250 Command Execute
> bridge(sofia/gateway/secretgw/1234)
> EXECUTE sofia/external/hemmeligt at 129.142.224.250
> bridge(sofia/gateway/secretgw/1234)
> 2009-08-26 13:43:29.684236 [NOTICE] switch_channel.c:602 New Channel
> sofia/external/1234 [705ad5d8-9246-11de-ac94-b736da546252]
> 2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:2809
> (sofia/external/1234) State Change CS_NEW -> CS_INIT
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/1234) Running State Change CS_INIT
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:481
> (sofia/external/1234) State INIT
> 2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:83 sofia/external/1234 SOFIA INIT
> 2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:111
> (sofia/external/1234) State Change CS_INIT -> CS_ROUTING
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:29.684236 [DEBUG] sofia.c:3302 Channel
> sofia/external/1234 entering state [calling][0]
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:481
> (sofia/external/1234) State INIT going to sleep
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/1234) Running State Change CS_ROUTING
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:484
> (sofia/external/1234) State ROUTING
> 2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:130 sofia/external/1234
> SOFIA ROUTING
> 2009-08-26 13:43:29.684236 [DEBUG] switch_ivr_originate.c:63
> (sofia/external/1234) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:484
> (sofia/external/1234) State ROUTING going to sleep
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/1234) Running State Change CS_CONSUME_MEDIA
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:503
> (sofia/external/1234) State CONSUME_MEDIA
> 2009-08-26 13:43:29.731221 [DEBUG] sofia.c:3302 Channel
> sofia/external/1234 entering state [calling][0]
> 2009-08-26 13:43:30.660221 [DEBUG] sofia.c:3302 Channel
> sofia/external/1234 entering state [proceeding][183]
> 2009-08-26 13:43:30.660221 [DEBUG] sofia.c:3309 Remote SDP:
> v=0
> o=root 18982 18982 IN IP4 129.142.224.250
> s=session
> c=IN IP4 129.142.224.250
> t=0 0
> m=audio 12480 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:3132 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
> 2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:2090 Set Codec
> sofia/external/1234 PCMA/8000 20 ms 160 samples
> 2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
> payload to 101
> 2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:2324 AUDIO RTP
> [sofia/external/1234] 10.226.231.225 port 25540 -> 129.142.224.250
> port 12480 codec: 8 ms: 20
> 2009-08-26 13:43:30.660221 [DEBUG] switch_rtp.c:1138 Starting timer
> [soft] 160 bytes per 20ms
> 2009-08-26 13:43:30.661084 [NOTICE] sofia_glue.c:2759 Pre-Answer
> sofia/external/1234!
> 2009-08-26 13:43:30.661084 [DEBUG] switch_channel.c:1778 Send signal
> sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_originate.c:2061
> Originate Resulted in Success: [sofia/external/1234]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_channel.c:182
> sofia/external/1234 receive message [AUDIO_SYNC]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_channel.c:182
> sofia/external/hemmeligt at 129.142.224.250 receive message [AUDIO_SYNC]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_bridge.c:889
> sofia/external/1234 receive message [BRIDGE]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_core_session.c:630 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_bridge.c:896
> sofia/external/hemmeligt at 129.142.224.250 receive message [BRIDGE]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_core_session.c:630 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_bridge.c:940
> (sofia/external/1234) State Change CS_CONSUME_MEDIA ->
> CS_EXCHANGE_MEDIA
> 2009-08-26 13:43:30.665170 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:30.705074 [DEBUG] switch_core_state_machine.c:503
> (sofia/external/1234) State CONSUME_MEDIA going to sleep
> 2009-08-26 13:43:30.705074 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/1234) Running State Change CS_EXCHANGE_MEDIA
> 2009-08-26 13:43:30.705074 [DEBUG] switch_core_state_machine.c:494
> (sofia/external/1234) State EXCHANGE_MEDIA
> 2009-08-26 13:43:30.705074 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK
> 2009-08-26 13:43:36.642173 [DEBUG] sofia.c:3302 Channel
> sofia/external/1234 entering state [ready][200]
> 2009-08-26 13:43:36.642173 [DEBUG] switch_channel.c:1891 Send signal
> sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:36.642173 [NOTICE] sofia.c:3752 Channel
> [sofia/external/1234] has been answered
> 2009-08-26 13:43:36.642173 [DEBUG] switch_channel.c:182
> sofia/external/1234 receive message [AUDIO_SYNC]
> 2009-08-26 13:43:40.309616 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF *:2000
> 2009-08-26 13:43:40.970513 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF 1:2000
> 2009-08-26 13:43:40.970513 [DEBUG] switch_ivr_async.c:1711
> sofia/external/hemmeligt at 129.142.224.250 Processing meta digit '1'
> [bridge::sofia/gateway/secretgw/8888]
> 2009-08-26 13:43:40.970513 [WARNING] switch_ivr_async.c:2310 Channel
> [sofia/external/hemmeligt at 129.142.224.250][bridge::sofia/gateway/secretgw/8888]
> already broadcasting...broadcast aborted
> 2009-08-26 13:43:46.272706 [NOTICE] sofia.c:327 Hangup
> sofia/external/1234 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2009-08-26 13:43:46.272706 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/1234 [KILL]
> 2009-08-26 13:43:46.272706 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:371
> sofia/external/1234 ending bridge by request from write function
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:426
> sofia/external/hemmeligt at 129.142.224.250 receive message [UNBRIDGE]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_session.c:630 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:452 BRIDGE
> THREAD DONE [sofia/external/hemmeligt at 129.142.224.250]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:454 Send signal
> sofia/external/1234 [BREAK]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:377
> sofia/external/1234 ending bridge by request from read function
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:452 BRIDGE
> THREAD DONE [sofia/external/1234]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:454 Send signal
> sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:494
> (sofia/external/1234) State EXCHANGE_MEDIA going to sleep
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/1234) Running State Change CS_HANGUP
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:434
> (sofia/external/1234) State HANGUP
> 2009-08-26 13:43:46.284704 [DEBUG] mod_sofia.c:338 Channel
> sofia/external/1234 hanging up, cause: NORMAL_CLEARING
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:46
> sofia/external/1234 Standard HANGUP, cause: NORMAL_CLEARING
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:434
> (sofia/external/1234) State HANGUP going to sleep
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:476
> (sofia/external/1234) State Change CS_HANGUP -> CS_REPORTING
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/1234) Running State Change CS_REPORTING
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:612
> (sofia/external/1234) State REPORTING
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:53
> sofia/external/1234 Standard REPORTING, cause: NORMAL_CLEARING
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:612
> (sofia/external/1234) State REPORTING going to sleep
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:411
> (sofia/external/1234) State Change CS_REPORTING -> CS_DESTROY
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_session.c:1068 Session
> 5 (sofia/external/1234) Locked, Waiting on external entities
> 2009-08-26 13:43:46.284704 [NOTICE] switch_core_session.c:1086 Session
> 5 (sofia/external/1234) Ended
> 2009-08-26 13:43:46.284704 [NOTICE] switch_core_session.c:1088 Close
> Channel sofia/external/1234 [CS_DESTROY]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:564
> (sofia/external/1234) State DESTROY
> 2009-08-26 13:43:46.284704 [DEBUG] mod_sofia.c:255 sofia/external/1234
> SOFIA DESTROY
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:60
> sofia/external/1234 Standard DESTROY
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:564
> (sofia/external/1234) State DESTROY going to sleep
> 2009-08-26 13:44:46.746524 [NOTICE] sofia.c:327 Hangup
> sofia/external/hemmeligt at 129.142.224.250 [CS_EXECUTE]
> [NORMAL_CLEARING]
> 2009-08-26 13:44:46.746524 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/hemmeligt at 129.142.224.250 [KILL]
> 2009-08-26 13:44:46.746524 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:491
> (sofia/external/hemmeligt at 129.142.224.250) State EXECUTE going to
> sleep
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/hemmeligt at 129.142.224.250) Running State Change
> CS_HANGUP
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:434
> (sofia/external/hemmeligt at 129.142.224.250) State HANGUP
> 2009-08-26 13:44:46.747796 [DEBUG] mod_sofia.c:338 Channel
> sofia/external/hemmeligt at 129.142.224.250 hanging up, cause:
> NORMAL_CLEARING
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:46
> sofia/external/hemmeligt at 129.142.224.250 Standard HANGUP, cause:
> NORMAL_CLEARING
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:434
> (sofia/external/hemmeligt at 129.142.224.250) State HANGUP going to sleep
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:476
> (sofia/external/hemmeligt at 129.142.224.250) State Change CS_HANGUP ->
> CS_REPORTING
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/hemmeligt at 129.142.224.250) Running State Change
> CS_REPORTING
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:612
> (sofia/external/hemmeligt at 129.142.224.250) State REPORTING
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:53
> sofia/external/hemmeligt at 129.142.224.250 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:612
> (sofia/external/hemmeligt at 129.142.224.250) State REPORTING going to
> sleep
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:411
> (sofia/external/hemmeligt at 129.142.224.250) State Change CS_REPORTING
> -> CS_DESTROY
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_session.c:1068 Session
> 4 (sofia/external/hemmeligt at 129.142.224.250) Locked, Waiting on
> external entities
> 2009-08-26 13:44:46.747796 [NOTICE] switch_core_session.c:1086 Session
> 4 (sofia/external/hemmeligt at 129.142.224.250) Ended
> 2009-08-26 13:44:46.747796 [NOTICE] switch_core_session.c:1088 Close
> Channel sofia/external/hemmeligt at 129.142.224.250 [CS_DESTROY]
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:564
> (sofia/external/hemmeligt at 129.142.224.250) State DESTROY
> 2009-08-26 13:44:46.747796 [DEBUG] mod_sofia.c:255
> sofia/external/hemmeligt at 129.142.224.250 SOFIA DESTROY
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:60
> sofia/external/hemmeligt at 129.142.224.250 Standard DESTROY
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:564
> (sofia/external/hemmeligt at 129.142.224.250) State DESTROY going to
> sleep
>
> Of which I mainly find these interesting:
>
> EXECUTE sofia/external/hemmeligt at 129.142.224.250 bind_meta_app(1 b a
> bridge::sofia/gateway/secretgw/8888)
> 2009-08-26 13:43:26.388737 [INFO] switch_ivr_async.c:1795 Bound B-Leg:
> 1 bridge::sofia/gateway/secretgw/8888
> [...]
> 2009-08-26 13:43:40.309616 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF *:2000
> 2009-08-26 13:43:40.970513 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF 1:2000
> 2009-08-26 13:43:40.970513 [DEBUG] switch_ivr_async.c:1711
> sofia/external/hemmeligt at 129.142.224.250 Processing meta digit '1'
> [bridge::sofia/gateway/secretgw/8888]
> 2009-08-26 13:43:40.970513 [WARNING] switch_ivr_async.c:2310 Channel
> [sofia/external/hemmeligt at 129.142.224.250][bridge::sofia/gateway/secretgw/8888]
> already broadcasting...broadcast aborted
>
> Can anybody explain what is going on here? It seems like the
> bind_meta_app is bound perfectly via event socket, but when it comes
> down, it doesn't work.
>
> Best regards,
> Harry
>


From foxb at abv.bg  Wed Aug 26 14:03:07 2009
From: foxb at abv.bg (Hristo Benev)
Date: Thu, 27 Aug 2009 00:03:07 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <784150205.123595.1251320595248.JavaMail.apache@mail22.abv.bg>

 
It seems that the problem is on kamailio configuration.
Will ask on their list.

But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.

I have it setup as friend in asterisk, but still ???

Any ideas?

Thanks,


 >-------- ?????????? ????? --------
 >??:  Hristo Benev 
 >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >??: freeswitch-users at lists.freeswitch.org
 >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST

 > I think that the problem is here:
 >-------------------------
 >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
 >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
 >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
 >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
 >--------------------------
 >
 >Actually Regex FAIL
 >
 >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
 >Here is my  default.xml:
 >----------------
 >
 >
 >
 >  
 >
 >     
 >      
 >         
 >         
 >         
 >         
 >         
 >         
 >         
 >      
 >    
 >
 >   
 >
 >  
 >
 >    
 >      
 >        
 >        
 >        
 >        
 >        
 >        
 >        
 >        
 >      
 >    
 >
 >  
 >
 >
 >
 >--------------------------
 >
 >
 >
 > >-------- ?????????? ????? --------
 > >??:  Brian West 
 > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 > >??: freeswitch-users at lists.freeswitch.org
 > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
 >
 > >We do not blindly follow 302's as that is a dangerous thing to do. 
 >You have to process all 302's in the dialplan. 
 >Set this on your sofia profile
 >You can set these variables sip_redirect_profile,
 >sip_redirect_context,
 >sip_redirect_dialplan,
 >When a redirect happens you get these variables - sip_redirect_contact_%d,
 >sip_redirected_to,
 >sip_redirect_contact_user_%d,
 >sip_redirect_contact_host_%d,
 >sip_redirect_contact_params_%d,
 >sip_redirect_dialstring_%d,
 >sip_redirect_dialstring,
 >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
 >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
 > >
 >
 >_______________________________________________
 >FreeSWITCH-users mailing list
 >FreeSWITCH-users at lists.freeswitch.org
 >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >http://www.freeswitch.org
 >


From pbd at suspiria.net  Wed Aug 26 14:36:41 2009
From: pbd at suspiria.net (Public Dump)
Date: Wed, 26 Aug 2009 23:36:41 +0200
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
	<191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>
Message-ID: <13C421883438EB42B9E2C30069FD4AB77004E8C0A9@crushinator.central.local>

I am quite sure that it's not a problem with the network connection, since the delay builds up gradually over time, while the host is reachable without any problems.



I have added the suggested config lines and will check the results.



By the way, can I remap the hash key, so that is does not end the conference ? I've tried remapping it, but freeswitch seems to ignore the setting (the console shows: hangup mapped to '*' ... but # still ends the call).

It is irritating for users, since most users try to enter the conference pin the #-key at the end.



Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 22:45
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] delay buildup in conference



There is no bug, it's all dependent on your network conditions.
I spend literally 8-12 hours a day on a conference and there is no delay.

The important param is
<param name="rtp-autoflush-during-bridge" value="true"/>

in the sofia profile in question.

if you have delay with that in place, then it's probably not FS



On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd at suspiria.net<mailto:pbd at suspiria.net>> wrote:

Hi,



I am on a quite recent version (i assume):



FreeSWITCH Version 1.0.trunk (14461)



Should the bug be fixed in this revision ? What config settings would a have to check to limit delay (even at the cost of reduction in quality).



Thanks



Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 20:57
An: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] delay buildup in conference



which revision are you on?
The defaults on the latest code and examples should be configured to minimize delay.
Some of the older revisions built up some delay issues from udp buffering when timers were not synced.

On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net<mailto:pbd at suspiria.net>> wrote:

When running conferences with users dialed in from a PSTN gateway (SIP) and directly from remote SIP endpoints there is an ever longer buildup in delay, reaching up to multiple seconds. Is there any way to limit the delay ?



I am not 100% sure whether the delays is caused by the SIP jitter buffer of freeswitch or directly by the conference module.



Any advice?



Thanks


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400

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From anthony.minessale at gmail.com  Wed Aug 26 15:00:32 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 17:00:32 -0500
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <13C421883438EB42B9E2C30069FD4AB77004E8C0A9@crushinator.central.local>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
	<191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A9@crushinator.central.local>
Message-ID: <191c3a030908261500q78cc50c6gaf4d469722e1edf4@mail.gmail.com>

being unreachable does not cause delay, sending the audio too fast does.
the timer in the rtp stack will slow down the audio stream to match the
clocked rate and if there is a difference
it can translate to delay over time.

This is been especially prevalent in cisco phones but other things have
caused a similar problem.

you can also try disabling rtp timer by setting rtp_timer_name=none before
you answer the call.

the keys are controlled by the mappings in the conference profile


it's a little unnerving talking to someone named public dump, do you have an
identity?



On Wed, Aug 26, 2009 at 4:36 PM, Public Dump <pbd at suspiria.net> wrote:

>  I am quite sure that it?s not a problem with the network connection,
> since the delay builds up gradually over time, while the host is reachable
> without any problems.
>
>
>
> I have added the suggested config lines and will check the results.
>
>
>
> By the way, can I remap the hash key, so that is does not end the
> conference ? I?ve tried remapping it, but freeswitch seems to ignore the
> setting (the console shows: hangup mapped to ?*? ? but # still ends the
> call).
>
> It is irritating for users, since most users try to enter the conference
> pin the #-key at the end.
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Anthony
> Minessale
> *Gesendet:* Mittwoch, 26. August 2009 22:45
>
> *An:* freeswitch-users at lists.freeswitch.org
> *Betreff:* Re: [Freeswitch-users] delay buildup in conference
>
>
>
> There is no bug, it's all dependent on your network conditions.
> I spend literally 8-12 hours a day on a conference and there is no delay.
>
> The important param is
> <param name="rtp-autoflush-during-bridge" value="true"/>
>
> in the sofia profile in question.
>
> if you have delay with that in place, then it's probably not FS
>
>  On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd at suspiria.net> wrote:
>
> Hi,
>
>
>
> I am on a quite recent version (i assume):
>
>
>
> FreeSWITCH Version 1.0.trunk (14461)
>
>
>
> Should the bug be fixed in this revision ? What config settings would a
> have to check to limit delay (even at the cost of reduction in quality).
>
>
>
> Thanks
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Anthony
> Minessale
> *Gesendet:* Mittwoch, 26. August 2009 20:57
> *An:* freeswitch-users at lists.freeswitch.org
> *Betreff:* Re: [Freeswitch-users] delay buildup in conference
>
>
>
> which revision are you on?
> The defaults on the latest code and examples should be configured to
> minimize delay.
> Some of the older revisions built up some delay issues from udp buffering
> when timers were not synced.
>
> On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net> wrote:
>
> When running conferences with users dialed in from a PSTN gateway (SIP) and
> directly from remote SIP endpoints there is an ever longer buildup in delay,
> reaching up to multiple seconds. Is there any way to limit the delay ?
>
>
>
> I am not 100% sure whether the delays is caused by the SIP jitter buffer of
> freeswitch or directly by the conference module.
>
>
>
> Any advice?
>
>
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jaybinks at gmail.com  Wed Aug 26 15:45:21 2009
From: jaybinks at gmail.com (jay binks)
Date: Thu, 27 Aug 2009 08:45:21 +1000
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908261042l42437ed9xc4fab87b757adc80@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com>
	<65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com>
	<7b197bef0908260125m56c3d681l26cb18a5e34aaa33@mail.gmail.com>
	<65d96fc80908260129xebbf460raffaa38c72c3281a@mail.gmail.com>
	<681a20520908260453q57877b1eg8f0949061cd129ee@mail.gmail.com>
	<65d96fc80908261042l42437ed9xc4fab87b757adc80@mail.gmail.com>
Message-ID: <ec2ce17e0908261545i2c2a8537pf36984945fe8b823@mail.gmail.com>

Just for fun...
Im personally interested how this holds up on Debian 5 - 64 bit..

Im more of a debian person... not a huge centos fan and it seems you had a
pref for debian at first also
so if you have the time to play, let us know how that goes :)
surly its just the change to a 64 bit OS....

Jay



On Thu, Aug 27, 2009 at 3:42 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Guys you made a monster!!
>
>
> so, i moved the machine to 64bit CentOS 5.3... recompiled the latest trunk
> and did my tests again.
>
> The old tests on 32bit debian 5 on the same hardware shown a CPS rate of
> 120 as 75 - 80% CPU.... and after some time on that 120 CPS rate the CPU
> goes to 100% without any chance FS recovers at all.
> New tests on 64bit CentOS shown a monster.... 400 CPS rate at 75% CPU....
> during the tests FS was really stable and responsive... i placed few calls
> that went through as a charm :).
>
>
> Also, i didn't optimize the machine at all ... as it is after CentOS
> install!.... not even db files are on ramdisk.
>
> What did it really happen? .. did you guys change something in the trunk
> overnight or it is just moving to CentOS 64bit that boosted drastically?
>
>
>
>
> Here are some details:
>
> ?nmon?12a??????[H for help]???Hostname=l01sipindir2?Refresh= 2secs
> ???19:17.48??????????????????????????????????????????????????????
> ? CPU
> +-------------------------------------------------------------------------+
> ?
> ?100%-|
> |
> ?
> ? 95%-|
> |
> ?
> ? 90%-|
> |
> ?
> ? 85%-|
> |
> ?
> ? 80%-|  | w              s  w      www  w        s   sw    s
> s           s                                                     ?
> ? 75%-|
> |ssssssssssssssssssssswssssssssssssssssssssssssssssssssssssssssssssss
> ?
> ? 70%-|
> +sssssssssssssssssssssssssssssssUssssssssUsssUssssssssssUssssssssssss
> ?
> ? 65%-|
> |UUUUUsUsUUUUUUUUUUUUUsUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUsUUU
> ?
> ? 60%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 55%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 50%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 45%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 40%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUw
> ?
> ? 35%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUs
> ?
> ? 30%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 25%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 20%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 15%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 10%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ?  5%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ?
> +-------------------------------------------------------------------------+
> ?
> ? CPU Utilisation
> ??????????????????????????????????????????????????????????????????????????????????????????????????????????????????
> ?
> +-------------------------------------------------+
> ?
> ?CPU  User%  Sys% Wait% Idle|0          |25         |50          |75
> 100|                                                    ?
> ? 1   1.0   0.5   0.0   98.5|
> >          |                                                    ?
> ? 2   1.5   1.0   0.0   97.5|
> >  |                                                    ?
> ?
> +-------------------------------------------------+
> ?
> ?Avg  1.2   0.5   0.0   98.3|
> >         |                                                    ?
> ?
> +-------------------------------------------------+
> ?
> ? Disk I/O ?????(/proc/diskstats)????????all data is Kbytes per
> second??????????????????????????????????????????????????????????????
> ?DiskName Busy  Read WriteKB|0          |25         |50          |75
> 100|                                                    ?
> ?iss/c0d0   0%    0.0    0.0|
> >   |                                                    ?
> ?s/c0d0p1   0%    0.0
> 0.0|>
> |                                                    ?
> ?s/c0d0p2   0%    0.0    0.0|
> >  |                                                    ?
> ?dm-0       0%    0.0    0.0|
> >  |                                                    ?
> ?dm-1       0%    0.0
> 0.0|>
> |                                                    ?
>
> ????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????
>
>
>
>
>
>
>
> ------------------------------ Scenario Screen -------- [1-9]: Change
> Screen --
>   Call-rate(length)   Port   Total-time  Total-calls  Remote-host
>   40.0(0 ms)/0.100s   5060     488.20 s       193555  10.4.4.252:5060(UDP)
>
>   402 new calls during 1.002 s period    0 ms scheduler resolution
>   3 calls (limit 4000)                   Peak was 53 calls, after 351 s
>   0 Running, 13216 Paused, 670 Woken up
>   0 dead call msg (discarded)            0 out-of-call msg
> (discarded)
>   3 open sockets
>
>                                  Messages  Retrans   Timeout
> Unexpected-Msg
>       INVITE ---------->  B-RTD1 193553    0         0
>          100 <----------  E-RTD1 193553    0         0         0
>          302 <----------  E-RTD2 193552    0         0         0
>          ACK ---------->         193552    0
> ------ [+|-|*|/]: Adjust rate ---- [q]: Soft exit ---- [p]: Pause traffic
> -----
>
> ------------------------------ Scenario Screen -------- [1-9]: Change
> Screen --
>   Call-rate(length)   Port   Total-time  Total-calls  Remote-host
>   40.0(0 ms)/0.100s   5060     489.10 s       193917  10.4.4.252:5060(UDP)
>
>   362 new calls during 0.906 s period    1 ms scheduler resolution
>   2 calls (limit 4000)                   Peak was 53 calls, after 351 s
>   0 Running, 13215 Paused, 623 Woken up
>   0 dead call msg (discarded)            0 out-of-call msg
> (discarded)
>   3 open sockets
>
>                                  Messages  Retrans   Timeout
> Unexpected-Msg
>       INVITE ---------->  B-RTD1 193917    0         0
>          100 <----------  E-RTD1 193917    0         0         0
>          302 <----------  E-RTD2 193915    0         0         0
>          ACK ---------->         193915    0
> ------------------------------ Test Terminated
> --------------------------------
>
>
> ----------------------------- Statistics Screen ------- [1-9]: Change
> Screen --
>   Start Time             | 2009-08-26   19:09:34:575
> 1251306574.575684
>   Last Reset Time        | 2009-08-26   19:17:42:779
> 1251307062.779468
>   Current Time           | 2009-08-26   19:17:43:685
> 1251307063.685281
>
> -------------------------+---------------------------+--------------------------
>   Counter Name           | Periodic value            | Cumulative value
>
> -------------------------+---------------------------+--------------------------
>   Elapsed Time           | 00:00:00:905              |
> 00:08:09:109
>   Call Rate              |  400.000 cps              |  396.470
> cps
>
> -------------------------+---------------------------+--------------------------
>   Incoming call created  |        0                  |
> 0
>   OutGoing call created  |      362                  |
> 193917
>   Total Call created     |                           |
> 193917
>   Current Call           |        2
> |
>
> -------------------------+---------------------------+--------------------------
>   Successful call        |      363                  |
> 193915
>   Failed call            |        0                  |
> 0
>
> -------------------------+---------------------------+--------------------------
>   Response Time 1        | 00:00:00:001              |
> 00:00:00:000
>   Response Time 2        | 00:00:00:010              |
> 00:00:00:008
>   Call Length            | 00:00:00:010              |
> 00:00:00:008
> ------------------------------ Test Terminated
> --------------------------------
>
>
>
> ...i didn't beleive to SIPp and i went to FS console issuing status command
> to conferm the results.
>
>
> freeswitch at l01sipindir2.ot.hr> status
> API CALL [status()] output:
> UP 0 years, 0 days, 0 hours, 8 minutes, 13 seconds, 703 milliseconds, 971
> microseconds
> 183382 session(s) since startup
> 1 session(s) 410/800
> 8000 session(s) max
>
> freeswitch at l01sipindir2.ot.hr> status
> API CALL [status()] output:
> UP 0 years, 0 days, 0 hours, 8 minutes, 15 seconds, 109 milliseconds, 891
> microseconds
> 183944 session(s) since startup
> 1 session(s) 401/800
> 8000 session(s) max
>
> freeswitch at l01sipindir2.ot.hr> status
> API CALL [status()] output:
> UP 0 years, 0 days, 0 hours, 8 minutes, 16 seconds, 139 milliseconds, 412
> microseconds
> 184356 session(s) since startup
> 2 session(s) 389/800
> 8000 session(s) max
>
> freeswitch at l01sipindir2.ot.hr> status
> API CALL [status()] output:
> UP 0 years, 0 days, 0 hours, 8 minutes, 17 seconds, 62 milliseconds, 16
> microseconds
> 184717 session(s) since startup
> 6 session(s) 410/800
> 8000 session(s) max
>
> freeswitch at l01sipindir2.ot.hr> status
> API CALL [status()] output:
> UP 0 years, 0 days, 0 hours, 8 minutes, 35 seconds, 150 milliseconds, 253
> microseconds
> 191959 session(s) since startup
> 1 session(s) 400/800
> 8000 session(s) max
>
> freeswitch at l01sipindir2.ot.hr> status
> API CALL [status()] output:
> UP 0 years, 0 days, 0 hours, 8 minutes, 36 seconds, 892 milliseconds, 672
> microseconds
> 192657 session(s) since startup
> 1 session(s) 393/800
> 8000 session(s) max
>
>
>
>
> On Wed, Aug 26, 2009 at 1:53 PM, Dmitry Kadantsev <kadantsev.d at gmail.com>wrote:
>
>> Hi all,
>>
>> is there same situation with FS for Windows? I mean 64bit is more
>> preferable than 32bit, isn't it?
>>
>> Any performance test on Win 32/64 were done?
>>
>> --
>> Best regards,
>> Dmitry Kadantsev
>>
>>
>>
>> On Wed, Aug 26, 2009 at 10:29 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> intanto e il centos che si sta installando :)
>>>
>>> grazie.
>>>
>>> T.
>>>
>>>
>>> On Wed, Aug 26, 2009 at 10:25 AM, Giovanni Maruzzelli <
>>> gmaruzz at celliax.org> wrote:
>>>
>>>> netbook remix
>>>>
>>>>
>>>> joking! Server 64bit :-)
>>>>
>>>> -gm
>>>>
>>>>
>>>>
>>>> On Wed, Aug 26, 2009 at 10:08 AM, Tihomir Culjaga<tculjaga at gmail.com>
>>>> wrote:
>>>> > Hi Giovanny,
>>>> >
>>>> > regarding ubuntu, did you mean 8.04 server or desktop ?
>>>> >
>>>> >
>>>> > On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli <
>>>> gmaruzz at celliax.org>
>>>> > wrote:
>>>> >>
>>>> >> Definitely go for 64 bit OS.
>>>> >>
>>>> >> If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
>>>> >> used both for development and for heavy duty production.
>>>> >>
>>>> >> Also Ubuntu 8.04 is good.
>>>> >>
>>>> >> Other versions/distros are less used by the community.
>>>> >>
>>>> >> Adding RAM and CPUs helps to scale up.
>>>> >>
>>>> >> -gm
>>>> >>
>>>> >>
>>>> >>
>>>> >> Sincerely,
>>>> >>
>>>> >> Giovanni Maruzzelli
>>>> >> Cell : +39-347-2665618
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >> On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com>
>>>> >> wrote:
>>>> >> > Hey Giovanni,
>>>> >> >
>>>> >> > thanks for the tip... indeed the db files were heavily used
>>>> regardless
>>>> >> > if i
>>>> >> > started freeswitch with nosql option (freeswitch -nosql)... FS was
>>>> not
>>>> >> > writing anything into that files ... instead it was just accessing
>>>> >> > it....
>>>> >> > This behaviour leads to a waste of 40% CPU time... waiting for
>>>> other
>>>> >> > processes (mainly disk access) to finish!!!
>>>> >> >
>>>> >> > I moved freeswitch/db/ to a ramdisk and the performance got a boost
>>>> to
>>>> >> > 140
>>>> >> > CPS with a CPU load of 80%. I was keeping the machine for a while
>>>> (20 -
>>>> >> > 30
>>>> >> > minutes) on that rate when i sow CPU suddenly went to 100% and FS
>>>> >> > becoming
>>>> >> > irresponsive :).
>>>> >> >
>>>> >> >
>>>> >> > What can be wrong?
>>>> >> > What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should
>>>> not
>>>> >> > cross?
>>>> >> > What fine tuning do we need in order to asure a long high load run?
>>>> >> >
>>>> >> >
>>>> >> >
>>>> >> > Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it
>>>> have
>>>> >> > sense
>>>> >> > to move my OS to 64 bit? ... will FS gain more preformance ?... I
>>>> mean
>>>> >> > will
>>>> >> > FS perofomr drastically better 20%+ ?
>>>> >> >
>>>> >> >
>>>> >> > Tihomir.
>>>> >> >
>>>> >> >
>>>> >> > On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
>>>> >> > <gmaruzz at celliax.org>
>>>> >> > wrote:
>>>> >> >>
>>>> >> >> Maybe your load comes from disk access?
>>>> >> >>
>>>> >> >> Try putting the sql and log directories on a ramdisk.
>>>> >> >>
>>>> >> >> OTH,
>>>> >> >>
>>>> >> >> -giovanni
>>>> >> >>
>>>> >> >> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<
>>>> tculjaga at gmail.com>
>>>> >> >> wrote:
>>>> >> >> > Hello,
>>>> >> >> >
>>>> >> >> > i'm trying to use freeswitch as a redirecting server meaning FS
>>>> has
>>>> >> >> > to
>>>> >> >> > receive an INVITE and according to some rules it will redirect
>>>> calls
>>>> >> >> > to
>>>> >> >> > other destinations.
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > CALLING_USER                FREESWITCH
>>>> >> >> > SOMEWHERE
>>>> >> >> >
>>>> >> >> > INVITE ------------------------------->
>>>> >> >> >            <------------------------------ 100 Trying
>>>> >> >> >            <------------------------------ 302 Moved Temporary
>>>> >> >> > ACK    ------------------------------->
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> INVITE--------------------------------------------------------------------------------->
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > Well, wverything works well except i have perfromance issues
>>>> .... on
>>>> >> >> > my
>>>> >> >> > HW
>>>> >> >> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
>>>> >> >> > Temporary).
>>>> >> >> > When
>>>> >> >> > i increase the rate, FS starts delaying 302 response. Right at
>>>> 50 CPS
>>>> >> >> > i
>>>> >> >> > see
>>>> >> >> > "calls" being build up in FS and the delay begining to grow.
>>>> >> >> >
>>>> >> >> > When i observe the machine, load average is almost nothing (load
>>>> >> >> > average:
>>>> >> >> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one
>>>> thread
>>>> >> >> > taking
>>>> >> >> > most load... all others are just sitting there with 1-5 % CPU
>>>> time.
>>>> >> >> > This looks to me as FS handles 302 messages in a single
>>>> thread?!?!
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>>>> >> >> >
>>>> >> >> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average:
>>>> 1.41,
>>>> >> >> > 0.61,
>>>> >> >> > 0.60
>>>> >> >> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0
>>>> zombie
>>>> >> >> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,
>>>> >> >> > 0.2%si,
>>>> >> >> > 0.0%st
>>>> >> >> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k
>>>> >> >> > buffers
>>>> >> >> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k
>>>> >> >> > cached
>>>> >> >> >
>>>> >> >> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
>>>> >> >> > COMMAND
>>>> >> >> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
>>>> >> >> > freeswitch
>>>> >> >> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
>>>> >> >> > freeswitch
>>>> >> >> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
>>>> >> >> > freeswitch
>>>> >> >> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
>>>> >> >> > freeswitch
>>>> >> >> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
>>>> >> >> > freeswitch
>>>> >> >> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
>>>> >> >> > freeswitch
>>>> >> >> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
>>>> >> >> > freeswitch
>>>> >> >> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
>>>> >> >> > freeswitch
>>>> >> >> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
>>>> >> >> > freeswitch
>>>> >> >> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50
>>>> >> >> > freeswitch
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > cat /proc/cpuinfo
>>>> >> >> > processor       : 0
>>>> >> >> > vendor_id       : GenuineIntel
>>>> >> >> > cpu family      : 6
>>>> >> >> > model           : 15
>>>> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @
>>>> 2.33GHz
>>>> >> >> > stepping        : 6
>>>> >> >> > cpu MHz         : 2333.560
>>>> >> >> > cache size      : 4096 KB
>>>> >> >> > physical id     : 0
>>>> >> >> > siblings        : 2
>>>> >> >> > core id         : 0
>>>> >> >> > cpu cores       : 2
>>>> >> >> > apicid          : 0
>>>> >> >> > initial apicid  : 0
>>>> >> >> > fdiv_bug        : no
>>>> >> >> > hlt_bug         : no
>>>> >> >> > f00f_bug        : no
>>>> >> >> > coma_bug        : no
>>>> >> >> > fpu             : yes
>>>> >> >> > fpu_exception   : yes
>>>> >> >> > cpuid level     : 10
>>>> >> >> > wp              : yes
>>>> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep
>>>> mtrr
>>>> >> >> > pge
>>>> >> >> > mca
>>>> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe
>>>> lm
>>>> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est
>>>> tm2
>>>> >> >> > ssse3
>>>> >> >> > cx16
>>>> >> >> > xtpr dca lahf_lm
>>>> >> >> > bogomips        : 4670.78
>>>> >> >> > clflush size    : 64
>>>> >> >> > power management:
>>>> >> >> >
>>>> >> >> > processor       : 1
>>>> >> >> > vendor_id       : GenuineIntel
>>>> >> >> > cpu family      : 6
>>>> >> >> > model           : 15
>>>> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @
>>>> 2.33GHz
>>>> >> >> > stepping        : 6
>>>> >> >> > cpu MHz         : 2333.560
>>>> >> >> > cache size      : 4096 KB
>>>> >> >> > physical id     : 0
>>>> >> >> > siblings        : 2
>>>> >> >> > core id         : 1
>>>> >> >> > cpu cores       : 2
>>>> >> >> > apicid          : 1
>>>> >> >> > initial apicid  : 1
>>>> >> >> > fdiv_bug        : no
>>>> >> >> > hlt_bug         : no
>>>> >> >> > f00f_bug        : no
>>>> >> >> > coma_bug        : no
>>>> >> >> > fpu             : yes
>>>> >> >> > fpu_exception   : yes
>>>> >> >> > cpuid level     : 10
>>>> >> >> > wp              : yes
>>>> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep
>>>> mtrr
>>>> >> >> > pge
>>>> >> >> > mca
>>>> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe
>>>> lm
>>>> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est
>>>> tm2
>>>> >> >> > ssse3
>>>> >> >> > cx16
>>>> >> >> > xtpr dca lahf_lm
>>>> >> >> > bogomips        : 4666.82
>>>> >> >> > clflush size    : 64
>>>> >> >> > power management:
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > uname -a
>>>> >> >> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC
>>>> 2009
>>>> >> >> > i686
>>>> >> >> > GNU/Linux
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > Of course, i've tuned the machine up
>>>> >> >> >
>>>> >> >> > ulimit -c unlimited
>>>> >> >> > ulimit -d unlimited
>>>> >> >> > ulimit -f unlimited
>>>> >> >> > ulimit -i unlimited
>>>> >> >> > ulimit -n 999999
>>>> >> >> > ulimit -q unlimited
>>>> >> >> > ulimit -u unlimited
>>>> >> >> > ulimit -v unlimited
>>>> >> >> > ulimit -x unlimited
>>>> >> >> > ulimit -s 240
>>>> >> >> > ulimit -l unlimited
>>>> >> >> > ulimit -a
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > Started FS with minimum modules but still 40 CPS seems to be the
>>>> >> >> > limit.
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > So, is there any way to improve performance?
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > Tihomir.
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > _______________________________________________
>>>> >> >> > FreeSWITCH-users mailing list
>>>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> >> >
>>>> >> >> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> >> > http://www.freeswitch.org
>>>> >> >> >
>>>> >> >> >
>>>> >> >>
>>>> >> >> _______________________________________________
>>>> >> >> FreeSWITCH-users mailing list
>>>> >> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> >>
>>>> >> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> >> http://www.freeswitch.org
>>>> >> >
>>>> >> >
>>>> >> > _______________________________________________
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> > http://www.freeswitch.org
>>>> >> >
>>>> >> >
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From larclap at yahoo.com  Wed Aug 26 18:00:00 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 26 Aug 2009 18:00:00 -0700
Subject: [Freeswitch-users] Dial phone number + extension?
Message-ID: <001c01ca26b1$b45bc360$1d134a20$@com>

Is there a way to dial an external 10-digit phone number, wait a second or
two after connecting, and then dial a 4-digit extension?

 

Thanks, Lars

 

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From Prometheus001 at gmx.net  Wed Aug 26 18:05:49 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 27 Aug 2009 03:05:49 +0200
Subject: [Freeswitch-users] sofia profile external register gwname via
	XML-Curl?
Message-ID: <4A95DBED.2060505@gmx.net>

Hello,

I am using XML-Curl to handle the configuration of freeswitch
When I try to register a gateway via event-socket with
    sofia profile external register <gw-name>
I receive back "invalid gateway".

After reload mod_sofia the gateway is there. Question: Does this command
work with xml-curl or only with local files??
At least I see no xml-curl request when grepping network traffic.

Best regards
Peter


From msc at freeswitch.org  Wed Aug 26 19:15:45 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 26 Aug 2009 19:15:45 -0700
Subject: [Freeswitch-users] Dial phone number + extension?
Message-ID: <A057C640-E152-4218-A4FB-CFF82FC48FE1@freeswitch.org>



Sent from my iPhone

On Aug 26, 2009, at 6:00 PM, "Lars Zeb" <larclap at yahoo.com> wrote:

> Is there a way to dial an external 10-digit phone number, wait a  
> second or two after connecting, and then dial a 4-digit extension?
>
>
 From the CLI:
originate sofia/gateway/mygw/1234567890 1234

As for the short pause you could put that in the Dialplan entry for  
ext 1234 or you could even do an inline Dialplan instead of just 1234  
as the second arg to originate.

originate sofia/foo/1234567890 sleep:2000,transfer:1234 inline

Search the wiki for "inline dialplan" to learn more about this handy  
feature
-MC
>
>
> Thanks, Lars
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From ahmedmunir007 at gmail.com  Wed Aug 26 20:57:51 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Thu, 27 Aug 2009 09:57:51 +0600
Subject: [Freeswitch-users] Define Condition in FreeSwitch
Message-ID: <a60fed5e0908262057k5c49cb91t68466a69d6a7df56@mail.gmail.com>

Hi,

   I'm newbie. How can we translate the asterisk's condition in freeswitch
as listed below;

   1. NoOp ("Remote Conference Call")
   2. GotoIf ($[${LEN(${DIALSTR})}=0]?3:4)
   3. Hangup()
   4. NoOp(Finish if-CONFERENCE-430)

   Kindly reply soon.

-- 
Regards,

Ahmed Munir
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From lakindia89 at gmail.com  Wed Aug 26 21:38:43 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Wed, 26 Aug 2009 21:38:43 -0700 (PDT)
Subject: [Freeswitch-users]  How to make a call back
Message-ID: <25166083.post@talk.nabble.com>


When I give the following from the command line it calls to 1010 extension
and once answered, it calls to 1000 and bridge the connection.
            originate user/1010 &bridge(user/1000)
But I want to do this in perl. So I have given as follows
            $session->originate($session,"user/1010 &bridge user/1000");
But it is not working. It says "user/1010 &bridge user/1000 is invalid
user". 
How to do this in perl. pls help.

-- 
View this message in context: http://www.nabble.com/How-to-make-a-call-back-tp25166083p25166083.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From delianspam at gmail.com  Wed Aug 26 23:35:02 2009
From: delianspam at gmail.com (delianSPAM)
Date: Thu, 27 Aug 2009 09:35:02 +0300
Subject: [Freeswitch-users] Timers/DTMFs During a Call
In-Reply-To: <87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>
	<87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAPzw61gTNaNJsjaEGYPk5BEBAAAAAA==@gmail.com>

Hello Michael!

 

Thank you for your reply! I know about sched_api and I use Python as a
scripting language. Also "sched_api" can be used inside python, using
session.execute. However the problem is that the "sched_api" timer starts
right after you initiate the SETUP of the second call leg. What I need is to
call something, after a call CONNECT instead. One workaround would be if I
can check what time it took to connect the call, but I do not know/see how
to do this. I do not see a CONNECT callback function either.

 

Best Regards, Delian Tashev

P.S. Dear enlightened people, thank you for providing help to the community
by replying to the list e-mails.

 

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Wednesday, August 26, 2009 11:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call

 

Do you actually need Python for the IVR, or is it that you're comfortable
using a scripting lang for an IVR? I like using XML for IVRs, but using
scripting langs does give you a bit more power & flexibility at the cost of
some resources.

For the record, you can do this in the dialplan using XML and sched_api
without touching a scripting language. Checkout the sched_api channel
variable on the wiki - it may give you the functionality you need.

-MC

On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM <delianspam at gmail.com> wrote:

Hello Everybody!

 

1.       Scenario.

I am writing an IVR in Python that gets a destination from the calling party
(party A) and then connects to the destination (party B).

When the call is CONNECTED, I want to:

-          Receive DTMFs

-          Have a timer that can call a certain function in my script. The
script will have to play a message to party A.

-          Have a timer that can call a certain function in my script. The
script will have to drop the call.

Please notice that I want to do the things after the two parties are
connected, and not after I send the Invite to party B.

 

2.       Problem.

I will be happy to receive help on:

-          Which methods should I look for to implement this.

 

3.       Details

Here is how I connect the call currently:

session.execute("bridge","sofia/internal/" + destination_number +
"@domain.com")

 

I have tried to create a timer callback function "my_method()" using:

ivr_timer =threading.Timer(30,my_method)

This never called the function "my_method()".

 

Maybe I am wrong in using threading.Timer and the "bridge" application?
Maybe I need to create a new thread and a new timer using the API of
freeswitch, plus to use the "session.setInputCallback", plus use a
conference rather than a bridge? Can you please provide any suggestions or
examples? 

 

Thank you!

Best Regards, Delian Tashev


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From kawarod at laposte.net  Wed Aug 26 23:41:05 2009
From: kawarod at laposte.net (rod)
Date: Thu, 27 Aug 2009 10:41:05 +0400
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
In-Reply-To: <784150205.123595.1251320595248.JavaMail.apache@mail22.abv.bg>
References: <784150205.123595.1251320595248.JavaMail.apache@mail22.abv.bg>
Message-ID: <4A962A81.3010709@laposte.net>

Hi Hristo,

I'm the author of this setup and wiki page. I did a lot of modifications 
on this setup (alternative routing if failure essentially) but don't 
have too much time to update the wiki.

May you please send me an ngrep trace when you call 1000:

ngrep -d any -nn -i '1000' port 5060 -W byline 

I will check what's happening.
Do you have an entry for 1000 in your mysql database ?

regards,
rod

Hristo Benev a ?crit :
>  
> It seems that the problem is on kamailio configuration.
> Will ask on their list.
>
> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
>
> I have it setup as friend in asterisk, but still ???
>
> Any ideas?
>
> Thanks,
>
>
>  >-------- ?????????? ????? --------
>  >??:  Hristo Benev 
>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >??: freeswitch-users at lists.freeswitch.org
>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
>
>  > I think that the problem is here:
>  >-------------------------
>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
>  >--------------------------
>  >
>  >Actually Regex FAIL
>  >
>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
>  >Here is my  default.xml:
>  >----------------
>  >
>  >
>  >
>  >  
>  >
>  >     
>  >      
>  >         
>  >         
>  >         
>  >         
>  >         
>  >         
>  >         
>  >      
>  >    
>  >
>  >   
>  >
>  >  
>  >
>  >    
>  >      
>  >        
>  >        
>  >        
>  >        
>  >        
>  >        
>  >        
>  >        
>  >      
>  >    
>  >
>  >  
>  >
>  >
>  >
>  >--------------------------
>  >
>  >
>  >
>  > >-------- ?????????? ????? --------
>  > >??:  Brian West 
>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  > >??: freeswitch-users at lists.freeswitch.org
>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
>  >
>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
>  >You have to process all 302's in the dialplan. 
>  >Set this on your sofia profile
>  >You can set these variables sip_redirect_profile,
>  >sip_redirect_context,
>  >sip_redirect_dialplan,
>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
>  >sip_redirected_to,
>  >sip_redirect_contact_user_%d,
>  >sip_redirect_contact_host_%d,
>  >sip_redirect_contact_params_%d,
>  >sip_redirect_dialstring_%d,
>  >sip_redirect_dialstring,
>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
>  > >
>  >
>  >_______________________________________________
>  >FreeSWITCH-users mailing list
>  >FreeSWITCH-users at lists.freeswitch.org
>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >http://www.freeswitch.org
>  >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From delianspam at gmail.com  Wed Aug 26 23:52:35 2009
From: delianspam at gmail.com (delianSPAM)
Date: Thu, 27 Aug 2009 09:52:35 +0300
Subject: [Freeswitch-users] Timers/DTMFs During a Call
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>
	<87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com> 
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAANZUwRmfIlpOtIAK9HYapUoBAAAAAA==@gmail.com>

Looks like I was wrong about using the native Python timers. Here is how
they can be used in your script:

 

# Imports - add these new imports

import time

import threading

 

# class definitions - add this new class

class Timer(threading.Thread):

                def __init__(self, seconds):

                               self.runTime = seconds

                               threading.Thread.__init__(self)

                def run(self):

                               time.sleep(self.runTime)

                               console_log("debug", "TIMER
********************")

 

# entry point - add two rows in the entry point function that is called from
freeswitch

def handler(session, args):

.            

 t = Timer(10)

 t.start()

.

 

So what is now remaining is to get when the call CONNECTS and how to get
DTMFs during the call. 

 

From: delianSPAM [mailto:delianspam at gmail.com] 
Sent: Thursday, August 27, 2009 9:35 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call

 

Hello Michael!

 

Thank you for your reply! I know about sched_api and I use Python as a
scripting language. Also "sched_api" can be used inside python, using
session.execute. However the problem is that the "sched_api" timer starts
right after you initiate the SETUP of the second call leg. What I need is to
call something, after a call CONNECT instead. One workaround would be if I
can check what time it took to connect the call, but I do not know/see how
to do this. I do not see a CONNECT callback function either.

 

Best Regards, Delian Tashev

P.S. Dear enlightened people, thank you for providing help to the community
by replying to the list e-mails.

 

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Wednesday, August 26, 2009 11:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call

 

Do you actually need Python for the IVR, or is it that you're comfortable
using a scripting lang for an IVR? I like using XML for IVRs, but using
scripting langs does give you a bit more power & flexibility at the cost of
some resources.

For the record, you can do this in the dialplan using XML and sched_api
without touching a scripting language. Checkout the sched_api channel
variable on the wiki - it may give you the functionality you need.

-MC

On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM <delianspam at gmail.com> wrote:

Hello Everybody!

 

1.       Scenario.

I am writing an IVR in Python that gets a destination from the calling party
(party A) and then connects to the destination (party B).

When the call is CONNECTED, I want to:

-          Receive DTMFs

-          Have a timer that can call a certain function in my script. The
script will have to play a message to party A.

-          Have a timer that can call a certain function in my script. The
script will have to drop the call.

Please notice that I want to do the things after the two parties are
connected, and not after I send the Invite to party B.

 

2.       Problem.

I will be happy to receive help on:

-          Which methods should I look for to implement this.

 

3.       Details

Here is how I connect the call currently:

session.execute("bridge","sofia/internal/" + destination_number +
"@domain.com")

 

I have tried to create a timer callback function "my_method()" using:

ivr_timer =threading.Timer(30,my_method)

This never called the function "my_method()".

 

Maybe I am wrong in using threading.Timer and the "bridge" application?
Maybe I need to create a new thread and a new timer using the API of
freeswitch, plus to use the "session.setInputCallback", plus use a
conference rather than a bridge? Can you please provide any suggestions or
examples? 

 

Thank you!

Best Regards, Delian Tashev


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From delianspam at gmail.com  Thu Aug 27 00:01:09 2009
From: delianspam at gmail.com (delianSPAM)
Date: Thu, 27 Aug 2009 10:01:09 +0300
Subject: [Freeswitch-users] Timers/DTMFs During a Call
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>
	<87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com> 
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAPlzV3BY/cZHrlu/Yf4kc9QBAAAAAA==@gmail.com>

I will try to execute and parse from python:

freeswitch at internal> show channels

uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,des
t,application,application_data,dialplan,context,read_codec,read_rate,write_c
odec,write_rate

53a62ebd-156c-4684-b616-740d7a5b609b,inbound,2009-04-23
11:14:09,1240510449,sofia/internal/1000 at ...,CS_EXECUTE,Mikey,1000,10.15.0.21
3,9999,playback,local_stream://moh,XML,default,PCMU,8000,PCMU,8000

 

Hoping that this will get the state of the call. If I call this check
frequently I will catch the call connect I trust.

 

From: delianSPAM [mailto:delianspam at gmail.com] 
Sent: Thursday, August 27, 2009 9:53 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call

 

Looks like I was wrong about using the native Python timers. Here is how
they can be used in your script:

 

# Imports - add these new imports

import time

import threading

 

# class definitions - add this new class

class Timer(threading.Thread):

                def __init__(self, seconds):

                               self.runTime = seconds

                               threading.Thread.__init__(self)

                def run(self):

                               time.sleep(self.runTime)

                               console_log("debug", "TIMER
********************")

 

# entry point - add two rows in the entry point function that is called from
freeswitch

def handler(session, args):

.            

 t = Timer(10)

 t.start()

.

 

So what is now remaining is to get when the call CONNECTS and how to get
DTMFs during the call. 

 

From: delianSPAM [mailto:delianspam at gmail.com] 
Sent: Thursday, August 27, 2009 9:35 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call

 

Hello Michael!

 

Thank you for your reply! I know about sched_api and I use Python as a
scripting language. Also "sched_api" can be used inside python, using
session.execute. However the problem is that the "sched_api" timer starts
right after you initiate the SETUP of the second call leg. What I need is to
call something, after a call CONNECT instead. One workaround would be if I
can check what time it took to connect the call, but I do not know/see how
to do this. I do not see a CONNECT callback function either.

 

Best Regards, Delian Tashev

P.S. Dear enlightened people, thank you for providing help to the community
by replying to the list e-mails.

 

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Wednesday, August 26, 2009 11:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call

 

Do you actually need Python for the IVR, or is it that you're comfortable
using a scripting lang for an IVR? I like using XML for IVRs, but using
scripting langs does give you a bit more power & flexibility at the cost of
some resources.

For the record, you can do this in the dialplan using XML and sched_api
without touching a scripting language. Checkout the sched_api channel
variable on the wiki - it may give you the functionality you need.

-MC

On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM <delianspam at gmail.com> wrote:

Hello Everybody!

 

1.       Scenario.

I am writing an IVR in Python that gets a destination from the calling party
(party A) and then connects to the destination (party B).

When the call is CONNECTED, I want to:

-          Receive DTMFs

-          Have a timer that can call a certain function in my script. The
script will have to play a message to party A.

-          Have a timer that can call a certain function in my script. The
script will have to drop the call.

Please notice that I want to do the things after the two parties are
connected, and not after I send the Invite to party B.

 

2.       Problem.

I will be happy to receive help on:

-          Which methods should I look for to implement this.

 

3.       Details

Here is how I connect the call currently:

session.execute("bridge","sofia/internal/" + destination_number +
"@domain.com")

 

I have tried to create a timer callback function "my_method()" using:

ivr_timer =threading.Timer(30,my_method)

This never called the function "my_method()".

 

Maybe I am wrong in using threading.Timer and the "bridge" application?
Maybe I need to create a new thread and a new timer using the API of
freeswitch, plus to use the "session.setInputCallback", plus use a
conference rather than a bridge? Can you please provide any suggestions or
examples? 

 

Thank you!

Best Regards, Delian Tashev


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From msc at freeswitch.org  Thu Aug 27 00:01:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 00:01:41 -0700
Subject: [Freeswitch-users] Define Condition in FreeSwitch
In-Reply-To: <a60fed5e0908262057k5c49cb91t68466a69d6a7df56@mail.gmail.com>
References: <a60fed5e0908262057k5c49cb91t68466a69d6a7df56@mail.gmail.com>
Message-ID: <87f2f3b90908270001o1e0611fsa7a2f61893571333@mail.gmail.com>

On Wed, Aug 26, 2009 at 8:57 PM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:

> Hi,
>
>    I'm newbie. How can we translate the asterisk's condition in freeswitch
> as listed below;
>
>    1. NoOp ("Remote Conference Call")
>    2. GotoIf ($[${LEN(${DIALSTR})}=0]?3:4)
>    3. Hangup()
>    4. NoOp(Finish if-CONFERENCE-430)
>
>    Kindly reply soon.


Before I answer this question I just want you to know that there's probably
a more elegant way of doing whatever it is you're trying to do. This
dialplan snippet is pretty short. My first question would be: how does a
call get to this point? Also, what is the big picture, that is, what's the
application you're creating? Remember the golden rule: Anything that you do
in Asterisk is easier to do in FreeSWITCH, but you need to learn the ropes a
bit.

The answer to your question is, of course, "It depends." :P Give us the
background on what you're doing so that we can give you an educated answer.
You could use the <condition> tags with actions and anti-actions. You could
also call a Lua/JavaScript/Perl/Python/etc. script to handle the logic but
that's probably overkill.

Tell us more and we'll tell you more. ;)

Thanks!
-Michael Collins
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From msc at freeswitch.org  Thu Aug 27 00:04:50 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 00:04:50 -0700
Subject: [Freeswitch-users] Timers/DTMFs During a Call
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAPzw61gTNaNJsjaEGYPk5BEBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>
	<87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com>
	<!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAPzw61gTNaNJsjaEGYPk5BEBAAAAAA==@gmail.com>
Message-ID: <87f2f3b90908270004u2845df79t4bf908c7556a4b6b@mail.gmail.com>

On Wed, Aug 26, 2009 at 11:35 PM, delianSPAM <delianspam at gmail.com> wrote:

>  Hello Michael!
>
>
>
> Thank you for your reply! I know about sched_api and I use Python as a
> scripting language. Also ?sched_api? can be used inside python, using
> session.execute. However the problem is that the ?sched_api? timer starts
> right after you initiate the SETUP of the second call leg. What I need is to
> call something, after a call CONNECT instead. One workaround would be if I
> can check what time it took to connect the call, but I do not know/see how
> to do this. I do not see a CONNECT callback function either.
>
>
>
Perhaps you could set this channel variable to the sched_api call?
http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer
-MC
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From delianspam at gmail.com  Thu Aug 27 00:35:03 2009
From: delianspam at gmail.com (delianSPAM)
Date: Thu, 27 Aug 2009 10:35:03 +0300
Subject: [Freeswitch-users] Timers/DTMFs During a Call
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAPlzV3BY/cZHrlu/Yf4kc9QBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>	<87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com>
	<!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAPlzV3BY/cZHrlu/Yf4kc9QBAAAAAA==@gmail.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAACPZt1oMIAxCnItIAzX5gwYBAAAAAA==@gmail.com>

To get the state use: 

session.getVariable("state")

 

I hope that I will solve the Timers puzzle soon. I am still looking on
getting the DTMFs during a bridged call.

 

Best Regards, Delian Tashev

 

From: delianSPAM [mailto:delianspam at gmail.com] 
Sent: Thursday, August 27, 2009 10:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call

 

I will try to execute and parse from python:

freeswitch at internal> show channels

uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,des
t,application,application_data,dialplan,context,read_codec,read_rate,write_c
odec,write_rate

53a62ebd-156c-4684-b616-740d7a5b609b,inbound,2009-04-23
11:14:09,1240510449,sofia/internal/1000 at ...,CS_EXECUTE,Mikey,1000,10.15.0.21
3,9999,playback,local_stream://moh,XML,default,PCMU,8000,PCMU,8000

 

Hoping that this will get the state of the call. If I call this check
frequently I will catch the call connect I trust.

 

From: delianSPAM [mailto:delianspam at gmail.com] 
Sent: Thursday, August 27, 2009 9:53 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call

 

Looks like I was wrong about using the native Python timers. Here is how
they can be used in your script:

 

# Imports - add these new imports

import time

import threading

 

# class definitions - add this new class

class Timer(threading.Thread):

                def __init__(self, seconds):

                               self.runTime = seconds

                               threading.Thread.__init__(self)

                def run(self):

                               time.sleep(self.runTime)

                               console_log("debug", "TIMER
********************")

 

# entry point - add two rows in the entry point function that is called from
freeswitch

def handler(session, args):

.            

 t = Timer(10)

 t.start()

.

 

So what is now remaining is to get when the call CONNECTS and how to get
DTMFs during the call. 

 

From: delianSPAM [mailto:delianspam at gmail.com] 
Sent: Thursday, August 27, 2009 9:35 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call

 

Hello Michael!

 

Thank you for your reply! I know about sched_api and I use Python as a
scripting language. Also "sched_api" can be used inside python, using
session.execute. However the problem is that the "sched_api" timer starts
right after you initiate the SETUP of the second call leg. What I need is to
call something, after a call CONNECT instead. One workaround would be if I
can check what time it took to connect the call, but I do not know/see how
to do this. I do not see a CONNECT callback function either.

 

Best Regards, Delian Tashev

P.S. Dear enlightened people, thank you for providing help to the community
by replying to the list e-mails.

 

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Wednesday, August 26, 2009 11:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call

 

Do you actually need Python for the IVR, or is it that you're comfortable
using a scripting lang for an IVR? I like using XML for IVRs, but using
scripting langs does give you a bit more power & flexibility at the cost of
some resources.

For the record, you can do this in the dialplan using XML and sched_api
without touching a scripting language. Checkout the sched_api channel
variable on the wiki - it may give you the functionality you need.

-MC

On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM <delianspam at gmail.com> wrote:

Hello Everybody!

 

1.       Scenario.

I am writing an IVR in Python that gets a destination from the calling party
(party A) and then connects to the destination (party B).

When the call is CONNECTED, I want to:

-          Receive DTMFs

-          Have a timer that can call a certain function in my script. The
script will have to play a message to party A.

-          Have a timer that can call a certain function in my script. The
script will have to drop the call.

Please notice that I want to do the things after the two parties are
connected, and not after I send the Invite to party B.

 

2.       Problem.

I will be happy to receive help on:

-          Which methods should I look for to implement this.

 

3.       Details

Here is how I connect the call currently:

session.execute("bridge","sofia/internal/" + destination_number +
"@domain.com")

 

I have tried to create a timer callback function "my_method()" using:

ivr_timer =threading.Timer(30,my_method)

This never called the function "my_method()".

 

Maybe I am wrong in using threading.Timer and the "bridge" application?
Maybe I need to create a new thread and a new timer using the API of
freeswitch, plus to use the "session.setInputCallback", plus use a
conference rather than a bridge? Can you please provide any suggestions or
examples? 

 

Thank you!

Best Regards, Delian Tashev


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From juanbackson at gmail.com  Thu Aug 27 01:56:09 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 27 Aug 2009 16:56:09 +0800
Subject: [Freeswitch-users] need help with mod_xml_odbc
Message-ID: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>

Hi,

I tried to make install mod_xml_odbc and load it in freeswitch, but I am
getting:

2009-08-28 00:46:55.848087 [CRIT] switch_loadable_module.c:871 Error Loading
module /usr/local/freeswitch/mod/mod_xml_odbc.so
**/usr/local/freeswitch/mod/mod_xml_odbc.so: invalid ELF header**

I had to manually copy the .so file from the mod_xml_odbc dir to
/usr/local/freeswitch/mod

Does anyone know what is wrong?

Thanks,
JB
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From sylver_b at yahoo.com  Wed Aug 26 17:14:55 2009
From: sylver_b at yahoo.com (sylver_b)
Date: Wed, 26 Aug 2009 17:14:55 -0700 (PDT)
Subject: [Freeswitch-users] upstream Registrar / Mirror proxy
Message-ID: <252645.39333.qm@web39706.mail.mud.yahoo.com>

Hello,

We would like to use FreeSwitch to test interoperability with our sip registrar platform - as our registrar doesnt handle NAT Traversal we would like to use FS as a mirror proxy . 

If a call goes through, FS should transparently handle nat traversal functionalities while forwarding the REGISTER/INVITE requests to our registrar server.

Below some detailed requirements:
Upper Registration is the capability of a SBC to proxy Registrations towards an upstream Registrar. While it is necessary that all SIP requests traverse the SBC for NAT continuity, Registrations should be allowed to be relayed towards our upstream Registrar while the SBC retains a copy of the AOR and masquerade on behalf of the UA that send the registration request.

Please let us know the best way to configure FS to achieve this type of configuration.

Thank you


      
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From krice at freeswitch.org  Thu Aug 27 02:02:14 2009
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 27 Aug 2009 04:02:14 -0500
Subject: [Freeswitch-users] upstream Registrar / Mirror proxy
In-Reply-To: <252645.39333.qm@web39706.mail.mud.yahoo.com>
Message-ID: <C6BBB5C6.5DDCE%krice@freeswitch.org>

FreeSWITCH is a B2BUA and NOT a proxy and will not proxy any requests
(REGISTER or INVITE)

If you want a Proxy you should look toward OpenSIPS



From: sylver_b <sylver_b at yahoo.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Wed, 26 Aug 2009 17:14:55 -0700 (PDT)
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] upstream Registrar / Mirror proxy

Hello,

We would like to use FreeSwitch to test interoperability with our sip
registrar platform - as our registrar doesnt handle NAT Traversal we would
like to use FS as a mirror proxy .

If a call goes through, FS should transparently handle nat traversal
functionalities while forwarding the REGISTER/INVITE requests to our
registrar server.

Below some detailed requirements:
Upper Registration is the capability of a SBC to proxy Registrations towards
an upstream Registrar. While it is necessary that all SIP requests traverse
the SBC for NAT continuity, Registrations should be allowed to be relayed
towards our upstream Registrar while the SBC retains a copy of the AOR and
masquerade on behalf of the UA that send the registration request.

Please let us know the best way to configure FS to achieve this type of
configuration.

Thank you

 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From krice at freeswitch.org  Thu Aug 27 02:03:50 2009
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 27 Aug 2009 04:03:50 -0500
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
Message-ID: <C6BBB626.5DDCF%krice@freeswitch.org>

You can not just copy the .so file from the directory it is just a pointer
file to the real .so ... You should always use the make install target or
make {module_name}-install target to get it properly installed



From: Juan Backson <juanbackson at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Thu, 27 Aug 2009 16:56:09 +0800
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] need help with mod_xml_odbc

Hi,

I tried to make install mod_xml_odbc and load it in freeswitch, but I am
getting:

2009-08-28 00:46:55.848087 [CRIT] switch_loadable_module.c:871 Error Loading
module /usr/local/freeswitch/mod/mod_xml_odbc.so
**/usr/local/freeswitch/mod/mod_xml_odbc.so: invalid ELF header**

I had to manually copy the .so file from the mod_xml_odbc dir to
/usr/local/freeswitch/mod

Does anyone know what is wrong?

Thanks,
JB


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From steve.kurzeja at gmail.com  Thu Aug 27 02:21:47 2009
From: steve.kurzeja at gmail.com (Steve Kurzeja)
Date: Thu, 27 Aug 2009 21:21:47 +1200
Subject: [Freeswitch-users] upstream Registrar / Mirror proxy
In-Reply-To: <C6BBB5C6.5DDCE%krice@freeswitch.org>
References: <252645.39333.qm@web39706.mail.mud.yahoo.com>
	<C6BBB5C6.5DDCE%krice@freeswitch.org>
Message-ID: <5f7152000908270221g5b37ffbdi851288874e28f95d@mail.gmail.com>

A proxy won't do what the original poster is asking for. Upper registration
is a special type of function performed by SBCs and not defined by any RFC
yet but there are drafts out there. This question comes up quite often in
various mailing lists and has been asked this list before. The answer is no
freeswitch can't be configured to do this as it stands today.

The only opensource SBC I know of that attempts to do upper registration at
the moment is OpenSBC.  Otherwise there's vendor SBCs like ACME packet,
Nextone (now Genband -  they call it Mirror Proxy mode) etc which do it.

Regards,
Steve

On Thu, Aug 27, 2009 at 9:02 PM, Ken Rice <krice at freeswitch.org> wrote:

>  FreeSWITCH is a B2BUA and NOT a proxy and will not proxy any requests
> (REGISTER or INVITE)
>
> If you want a Proxy you should look toward OpenSIPS
>
>
> ------------------------------
> *From: *sylver_b <sylver_b at yahoo.com>
> *Reply-To: *<freeswitch-users at lists.freeswitch.org>
> *Date: *Wed, 26 Aug 2009 17:14:55 -0700 (PDT)
> *To: *<freeswitch-users at lists.freeswitch.org>
> *Subject: *[Freeswitch-users] upstream Registrar / Mirror proxy
>
> Hello,
>
> We would like to use FreeSwitch to test interoperability with our sip
> registrar platform - as our registrar doesnt handle NAT Traversal we would
> like to use FS as a mirror proxy .
>
> If a call goes through, FS should transparently handle nat traversal
> functionalities while forwarding the REGISTER/INVITE requests to our
> registrar server.
>
> Below some detailed requirements:
> Upper Registration is the capability of a SBC to proxy Registrations
> towards an upstream Registrar. While it is necessary that all SIP requests
> traverse the SBC for NAT continuity, Registrations should be allowed to be
> relayed towards our upstream Registrar while the SBC retains a copy of the
> AOR and masquerade on behalf of the UA that send the registration request.
>
> Please let us know the best way to configure FS to achieve this type of
> configuration.
>
> Thank you
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From se02005-mec at yahoo.com  Thu Aug 27 02:46:47 2009
From: se02005-mec at yahoo.com (Merle J. Ebbert)
Date: Thu, 27 Aug 2009 02:46:47 -0700
Subject: [Freeswitch-users] Can an Instant Message (IM) be sent to a cell
	phone with FS?
Message-ID: <4A965607.7000003@yahoo.com>

Can an Instant Message (IM) be sent to a cell phone with FS?
Any of these:
    AOL Instant Messenger (AIM)
    Yahoo (Y!)
    Blackberry Messenger
    Google Talk
    Windows Live Messenger

Thanks,
Merle



From se02005-mec at yahoo.com  Thu Aug 27 02:50:54 2009
From: se02005-mec at yahoo.com (Merle J. Ebbert)
Date: Thu, 27 Aug 2009 02:50:54 -0700
Subject: [Freeswitch-users] Can a chat message be sent to a cell phone with
	FS?
Message-ID: <4A9656FE.9030504@yahoo.com>

Can a chat message be sent to a cell phone with FS?

Thanks,
Merle


From delianspam at gmail.com  Thu Aug 27 03:23:49 2009
From: delianspam at gmail.com (delianSPAM)
Date: Thu, 27 Aug 2009 13:23:49 +0300
Subject: [Freeswitch-users] Get the Session State
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAI0pCHFyxT5Mn9pdicKxxSMBAAAAAA==@gmail.com>

Hello Everybody!

 

How do you get the call state in Python?

 

I have tried:

.

session.answer()

state_result=str(session.getVariable("state"))

console_log("debug",state_result)

.

 

But it returns: "None"

 

Thank you!

 

Best Regards, Delian Tashev

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From juanbackson at gmail.com  Thu Aug 27 03:37:41 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 27 Aug 2009 18:37:41 +0800
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <C6BBB626.5DDCF%krice@freeswitch.org>
References: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
	<C6BBB626.5DDCF%krice@freeswitch.org>
Message-ID: <27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>

Hi,

ok, the module can be loaded, but it now complains about odbc.  I can't find
anything missing in my odbc.ini.  Could someone please point me to the right
direction?

2009-08-28 02:22:35.670284 [ERR] switch_odbc.c:188 STATE: IM002 CODE 201
ERROR: [unixODBC]Missing server name, port, or database name in call to
CC_connect.

2009-08-28 02:22:35.670312 [CRIT] mod_xml_odbc.c:577 Cannot Open ODBC
Database!
2009-08-28 02:22:35.670319 [ERR] mod_xml_odbc.c:612 Unable to load xml_odbc
config file
2009-08-28 02:22:35.670326 [CRIT] switch_loadable_module.c:871 Error Loading
module /usr/local/freeswitch/mod/mod_xml_odbc.so
**Module load routine returned an error**


I already have xml_odbc.conf
 cat /etc/odbc.ini

; begin odbc.ini
[ODBC Data Sources]
test = PostgreSQL ODBC Driver

[test]
Driver = /usr/local/lib/libodbcpsql.so
Description = PostgreSQL Data Source
DSN = test
Servername = 192.168.1.133
Server = 192.168.1.133
Port = 5432
;Socket = 4096
Protocol = 6.4 # 7.2 or other values
UserName = root
Password = JdqB-S
Database = freeswitch
ReadOnly = no
ServerType = Postgres
FetchBufferSize = 99
ServerOptions =
ConnectOptions =
;Options = 3
Trace = 0
TraceFile = /var/log/PostgreSQL_test_trace.log
Debug = 0
DebugFile = /var/log/PostgreSQL_test_debug.log
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From odermann at googlemail.com  Thu Aug 27 04:06:08 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 13:06:08 +0200
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <5e414ed0908260910h55122d77o5013ef1795d0c75@mail.gmail.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
	<5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
	<274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>
	<5e414ed0908260910h55122d77o5013ef1795d0c75@mail.gmail.com>
Message-ID: <5e414ed0908270406m628342d5ya0e9f25514ed09d2@mail.gmail.com>

sorry, but i do not know i which category i have to set this problem.
could you help me with that?


From juanbackson at gmail.com  Thu Aug 27 04:40:37 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 27 Aug 2009 19:40:37 +0800
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>
References: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
	<C6BBB626.5DDCF%krice@freeswitch.org>
	<27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>
Message-ID: <27c25bc40908270440o5f1b7a6csf675716677baef48@mail.gmail.com>

Hi,

Finally, I got xml_odbc running, but it does not really work well for me.  I
am getting:
2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:325 Stopped rendering
template, called xml_odbc_render_template more than [32] times, probably
looping.
2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:408 Something went horribly
wrong while generating an XML template!

My config is:
<configuration name="xml_odbc.conf" description="XML ODBC Configuration">

  <settings>
        <param name="binding" value="directory"/>
    <param name="odbc-dsn" value="class5_odbc:root:JdqB-S"/>
    <param name="debug" value="true"/>
    <param name="keep-files-around" value="true"/>
  </settings>

  <templates>

    <template name="default">
      <document type="freeswitch/xml">
        <section name="directory">
     <xml-odbc-do name="query" on-empty-result-break-to="not-found"
value="SELECT enabled, sip_password FROM agent WHERE sip_user =  ${user}"/>
        <domain name="${domain}">
          <user id="${user}">
            <params>
              <xml-odbc-do name="query" value="SELECT sip_password FROM
agent WHERE sip_user = ${user}">
                <param name="password", value="${sip_password}"/>
              </xml-odbc-do>
            </params>
              </user>
        </domain>


        </section>
      </document>
    </template>

  </templates>

</configuration>


Since these two queries get data from the same table, I tried to merge them,
but could not get it to work.

Anyone has any idea?

Thanks,
JB
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From foxb at abv.bg  Thu Aug 27 04:58:53 2009
From: foxb at abv.bg (Hristo Benev)
Date: Thu, 27 Aug 2009 14:58:53 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <877409356.2721.1251374341513.JavaMail.apache@mail23.abv.bg>

 
Bojnour,

I'll send a trace ASAP.

What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)

I'm really interested to see the updates of the project.

Thank you for the good tutorial.

Hristo


 >-------- ?????????? ????? --------
 >??:  rod 
 >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >??: freeswitch-users at lists.freeswitch.org
 >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST

 >Hi Hristo,
 >
 >I'm the author of this setup and wiki page. I did a lot of modifications 
 >on this setup (alternative routing if failure essentially) but don't 
 >have too much time to update the wiki.
 >
 >May you please send me an ngrep trace when you call 1000:
 >
 >ngrep -d any -nn -i '1000' port 5060 -W byline 
 >
 >I will check what's happening.
 >Do you have an entry for 1000 in your mysql database ?
 >
 >regards,
 >rod
 >
 >Hristo Benev a ?crit :
 >>  
 >> It seems that the problem is on kamailio configuration.
 >> Will ask on their list.
 >>
 >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
 >>
 >> I have it setup as friend in asterisk, but still ???
 >>
 >> Any ideas?
 >>
 >> Thanks,
 >>
 >>
 >>  >-------- ?????????? ????? --------
 >>  >??:  Hristo Benev 
 >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >??: freeswitch-users at lists.freeswitch.org
 >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
 >>
 >>  > I think that the problem is here:
 >>  >-------------------------
 >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
 >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
 >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
 >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
 >>  >--------------------------
 >>  >
 >>  >Actually Regex FAIL
 >>  >
 >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
 >>  >Here is my  default.xml:
 >>  >----------------
 >>  >
 >>  >
 >>  >
 >>  >  
 >>  >
 >>  >     
 >>  >      
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >      
 >>  >    
 >>  >
 >>  >   
 >>  >
 >>  >  
 >>  >
 >>  >    
 >>  >      
 >>  >        
 >>  >        
 >>  >        
 >>  >        
 >>  >        
 >>  >        
 >>  >        
 >>  >        
 >>  >      
 >>  >    
 >>  >
 >>  >  
 >>  >
 >>  >
 >>  >
 >>  >--------------------------
 >>  >
 >>  >
 >>  >
 >>  > >-------- ?????????? ????? --------
 >>  > >??:  Brian West 
 >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  > >??: freeswitch-users at lists.freeswitch.org
 >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
 >>  >
 >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
 >>  >You have to process all 302's in the dialplan. 
 >>  >Set this on your sofia profile
 >>  >You can set these variables sip_redirect_profile,
 >>  >sip_redirect_context,
 >>  >sip_redirect_dialplan,
 >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
 >>  >sip_redirected_to,
 >>  >sip_redirect_contact_user_%d,
 >>  >sip_redirect_contact_host_%d,
 >>  >sip_redirect_contact_params_%d,
 >>  >sip_redirect_dialstring_%d,
 >>  >sip_redirect_dialstring,
 >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
 >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
 >>  > >
 >>  >
 >>  >_______________________________________________
 >>  >FreeSWITCH-users mailing list
 >>  >FreeSWITCH-users at lists.freeswitch.org
 >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >http://www.freeswitch.org
 >>  >
 >>
 >> _______________________________________________
 >> FreeSWITCH-users mailing list
 >> FreeSWITCH-users at lists.freeswitch.org
 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >> http://www.freeswitch.org
 >>   
 >
 >_______________________________________________
 >FreeSWITCH-users mailing list
 >FreeSWITCH-users at lists.freeswitch.org
 >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >http://www.freeswitch.org
 >


From rs at runsolutions.com  Thu Aug 27 05:10:36 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Thu, 27 Aug 2009 14:10:36 +0200
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
	<191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>
Message-ID: <225AA30C-8455-4D4A-A888-5AC7D8C64E7B@runsolutions.com>

very offtopic, but, if this is not to personally, but how do you get  
to spend literally 8-12 hours a day on a conference?

:-)

best

-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-
Email: rs at runsolutions.com
tel: 625 40 32 08

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 26, 2009, at 10:45 PM, Anthony Minessale wrote:

> There is no bug, it's all dependent on your network conditions.
> I spend literally 8-12 hours a day on a conference and there is no  
> delay.
>
> The important param is
> <param name="rtp-autoflush-during-bridge" value="true"/>
>
> in the sofia profile in question.
>
> if you have delay with that in place, then it's probably not FS
>
>
> On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd at suspiria.net> wrote:
> Hi,
>
>
> I am on a quite recent version (i assume):
>
>
> FreeSWITCH Version 1.0.trunk (14461)
>
>
> Should the bug be fixed in this revision ? What config settings  
> would a have to check to limit delay (even at the cost of reduction  
> in quality).
>
>
> Thanks
>
>
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] Im Auftrag von Anthony Minessale
> Gesendet: Mittwoch, 26. August 2009 20:57
> An: freeswitch-users at lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] delay buildup in conference
>
>
> which revision are you on?
> The defaults on the latest code and examples should be configured to  
> minimize delay.
> Some of the older revisions built up some delay issues from udp  
> buffering when timers were not synced.
>
> On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net>  
> wrote:
>
> When running conferences with users dialed in from a PSTN gateway  
> (SIP) and directly from remote SIP endpoints there is an ever longer  
> buildup in delay, reaching up to multiple seconds. Is there any way  
> to limit the delay ?
>
>
> I am not 100% sure whether the delays is caused by the SIP jitter  
> buffer of freeswitch or directly by the conference module.
>
>
> Any advice?
>
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From leon at scarlet-internet.nl  Thu Aug 27 05:57:20 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 27 Aug 2009 14:57:20 +0200
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <27c25bc40908270440o5f1b7a6csf675716677baef48@mail.gmail.com>
References: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
	<C6BBB626.5DDCF%krice@freeswitch.org>
	<27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>
	<27c25bc40908270440o5f1b7a6csf675716677baef48@mail.gmail.com>
Message-ID: <352125CC-DE10-4665-AD62-ABB3B4B2A1DE@scarlet-internet.nl>

Hi Juan,

Perhaps it loops because you didn't include the "not-found" template ?  
Actually, I see there's a bug in the example xml_odbc.conf.xml file  
where it's defined with an underscore instead of a dash, will change  
that tonight..

The not-found template needs to be specified as a template in the  
configuration. I think I'll define that template statically in the  
module itself later.
Because it's the 'fall-through' template when it can't find a  
template, you get a loop.

So, something like this should probably work for you:

<templates>

   <template name="default">
     <xml-odbc-do name="query" on-empty-result-break-to="not-found"  
value="SELECT sip_password FROM agent WHERE sip_user = '${user}';"/>
     <document type="freeswitch/xml">
       <section name="directory">
         <domain name="${domain}">
           <user id="${user}">
             <params>
               <param name="password", value="${sip_password}"/>
             </params>
           </user>
         </domain>
       </section>
     </document>
   </template>

   <template name="not-found">
     <document type="freeswitch/xml">
       <section name="result">
         <result status="not found"/>
       </section>
     </document>
   </template>

</templates>

(I didn't include the enabled field in your select statement, as you  
don't use it later, perhaps you need it in the where clause ?)

Also, note that this way the template will also be used at startup  
when FS tries to get a list of all ACL's - I believe for something  
else as well, have to check it - but those lookups probably don't give  
a ${user} so will render the "not-found" anyway..

One last thing, you didn't have ${user} enclosed in quotes in your  
query, so if no ${user} was given with the lookup to the module, then  
your query becomes invalid, which probably breaks things as well.

Let me know if it works..

regards,

Leon



On Aug 27, 2009, at 1:40 PM, Juan Backson wrote:

> Hi,
>
> Finally, I got xml_odbc running, but it does not really work well  
> for me.  I am getting:
> 2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:325 Stopped  
> rendering template, called xml_odbc_render_template more than [32]  
> times, probably looping.
> 2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:408 Something went  
> horribly wrong while generating an XML template!
>
> My config is:
> <configuration name="xml_odbc.conf" description="XML ODBC  
> Configuration">
>
>   <settings>
>         <param name="binding" value="directory"/>
>     <param name="odbc-dsn" value="class5_odbc:root:JdqB-S"/>
>     <param name="debug" value="true"/>
>     <param name="keep-files-around" value="true"/>
>   </settings>
>
>   <templates>
>
>     <template name="default">
>       <document type="freeswitch/xml">
>         <section name="directory">
>      <xml-odbc-do name="query" on-empty-result-break-to="not-found"  
> value="SELECT enabled, sip_password FROM agent WHERE sip_user =  $ 
> {user}"/>
>         <domain name="${domain}">
>           <user id="${user}">
>             <params>
>               <xml-odbc-do name="query" value="SELECT sip_password  
> FROM agent WHERE sip_user = ${user}">
>                 <param name="password", value="${sip_password}"/>
>               </xml-odbc-do>
>             </params>
>               </user>
>         </domain>
>
>
>         </section>
>       </document>
>     </template>
>
>   </templates>
>
> </configuration>
>
>
> Since these two queries get data from the same table, I tried to  
> merge them, but could not get it to work.
>
> Anyone has any idea?
>
> Thanks,
> JB
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From fraunhofer.lists.freeswitch-001 at traced.net  Thu Aug 27 06:15:21 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Thu, 27 Aug 2009 15:15:21 +0200
Subject: [Freeswitch-users] memory leak
Message-ID: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>

Hello *,

a memory leak showed up in our loadtests. It's (still) the same setup as in the
http://jira.freeswitch.org/browse/MODSOFIA-22 bugfix.

One thing I'd like to add is that "fsctl shutdown restart" was unable
to shutdown freeswitch.
The last line printed is "switch_core_memory.c:567 Stopping memory pool queue."

attached file is a the collected and graphed output of some "ps waux"
command. "sz" should be
in "physical pages" (2k?).

I rerun the test and this time it coredumped trying to malloc() space
for some playback.

Anything else you need (full backtraces?) to dig into it?

Cheers
  Beni.
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From anatoliy at kounitskiy.com  Thu Aug 27 06:19:27 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Thu, 27 Aug 2009 16:19:27 +0300
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
Message-ID: <1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>

Just for information - the idea :
A---calls---> B ---att_xfer---> C    , B hangs C up and goes back to A
( I`m sorry C is not answering :) )

dialplan:
-----------
 <extension name="local_number">
    <condition field="${toll_allow}" expression="local"/>
    <condition field="destination_number" expression="^(\d{3})$">
      <action application="set" data="dialed_extension=$1"/>
      <action application="export" data="dialed_extension=$1"/>
      <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
      <action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
      <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
      <action application="bind_meta_app" data="4 b s
execute_extension::attented_xfer XML features"/>
      <action application="set" data="transfer_ringback=$${hold_music}"/>
      <action application="set" data="call_timeout=10"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set"
data="continue_on_fail=USER_NOT_REGISTERED,SUBSCRIBER_ABSENT"/>
      <action application="info"/>
      <action application="set"
data="bringback=${user_data(${dialed_extension}@${domain_name} var
cringback )}"/>
      <action application="set" data="ringback=${${bringback}}"/>
      <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>
      <action application="info"/>
      ...
    </condition>
  </extension>

features.xml
---------------------
  <extension name="attented_xfer">
      <condition field="${toll_allow}" expression="local"/>
      <condition field="destination_number" expression="^attented_xfer$">
        <action application="set" data="continue_on_fail=true"/>
        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 30000 #"/>
        <action application="set"
data="bringback=${user_data(${attxfer_callthis}@${domain_name} var
cringback )}"/>
        <action application="set" data="ringback=${${bringback}}"/>
        <action application="info"/>
        <action application="set" data="origination_cancel_key=#"/>
        <action application="att_xfer"
data="user/${attxfer_callthis}@${domain_name}"/>
      </condition>
    </extension>

After testing the patch the results are:

1) B answers the call from A and executes the feature code *4
(attented_xfer), sends the desired number ( in this case the number of
C). A goes to MusicOnHold, B is waiting for C to pick up - B sends # -
sends SIP CANCEL to C and SIP BUY to A - so all calls are dropped.

2) B answers the call from A and executes the feature code *4
(attented_xfer), but without entering the number of C - so the read
aplication timeouts after 30 sec - after that att_xfer is executed
with empty string (nothing is entered) - B and A are bridged together.

On the console without entering the extension of C:
2009-08-27 16:00:58.929756 [NOTICE] switch_core_session.c:1576 Execute
set(origination_cancel_key=#)
EXECUTE sofia/internal/sip:102 at 10.17.4.107:5060 set(origination_cancel_key=#)
2009-08-27 16:00:58.929756 [DEBUG] mod_dptools.c:748
sofia/internal/sip:userB at 10.17.4.107:5060 SET
[origination_cancel_key]=[#]
2009-08-27 16:00:58.929756 [NOTICE] switch_core_session.c:1576 Execute
att_xfer(user/${attxfer_callthis}@${domain_name})
EXECUTE sofia/internal/sip:102 at 10.17.4.107:5060 att_xfer(user/@1.1.1.1)
2009-08-27 16:00:58.929756 [WARNING] mod_dptools.c:2373 Can't find
user [@1.1.1.1]
2009-08-27 16:00:58.929756 [ERR] switch_ivr_originate.c:1527 Cannot
create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2009-08-27 16:00:58.929756 [DEBUG] switch_ivr_originate.c:2167
Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT]
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_play_say.c:1402 done playing file
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:231
sofia/internal/userA at 1.1.1.1 receive message [BRIDGE]
2009-08-27 16:00:58.949842 [DEBUG] switch_core_session.c:630 Send
signal sofia/internal/userA at 1.1.1.1 [BREAK]
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:233 Send signal
sofia/internal/sip:userB at 10.17.4.107:5060 [BREAK]
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:231
sofia/internal/sip:userB at 10.17.4.107:5060 receive message [BRIDGE]
2009-08-27 16:00:58.949842 [DEBUG] switch_core_session.c:630 Send
signal sofia/internal/sip:userB at 10.17.4.107:5060 [BREAK]
2009-08-27 16:00:58.954052 [DEBUG] switch_ivr_bridge.c:233 Send signal
sofia/internal/userA at 1.1.1.1 [BREAK]
2009-08-27 16:01:22.881501 [DEBUG] sofia.c:3302 Channel
sofia/internal/userA at 1.1.1.1 entering state [calling][0]
2009-08-27 16:01:22.886122 [DEBUG] sofia.c:3302 Channel
sofia/internal/userA at 1.1.1.1 entering state [ready][200]

I'll try to summarize my idea - could it be possible when the # (in
the att_xfer) is executed to behave as if you're trying to make
attended transfer to non existing subscriber( SUBSCRIBER_ABSENT ), so
B can be bridged back with A.

Thank you in advance,
Anatoliy Kounitskiy

On Wed, Aug 26, 2009 at 11:11 PM, Michael Collins<msc at freeswitch.org> wrote:
>
>
> On Wed, Aug 26, 2009 at 12:59 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> i added a patch to attempt to do this so try adding
>
> Nice work!
>>
>> {origination_cancel_key=#} before the dial string
>> or
>> <action application="set" data="origination_cancel_key=#"/>
>> before you bridge.
>
> Anatoliy,
>
> Please try this and let us know if it works for you as expected. If so,
> please write up the exact procedure that you used and include dialplan
> examples. I'd like to get this on the wiki while it's fresh in my mind. If
> you feel comfortable editing the wiki on your own then just let me know so
> that we can coordinate efforts.
>
> Thanks,
> MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From leon at scarlet-internet.nl  Thu Aug 27 06:31:21 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 27 Aug 2009 15:31:21 +0200
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <352125CC-DE10-4665-AD62-ABB3B4B2A1DE@scarlet-internet.nl>
References: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
	<C6BBB626.5DDCF%krice@freeswitch.org>
	<27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>
	<27c25bc40908270440o5f1b7a6csf675716677baef48@mail.gmail.com>
	<352125CC-DE10-4665-AD62-ABB3B4B2A1DE@scarlet-internet.nl>
Message-ID: <F062600C-BDA4-4B70-B684-5837EF5E4066@scarlet-internet.nl>

Made a typo in the <param>, you have to leave out the comma..

regards,

Leon


On Aug 27, 2009, at 2:57 PM, Leon de Rooij wrote:

> Hi Juan,
>
> Perhaps it loops because you didn't include the "not-found" template ?
> Actually, I see there's a bug in the example xml_odbc.conf.xml file
> where it's defined with an underscore instead of a dash, will change
> that tonight..
>
> The not-found template needs to be specified as a template in the
> configuration. I think I'll define that template statically in the
> module itself later.
> Because it's the 'fall-through' template when it can't find a
> template, you get a loop.
>
> So, something like this should probably work for you:
>
> <templates>
>
>   <template name="default">
>     <xml-odbc-do name="query" on-empty-result-break-to="not-found"
> value="SELECT sip_password FROM agent WHERE sip_user = '${user}';"/>
>     <document type="freeswitch/xml">
>       <section name="directory">
>         <domain name="${domain}">
>           <user id="${user}">
>             <params>
>               <param name="password", value="${sip_password}"/>
>             </params>
>           </user>
>         </domain>
>       </section>
>     </document>
>   </template>
>
>   <template name="not-found">
>     <document type="freeswitch/xml">
>       <section name="result">
>         <result status="not found"/>
>       </section>
>     </document>
>   </template>
>
> </templates>
>
> (I didn't include the enabled field in your select statement, as you
> don't use it later, perhaps you need it in the where clause ?)
>
> Also, note that this way the template will also be used at startup
> when FS tries to get a list of all ACL's - I believe for something
> else as well, have to check it - but those lookups probably don't give
> a ${user} so will render the "not-found" anyway..
>
> One last thing, you didn't have ${user} enclosed in quotes in your
> query, so if no ${user} was given with the lookup to the module, then
> your query becomes invalid, which probably breaks things as well.
>
> Let me know if it works..
>
> regards,
>
> Leon
>
>
>
> On Aug 27, 2009, at 1:40 PM, Juan Backson wrote:
>
>> Hi,
>>
>> Finally, I got xml_odbc running, but it does not really work well
>> for me.  I am getting:
>> 2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:325 Stopped
>> rendering template, called xml_odbc_render_template more than [32]
>> times, probably looping.
>> 2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:408 Something went
>> horribly wrong while generating an XML template!
>>
>> My config is:
>> <configuration name="xml_odbc.conf" description="XML ODBC
>> Configuration">
>>
>>  <settings>
>>        <param name="binding" value="directory"/>
>>    <param name="odbc-dsn" value="class5_odbc:root:JdqB-S"/>
>>    <param name="debug" value="true"/>
>>    <param name="keep-files-around" value="true"/>
>>  </settings>
>>
>>  <templates>
>>
>>    <template name="default">
>>      <document type="freeswitch/xml">
>>        <section name="directory">
>>     <xml-odbc-do name="query" on-empty-result-break-to="not-found"
>> value="SELECT enabled, sip_password FROM agent WHERE sip_user =  $
>> {user}"/>
>>        <domain name="${domain}">
>>          <user id="${user}">
>>            <params>
>>              <xml-odbc-do name="query" value="SELECT sip_password
>> FROM agent WHERE sip_user = ${user}">
>>                <param name="password", value="${sip_password}"/>
>>              </xml-odbc-do>
>>            </params>
>>              </user>
>>        </domain>
>>
>>
>>        </section>
>>      </document>
>>    </template>
>>
>>  </templates>
>>
>> </configuration>
>>
>>
>> Since these two queries get data from the same table, I tried to
>> merge them, but could not get it to work.
>>
>> Anyone has any idea?
>>
>> Thanks,
>> JB
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From woodydickson at gmail.com  Thu Aug 27 06:49:35 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Thu, 27 Aug 2009 21:49:35 +0800
Subject: [Freeswitch-users] mod_limit and memcache
Message-ID: <a44688a0908270649y6cde3c26pb739c7d144e1b86b@mail.gmail.com>

Hello,

I read something that talks about using memcache for mod_limit before.   Is
it something that is available now?

If I have multiple instances of freeswitch that need to share the same limit
status, it there any existing solution?

If no existing solution is available, what is the best way to go about
modifying mod_limit to accomplish limiting for multiple freeswitch servers
together?

Thanks,
Woody
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From juanbackson at gmail.com  Thu Aug 27 06:53:42 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 27 Aug 2009 21:53:42 +0800
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <F062600C-BDA4-4B70-B684-5837EF5E4066@scarlet-internet.nl>
References: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
	<C6BBB626.5DDCF%krice@freeswitch.org>
	<27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>
	<27c25bc40908270440o5f1b7a6csf675716677baef48@mail.gmail.com>
	<352125CC-DE10-4665-AD62-ABB3B4B2A1DE@scarlet-internet.nl>
	<F062600C-BDA4-4B70-B684-5837EF5E4066@scarlet-internet.nl>
Message-ID: <27c25bc40908270653q47345f17s8d78e99ee5cc4b35@mail.gmail.com>

Hi Leon,

Thanks for your help.

I have changed it according to your comment but I am still getting the
looping error.

Would you please take a look see what else I did wrong?

Also, sip_user is an integer field, so I can't really use ''.  Is there
anyway to get around that?


<configuration name="xml_odbc.conf" description="XML ODBC Configuration">

  <settings>
        <param name="binding" value="directory"/>
    <param name="odbc-dsn" value="class5_odbc:root:abcd"/>
    <param name="debug" value="true"/>
    <param name="keep-files-around" value="true"/>
  </settings>

  <templates>

    <template name="default">
      <document type="freeswitch/xml">
        <section name="directory">
     <xml-odbc-do name="query" on-empty-result-break-to="not-found"
value="SELECT enabled, sip_password FROM agent WHERE sip_user =  ${user} and
enabled='t'"/>
        <domain name="${domain}">
          <user id="${user}">
            <params>
              <xml-odbc-do name="query" value="SELECT sip_password FROM
agent WHERE sip_user = ${user}">
                <param name="password" value="${sip_password}"/>
              </xml-odbc-do>
            </params>
              </user>
        </domain>


        </section>
      </document>
    </template>


    <template>

        <document type="freeswitch/xml">
            <section name="result">
                <result status="not found"/>
            </section>
        </document>
    </template>
  </templates>

</configuration>
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From fraunhofer.lists.freeswitch-001 at traced.net  Thu Aug 27 07:08:23 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Thu, 27 Aug 2009 16:08:23 +0200
Subject: [Freeswitch-users] fscore mutex locking question
Message-ID: <a37510170908270708g1e3236ao1f16a6835a87e8d7@mail.gmail.com>

Hello *,

while looking at the code i came across a region of code which is
unclear to me regarding locking issues.
One example is switch_ivr_broadcast in switch_ivr_async.c. This should
be the function called by
uuid_broadcast() and others.

in line 2341 it tries to queue an event to the bleg if it has to...
-----
.   if ((flags & SMF_ECHO_BLEG) && (other_uuid =
switch_channel_get_variable(channel, SWITCH_SIGNAL_BOND_VARIABLE))
.   .   && (other_session = switch_core_session_locate(other_uuid))) {

---
switch_core_session_locate() does a "readonly trylock" on the channel
mutex returning NULL if it's unable to
aquire the lock, which brings up the following question:
If some other thread is currently holding a writelock on the channel,
the broadcast is not queued and not retried at a later time at all?

I guess it's pretty easy to cause some unexpected behaviour using some
endless loop calling "uuid_setvar" or some other race condition where
the channel-mutex is write-locked while calling uuid_broadcast (eg.
uuid_media?).

Could this lead to a problem in "real live" scenarios, or are there
other countermeasures despite "chances are one in a million that you
hit that small time frame"?

thx in advance
  Beni.


From anthony.minessale at gmail.com  Thu Aug 27 07:32:27 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 09:32:27 -0500
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <225AA30C-8455-4D4A-A888-5AC7D8C64E7B@runsolutions.com>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
	<191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>
	<225AA30C-8455-4D4A-A888-5AC7D8C64E7B@runsolutions.com>
Message-ID: <191c3a030908270732q114fc3d4h5f2c1f407182e43e@mail.gmail.com>

Our whole development team works from 6 different states so we use a 24x7
conference as part of our virtual office.
There is a public conference for freeswitch users as well.

http://conference.freeswitch.org
sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>



On Thu, Aug 27, 2009 at 7:10 AM, Raimund Sacherer <rs at runsolutions.com>wrote:

> very offtopic, but, if this is not to personally, but how do you get to
> spend literally 8-12 hours a day on a conference?
> :-)
>
> best
>
> --
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
> Email: rs at runsolutions.com
> tel: 625 40 32 08
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
> On Aug 26, 2009, at 10:45 PM, Anthony Minessale wrote:
>
> There is no bug, it's all dependent on your network conditions.
> I spend literally 8-12 hours a day on a conference and there is no delay.
>
> The important param is
> <param name="rtp-autoflush-during-bridge" value="true"/>
>
> in the sofia profile in question.
>
> if you have delay with that in place, then it's probably not FS
>
>
> On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd at suspiria.net> wrote:
>
>>  Hi,
>>
>>
>> I am on a quite recent version (i assume):
>>
>>
>> FreeSWITCH Version 1.0.trunk (14461)
>>
>>
>> Should the bug be fixed in this revision ? What config settings would a
>> have to check to limit delay (even at the cost of reduction in quality).
>>
>>
>> Thanks
>>
>>
>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Anthony
>> Minessale
>> *Gesendet:* Mittwoch, 26. August 2009 20:57
>> *An:* freeswitch-users at lists.freeswitch.org
>> *Betreff:* Re: [Freeswitch-users] delay buildup in conference
>>
>>
>> which revision are you on?
>> The defaults on the latest code and examples should be configured to
>> minimize delay.
>> Some of the older revisions built up some delay issues from udp buffering
>> when timers were not synced.
>>
>>  On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net> wrote:
>>
>> When running conferences with users dialed in from a PSTN gateway (SIP)
>> and directly from remote SIP endpoints there is an ever longer buildup in
>> delay, reaching up to multiple seconds. Is there any way to limit the delay
>> ?
>>
>>
>> I am not 100% sure whether the delays is caused by the SIP jitter buffer
>> of freeswitch or directly by the conference module.
>>
>>
>> Any advice?
>>
>>
>> Thanks
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Aug 27 07:38:58 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 09:38:58 -0500
Subject: [Freeswitch-users] memory leak
In-Reply-To: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
Message-ID: <191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>

600k is not a leak? FS can use as much as a gig of ram or more depending on
what you are doing.
you may want to install a fresh copy of FS, removing all your old files etc
and make sure they build clean.
We also have not had much luck running on ubuntu which is more of a desktop
centric OS.
I recommend you try your application on 64 bit CentOS which is the platform
all of our paid customers use.





On Thu, Aug 27, 2009 at 8:15 AM, Benedikt Fraunhofer <
fraunhofer.lists.freeswitch-001 at traced.net> wrote:

> Hello *,
>
> a memory leak showed up in our loadtests. It's (still) the same setup as in
> the
> http://jira.freeswitch.org/browse/MODSOFIA-22 bugfix.
>
> One thing I'd like to add is that "fsctl shutdown restart" was unable
> to shutdown freeswitch.
> The last line printed is "switch_core_memory.c:567 Stopping memory pool
> queue."
>
> attached file is a the collected and graphed output of some "ps waux"
> command. "sz" should be
> in "physical pages" (2k?).
>
> I rerun the test and this time it coredumped trying to malloc() space
> for some playback.
>
> Anything else you need (full backtraces?) to dig into it?
>
> Cheers
>  Beni.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From rupa at rupa.com  Thu Aug 27 07:42:38 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 27 Aug 2009 09:42:38 -0500
Subject: [Freeswitch-users] mod_limit and memcache
In-Reply-To: <a44688a0908270649y6cde3c26pb739c7d144e1b86b@mail.gmail.com>
References: <a44688a0908270649y6cde3c26pb739c7d144e1b86b@mail.gmail.com>
Message-ID: <b39273530908270742i1e018ed2x953203e4eee7bb2f@mail.gmail.com>

limit and memcache haven't been introduced to each other yet -- it is
on my (semi-long) list of things to do.

If you want it you can:

1) do it yourself and submit the patches
2) open a jira and hope someone does it
3) open a jira + bounty and someone will probably do it

It will get done eventually, just hasn't been a itch for ME to scratch yet.


To do it:

1) I need to make it possible to call inc/dec methods of mod_memcache
to support an expiration time.

2) mod_limit.c - use the hash limit as a guide

Initial pitfalls:

hash limits concurrent access/modification of the hash and by
implication limit_hash_item_t (hash data) by using a mutex.  We can't
mutex across FS instances.

So perhaps split up limit_has_item_t and spread it across multiple
keys.  So instead of one key marked as realm_id, we could have
realm_id_total_usage realm_id_rate_usage and realm_id_last_check.

This does mean that rate_usage and total_usage can inc/dec independent
of each other, but I think the logic will still be "ok" *IF* we
remember to decrement earlier incremented items in the event a later
item is failed.   (so, say we increment rate but fail on total we need
to remember to decrement rate so that we have no net effect on the
counters)

Alternatively, we could use CAS support and pull the limit_hash_item_t
item from memcache, twiddle it and then try to put it back only if the
check info is the same (no one else has changed the entry).  If the
entry has changed, pull the new version, do the limit logic, and try
again.  Loop that a few times until you succeed or give up.  Problem
is that CAS needs to be explicitly turned on in memcache (some distros
compile with it off), is relatively new in memcache (hint: may have
issues) and has some performance/memory downsides though by how much
I'm not sure.

Thoughts?

On Thu, Aug 27, 2009 at 8:49 AM, Woody Dickson<woodydickson at gmail.com> wrote:
> Hello,
>
> I read something that talks about using memcache for mod_limit before.?? Is
> it something that is available now?
>
> If I have multiple instances of freeswitch that need to share the same limit
> status, it there any existing solution?
>
> If no existing solution is available, what is the best way to go about
> modifying mod_limit to accomplish limiting for multiple freeswitch servers
> together?
>
> Thanks,
> Woody
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa


From sylver_b at yahoo.com  Thu Aug 27 07:56:23 2009
From: sylver_b at yahoo.com (sylver_b)
Date: Thu, 27 Aug 2009 07:56:23 -0700 (PDT)
Subject: [Freeswitch-users] Re :  upstream Registrar / Mirror proxy
In-Reply-To: <5f7152000908270221g5b37ffbdi851288874e28f95d@mail.gmail.com>
References: <252645.39333.qm@web39706.mail.mud.yahoo.com>
	<C6BBB5C6.5DDCE%krice@freeswitch.org>
	<5f7152000908270221g5b37ffbdi851288874e28f95d@mail.gmail.com>
Message-ID: <608767.97175.qm@web39705.mail.mud.yahoo.com>

Thanks for the feedback

Steve: we are using Genband/Nextone MSX , but we're looking for a cheaper alternative to avoid using our vports for this purpose.

We'll check OpenSBC .. thanks



________________________________
De : Steve Kurzeja <steve.kurzeja at gmail.com>
? : freeswitch-users at lists.freeswitch.org
Envoy? le : Jeudi, 27 Ao?t 2009, 10h21mn 47s
Objet : Re: [Freeswitch-users] upstream Registrar / Mirror proxy

A proxy won't do what the original poster is asking for. Upper registration is a special type of function performed by SBCs and not defined by any RFC yet but there are drafts out there. This question comes up quite often in various mailing lists and has been asked this list before. The answer is no freeswitch can't be configured to do this as it stands today.

The only opensource SBC I know of that attempts to do upper registration at the moment is OpenSBC.  Otherwise there's vendor SBCs like ACME packet, Nextone (now Genband -  they call it Mirror Proxy mode) etc which do it.

Regards,
Steve


On Thu, Aug 27, 2009 at 9:02 PM, Ken Rice <krice at freeswitch.org> wrote:

FreeSWITCH is a B2BUA and NOT a proxy and will not proxy any requests (REGISTER or INVITE) 
>
>>If you want a Proxy you should look toward OpenSIPS 
>
>
>________________________________
From: sylver_b <sylver_b at yahoo.com>
>Reply-To: <freeswitch-users at lists.freeswitch.org>
>Date: Wed, 26 Aug 2009 17:14:55 -0700 (PDT)
>To: <freeswitch-users at lists.freeswitch.org>
>Subject: [Freeswitch-users] upstream Registrar / Mirror proxy
>
>
>Hello,
>
>>We would like to use FreeSwitch to test interoperability with our sip registrar platform - as our registrar doesnt handle NAT Traversal we would like to use FS as a mirror proxy . 
>
>>If a call goes through, FS should transparently handle nat traversal functionalities while forwarding the REGISTER/INVITE requests to our registrar server.
>
>>Below some detailed requirements:
>>Upper Registration is the capability of a SBC to proxy Registrations towards an upstream Registrar. While it is necessary that all SIP requests traverse the SBC for NAT continuity, Registrations should be allowed to be relayed towards our upstream Registrar while the SBC retains a copy of the AOR and masquerade on behalf of the UA that send the registration request.
>
>>Please let us know the best way to configure FS to achieve this type of configuration.
>
>>Thank you
>
>> 
>________________________________
_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
> 
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>



      
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From leon at scarlet-internet.nl  Thu Aug 27 07:57:08 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 27 Aug 2009 16:57:08 +0200
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <27c25bc40908270653q47345f17s8d78e99ee5cc4b35@mail.gmail.com>
References: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
	<C6BBB626.5DDCF%krice@freeswitch.org>
	<27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>
	<27c25bc40908270440o5f1b7a6csf675716677baef48@mail.gmail.com>
	<352125CC-DE10-4665-AD62-ABB3B4B2A1DE@scarlet-internet.nl>
	<F062600C-BDA4-4B70-B684-5837EF5E4066@scarlet-internet.nl>
	<27c25bc40908270653q47345f17s8d78e99ee5cc4b35@mail.gmail.com>
Message-ID: <E2C7D974-6301-4AC5-BC57-E854C35400F1@scarlet-internet.nl>

Hi Juan,

With debug=true you should be able to see what template it's trying to  
render in a loop, can you tell which one that is ? (I'm guessing it  
says 32 times it wants to render "not-found")

In the xml you pasted in your mail, you didn't specify the name of the  
"not-found" template, just <template>, that could be the problem..

To get around the problem of generating a wrong query when the ${user}  
is not given, you could simply encapsulate everything with a check- 
event-header, like this:

<template name="default">
   <xml-odbc-do name="check-event-header" if-name="user">
     <xml-odbc-do name="query" on-empty-result-break-to="not-found"  
value="...etc">
       ..rest of your template goes here..
     </xml-odbc-do>
   </xml-odbc-do>
</template>

But that would mean that an "empty" xml is returned whenever no $ 
{user} is provided. I didn't test how fs will react to that when it's  
looking up all the users for generating the acl list..

A way around that would be to only render your template when ${user}  
is set and otherwise render the not-found template, like this:

<template name="default">
   <xml-odbc-do name="check-event-header" if-name="user">
     <xml-odbc-do name="break-to" value="directory-user"/>
   </xml-odbc-do>
   <xml-odbc-do name="break-to" value="not-found"/>
</template>

<template name="directory-user">
   <xml-odbc-do name="query" on-empty-result-break-to="not-found"  
value="...etc">
     ..rest of your template goes here..
   </xml-odbc-do>
</template>

<template name="not-found">
etc...
</template>

---

Which comes close to how it's defined in the xml_odbc.conf.xml  
example, which solves it the other way around:

...
     <template name="default">
       <xml-odbc-do name="break-to" value="${section}"/>
     </template>

     <template name="directory"> -->
       <xml-odbc-do name="check-event-header" if-name="purpose">
         <!-- catches purpose gateways and network-list (any more?) -->
         <xml-odbc-do name="break-to" value="directory-${purpose}"/>
       </xml-odbc-do>
       <xml-odbc-do name="break-to" value="directory-user"/>
     </template>
...

- First it checks what section is requested, which is nice but not  
really necessary in your case, since you only bind to directory.
- Then it checks whether a purpose header is set (which was only the  
case for looking up acl's and gateways and NOT for regular user  
lookups last time I checked out the svn)
- If a purpose header IS set, then it tries to render directory-$ 
{purpose}, which isn't specified, so it falls back to "not-found"  
which is what you want in this case
- And as a last resort - this is where regular user lookups end up, it  
renders "directory-user" where you can put the template as specified  
above..

Hope that helps ?

regards,

Leon


On Aug 27, 2009, at 3:53 PM, Juan Backson wrote:

> Hi Leon,
>
> Thanks for your help.
>
> I have changed it according to your comment but I am still getting  
> the looping error.
>
> Would you please take a look see what else I did wrong?
>
> Also, sip_user is an integer field, so I can't really use ''.  Is  
> there anyway to get around that?
>
>
> <configuration name="xml_odbc.conf" description="XML ODBC  
> Configuration">
>
>   <settings>
>         <param name="binding" value="directory"/>
>     <param name="odbc-dsn" value="class5_odbc:root:abcd"/>
>     <param name="debug" value="true"/>
>     <param name="keep-files-around" value="true"/>
>   </settings>
>
>   <templates>
>
>     <template name="default">
>       <document type="freeswitch/xml">
>         <section name="directory">
>      <xml-odbc-do name="query" on-empty-result-break-to="not-found"  
> value="SELECT enabled, sip_password FROM agent WHERE sip_user =  $ 
> {user} and enabled='t'"/>
>         <domain name="${domain}">
>           <user id="${user}">
>             <params>
>               <xml-odbc-do name="query" value="SELECT sip_password  
> FROM agent WHERE sip_user = ${user}">
>                 <param name="password" value="${sip_password}"/>
>               </xml-odbc-do>
>             </params>
>               </user>
>         </domain>
>
>
>         </section>
>       </document>
>     </template>
>
>
>     <template>
>
>         <document type="freeswitch/xml">
>             <section name="result">
>                 <result status="not found"/>
>             </section>
>         </document>
>     </template>
>   </templates>
>
> </configuration>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From gregt at cgicommunications.com  Thu Aug 27 08:06:26 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Thu, 27 Aug 2009 11:06:26 -0400
Subject: [Freeswitch-users] Odd sonus warning
Message-ID: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>

Hi, I have some DIDs from Bandwidth.com and when they call in I see  
this in the console:

2009-08-27 11:01:47 [WARNING] sofia_glue.c:2701  
sofia_glue_negotiate_sdp() Hello,
I see you have a Sonus!
FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
Sadly, my creator had to spend several hours figuring this out so I  
thought you'd like to know that!
Don't worry, DTMF will work but you may want to ask them to fix it......

Anything I should worry about?
--
Greg


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From damin at nacs.net  Thu Aug 27 08:14:55 2009
From: damin at nacs.net (Gregory Boehnlein)
Date: Thu, 27 Aug 2009 11:14:55 -0400
Subject: [Freeswitch-users] Odd sonus warning
In-Reply-To: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>
References: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>
Message-ID: <00e801ca2729$254f0b00$6fed2100$@net>

> Anything I should worry about?
 
Nothing to worry about. Freeswitch will deal with the issue, but there are a
few caveats.

I believe this is the most up-to date resource:

http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus



From brian at freeswitch.org  Thu Aug 27 08:15:36 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 27 Aug 2009 10:15:36 -0500
Subject: [Freeswitch-users] Odd sonus warning
In-Reply-To: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>
References: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>
Message-ID: <03399852-40FF-45E8-A69A-FE9DBEA963EB@freeswitch.org>

Does your DTMF work?

/b

On Aug 27, 2009, at 10:06 AM, Greg Thoen wrote:

> Anything I should worry about?



From foxb at abv.bg  Thu Aug 27 08:23:00 2009
From: foxb at abv.bg (Hristo Benev)
Date: Thu, 27 Aug 2009 18:23:00 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <999373713.8871.1251386588649.JavaMail.apache@mail22.abv.bg>

 I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).

If you need additional info I'll provide it.

Here is trace:

ngrep -d any -nn -i '1000' port 5062 -W byline
interface: any
filter: (ip or ip6) and ( port 5062 )
match: 1000
#
U 10.10.10.10:5090 -> 127.0.0.1:5062
INVITE sip:1000 at 127.0.0.1:5062 SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
Max-Forwards: 69.
From: "Extension 1001" <sip:1001 at 10.10.10.10>;tag=978g69jZaFpBD.
To: <sip:1000 at 127.0.0.1:5062>.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 INVITE.
Contact: <sip:mod_sofia at 10.10.10.10:5090>.
User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 429.
X-ROUTE: LOOKUP.
Remote-Party-ID: "Extension 1001" <sip:1001 at 10.10.10.10>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
s=FreeSWITCH.
c=IN IP4 10.10.10.10.
t=0 0.
m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:115 G7221/32000.
a=fmtp:115 bitrate=48000.
a=rtpmap:107 G7221/16000.
a=fmtp:107 bitrate=32000.
a=rtpmap:9 G722/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
U 127.0.0.1:5062 -> 10.10.10.10:5090
SIP/2.0 302 PEER_01.
Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
From: "Extension 1001" <sip:1001 at 10.10.10.10>;tag=978g69jZaFpBD.
To: <sip:1000 at 127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 INVITE.
Contact: sip:France at PEER_01.
Server: Kamailio (1.5.2-notls (i386/linux)).
Content-Length: 0.
.

#
U 10.10.10.10:5090 -> 127.0.0.1:5062
ACK sip:1000 at 127.0.0.1:5062 SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
Max-Forwards: 69.
From: "Extension 1001" <sip:1001 at 10.10.10.10>;tag=978g69jZaFpBD.
To: <sip:1000 at 127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 ACK.
Content-Length: 0.
.





 >-------- ?????????? ????? --------
 >??:  Hristo Benev 
 >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >??: freeswitch-users at lists.freeswitch.org
 >????????? ??: ?????????, 2009, ?????? 27 14:58:53 EEST

 > 
 >Bojnour,
 >
 >I'll send a trace ASAP.
 >
 >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
 >
 >I'm really interested to see the updates of the project.
 >
 >Thank you for the good tutorial.
 >
 >Hristo
 >
 >
 > >-------- ?????????? ????? --------
 > >??:  rod 
 > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 > >??: freeswitch-users at lists.freeswitch.org
 > >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST
 >
 > >Hi Hristo,
 > >
 > >I'm the author of this setup and wiki page. I did a lot of modifications 
 > >on this setup (alternative routing if failure essentially) but don't 
 > >have too much time to update the wiki.
 > >
 > >May you please send me an ngrep trace when you call 1000:
 > >
 > >ngrep -d any -nn -i '1000' port 5060 -W byline 
 > >
 > >I will check what's happening.
 > >Do you have an entry for 1000 in your mysql database ?
 > >
 > >regards,
 > >rod
 > >
 > >Hristo Benev a ?crit :
 > >>  
 > >> It seems that the problem is on kamailio configuration.
 > >> Will ask on their list.
 > >>
 > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
 > >>
 > >> I have it setup as friend in asterisk, but still ???
 > >>
 > >> Any ideas?
 > >>
 > >> Thanks,
 > >>
 > >>
 > >>  >-------- ?????????? ????? --------
 > >>  >??:  Hristo Benev 
 > >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 > >>  >??: freeswitch-users at lists.freeswitch.org
 > >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
 > >>
 > >>  > I think that the problem is here:
 > >>  >-------------------------
 > >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
 > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
 > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
 > >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
 > >>  >--------------------------
 > >>  >
 > >>  >Actually Regex FAIL
 > >>  >
 > >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
 > >>  >Here is my  default.xml:
 > >>  >----------------
 > >>  >
 > >>  >
 > >>  >
 > >>  >  
 > >>  >
 > >>  >     
 > >>  >      
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >      
 > >>  >    
 > >>  >
 > >>  >   
 > >>  >
 > >>  >  
 > >>  >
 > >>  >    
 > >>  >      
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >      
 > >>  >    
 > >>  >
 > >>  >  
 > >>  >
 > >>  >
 > >>  >
 > >>  >--------------------------
 > >>  >
 > >>  >
 > >>  >
 > >>  > >-------- ?????????? ????? --------
 > >>  > >??:  Brian West 
 > >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 > >>  > >??: freeswitch-users at lists.freeswitch.org
 > >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
 > >>  >
 > >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
 > >>  >You have to process all 302's in the dialplan. 
 > >>  >Set this on your sofia profile
 > >>  >You can set these variables sip_redirect_profile,
 > >>  >sip_redirect_context,
 > >>  >sip_redirect_dialplan,
 > >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
 > >>  >sip_redirected_to,
 > >>  >sip_redirect_contact_user_%d,
 > >>  >sip_redirect_contact_host_%d,
 > >>  >sip_redirect_contact_params_%d,
 > >>  >sip_redirect_dialstring_%d,
 > >>  >sip_redirect_dialstring,
 > >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
 > >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
 > >>  > >
 > >>  >
 > >>  >_______________________________________________
 > >>  >FreeSWITCH-users mailing list
 > >>  >FreeSWITCH-users at lists.freeswitch.org
 > >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > >>  >http://www.freeswitch.org
 > >>  >
 > >>
 > >> _______________________________________________
 > >> FreeSWITCH-users mailing list
 > >> FreeSWITCH-users at lists.freeswitch.org
 > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > >> http://www.freeswitch.org
 > >>   
 > >
 > >_______________________________________________
 > >FreeSWITCH-users mailing list
 > >FreeSWITCH-users at lists.freeswitch.org
 > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > >http://www.freeswitch.org
 > >
 >
 >_______________________________________________
 >FreeSWITCH-users mailing list
 >FreeSWITCH-users at lists.freeswitch.org
 >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >http://www.freeswitch.org
 >


From dave at 3c.co.uk  Thu Aug 27 08:38:13 2009
From: dave at 3c.co.uk (David Knell)
Date: Thu, 27 Aug 2009 18:38:13 +0300
Subject: [Freeswitch-users] Odd sonus warning
In-Reply-To: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>
References: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>
Message-ID: <1251387493.27240.105.camel@dk-d820>

On Thu, 2009-08-27 at 11:06 -0400, Greg Thoen wrote:
> Hi, I have some DIDs from Bandwidth.com and when they call in I see
> this in the console:
> 
> 
> 2009-08-27 11:01:47 [WARNING] sofia_glue.c:2701
> sofia_glue_negotiate_sdp() Hello,
> I see you have a Sonus!
> FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
> Sadly, my creator had to spend several hours figuring this out so I
> thought you'd like to know that!
> Don't worry, DTMF will work but you may want to ask them to fix
> it......
> 
> 
> Anything I should worry about?

Not unless you're a Sonus shareholder - see
http://messages.finance.yahoo.com/Stocks_(A_to_Z)/Stocks_S/threadview?m=te&bn=16942&tid=827769&mid=-1&tof=15&rt=1&frt=2
for an amusing read ;-)

--Dave


-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From odermann at googlemail.com  Thu Aug 27 08:39:16 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 17:39:16 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
Message-ID: <5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>

> Is the snom firmware up to the latest?
yes, the firmware is the latest.

> I believe session timers should work properly with snom?
i don't know, but i think so. we tested with session timers "off" and
with setting the session timer to 0. both does not change anything.


we played with using a stun-server and had a small success. if we are
connected with a stund-server, the hangup will not come after 120
seconds. we can also call the 5900 and listen to the moh. this works
fine.
here we have a totally different problem. if another voip phone is
connected to the server using stun (for example the 1000), both sides
can not talk to each other. one won't hear, if the other side is
talking.

very strange (for us), but we have no idea, what we can do about that.


From brian at freeswitch.org  Thu Aug 27 08:42:28 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 27 Aug 2009 10:42:28 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
Message-ID: <32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>

Are the phones behind the same nat as FS?

/b

On Aug 27, 2009, at 10:39 AM, Dennis wrote:

> we played with using a stun-server and had a small success. if we are
> connected with a stund-server, the hangup will not come after 120
> seconds. we can also call the 5900 and listen to the moh. this works
> fine.
> here we have a totally different problem. if another voip phone is
> connected to the server using stun (for example the 1000), both sides
> can not talk to each other. one won't hear, if the other side is
> talking.



From odermann at googlemail.com  Thu Aug 27 09:01:06 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 18:01:06 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
Message-ID: <5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>

> Are the phones behind the same nat as FS?

no, both phones are behind the same nat, fs is behind another nat (and
the internet is inbetween).

both phones are in our office and we are sitting behind a nat. we
connect through our nat over the internet into the other nat, where fs
is behind.


From csa at nowthor.com  Thu Aug 27 09:04:56 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 27 Aug 2009 12:04:56 -0400
Subject: [Freeswitch-users] Who is "callie"?
Message-ID: <4A96AEA8.3030308@nowthor.com>

Hi!

I searched the wiki but couldn't find the answer. Is "callie" a real 
woman one may ask to record additional sounds?

Thanks!

Carlos


From msc at freeswitch.org  Thu Aug 27 09:16:30 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 09:16:30 -0700
Subject: [Freeswitch-users] How to make a call back
In-Reply-To: <25166083.post@talk.nabble.com>
References: <25166083.post@talk.nabble.com>
Message-ID: <87f2f3b90908270916l7cc7f3cap36bcf8725fa5fa5f@mail.gmail.com>

On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com> wrote:

>
> When I give the following from the command line it calls to 1010 extension
> and once answered, it calls to 1000 and bridge the connection.
>            originate user/1010 &bridge(user/1000)
> But I want to do this in perl. So I have given as follows
>            $session->originate($session,"user/1010 &bridge user/1000");
> But it is not working. It says "user/1010 &bridge user/1000 is invalid
> user".
> How to do this in perl. pls help.


Are you calling this perl script from the CLI? If so you won't have the
$session object because a channel does not exist for a simple API call.
-MC
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From msc at freeswitch.org  Thu Aug 27 09:18:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 09:18:10 -0700
Subject: [Freeswitch-users] Who is "callie"?
In-Reply-To: <4A96AEA8.3030308@nowthor.com>
References: <4A96AEA8.3030308@nowthor.com>
Message-ID: <87f2f3b90908270918u66439bd2wd046a4dc485caeb8@mail.gmail.com>

On Thu, Aug 27, 2009 at 9:04 AM, Carlos S. Antunes <csa at nowthor.com> wrote:

> Hi!
>
> I searched the wiki but couldn't find the answer. Is "callie" a real
> woman one may ask to record additional sounds?
>

"Callie" is one of the voices from GM Voices. She is definitely available
for custom work. Visit www.gmvoices.com for more info. Tell them that the
FreeSWITCH project sent you. :)
-MC


>
> Thanks!
>
> Carlos
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fraunhofer.lists.freeswitch-001 at traced.net  Thu Aug 27 09:22:15 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Thu, 27 Aug 2009 18:22:15 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
Message-ID: <a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>

Hello Anthony,

2009/8/27 Anthony Minessale <anthony.minessale at gmail.com>:

> 600k is not a leak? FS can use as much as a gig of ram or more depending on
> what you are doing.

it's 600*1024 * 4* 1024 /1024/1024 => ~ 2.4 Gig
that's what i tried to express with the "physical pages" in my first post.

> you may want to install a fresh copy of FS, removing all your old files etc
> and make sure they build clean.

i "make current" to get the SOFIA-22 bugfix.

> We also have not had much luck running on ubuntu which is more of a desktop
> centric OS.

really? even the "server" edition? (not the netbook remix :)
Is 64bit so grave different if i don't need to address more than 4 Gig
of ram per process?

> I recommend you try your application on 64 bit CentOS which is the platform
> all of our paid customers use.

hmm. our operations-people wont be happy with that.

But as i carefully followed the other mailing-list-threads it came to
my mind that i could be the culprit: somehow the original example.xml
came back into the directory tree and included a lot of things, most
notably the hash-inserts for last-dailed-numbers and stuff.
I stripped it down to what it was before and gave it another try.

thx again and (hopefully) sorry

  Beni.


From msc at freeswitch.org  Thu Aug 27 09:25:46 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 09:25:46 -0700
Subject: [Freeswitch-users] Get the Session State
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAI0pCHFyxT5Mn9pdicKxxSMBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAI0pCHFyxT5Mn9pdicKxxSMBAAAAAA==@gmail.com>
Message-ID: <87f2f3b90908270925i8815f33qdbe60758f4645631@mail.gmail.com>

When you say call state, do you mean something like "ringing" or "answered"?
-MC

On Thu, Aug 27, 2009 at 3:23 AM, delianSPAM <delianspam at gmail.com> wrote:

>  Hello Everybody!
>
>
>
> How do you get the call state in Python?
>
>
>
> I have tried:
>
> ?
>
> session.answer()
>
> state_result=str(session.getVariable("state"))
>
> console_log("debug",state_result)
>
> ?
>
>
>
> But it returns: ?None?
>
>
>
> Thank you!
>
>
>
> Best Regards, Delian Tashev
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Thu Aug 27 09:32:23 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 27 Aug 2009 13:32:23 -0300
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <5e414ed0908270406m628342d5ya0e9f25514ed09d2@mail.gmail.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
	<5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
	<274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>
	<5e414ed0908260910h55122d77o5013ef1795d0c75@mail.gmail.com>
	<5e414ed0908270406m628342d5ya0e9f25514ed09d2@mail.gmail.com>
Message-ID: <ebbc2d550908270932q391deaffwd03fc7a6aa42dde4@mail.gmail.com>

Open on endpoint modules. We will relocate if needed.

jmesquita

On Thu, Aug 27, 2009 at 8:06 AM, Dennis <odermann at googlemail.com> wrote:

> sorry, but i do not know i which category i have to set this problem.
> could you help me with that?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mrene_lists at avgs.ca  Thu Aug 27 09:41:25 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 27 Aug 2009 12:41:25 -0400
Subject: [Freeswitch-users] fscore mutex locking question
In-Reply-To: <a37510170908270708g1e3236ao1f16a6835a87e8d7@mail.gmail.com>
References: <a37510170908270708g1e3236ao1f16a6835a87e8d7@mail.gmail.com>
Message-ID: <83AEC845-DFE5-47EF-8C67-80A9CD73F1B1@avgs.ca>

The only time the session is  write locked is when its about to be  
free'd, to make sure nothing holds a valid pointer.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 27-Aug-09, at 10:08 AM, Benedikt Fraunhofer wrote:

> Hello *,
>
> while looking at the code i came across a region of code which is
> unclear to me regarding locking issues.
> One example is switch_ivr_broadcast in switch_ivr_async.c. This should
> be the function called by
> uuid_broadcast() and others.
>
> in line 2341 it tries to queue an event to the bleg if it has to...
> -----
> .   if ((flags & SMF_ECHO_BLEG) && (other_uuid =
> switch_channel_get_variable(channel, SWITCH_SIGNAL_BOND_VARIABLE))
> .   .   && (other_session = switch_core_session_locate(other_uuid))) {
>
> ---
> switch_core_session_locate() does a "readonly trylock" on the channel
> mutex returning NULL if it's unable to
> aquire the lock, which brings up the following question:
> If some other thread is currently holding a writelock on the channel,
> the broadcast is not queued and not retried at a later time at all?
>
> I guess it's pretty easy to cause some unexpected behaviour using some
> endless loop calling "uuid_setvar" or some other race condition where
> the channel-mutex is write-locked while calling uuid_broadcast (eg.
> uuid_media?).
>
> Could this lead to a problem in "real live" scenarios, or are there
> other countermeasures despite "chances are one in a million that you
> hit that small time frame"?
>
> thx in advance
>  Beni.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From anthony.minessale at gmail.com  Thu Aug 27 09:48:37 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 11:48:37 -0500
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
	<1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
Message-ID: <191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>

please retest with latest SVN trunk

On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy <
anatoliy at kounitskiy.com> wrote:

> Just for information - the idea :
> A---calls---> B ---att_xfer---> C    , B hangs C up and goes back to A
> ( I`m sorry C is not answering :) )
>
>
-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Aug 27 09:51:33 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 11:51:33 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
Message-ID: <191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>

Can you paste in the line that actually hangs up your call
it's BLUE and says something like

2009-08-27 11:39:04.903202 [NOTICE] sofia.c:327 Hangup sofia/internal/
1004 at dev.bkw.org [CS_EXECUTE] [NORMAL_CLEARING]

find the line like this for each leg of you call and put them in your reply


On Thu, Aug 27, 2009 at 11:01 AM, Dennis <odermann at googlemail.com> wrote:

> > Are the phones behind the same nat as FS?
>
> no, both phones are behind the same nat, fs is behind another nat (and
> the internet is inbetween).
>
> both phones are in our office and we are sitting behind a nat. we
> connect through our nat over the internet into the other nat, where fs
> is behind.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Aug 27 09:54:22 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 11:54:22 -0500
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <ebbc2d550908270932q391deaffwd03fc7a6aa42dde4@mail.gmail.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
	<5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
	<274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>
	<5e414ed0908260910h55122d77o5013ef1795d0c75@mail.gmail.com>
	<5e414ed0908270406m628342d5ya0e9f25514ed09d2@mail.gmail.com>
	<ebbc2d550908270932q391deaffwd03fc7a6aa42dde4@mail.gmail.com>
Message-ID: <191c3a030908270954n4b24fea4tf6a7b8815943193c@mail.gmail.com>

are you setting presence_id=user at domain variable on the outbound leg?
This is done for you in the DP via the user/ channel in the defaults but if
you are not using this
you have to set it manually.


2009/8/27 Jo?o Mesquita <jmesquita at freeswitch.org>

> Open on endpoint modules. We will relocate if needed.
>
> jmesquita
>
>
> On Thu, Aug 27, 2009 at 8:06 AM, Dennis <odermann at googlemail.com> wrote:
>
>> sorry, but i do not know i which category i have to set this problem.
>> could you help me with that?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Thu Aug 27 10:02:52 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 19:02:52 +0200
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <191c3a030908270954n4b24fea4tf6a7b8815943193c@mail.gmail.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
	<5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
	<274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>
	<5e414ed0908260910h55122d77o5013ef1795d0c75@mail.gmail.com>
	<5e414ed0908270406m628342d5ya0e9f25514ed09d2@mail.gmail.com>
	<ebbc2d550908270932q391deaffwd03fc7a6aa42dde4@mail.gmail.com>
	<191c3a030908270954n4b24fea4tf6a7b8815943193c@mail.gmail.com>
Message-ID: <5e414ed0908271002y6f30ab24w2ed5b5532df758a8@mail.gmail.com>

> are you setting presence_id=user at domain variable on the outbound leg?
> This is done for you in the DP via the user/ channel in the defaults but if
> you are not using this
> you have to set it manually.

in directory default we have the following:

<params>
     <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
   </params>

everthing works fine with the led lights - the only problem is the
described above.


From msc at freeswitch.org  Thu Aug 27 10:21:09 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 10:21:09 -0700
Subject: [Freeswitch-users] fifo question
In-Reply-To: <27c25bc40908240725k5e0c92a2u2def5bd679b7b666@mail.gmail.com>
References: <27c25bc40908240725k5e0c92a2u2def5bd679b7b666@mail.gmail.com>
Message-ID: <87f2f3b90908271021i2ca380euf41e233c6a8ea232@mail.gmail.com>

On Mon, Aug 24, 2009 at 7:25 AM, Juan Backson <juanbackson at gmail.com> wrote:

> Hi,
>
> Does anyone know the purpose of fifo_orbit_announce?
>
> When does fifo_orbit_announce get played?
>

I believe this is what gets played to a caller when he/she has been in queue
too long and it times out. I'm assuming this gets played and then the caller
is transferred to the fifo_orbit_extension. (See also fifo_orbit_context and
fifo_orbit_dialplan.)
-MC
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From msc at freeswitch.org  Thu Aug 27 10:23:21 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 10:23:21 -0700
Subject: [Freeswitch-users] FXO and analogue phones
In-Reply-To: <C281504B-3D79-465E-ADA9-25C140FCF50F@mac.com>
References: <C281504B-3D79-465E-ADA9-25C140FCF50F@mac.com>
Message-ID: <87f2f3b90908271023u10abd4c3n6012bc7817a73782@mail.gmail.com>

Merul,

My apologies, I had meant to follow up much earlier. Would you mind going
ahead and capturing full debug log output of a call in as well as a call
out? Please use pastebin.freeswitch.org to post the logs and then reply here
with the link to the pb post.

Thanks,
MC

On Sun, Aug 23, 2009 at 10:27 AM, Merul Patel <merul at mac.com> wrote:

> I have a Freeswitch setup working on an Alix embedded platform in
> conjunction with a USB FXO device from Sangoma. My goal is to be able to
> either answer incoming calls on a softphone or on a POTS handset elsewhere
> in the building, and to also be able to make outgoing calls from either. For
> clarity, the analogue line has two physical extensions, one connected to the
> POTS and the other to the FXO.
>
> I can make and receive calls fine, but have problems when the call is
> answered on the POTS handset.
>
> Here is the dialplan I initially used  in
> /opt/freeswitch/conf/dialplans/public/01_incoming.xml:
>
> <include>
>  <extension name="public_did">
>    <condition field="${strftime(%w)}" expression="^(\d)$">
>      <!-- There seems to be a delay of 7 seconds from when FS starts
> dealing with the call and from when it starts ringing -->
>      <action application="sleep" data="23000"/>
>      <action application="set" data="domain_name=$${domain}"/>
>      <action application="transfer" data="1001 XML default"/>
>    </condition>
>  </extension>
> </include>
>
> It's pretty basic, and if the softphone is not registered or does not
> answer then the call goes to voicemail. However the call will always go to
> voicemail, and the voicemail application will begin to execute after the
> call has been answered on the POTS handset.
>
> I've been trying to make the dialplan more useful, by having it ring the
> softphone immediately, and only transfer the call to the voicemail
> application if the line is still ringing. I'm in the UK, hence my choice of
> frequencies in the tone_detect application:
>
> <include>
>  <extension name="public_did">
>    <condition field="${strftime(%w)}" expression="^(\d)$">
>      <!-- There seems to be a delay of 7 seconds from when FS starts
> dealing with the call and from when it starts ringing -->
>      <action application="set" data="call_timeout=23"/>
>      <action application="set" data="continue_on_fail=true"/>
>      <action application="set" data="hangup_after_bridge=true"/>
>      <action application="bridge" data="sofia/internal/1001%$${domain}"/>
>      <action application="sleep" data="23000"/>
>      <action application="tone_detect" data="ring 400,450 r +5000 set
> RING=true"/>
>      <action application="transfer" data="public_answer_and_email"/>
>    </condition>
>  </extension>
>
> <extension name="public_answer_and_email">
>    <condition field="RING" expression="true">
>       <action application="answer"/>
>       <action application="voicemail" data="default $${domain} 1001"/>
>    </condition>
>  </extension>
> </include>
>
> Unfortunately, this is not working, and the logs are not yielding anything
> the is helpful to me. Is my use of the tone_detect application and the basic
> dialplan correct?
>
> Merul
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From odermann at googlemail.com  Thu Aug 27 10:25:56 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 19:25:56 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
Message-ID: <5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>

this is the line (without stun - so we only have one leg) and we
called the 5900 to moh:

2009-08-27 19:18:02.348232 [NOTICE] sofia.c:3863 Hangup
sofia/internal/1004 at 212.18.215.102 [CS_EXECUTE]
[RECOVERY_ON_TIMER_EXPIRE]

we called the 5900 and waited 2 minutes...

or did you mean something different?


From Nick.Lemberger at lkfd.net  Thu Aug 27 10:41:39 2009
From: Nick.Lemberger at lkfd.net (Nick Lemberger)
Date: Thu, 27 Aug 2009 12:41:39 -0500
Subject: [Freeswitch-users] mod_voicemail email template variables
Message-ID: <4A967F21.2C9A.00FE.0@lkfd.net>

Is there a way to use dialplan variables in the email that gets sent with the voicemail attachement.  I tried using some but nothing seems to show up, I'm guessing it's a different channel or something...

Any ideas?

Thanks,
-Nick



From anthony.minessale at gmail.com  Thu Aug 27 10:41:18 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 12:41:18 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
Message-ID: <191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>

do the following:

1) restart FS
2) type sofia profile internal siptrace on
3) type console loglevel debug
4) reproduce the issue
5) upload the resulting output to http://pastebin.freeswitch.org (*hint* the
pass is in the dialog box)
6) let me know the pastebin number


On Thu, Aug 27, 2009 at 12:25 PM, Dennis <odermann at googlemail.com> wrote:

> this is the line (without stun - so we only have one leg) and we
> called the 5900 to moh:
>
> 2009-08-27 19:18:02.348232 [NOTICE] sofia.c:3863 Hangup
> sofia/internal/1004 at 212.18.215.102 [CS_EXECUTE]
> [RECOVERY_ON_TIMER_EXPIRE]
>
> we called the 5900 and waited 2 minutes...
>
> or did you mean something different?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Aug 27 10:44:22 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 12:44:22 -0500
Subject: [Freeswitch-users] mod_voicemail email template variables
In-Reply-To: <4A967F21.2C9A.00FE.0@lkfd.net>
References: <4A967F21.2C9A.00FE.0@lkfd.net>
Message-ID: <191c3a030908271044k63973088xeec12c578d02ef14@mail.gmail.com>

all variables referenced in the template should expand when sending the
email.


On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger <Nick.Lemberger at lkfd.net>wrote:

> Is there a way to use dialplan variables in the email that gets sent with
> the voicemail attachement.  I tried using some but nothing seems to show up,
> I'm guessing it's a different channel or something...
>
> Any ideas?
>
> Thanks,
> -Nick
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From Prometheus001 at gmx.net  Thu Aug 27 10:53:42 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 27 Aug 2009 19:53:42 +0200
Subject: [Freeswitch-users] sofia profile external register gwname via
 XML-Curl?
In-Reply-To: <4A95DBED.2060505@gmx.net>
References: <4A95DBED.2060505@gmx.net>
Message-ID: <4A96C826.3020204@gmx.net>

I got it,

gateways have to be preloaded (rescanned) before they can be registered.

Best regards
peter

Peter P GMX schrieb:
> Hello,
>
> I am using XML-Curl to handle the configuration of freeswitch
> When I try to register a gateway via event-socket with
>     sofia profile external register <gw-name>
> I receive back "invalid gateway".
>
> After reload mod_sofia the gateway is there. Question: Does this command
> work with xml-curl or only with local files??
> At least I see no xml-curl request when grepping network traffic.
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   


From Prometheus001 at gmx.net  Thu Aug 27 11:02:31 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 27 Aug 2009 20:02:31 +0200
Subject: [Freeswitch-users] Calls from registered gateway try to lookup
	Directory
Message-ID: <4A96CA37.8060303@gmx.net>

I have found a strange thing in my FS installation,

FS is registered via a Gateway to an external provider (QSC) in the
external context.
But when a call is coming in, FS does not seem to go to any context, but
tries to lookup the user, as I receive the following message
    2009-08-27 18:44:23.287782 [WARNING] sofia_reg.c:1773 Can't find
user [026xxxxxxxxx at my.domain@my.domain]
    You must define a domain called 'my.domain' in your directory and
add a user with the id="026xxxxxxxxx at my.domain" attribute
    and you must configure your device to use the proper domain in it's
authentication credentials.

I learnt that a call from an external gateway should go to the public
context. But (in CLI debug mode) there are no other messages, except the
3 lines above.

What am I doing wrong?
Best regards
Peter

Here the invite message.

INVITE sip:gw+gw_xxxxxxxxxx at xx.xxx.xx.xxx:5080;transport=udp SIP/2.0.
Via:SIP/2.0/UDP
62.206.3.xxx;branch=z9hG4bK-BroadWorks.as1-xx.xxx.xx.xxxV5080-0-778271239-1616003581-1251392025611-.
From:"0xxxxXXXXXX"<sip:0xxxxXXXXXX at 62.206.3.xxx;user=phone>;tag=1616003581-1251392025611-.
To:"Me"<sip:026xxxxxxxxx at my.domain>.
Call-ID:BW185345611270809356816303 at 62.206.3.xxx.
CSeq:778271239 INVITE.
Contact:<sip:62.206.3.xxx:5060>.
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE.
Accept:multipart/mixed,application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp.
Supported:.
Max-Forwards:20.
Proxy-Authorization:DIGEST
cnonce="fyvqi2pf",qop=auth,uri="sip:gw+gw_026xxxxxxxxx at xx.xxx.xx.xxx:5080;transport=udp",realm="my.domain",username="026xxxxxxxxx at my.domain",nonce="21bbe70c-932a-11de-b94d-bbade892ded3",algorithm=MD5,response="6d39a2546a4aa9a1fc39e2dc07c1e934",nc=00000001.
Content-Type:application/sdp.
Content-Length:344.
.
v=0.
o=BroadWorks 1271473 1 IN IP4 87.234.9.178.
s=-.
c=IN IP4 87.234.9.178.
t=0 0.
m=audio 18534 RTP/AVP 8 0 2 99 18 110.
a=rtpmap:99 G726-24/8000.
a=rtpmap:110 X-NSE/8000.
a=fmtp:110 192-194,200-202.
a=X-sqn:0.
a=X-cap: 1 audio RTP/AVP 110.
a=X-cpar: a=rtpmap:110 X-NSE/8000.
a=X-cpar: a=fmtp:110 192-194,200-202.
a=X-cap: 2 image udptl t38.


From Nick.Lemberger at lkfd.net  Thu Aug 27 11:06:01 2009
From: Nick.Lemberger at lkfd.net (Nick Lemberger)
Date: Thu, 27 Aug 2009 13:06:01 -0500
Subject: [Freeswitch-users] mod_voicemail email template variables
In-Reply-To: <191c3a030908271044k63973088xeec12c578d02ef14@mail.gmail.com>
References: <4A967F21.2C9A.00FE.0@lkfd.net>
	<191c3a030908271044k63973088xeec12c578d02ef14@mail.gmail.com>
Message-ID: <4A9684D7.2C9A.00FE.0@lkfd.net>

Thanks for the fast reply!

I just tried 10 random variables from http://wiki.freeswitch.org/wiki/Channel_Variables  and I only see the whitespace where the variable should be.  I've only been able to get the ones that are set in mod_voicemail.c circa line 1600 to work.

-Nick

>>> On 8/27/2009 at 12:44 PM, in message
<191c3a030908271044k63973088xeec12c578d02ef14 at mail.gmail.com>, Anthony
Minessale <anthony.minessale at gmail.com> wrote:
> all variables referenced in the template should expand when sending the
> email.
> 
> 
> On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger 
> <Nick.Lemberger at lkfd.net>wrote:
> 
>> Is there a way to use dialplan variables in the email that gets sent with
>> the voicemail attachement.  I tried using some but nothing seems to show up,
>> I'm guessing it's a different channel or something...
>>
>> Any ideas?
>>
>> Thanks,
>> -Nick
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org 
>>
> 
> 



From odermann at googlemail.com  Thu Aug 27 11:11:36 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 20:11:36 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
Message-ID: <5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>

the pastebin number is 10129


From odermann at googlemail.com  Thu Aug 27 11:11:36 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 20:11:36 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
Message-ID: <5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>

the pastebin number is 10129


From asterisk at dotr.com  Thu Aug 27 11:16:33 2009
From: asterisk at dotr.com (Julian Lyndon-Smith)
Date: Thu, 27 Aug 2009 19:16:33 +0100
Subject: [Freeswitch-users] newbie alert. Help with dialplan
Message-ID: <f9c510070908271116n33c3f35di44eea1790f62bfc8@mail.gmail.com>

I am a long-time asterisk user (2005), so I come from an unfortunate
position of having to unlearn an awful lot of stuff in order to make
freeswitch do the things I want ;(

I *think* i've got my head around using mod_curl_xml (?) to read all
the config stuff from my webserver. I *think* I've got my head around
setting up the sip clients etc etc ...

However, where I am really struggling is the dialplan. For the life of
me I simply cannot seem to grasp the fs way - that's no disrespect to
the fs way, but perhaps the failure of this old brain to change !

 have pastebinned an example of show 1234546 at inboundq at
http://www.pastebin.ca/1544890. If possible, would someone be able to
show me how to convert this dialplan to the fs way ? If I could be
given a little foot-up I'm sure that I would be able to convert the
rest of the dialplan !

Thanks in advance, and please go easy !

Julian


From anatoliy at kounitskiy.com  Thu Aug 27 11:18:53 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Thu, 27 Aug 2009 21:18:53 +0300
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
	<1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
	<191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>
Message-ID: <1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>

Thank you all for the great job :)
Now it works as I wanted!!

Tomorrow I'll  will try to update the wiki :)

But with few words ( I tried to strip the extensions to minimum for
more easy read for users).
In the dialplan I have created the extension showed below:

 <extension name="local_number">
    <condition field="destination_number" expression="^(\d{3})$">
      <action application="set" data="dialed_extension=$1"/>
      <action application="export" data="dialed_extension=$1"/>
      <action application="bind_meta_app" data="1 b s
execute_extension::attented_xfer XML features"/>
      <action application="set" data="transfer_ringback=$${hold_music}"/>
      <action application="set" data="call_timeout=10"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>
    </condition>
  </extension>

In the features.xml file I have extension attented_xfer, that is using
the att_xfer application :

 <extension name="attented_xfer">
      <condition field="destination_number" expression="^attented_xfer$">
        <action application="set" data="continue_on_fail=true"/>
        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 30000 #"/>
        <action application="set" data="origination_cancel_key=#"/>
        <action application="att_xfer"
data="user/${attxfer_callthis}@${domain_name}"/>
      </condition>
    </extension>

Also I have set the the origination cancel key will be # - so if B
wants to cancel the call to C and to go back to A - just uses the #
key :)

Again - thank you very much :) :)

Anatoliy Kounitskiy

On Thu, Aug 27, 2009 at 7:48 PM, Anthony
Minessale<anthony.minessale at gmail.com> wrote:
> please retest with latest SVN trunk
>
> On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy
> <anatoliy at kounitskiy.com> wrote:
>>
>> Just for information - the idea :
>> A---calls---> B ---att_xfer---> C ? ?, B hangs C up and goes back to A
>> ( I`m sorry C is not answering :) )
>>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540


From Prometheus001 at gmx.net  Thu Aug 27 11:23:12 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 27 Aug 2009 20:23:12 +0200
Subject: [Freeswitch-users] Calls from registered gateway try to lookup
	Directory
In-Reply-To: <4A96CA37.8060303@gmx.net>
References: <4A96CA37.8060303@gmx.net>
Message-ID: <4A96CF10.8090009@gmx.net>

And yes,

external profile is on Port 5080 and all request go to 5080.

Best regards
Peter

Peter P GMX schrieb:
> I have found a strange thing in my FS installation,
>
> FS is registered via a Gateway to an external provider (QSC) in the
> external context.
> But when a call is coming in, FS does not seem to go to any context, but
> tries to lookup the user, as I receive the following message
>     2009-08-27 18:44:23.287782 [WARNING] sofia_reg.c:1773 Can't find
> user [026xxxxxxxxx at my.domain@my.domain]
>     You must define a domain called 'my.domain' in your directory and
> add a user with the id="026xxxxxxxxx at my.domain" attribute
>     and you must configure your device to use the proper domain in it's
> authentication credentials.
>
> I learnt that a call from an external gateway should go to the public
> context. But (in CLI debug mode) there are no other messages, except the
> 3 lines above.
>
> What am I doing wrong?
> Best regards
> Peter
>
> Here the invite message.
>
> INVITE sip:gw+gw_xxxxxxxxxx at xx.xxx.xx.xxx:5080;transport=udp SIP/2.0.
> Via:SIP/2.0/UDP
> 62.206.3.xxx;branch=z9hG4bK-BroadWorks.as1-xx.xxx.xx.xxxV5080-0-778271239-1616003581-1251392025611-.
> From:"0xxxxXXXXXX"<sip:0xxxxXXXXXX at 62.206.3.xxx;user=phone>;tag=1616003581-1251392025611-.
> To:"Me"<sip:026xxxxxxxxx at my.domain>.
> Call-ID:BW185345611270809356816303 at 62.206.3.xxx.
> CSeq:778271239 INVITE.
> Contact:<sip:62.206.3.xxx:5060>.
> Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE.
> Accept:multipart/mixed,application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp.
> Supported:.
> Max-Forwards:20.
> Proxy-Authorization:DIGEST
> cnonce="fyvqi2pf",qop=auth,uri="sip:gw+gw_026xxxxxxxxx at xx.xxx.xx.xxx:5080;transport=udp",realm="my.domain",username="026xxxxxxxxx at my.domain",nonce="21bbe70c-932a-11de-b94d-bbade892ded3",algorithm=MD5,response="6d39a2546a4aa9a1fc39e2dc07c1e934",nc=00000001.
> Content-Type:application/sdp.
> Content-Length:344.
> .
> v=0.
> o=BroadWorks 1271473 1 IN IP4 87.234.9.178.
> s=-.
> c=IN IP4 87.234.9.178.
> t=0 0.
> m=audio 18534 RTP/AVP 8 0 2 99 18 110.
> a=rtpmap:99 G726-24/8000.
> a=rtpmap:110 X-NSE/8000.
> a=fmtp:110 192-194,200-202.
> a=X-sqn:0.
> a=X-cap: 1 audio RTP/AVP 110.
> a=X-cpar: a=rtpmap:110 X-NSE/8000.
> a=X-cpar: a=fmtp:110 192-194,200-202.
> a=X-cap: 2 image udptl t38.
>
>   


From brian at freeswitch.org  Thu Aug 27 11:23:39 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 27 Aug 2009 13:23:39 -0500
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
	<1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
	<191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>
	<1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>
Message-ID: <417A7E0B-E053-4E1C-9848-93E101656E49@freeswitch.org>

/me points to the first question on the FAQ  ;)

/b

On Aug 27, 2009, at 1:18 PM, Anatoliy Kounitskiy wrote:

> Thank you all for the great job :)
> Now it works as I wanted!!
>
> Tomorrow I'll  will try to update the wiki :)



From msc at freeswitch.org  Thu Aug 27 11:30:11 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 11:30:11 -0700
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
	<1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
	<191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>
	<1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>
Message-ID: <87f2f3b90908271130v4bb7deads45d5f1c190ea0661@mail.gmail.com>

Thanks! I'll be in touch to help you if you need any wiki editing
assistance.
-MC

On Thu, Aug 27, 2009 at 11:18 AM, Anatoliy Kounitskiy <
anatoliy at kounitskiy.com> wrote:

> Thank you all for the great job :)
> Now it works as I wanted!!
>
> Tomorrow I'll  will try to update the wiki :)
>
> But with few words ( I tried to strip the extensions to minimum for
> more easy read for users).
> In the dialplan I have created the extension showed below:
>
>  <extension name="local_number">
>     <condition field="destination_number" expression="^(\d{3})$">
>      <action application="set" data="dialed_extension=$1"/>
>      <action application="export" data="dialed_extension=$1"/>
>      <action application="bind_meta_app" data="1 b s
> execute_extension::attented_xfer XML features"/>
>      <action application="set" data="transfer_ringback=$${hold_music}"/>
>      <action application="set" data="call_timeout=10"/>
>      <action application="set" data="hangup_after_bridge=true"/>
>       <action application="bridge"
> data="user/${dialed_extension}@${domain_name}"/>
>     </condition>
>  </extension>
>
> In the features.xml file I have extension attented_xfer, that is using
> the att_xfer application :
>
>  <extension name="attented_xfer">
>       <condition field="destination_number" expression="^attented_xfer$">
>        <action application="set" data="continue_on_fail=true"/>
>        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
> attxfer_callthis 30000 #"/>
>         <action application="set" data="origination_cancel_key=#"/>
>        <action application="att_xfer"
> data="user/${attxfer_callthis}@${domain_name}"/>
>      </condition>
>    </extension>
>
> Also I have set the the origination cancel key will be # - so if B
> wants to cancel the call to C and to go back to A - just uses the #
> key :)
>
> Again - thank you very much :) :)
>
> Anatoliy Kounitskiy
>
> On Thu, Aug 27, 2009 at 7:48 PM, Anthony
> Minessale<anthony.minessale at gmail.com> wrote:
> > please retest with latest SVN trunk
> >
> > On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy
> > <anatoliy at kounitskiy.com> wrote:
> >>
> >> Just for information - the idea :
> >> A---calls---> B ---att_xfer---> C    , B hangs C up and goes back to A
> >> ( I`m sorry C is not answering :) )
> >>
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anatoliy Kounitskiy
> -------------------------
> E-mail: anatoliy at kounitskiy.com
> Mobile: +359898913540
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Thu Aug 27 11:44:30 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 27 Aug 2009 14:44:30 -0400
Subject: [Freeswitch-users] Who is "callie"?
In-Reply-To: <87f2f3b90908270918u66439bd2wd046a4dc485caeb8@mail.gmail.com>
References: <4A96AEA8.3030308@nowthor.com>
	<87f2f3b90908270918u66439bd2wd046a4dc485caeb8@mail.gmail.com>
Message-ID: <D206B5F9-2D91-4652-A944-09CAD425ABBD@jerris.com>

Also, note, we have a few sound files we would like to add to our  
default set.  If you are doing a session anyways and are willing  
please let me know if we can throw a few other prompts in your list to  
be recorded.

Mike

On Aug 27, 2009, at 12:18 PM, Michael Collins wrote:

>
>
> On Thu, Aug 27, 2009 at 9:04 AM, Carlos S. Antunes <csa at nowthor.com>  
> wrote:
> Hi!
>
> I searched the wiki but couldn't find the answer. Is "callie" a real
> woman one may ask to record additional sounds?
>
> "Callie" is one of the voices from GM Voices. She is definitely  
> available for custom work. Visit www.gmvoices.com for more info.  
> Tell them that the FreeSWITCH project sent you. :)
> -MC
>
>
> Thanks!
>
> Carlos
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Thu Aug 27 11:45:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 13:45:03 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
	<5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>
Message-ID: <191c3a030908271145g1995e642s92f6979c207987ef@mail.gmail.com>

I think the problem is the session-timeout is too long and your nat mapping
is being deleted.

try setting it to a smaller value like 20 or 40 sec
or try setting those phones to register more frequently (every 10 or 20 sec)


On Thu, Aug 27, 2009 at 1:11 PM, Dennis <odermann at googlemail.com> wrote:

> the pastebin number is 10129
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Aug 27 11:54:09 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 13:54:09 -0500
Subject: [Freeswitch-users] mod_voicemail email template variables
In-Reply-To: <4A9684D7.2C9A.00FE.0@lkfd.net>
References: <4A967F21.2C9A.00FE.0@lkfd.net>
	<191c3a030908271044k63973088xeec12c578d02ef14@mail.gmail.com>
	<4A9684D7.2C9A.00FE.0@lkfd.net>
Message-ID: <191c3a030908271154t155e844dy459acb700316182c@mail.gmail.com>

you should be able to for instance put

<action application="set" data="test_var=this is a test"/>

right before the voicemail app is called

then put

${test_var} in your template.

making sure to issue reloadxml or restart FS


On Thu, Aug 27, 2009 at 1:06 PM, Nick Lemberger <Nick.Lemberger at lkfd.net>wrote:

> Thanks for the fast reply!
>
> I just tried 10 random variables from
> http://wiki.freeswitch.org/wiki/Channel_Variables  and I only see the
> whitespace where the variable should be.  I've only been able to get the
> ones that are set in mod_voicemail.c circa line 1600 to work.
>
> -Nick
>
> >>> On 8/27/2009 at 12:44 PM, in message
> <191c3a030908271044k63973088xeec12c578d02ef14 at mail.gmail.com>, Anthony
> Minessale <anthony.minessale at gmail.com> wrote:
> > all variables referenced in the template should expand when sending the
> > email.
> >
> >
> > On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger
> > <Nick.Lemberger at lkfd.net>wrote:
> >
> >> Is there a way to use dialplan variables in the email that gets sent
> with
> >> the voicemail attachement.  I tried using some but nothing seems to show
> up,
> >> I'm guessing it's a different channel or something...
> >>
> >> Any ideas?
> >>
> >> Thanks,
> >> -Nick
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Thu Aug 27 12:09:12 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 21:09:12 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908271145g1995e642s92f6979c207987ef@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
	<5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>
	<191c3a030908271145g1995e642s92f6979c207987ef@mail.gmail.com>
Message-ID: <5e414ed0908271209j15cb8cc7v2a21d02b6831eb1f@mail.gmail.com>

> I think the problem is the session-timeout is too long and your nat mapping
> is being deleted.

we changed the param session-timeout (in internal) to 20 and 40,
without success.

we changed the minimum-session-expires to 20, although we knew, the
allowed minimum is 90 and 90 was shown in the console - no success.


> try setting it to a smaller value like 20 or 40 sec
> or try setting those phones to register more frequently (every 10 or 20 sec)

we changed all possible options to 10 and 20, again without success :(


From gmaruzz at celliax.org  Thu Aug 27 12:20:36 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 27 Aug 2009 21:20:36 +0200
Subject: [Freeswitch-users] Can a chat message be sent to a cell phone
	with FS?
In-Reply-To: <4A9656FE.9030504@yahoo.com>
References: <4A9656FE.9030504@yahoo.com>
Message-ID: <7b197bef0908271220w2dca29fase62f50fc5419d073@mail.gmail.com>

In a short while (for any value of "short") will be available for
testing mod-celliax, an interface to the cellular phones networks for
voice calls and SMSs.

-giovanni


On 8/27/09, Merle J. Ebbert <se02005-mec at yahoo.com> wrote:
> Can a chat message be sent to a cell phone with FS?
>
> Thanks,
> Merle
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618


From anthony.minessale at gmail.com  Thu Aug 27 12:21:59 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 14:21:59 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908271209j15cb8cc7v2a21d02b6831eb1f@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
	<5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>
	<191c3a030908271145g1995e642s92f6979c207987ef@mail.gmail.com>
	<5e414ed0908271209j15cb8cc7v2a21d02b6831eb1f@mail.gmail.com>
Message-ID: <191c3a030908271221tdf76f28g657cb8fd2b39406@mail.gmail.com>

If you look at your trace the call is sending a re-invite over and over and
over again with no reply

you need to examine your network topology and find out why the packets FS is
sending to your phone
never make it.

also try disabling session-timers on the snom


On Thu, Aug 27, 2009 at 2:09 PM, Dennis <odermann at googlemail.com> wrote:

> > I think the problem is the session-timeout is too long and your nat
> mapping
> > is being deleted.
>
> we changed the param session-timeout (in internal) to 20 and 40,
> without success.
>
> we changed the minimum-session-expires to 20, although we knew, the
> allowed minimum is 90 and 90 was shown in the console - no success.
>
>
> > try setting it to a smaller value like 20 or 40 sec
> > or try setting those phones to register more frequently (every 10 or 20
> sec)
>
> we changed all possible options to 10 and 20, again without success :(
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Thu Aug 27 12:42:34 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 21:42:34 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908271221tdf76f28g657cb8fd2b39406@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
	<5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>
	<191c3a030908271145g1995e642s92f6979c207987ef@mail.gmail.com>
	<5e414ed0908271209j15cb8cc7v2a21d02b6831eb1f@mail.gmail.com>
	<191c3a030908271221tdf76f28g657cb8fd2b39406@mail.gmail.com>
Message-ID: <5e414ed0908271242v214eeb4eofa47f796290b41df@mail.gmail.com>

> If you look at your trace the call is sending a re-invite over and over and
> over again with no reply
>
> you need to examine your network topology and find out why the packets FS is
> sending to your phone
> never make it.
>
> also try disabling session-timers on the snom

are you talking about our problem, that we get a hangup after 120
seconds? i just ask to avoid, that we are talking about different
things.

if we connect without stun and call the 5900, we can hear the music,
so i assume, that we are receiving packets from fs!? but after about
120 seconds, we receive the hangup. before everything was fine and as
expected.

we also tried disabling session-timers in the snom - same problem.

thant's the problem, we tried soooooo many things, but simply nothing
changes the problem or makes something different (like sending the
hangup later or earlier).


From anthony.minessale at gmail.com  Thu Aug 27 13:06:24 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 15:06:24 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908271242v214eeb4eofa47f796290b41df@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
	<5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>
	<191c3a030908271145g1995e642s92f6979c207987ef@mail.gmail.com>
	<5e414ed0908271209j15cb8cc7v2a21d02b6831eb1f@mail.gmail.com>
	<191c3a030908271221tdf76f28g657cb8fd2b39406@mail.gmail.com>
	<5e414ed0908271242v214eeb4eofa47f796290b41df@mail.gmail.com>
Message-ID: <191c3a030908271306s6e656caeqea86343184e4bc33@mail.gmail.com>

yes i am talking about that same problem.
I am not talking about audio I am talking about sip messages.

look at the trace you made from my earlier request
You can see the same invite being sent 10 times and never got any answer.
so you need to investigate it.


On Thu, Aug 27, 2009 at 2:42 PM, Dennis <odermann at googlemail.com> wrote:

> > If you look at your trace the call is sending a re-invite over and over
> and
> > over again with no reply
> >
> > you need to examine your network topology and find out why the packets FS
> is
> > sending to your phone
> > never make it.
> >
> > also try disabling session-timers on the snom
>
> are you talking about our problem, that we get a hangup after 120
> seconds? i just ask to avoid, that we are talking about different
> things.
>
> if we connect without stun and call the 5900, we can hear the music,
> so i assume, that we are receiving packets from fs!? but after about
> 120 seconds, we receive the hangup. before everything was fine and as
> expected.
>
> we also tried disabling session-timers in the snom - same problem.
>
> thant's the problem, we tried soooooo many things, but simply nothing
> changes the problem or makes something different (like sending the
> hangup later or earlier).
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From pjintheusa at gmail.com  Thu Aug 27 13:09:10 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 27 Aug 2009 16:09:10 -0400
Subject: [Freeswitch-users] Can I stream a file to a parked call.
Message-ID: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>

Hi there,

I know there are other ways of doing this. I am just trying get to
know have fun with the FreeSWITCH API.

I am using originate and park a call:

fsApi.Execute("originate",
string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
&park", blegSession.Uuid, OutgoingCallerID, NumberToDial));

That works great, the phone rings. I want to play something to this
called party when they pick up.

 while (!blegSession.answered())
 {
        Log.WriteLine(LogLevel.Alert,
"Inside::CallReturns:!Session.answered::Loop");
        blegSession.sleep(500, 1);
 }

string promptFile = "prompts/whisper.wav";
blegSession.StreamFile(promptFile, 0);


This works - but the audio is choppy and slow.

Is there something I need to do to that parked call before streaming that file?


Thanks for any input.

Phil


From csa at nowthor.com  Thu Aug 27 13:13:37 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 27 Aug 2009 16:13:37 -0400
Subject: [Freeswitch-users] Who is "callie"?
In-Reply-To: <D206B5F9-2D91-4652-A944-09CAD425ABBD@jerris.com>
References: <4A96AEA8.3030308@nowthor.com>	<87f2f3b90908270918u66439bd2wd046a4dc485caeb8@mail.gmail.com>
	<D206B5F9-2D91-4652-A944-09CAD425ABBD@jerris.com>
Message-ID: <4A96E8F1.8010106@nowthor.com>

Mike,

Sure! I am planning on doing a session "soon", maybe in a couple of 
weeks or so. My only question is whether GM Voices license will allow 
this kind of thing. Do you or anyone else know?

Carlos

Michael Jerris wrote:
> Also, note, we have a few sound files we would like to add to our 
> default set.  If you are doing a session anyways and are willing 
> please let me know if we can throw a few other prompts in your list to 
> be recorded.
>
> Mike
>
> On Aug 27, 2009, at 12:18 PM, Michael Collins wrote:
>
>>
>>
>> On Thu, Aug 27, 2009 at 9:04 AM, Carlos S. Antunes <csa at nowthor.com 
>> <mailto:csa at nowthor.com>> wrote:
>>
>>     Hi!
>>
>>     I searched the wiki but couldn't find the answer. Is "callie" a real
>>     woman one may ask to record additional sounds?
>>
>>
>> "Callie" is one of the voices from GM Voices. She is definitely 
>> available for custom work. Visit www.gmvoices.com 
>> <http://www.gmvoices.com/> for more info. Tell them that the 
>> FreeSWITCH project sent you. :)
>> -MC
>>  
>>
>>
>>     Thanks!
>>
>>     Carlos
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
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From anthony.minessale at gmail.com  Thu Aug 27 13:30:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 15:30:03 -0500
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
Message-ID: <191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>

try wedging {ignore_early_media=true} before the first [ in your dial string
and eliminate the code waiting for answer.


On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi there,
>
> I know there are other ways of doing this. I am just trying get to
> know have fun with the FreeSWITCH API.
>
> I am using originate and park a call:
>
> fsApi.Execute("originate",
>
> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
>
> That works great, the phone rings. I want to play something to this
> called party when they pick up.
>
>  while (!blegSession.answered())
>  {
>        Log.WriteLine(LogLevel.Alert,
> "Inside::CallReturns:!Session.answered::Loop");
>        blegSession.sleep(500, 1);
>  }
>
> string promptFile = "prompts/whisper.wav";
> blegSession.StreamFile(promptFile, 0);
>
>
> This works - but the audio is choppy and slow.
>
> Is there something I need to do to that parked call before streaming that
> file?
>
>
> Thanks for any input.
>
> Phil
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Thu Aug 27 13:40:04 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 13:40:04 -0700
Subject: [Freeswitch-users] Can a chat message be sent to a cell phone
	with FS?
In-Reply-To: <7b197bef0908271220w2dca29fase62f50fc5419d073@mail.gmail.com>
References: <4A9656FE.9030504@yahoo.com>
	<7b197bef0908271220w2dca29fase62f50fc5419d073@mail.gmail.com>
Message-ID: <87f2f3b90908271340l1a794686x55100f4da32c73f8@mail.gmail.com>

On Thu, Aug 27, 2009 at 12:20 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> In a short while (for any value of "short") will be available for
> testing mod-celliax, an interface to the cellular phones networks for
> voice calls and SMSs.
>
> -giovanni


We eagerly anticipate it's arrival! :)
-MC
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From gregt at cgicommunications.com  Thu Aug 27 14:13:42 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Thu, 27 Aug 2009 17:13:42 -0400
Subject: [Freeswitch-users] Mixing mod_curl_xml dynamic dialplans and static
	ones
Message-ID: <C62AA2C3-DBE4-47D1-95FA-A4DF92C2CE9E@cgicommunications.com>

I have mod_curl_xml working, but I also have several static dialplans  
in dialplan/public/ and I can't seem to get it to search those if my  
php generated xml page does not return a result.

I know in the static public.xml there is this
     <extension name="unloop">
       <condition field="${unroll_loops}" expression="^true$"/>
       <condition field="${sip_looped_call}" expression="^true$">
         <action application="deflect" data="${destination_number}"/>
       </condition>
     </extension>

Do I need my php page to generate something like that so it will  
continue looking in the static pages if it does not find a match in  
the xml_curl dynamic dialplan?

I don't know if I have given enough data for someone to point me in  
the right direction...
This is in the console when I have mod_xml_curl enabled:

2009-08-27 16:39:52 [INFO] mod_dialplan_xml.c:252 dialplan_hunt()  
Processing  ->15854199896 in context public
Dialplan: sofia/internal/585xxxxxxx at 208.34.86.39 parsing [public- 
 >curl_test] continue=false
Dialplan: sofia/internal/585xxxxxxx at 208.34.86.39 Regex (FAIL)  
[curl_test] destination_number(15854199896) =~ /^(18775844111)$/  
break=on-false
2009-08-27 16:39:52 [INFO] switch_core_state_machine.c:136  
switch_core_standard_on_routing() No Route, Aborting

--
Greg



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From krice at freeswitch.org  Thu Aug 27 14:32:28 2009
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 27 Aug 2009 16:32:28 -0500
Subject: [Freeswitch-users] Mixing mod_curl_xml dynamic dialplans and
 static ones
In-Reply-To: <C62AA2C3-DBE4-47D1-95FA-A4DF92C2CE9E@cgicommunications.com>
Message-ID: <C6BC659C.5DE68%krice@freeswitch.org>

Your php should return a not found, also you can stack static and non-static
XML dialplans on your technology profile settings

For example, on your internal.xml (in the default sip profiles) the dialplan
setting can be set to ?XML,XML:/path/to/some/static.xml? . Once it processes
thru the xml_curl responses it will hit the static.xml



From: Greg Thoen <gregt at cgicommunications.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Thu, 27 Aug 2009 17:13:42 -0400
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Mixing mod_curl_xml dynamic dialplans and static
ones

I have mod_curl_xml working, but I also have several static dialplans in
dialplan/public/ and I can't seem to get it to search those if my php
generated xml page does not return a result.

I know in the static public.xml there is this
    <extension name="unloop">
      <condition field="${unroll_loops}" expression="^true$"/>
      <condition field="${sip_looped_call}" expression="^true$">
        <action application="deflect" data="${destination_number}"/>
      </condition>
    </extension>

Do I need my php page to generate something like that so it will continue
looking in the static pages if it does not find a match in the xml_curl
dynamic dialplan?

I don't know if I have given enough data for someone to point me in the
right direction...
This is in the console when I have mod_xml_curl enabled:

2009-08-27 16:39:52 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing
->15854199896 in context public
Dialplan: sofia/internal/585xxxxxxx at 208.34.86.39 parsing [public->curl_test]
continue=false
Dialplan: sofia/internal/585xxxxxxx at 208.34.86.39 Regex (FAIL) [curl_test]
destination_number(15854199896) =~ /^(18775844111)$/ break=on-false
2009-08-27 16:39:52 [INFO] switch_core_state_machine.c:136
switch_core_standard_on_routing() No Route, Aborting

 
--

Greg 



 



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From msc at freeswitch.org  Thu Aug 27 14:39:44 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 14:39:44 -0700
Subject: [Freeswitch-users] Mixing mod_curl_xml dynamic dialplans and
	static ones
In-Reply-To: <C6BC659C.5DE68%krice@freeswitch.org>
References: <C62AA2C3-DBE4-47D1-95FA-A4DF92C2CE9E@cgicommunications.com>
	<C6BC659C.5DE68%krice@freeswitch.org>
Message-ID: <87f2f3b90908271439u466a2634xd5b1ff62dd41ebfe@mail.gmail.com>

On Thu, Aug 27, 2009 at 2:32 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Your php should return a not found, also you can stack static and
> non-static XML dialplans on your technology profile settings
>
> For example, on your internal.xml (in the default sip profiles) the
> dialplan setting can be set to ?XML,XML:/path/to/some/static.xml? . Once it
> processes thru the xml_curl responses it will hit the static.xml
>

Ken,

That's a very cool feature. I wasn't aware of it. I will see about adding
this knowledge to the wiki if it isn't there already.
-MC
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From tina at a2unlimited.com  Thu Aug 27 14:47:35 2009
From: tina at a2unlimited.com (Tina Martinez)
Date: Thu, 27 Aug 2009 17:47:35 -0400
Subject: [Freeswitch-users] Using a "Virtual" Extension
Message-ID: <47111.1251409655@a2unlimited.com>

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From jaybinks at gmail.com  Thu Aug 27 14:55:27 2009
From: jaybinks at gmail.com (Jay Binks)
Date: Fri, 28 Aug 2009 07:55:27 +1000
Subject: [Freeswitch-users] memory leak
In-Reply-To: <a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
Message-ID: <3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>

Anthony can you ( or anyone else alao ). Please elaborate on what  
makes centos 5.3 o much better for Freeswitch.

Is there some specific library vesiion on centos that makes a massive  
difference ?

Reason I ask ...   I personally only have a preference for debian,   
but others may have policy mandated Os's
For their companies, and it would be great to have some info about this.

I'd imagine this actually boils down to a requirement for libfoo  
version X ...  And if we ran those library versions on other OS's we'd  
be fine ?

Jay



>
>> I recommend you try your application on 64 bit CentOS which is the  
>> platform
>> all of our paid customers use.
>
> hmm. our operations-people wont be happy with that.


From msc at freeswitch.org  Thu Aug 27 14:59:56 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 14:59:56 -0700
Subject: [Freeswitch-users] Using a "Virtual" Extension
In-Reply-To: <47111.1251409655@a2unlimited.com>
References: <47111.1251409655@a2unlimited.com>
Message-ID: <87f2f3b90908271459s57e571d3ldb716d1913549edc@mail.gmail.com>

> So far this works great!  However, my question is whether or not using the
> same extension is a bad idea?  does it matter how many times I setup the
> link between the servers using the same extension for every conference?
>
> - T
>

Tina,

Glad to hear your multi-machine conference thing is working. Most likely I'd
say that you are okay to use the same extension number. It's really just an
administrative thing. As long as people know what number to dial it should
be just fine.

-MC
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From anthony.minessale at gmail.com  Thu Aug 27 15:43:39 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 17:43:39 -0500
Subject: [Freeswitch-users] memory leak
In-Reply-To: <3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
	<3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
Message-ID: <191c3a030908271543r70c0f03cw88e4fba21df2b61b@mail.gmail.com>

Mostly based on that is what we developed it with and continue to use and
have had the best luck and least problems with.

I am sure you are right about it being certain version of the core toolchain
I will guess it's the combo of libc and kernel and the fact that CentOS has
a purist attitude towards patching such code.

Some people use stable debian with no issues, other have reported problems
resolved by switching to CentOS

so presumably you could roll your own debian that was similar but we don't
have the resources to try to find out what makes what happen etc.

If you can configure debian to work, it's fine, if we get some inexplicable
problem on debian we will start to wonder though.
it's happened more than once with people doing nothing other than changing
OS =D



On Thu, Aug 27, 2009 at 4:55 PM, Jay Binks <jaybinks at gmail.com> wrote:

> Anthony can you ( or anyone else alao ). Please elaborate on what
> makes centos 5.3 o much better for Freeswitch.
>
> Is there some specific library vesiion on centos that makes a massive
> difference ?
>
> Reason I ask ...   I personally only have a preference for debian,
> but others may have policy mandated Os's
> For their companies, and it would be great to have some info about this.
>
> I'd imagine this actually boils down to a requirement for libfoo
> version X ...  And if we ran those library versions on other OS's we'd
> be fine ?
>
> Jay
>
>
>
> >
> >> I recommend you try your application on 64 bit CentOS which is the
> >> platform
> >> all of our paid customers use.
> >
> > hmm. our operations-people wont be happy with that.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jason at jasonjgw.net  Thu Aug 27 17:10:50 2009
From: jason at jasonjgw.net (Jason White)
Date: Fri, 28 Aug 2009 10:10:50 +1000
Subject: [Freeswitch-users] memory leak
In-Reply-To: <3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
	<3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
Message-ID: <20090828001050.GA8108@jdc.jasonjgw.net>

Jay Binks <jaybinks at gmail.com> wrote:
 
> Reason I ask ...   I personally only have a preference for debian,   
> but others may have policy mandated Os's
> For their companies, and it would be great to have some info about this.

The only problem I've had with FreeSWITCH under Debian Squeeze and Sid
involves TLS-related segmentation faults that appear to be related to
something in the version of libssl supplied with Debian. The same problem
can't be reproduced on Fedora, for example, but it does occur under Debian
Lenny as well as Debian Squeeze and Sid (i.e., "testing" and "unstable",
respectively).

besides this, all of the issues that I have encountered turned out to be
(usually short-lived) bugs in FreeSWITCH or one of the libraries included in
the source tree - they're mostly FreeSWITCH issues.

I should point out that the FreeSWITCH developers are very good at avoiding
the introduction of bugs into their code and that known bugs get fixed. It
appears to be an unchanging fact about programming that with a large and
complex project, even given highly knowledgeable, experienced, committed and
talented developers (as we have with FreeSWITCH), sometimes, bugs do slip
through.



From lakindia89 at gmail.com  Thu Aug 27 22:01:44 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 28 Aug 2009 10:31:44 +0530
Subject: [Freeswitch-users] How to make a call back
In-Reply-To: <87f2f3b90908270916l7cc7f3cap36bcf8725fa5fa5f@mail.gmail.com>
References: <25166083.post@talk.nabble.com>
	<87f2f3b90908270916l7cc7f3cap36bcf8725fa5fa5f@mail.gmail.com>
Message-ID: <7d79b3930908272201i3267aa54nafcc0171975cb023@mail.gmail.com>

No. In the dial plan I said, application="perl" data="The perl script".
I also checked $session->execute("bridge","user/1010"). This is working
fine.
But originate is not working as I expected.

On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com> wrote:
>
>>
>> When I give the following from the command line it calls to 1010 extension
>> and once answered, it calls to 1000 and bridge the connection.
>>            originate user/1010 &bridge(user/1000)
>> But I want to do this in perl. So I have given as follows
>>            $session->originate($session,"user/1010 &bridge user/1000");
>> But it is not working. It says "user/1010 &bridge user/1000 is invalid
>> user".
>> How to do this in perl. pls help.
>
>
> Are you calling this perl script from the CLI? If so you won't have the
> $session object because a channel does not exist for a simple API call.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From csa at nowthor.com  Thu Aug 27 22:13:38 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Fri, 28 Aug 2009 01:13:38 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <DEBF02AF-E36C-47FA-88C7-363BCC91282F@freeswitch.org>
References: <4A8D87D8.8080001@nowthor.com>	<700500D9-4508-4848-A198-76692574AF15@jerris.com>
	<DEBF02AF-E36C-47FA-88C7-363BCC91282F@freeswitch.org>
Message-ID: <4A976782.5000101@nowthor.com>

Brian,

You've been vindicated. Callcentric is now advertising zero weighted SRV 
records! :)

I've re-enabled SRV lookups for the Callcentric profile and will monitor 
to see if I get any errors.

Carlos

Brian West wrote:
> Or as I have argued today they should fix their SRV records to be zero  
> weighted.
>
> /b
>
> On Aug 20, 2009, at 1:36 PM, Michael Jerris wrote:
>
>   
>> You can bypass the srv records if you like by passing a :port with the
>> hostname where you use it in freeswitch.
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
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From msc at freeswitch.org  Thu Aug 27 22:57:59 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 27 Aug 2009 22:57:59 -0700
Subject: [Freeswitch-users] How to make a call back
Message-ID: <5F197C4F-2AF5-4DFA-AED5-045B8B08E342@freeswitch.org>



Sent from my iPhone

On Aug 27, 2009, at 10:01 PM, lakshmanan ganapathy  
<lakindia89 at gmail.com> wrote:

> No. In the dial plan I said, application="perl" data="The perl  
> script".
> I also checked $session->execute("bridge","user/1010"). This is  
> working fine.
> But originate is not working as I expected.
>
I think you might be confusing Dialplan apps with API commands. The  
$session object represents an existing channel and therefore it uses  
Dialplan apps. Originate is an API, that is, it is a command that you  
type at the CLI.

You need an API object to use originate from a script:
my $api = new $freeswitch::API();
my $res = $api->executeString("originate user/1010 &bridge(user/1000);

What kind of application are you developing? I'm curious why you need  
an originate to create a whole new call.

-MC

> On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com>  
> wrote:
>
> When I give the following from the command line it calls to 1010  
> extension
> and once answered, it calls to 1000 and bridge the connection.
>            originate user/1010 &bridge(user/1000)
> But I want to do this in perl. So I have given as follows
>            $session->originate($session,"user/1010 &bridge user/ 
> 1000");
> But it is not working. It says "user/1010 &bridge user/1000 is invalid
> user".
> How to do this in perl. pls help.
>
> Are you calling this perl script from the CLI? If so you won't have  
> the $session object because a channel does not exist for a simple  
> API call.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jmesquita at freeswitch.org  Thu Aug 27 23:09:26 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 28 Aug 2009 03:09:26 -0300
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A976782.5000101@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>
	<700500D9-4508-4848-A198-76692574AF15@jerris.com>
	<DEBF02AF-E36C-47FA-88C7-363BCC91282F@freeswitch.org>
	<4A976782.5000101@nowthor.com>
Message-ID: <ebbc2d550908272309i49620e97qdbf01b0efb2c8061@mail.gmail.com>

Bkw, I would recommend charging a fee from callcentric for the
consultancy. This consultant thing can get you going someday! LOL

Jmesquita funny joke

On 8/28/09, Carlos S. Antunes <csa at nowthor.com> wrote:
> Brian,
>
> You've been vindicated. Callcentric is now advertising zero weighted SRV
> records! :)
>
> I've re-enabled SRV lookups for the Callcentric profile and will monitor
> to see if I get any errors.
>
> Carlos
>
> Brian West wrote:
>> Or as I have argued today they should fix their SRV records to be zero
>> weighted.
>>
>> /b
>>
>> On Aug 20, 2009, at 1:36 PM, Michael Jerris wrote:
>>
>>
>>> You can bypass the srv records if you like by passing a :port with the
>>> hostname where you use it in freeswitch.
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

-- 
Sent from my mobile device


From lakindia89 at gmail.com  Thu Aug 27 23:43:08 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Thu, 27 Aug 2009 23:43:08 -0700 (PDT)
Subject: [Freeswitch-users] How to make a call back
In-Reply-To: <5F197C4F-2AF5-4DFA-AED5-045B8B08E342@freeswitch.org>
References: <25166083.post@talk.nabble.com>
	<5F197C4F-2AF5-4DFA-AED5-045B8B08E342@freeswitch.org>
Message-ID: <25184832.post@talk.nabble.com>


The following is the requirement.
Let say I have 3 extensions in my freeswitch. (777,1000,1010).
When 1000 made a call to 777, I'll execute a perl application.
In that, I'll get the DTMF (1010) from the caller. Then I'll make a call to
the entered digit and I've to bridge 1000 and 1010.

This is what my requirement.
I also tried the following.
#!/usr/bin/perl
use strict;
use freeswitch;
our $session;
my $sess=new freeswitch::Session("user/1000");
$sess->answer();
if($sess->ready())
{
        freeswitch::consoleLog("INFO","Session is answered\n");
       
$sess->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav");
        my $dtmf = $sess->getDigits(4,"", 5000);
        freeswitch::consoleLog("INFO","I received $dtmf\n");
        my $new_sess=new freeswitch::Session("user/$dtmf");
        $new_sess->answer();
        if($new_sess->ready())
        {
                $session->bridge($sess,$new_sess);
        }
}

In the above program I got usage error on $session->bridge.



mercutioviz wrote:
> 
> 
> 
> Sent from my iPhone
> 
> On Aug 27, 2009, at 10:01 PM, lakshmanan ganapathy  
> <lakindia89 at gmail.com> wrote:
> 
>> No. In the dial plan I said, application="perl" data="The perl  
>> script".
>> I also checked $session->execute("bridge","user/1010"). This is  
>> working fine.
>> But originate is not working as I expected.
>>
> I think you might be confusing Dialplan apps with API commands. The  
> $session object represents an existing channel and therefore it uses  
> Dialplan apps. Originate is an API, that is, it is a command that you  
> type at the CLI.
> 
> You need an API object to use originate from a script:
> my $api = new $freeswitch::API();
> my $res = $api->executeString("originate user/1010 &bridge(user/1000);
> 
> What kind of application are you developing? I'm curious why you need  
> an originate to create a whole new call.
> 
> -MC
> 
>> On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins  
>> <msc at freeswitch.org> wrote:
>>
>>
>> On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com>  
>> wrote:
>>
>> When I give the following from the command line it calls to 1010  
>> extension
>> and once answered, it calls to 1000 and bridge the connection.
>>            originate user/1010 &bridge(user/1000)
>> But I want to do this in perl. So I have given as follows
>>            $session->originate($session,"user/1010 &bridge user/ 
>> 1000");
>> But it is not working. It says "user/1010 &bridge user/1000 is invalid
>> user".
>> How to do this in perl. pls help.
>>
>> Are you calling this perl script from the CLI? If so you won't have  
>> the $session object because a channel does not exist for a simple  
>> API call.
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/How-to-make-a-call-back-tp25166083p25184832.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From delianspam at gmail.com  Thu Aug 27 23:43:48 2009
From: delianspam at gmail.com (delianSPAM)
Date: Fri, 28 Aug 2009 09:43:48 +0300
Subject: [Freeswitch-users] How to make a call back
In-Reply-To: <7d79b3930908272201i3267aa54nafcc0171975cb023@mail.gmail.com>
References: <25166083.post@talk.nabble.com>	<87f2f3b90908270916l7cc7f3cap36bcf8725fa5fa5f@mail.gmail.com>
	<7d79b3930908272201i3267aa54nafcc0171975cb023@mail.gmail.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAALucXIH9fG1HosZ+if323WMBAAAAAA==@gmail.com>

Hello Michael!

 

#Does this Python code helps:

new_api_obj = API()

result = new_api_obj.executeString("show channels"))             # Replace
"show channels" with a command of your choice

 

# By the way, I use the command below to connect an existing and answered
leg to a second one:

destination=1000

session.execute("bridge", "sofia/internal/" + destination +
"@yourpbx.domain.com")

 

Best Regards, 

Delian Tashev

 

 

From: lakshmanan ganapathy [mailto:lakindia89 at gmail.com] 
Sent: Friday, August 28, 2009 8:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How to make a call back

 

No. In the dial plan I said, application="perl" data="The perl script".
I also checked $session->execute("bridge","user/1010"). This is working
fine.
But originate is not working as I expected.

On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins <msc at freeswitch.org> wrote:

 

On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com> wrote:


When I give the following from the command line it calls to 1010 extension
and once answered, it calls to 1000 and bridge the connection.
           originate user/1010 &bridge(user/1000)
But I want to do this in perl. So I have given as follows
           $session->originate($session,"user/1010 &bridge user/1000");
But it is not working. It says "user/1010 &bridge user/1000 is invalid
user".
How to do this in perl. pls help.


Are you calling this perl script from the CLI? If so you won't have the
$session object because a channel does not exist for a simple API call. 
-MC


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From juanbackson at gmail.com  Thu Aug 27 23:51:03 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Fri, 28 Aug 2009 14:51:03 +0800
Subject: [Freeswitch-users] freeswitch odbc problem
Message-ID: <27c25bc40908272351g345ad1e2w31dc945e8f9a0331@mail.gmail.com>

Hi,
In a new environment, I am getting the following error when building the
latest freeswitch from svn.  Does anyone know how to resolve it?

gcc -I/usr/src/freeswitch-snapshot/src/include
-I/usr/src/freeswitch-snapshot/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-g -O2 -Wall -std=c99 -pedantic -o .libs/freeswitch freeswitch-switch.o  -lm
./.libs/libfreeswitch.so
-L/usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib
/usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/src/freeswitch-snapshot/libs/apr/.libs/libapr-1.a
-L/usr/src/freeswitch-snapshot/libs/srtp -L/usr/kerberos/lib64
libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt -lpthread
libs/libedit/src/.libs/libedit.a -lssl -lcrypto -ldl -lz -lncurses
-Wl,--rpath -Wl,/usr/local/freeswitch/lib
./.libs/libfreeswitch.so: undefined reference to `SQLGetData'
./.libs/libfreeswitch.so: undefined reference to `SQLDisconnect'
./.libs/libfreeswitch.so: undefined reference to `SQLSetConnectAttr'
./.libs/libfreeswitch.so: undefined reference to `SQLGetDiagRec'
./.libs/libfreeswitch.so: undefined reference to `SQLConnect'
./.libs/libfreeswitch.so: undefined reference to `SQLExecute'
./.libs/libfreeswitch.so: undefined reference to `SQLNumResultCols'
./.libs/libfreeswitch.so: undefined reference to `SQLFreeHandle'
./.libs/libfreeswitch.so: undefined reference to `SQLGetInfo'
./.libs/libfreeswitch.so: undefined reference to `SQLAllocHandle'
./.libs/libfreeswitch.so: undefined reference to `SQLDriverConnect'
./.libs/libfreeswitch.so: undefined reference to `SQLPrepare'
./.libs/libfreeswitch.so: undefined reference to `SQLError'
./.libs/libfreeswitch.so: undefined reference to `SQLRowCount'
./.libs/libfreeswitch.so: undefined reference to `SQLDescribeCol'
./.libs/libfreeswitch.so: undefined reference to `SQLFetch'
./.libs/libfreeswitch.so: undefined reference to `SQLSetEnvAttr'
collect2: ld returned 1 exit status
make[2]: *** [freeswitch] Error 1
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From kawarod at laposte.net  Thu Aug 27 23:54:00 2009
From: kawarod at laposte.net (rod)
Date: Fri, 28 Aug 2009 10:54:00 +0400
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
In-Reply-To: <999373713.8871.1251386588649.JavaMail.apache@mail22.abv.bg>
References: <999373713.8871.1251386588649.JavaMail.apache@mail22.abv.bg>
Message-ID: <4A977F08.9040005@laposte.net>

Hello,

the trace seems good.
If you check the answer from Kamailio, you'll see that Kamailio answers 
with "302 PEER_01".

As Michael Collins stated before, you can get the variable containing 
"PEER_01", then this variable is stored in a custom variable.

In your dialplan, may you please add:

<action application="info"/>, just before the transfer line, eg:

     <condition field="destination_number" expression="(\d+)$">
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" data="continue_on_fail=true"/>
         <action application="export" data="sip_h_X-ROUTE=LOOKUP"/>
         <action application="bridge" data="sofia/internal/${destination_number}@127.0.0.1:5062"/>
         <action application="set" data="ROUTE_GW=${sip_redirect_contact_user_0}"/>
         <action application="set" data="AREA=${sip_redirect_contact_user_0}"/>
	 <action application="info"/>
         <action application="transfer" data="${destination_number} XML ROUTING"/>
      </condition>


Using application Info, you'll see on the console (or CDR) the list of 
variables used for this call.  You should see the content of 
"${sip_redirect_contact_user_0}" that should contain  the value 
"PEER_01". Please check this and let me know.

For the new configuration file, no problem for sharing. But as I wrote 
on the wiki page, I worked on this setup cause LCR module was not 
available when I start working on FS, nor I'm a good programmer to write 
a server side HTTP script (used by xml_curl) that could scale to my needs.

I enhanced a bit this configuration with support for fallback routing, 
almost realtime graph (every minutes using www.cacti.net and some 
functions in FS like limit_hash) of number of concurrent calls per AREA, 
PEER...

Thanks for "the good tutorial", but don't forget the dev team who did 
this great product ;-)

rod.


Hristo Benev a ?crit :
>  I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).
>
> If you need additional info I'll provide it.
>
> Here is trace:
>
> ngrep -d any -nn -i '1000' port 5062 -W byline
> interface: any
> filter: (ip or ip6) and ( port 5062 )
> match: 1000
> #
> U 10.10.10.10:5090 -> 127.0.0.1:5062
> INVITE sip:1000 at 127.0.0.1:5062 SIP/2.0.
> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
> Max-Forwards: 69.
> From: "Extension 1001" <sip:1001 at 10.10.10.10>;tag=978g69jZaFpBD.
> To: <sip:1000 at 127.0.0.1:5062>.
> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
> CSeq: 119574771 INVITE.
> Contact: <sip:mod_sofia at 10.10.10.10:5090>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 429.
> X-ROUTE: LOOKUP.
> Remote-Party-ID: "Extension 1001" <sip:1001 at 10.10.10.10>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
> s=FreeSWITCH.
> c=IN IP4 10.10.10.10.
> t=0 0.
> m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:115 G7221/32000.
> a=fmtp:115 bitrate=48000.
> a=rtpmap:107 G7221/16000.
> a=fmtp:107 bitrate=32000.
> a=rtpmap:9 G722/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
>
> #
> U 127.0.0.1:5062 -> 10.10.10.10:5090
> SIP/2.0 302 PEER_01.
> Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
> From: "Extension 1001" <sip:1001 at 10.10.10.10>;tag=978g69jZaFpBD.
> To: <sip:1000 at 127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8.
> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
> CSeq: 119574771 INVITE.
> Contact: sip:France at PEER_01.
> Server: Kamailio (1.5.2-notls (i386/linux)).
> Content-Length: 0.
> .
>
> #
> U 10.10.10.10:5090 -> 127.0.0.1:5062
> ACK sip:1000 at 127.0.0.1:5062 SIP/2.0.
> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
> Max-Forwards: 69.
> From: "Extension 1001" <sip:1001 at 10.10.10.10>;tag=978g69jZaFpBD.
> To: <sip:1000 at 127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8.
> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
> CSeq: 119574771 ACK.
> Content-Length: 0.
> .
>
>
>
>
>
>  >-------- ?????????? ????? --------
>  >??:  Hristo Benev 
>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >??: freeswitch-users at lists.freeswitch.org
>  >????????? ??: ?????????, 2009, ?????? 27 14:58:53 EEST
>
>  > 
>  >Bojnour,
>  >
>  >I'll send a trace ASAP.
>  >
>  >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
>  >
>  >I'm really interested to see the updates of the project.
>  >
>  >Thank you for the good tutorial.
>  >
>  >Hristo
>  >
>  >
>  > >-------- ?????????? ????? --------
>  > >??:  rod 
>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  > >??: freeswitch-users at lists.freeswitch.org
>  > >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST
>  >
>  > >Hi Hristo,
>  > >
>  > >I'm the author of this setup and wiki page. I did a lot of modifications 
>  > >on this setup (alternative routing if failure essentially) but don't 
>  > >have too much time to update the wiki.
>  > >
>  > >May you please send me an ngrep trace when you call 1000:
>  > >
>  > >ngrep -d any -nn -i '1000' port 5060 -W byline 
>  > >
>  > >I will check what's happening.
>  > >Do you have an entry for 1000 in your mysql database ?
>  > >
>  > >regards,
>  > >rod
>  > >
>  > >Hristo Benev a ?crit :
>  > >>  
>  > >> It seems that the problem is on kamailio configuration.
>  > >> Will ask on their list.
>  > >>
>  > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
>  > >>
>  > >> I have it setup as friend in asterisk, but still ???
>  > >>
>  > >> Any ideas?
>  > >>
>  > >> Thanks,
>  > >>
>  > >>
>  > >>  >-------- ?????????? ????? --------
>  > >>  >??:  Hristo Benev 
>  > >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  > >>  >??: freeswitch-users at lists.freeswitch.org
>  > >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
>  > >>
>  > >>  > I think that the problem is here:
>  > >>  >-------------------------
>  > >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
>  > >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
>  > >>  >--------------------------
>  > >>  >
>  > >>  >Actually Regex FAIL
>  > >>  >
>  > >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
>  > >>  >Here is my  default.xml:
>  > >>  >----------------
>  > >>  >
>  > >>  >
>  > >>  >
>  > >>  >  
>  > >>  >
>  > >>  >     
>  > >>  >      
>  > >>  >         
>  > >>  >         
>  > >>  >         
>  > >>  >         
>  > >>  >         
>  > >>  >         
>  > >>  >         
>  > >>  >      
>  > >>  >    
>  > >>  >
>  > >>  >   
>  > >>  >
>  > >>  >  
>  > >>  >
>  > >>  >    
>  > >>  >      
>  > >>  >        
>  > >>  >        
>  > >>  >        
>  > >>  >        
>  > >>  >        
>  > >>  >        
>  > >>  >        
>  > >>  >        
>  > >>  >      
>  > >>  >    
>  > >>  >
>  > >>  >  
>  > >>  >
>  > >>  >
>  > >>  >
>  > >>  >--------------------------
>  > >>  >
>  > >>  >
>  > >>  >
>  > >>  > >-------- ?????????? ????? --------
>  > >>  > >??:  Brian West 
>  > >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  > >>  > >??: freeswitch-users at lists.freeswitch.org
>  > >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
>  > >>  >
>  > >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
>  > >>  >You have to process all 302's in the dialplan. 
>  > >>  >Set this on your sofia profile
>  > >>  >You can set these variables sip_redirect_profile,
>  > >>  >sip_redirect_context,
>  > >>  >sip_redirect_dialplan,
>  > >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
>  > >>  >sip_redirected_to,
>  > >>  >sip_redirect_contact_user_%d,
>  > >>  >sip_redirect_contact_host_%d,
>  > >>  >sip_redirect_contact_params_%d,
>  > >>  >sip_redirect_dialstring_%d,
>  > >>  >sip_redirect_dialstring,
>  > >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
>  > >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
>  > >>  > >
>  > >>  >
>  > >>  >_______________________________________________
>  > >>  >FreeSWITCH-users mailing list
>  > >>  >FreeSWITCH-users at lists.freeswitch.org
>  > >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  > >>  >http://www.freeswitch.org
>  > >>  >
>  > >>
>  > >> _______________________________________________
>  > >> FreeSWITCH-users mailing list
>  > >> FreeSWITCH-users at lists.freeswitch.org
>  > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  > >> http://www.freeswitch.org
>  > >>   
>  > >
>  > >_______________________________________________
>  > >FreeSWITCH-users mailing list
>  > >FreeSWITCH-users at lists.freeswitch.org
>  > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  > >http://www.freeswitch.org
>  > >
>  >
>  >_______________________________________________
>  >FreeSWITCH-users mailing list
>  >FreeSWITCH-users at lists.freeswitch.org
>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >http://www.freeswitch.org
>  >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From lakindia89 at gmail.com  Fri Aug 28 00:06:40 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Fri, 28 Aug 2009 00:06:40 -0700 (PDT)
Subject: [Freeswitch-users] How to make a call back
In-Reply-To: <25184832.post@talk.nabble.com>
References: <25166083.post@talk.nabble.com>
	<5F197C4F-2AF5-4DFA-AED5-045B8B08E342@freeswitch.org>
	<25184832.post@talk.nabble.com>
Message-ID: <25185052.post@talk.nabble.com>


Hey I'm sorry. I've solved this as follows
my $sess=new freeswitch::Session("user/1000");
$sess->answer();
#$sess->waitForAnswer($session);
if($sess->ready())
{
    freeswitch::consoleLog("INFO","Session is answered\n");
   
$sess->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav");
    my $dtmf = $sess->getDigits(4,"", 5000);
    freeswitch::consoleLog("INFO","I received $dtmf\n");
    $sess->execute("bridge","user/$dtmf");
}


lakshmanan wrote:
> 
> I also tried the following.
> #!/usr/bin/perl
> use strict;
> use freeswitch;
> our $session;
> my $sess=new freeswitch::Session("user/1000");
> $sess->answer();
> if($sess->ready())
> {
>         freeswitch::consoleLog("INFO","Session is answered\n");
>        
> $sess->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav");
>         my $dtmf = $sess->getDigits(4,"", 5000);
>         freeswitch::consoleLog("INFO","I received $dtmf\n");
>         my $new_sess=new freeswitch::Session("user/$dtmf");
>         $new_sess->answer();
>         if($new_sess->ready())
>         {
>                 $session->bridge($sess,$new_sess);
>         }
> }
> 
> In the above program I got usage error on $session->bridge.
> 
> 
> 
> mercutioviz wrote:
>> 
>> 
>> 
>> Sent from my iPhone
>> 
>> On Aug 27, 2009, at 10:01 PM, lakshmanan ganapathy  
>> <lakindia89 at gmail.com> wrote:
>> 
>>> No. In the dial plan I said, application="perl" data="The perl  
>>> script".
>>> I also checked $session->execute("bridge","user/1010"). This is  
>>> working fine.
>>> But originate is not working as I expected.
>>>
>> I think you might be confusing Dialplan apps with API commands. The  
>> $session object represents an existing channel and therefore it uses  
>> Dialplan apps. Originate is an API, that is, it is a command that you  
>> type at the CLI.
>> 
>> You need an API object to use originate from a script:
>> my $api = new $freeswitch::API();
>> my $res = $api->executeString("originate user/1010 &bridge(user/1000);
>> 
>> What kind of application are you developing? I'm curious why you need  
>> an originate to create a whole new call.
>> 
>> -MC
>> 
>>> On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins  
>>> <msc at freeswitch.org> wrote:
>>>
>>>
>>> On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com>  
>>> wrote:
>>>
>>> When I give the following from the command line it calls to 1010  
>>> extension
>>> and once answered, it calls to 1000 and bridge the connection.
>>>            originate user/1010 &bridge(user/1000)
>>> But I want to do this in perl. So I have given as follows
>>>            $session->originate($session,"user/1010 &bridge user/ 
>>> 1000");
>>> But it is not working. It says "user/1010 &bridge user/1000 is invalid
>>> user".
>>> How to do this in perl. pls help.
>>>
>>> Are you calling this perl script from the CLI? If so you won't have  
>>> the $session object because a channel does not exist for a simple  
>>> API call.
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 

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From lakindia89 at gmail.com  Fri Aug 28 00:08:50 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Fri, 28 Aug 2009 00:08:50 -0700 (PDT)
Subject: [Freeswitch-users] How to make a call back
Message-ID: <25185052.post@talk.nabble.com>


Hey I'm sorry. I've solved this as follows
my $sess=new freeswitch::Session("user/1000");
$sess->answer();
#$sess->waitForAnswer($session);
if($sess->ready())
{
    freeswitch::consoleLog("INFO","Session is answered\n");
   
$sess->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav");
    my $dtmf = $sess->getDigits(4,"", 5000);
    freeswitch::consoleLog("INFO","I received $dtmf\n");
    $sess->execute("bridge","user/$dtmf");
}




mercutioviz wrote:
> 
> 
> 
> Sent from my iPhone
> 
> On Aug 27, 2009, at 10:01 PM, lakshmanan ganapathy  
> <lakindia89 at gmail.com> wrote:
> 
>> No. In the dial plan I said, application="perl" data="The perl  
>> script".
>> I also checked $session->execute("bridge","user/1010"). This is  
>> working fine.
>> But originate is not working as I expected.
>>
> I think you might be confusing Dialplan apps with API commands. The  
> $session object represents an existing channel and therefore it uses  
> Dialplan apps. Originate is an API, that is, it is a command that you  
> type at the CLI.
> 
> You need an API object to use originate from a script:
> my $api = new $freeswitch::API();
> my $res = $api->executeString("originate user/1010 &bridge(user/1000);
> 
> What kind of application are you developing? I'm curious why you need  
> an originate to create a whole new call.
> 
> -MC
> 
>> On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins  
>> <msc at freeswitch.org> wrote:
>>
>>
>> On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com>  
>> wrote:
>>
>> When I give the following from the command line it calls to 1010  
>> extension
>> and once answered, it calls to 1000 and bridge the connection.
>>            originate user/1010 &bridge(user/1000)
>> But I want to do this in perl. So I have given as follows
>>            $session->originate($session,"user/1010 &bridge user/ 
>> 1000");
>> But it is not working. It says "user/1010 &bridge user/1000 is invalid
>> user".
>> How to do this in perl. pls help.
>>
>> Are you calling this perl script from the CLI? If so you won't have  
>> the $session object because a channel does not exist for a simple  
>> API call.
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 



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From cstomi.levlist at gmail.com  Fri Aug 28 03:59:57 2009
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Fri, 28 Aug 2009 12:59:57 +0200
Subject: [Freeswitch-users] timeout while waiting for CF_BRIDGED flag
Message-ID: <4A97B8AD.3030009@gmail.com>

Hello,

we have a strange problem, sometimes calls are hung up with 
DESTINATION_OUT_OF_ORDER cause.
I added some extra log lines, and it turned out that

switch_ivr_bridge.c:140 a timeout occurs while waiting for CF_BRIDGED flag
switch_channel_wait_for_flag(chan_b, CF_BRIDGED, SWITCH_TRUE, 10000, 
chan_a); returns SWITCH_STATUS_TIMEOUT

As far as I understand the flag on B channel is set in another thread, 
so maybe it happens slow sometimes.
We are using r14144. I know it is quiet old, but maybe this code hasn't 
changed since them

Do you have any idea, what could cause this?
Do you think it is a bug?

Thanks in advance,
Tamas



From diego.viola at gmail.com  Fri Aug 28 03:59:50 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 28 Aug 2009 10:59:50 +0000
Subject: [Freeswitch-users] memory leak
In-Reply-To: <20090828001050.GA8108@jdc.jasonjgw.net>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
	<3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
	<20090828001050.GA8108@jdc.jasonjgw.net>
Message-ID: <86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com>

Cool, feel free to report these bugs in Jira. If they are indeed FreeSWITCH
bugs.

On Fri, Aug 28, 2009 at 12:10 AM, Jason White <jason at jasonjgw.net> wrote:

> Jay Binks <jaybinks at gmail.com> wrote:
>
> > Reason I ask ...   I personally only have a preference for debian,
> > but others may have policy mandated Os's
> > For their companies, and it would be great to have some info about this.
>
> The only problem I've had with FreeSWITCH under Debian Squeeze and Sid
> involves TLS-related segmentation faults that appear to be related to
> something in the version of libssl supplied with Debian. The same problem
> can't be reproduced on Fedora, for example, but it does occur under Debian
> Lenny as well as Debian Squeeze and Sid (i.e., "testing" and "unstable",
> respectively).
>
> besides this, all of the issues that I have encountered turned out to be
> (usually short-lived) bugs in FreeSWITCH or one of the libraries included
> in
> the source tree - they're mostly FreeSWITCH issues.
>
> I should point out that the FreeSWITCH developers are very good at avoiding
> the introduction of bugs into their code and that known bugs get fixed. It
> appears to be an unchanging fact about programming that with a large and
> complex project, even given highly knowledgeable, experienced, committed
> and
> talented developers (as we have with FreeSWITCH), sometimes, bugs do slip
> through.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anatoliy at kounitskiy.com  Fri Aug 28 04:13:14 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Fri, 28 Aug 2009 14:13:14 +0300
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <87f2f3b90908271130v4bb7deads45d5f1c190ea0661@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
	<1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
	<191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>
	<1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>
	<87f2f3b90908271130v4bb7deads45d5f1c190ea0661@mail.gmail.com>
Message-ID: <1cd828b60908280413v6e5726cck72a5dcaa94ce3705@mail.gmail.com>

The wiki page is updated (
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer ). Hope
it is understandable :)

Anatoliy Kounitskiy

On Thu, Aug 27, 2009 at 9:30 PM, Michael Collins<msc at freeswitch.org> wrote:
> Thanks! I'll be in touch to help you if you need any wiki editing
> assistance.
> -MC
>
> On Thu, Aug 27, 2009 at 11:18 AM, Anatoliy Kounitskiy
> <anatoliy at kounitskiy.com> wrote:
>>
>> Thank you all for the great job :)
>> Now it works as I wanted!!
>>
>> Tomorrow I'll ?will try to update the wiki :)
>>
>> But with few words ( I tried to strip the extensions to minimum for
>> more easy read for users).
>> In the dialplan I have created the extension showed below:
>>
>> ?<extension name="local_number">
>> ? ?<condition field="destination_number" expression="^(\d{3})$">
>> ? ? ?<action application="set" data="dialed_extension=$1"/>
>> ? ? ?<action application="export" data="dialed_extension=$1"/>
>> ? ? ?<action application="bind_meta_app" data="1 b s
>> execute_extension::attented_xfer XML features"/>
>> ? ? ?<action application="set" data="transfer_ringback=$${hold_music}"/>
>> ? ? ?<action application="set" data="call_timeout=10"/>
>> ? ? ?<action application="set" data="hangup_after_bridge=true"/>
>> ? ? ?<action application="bridge"
>> data="user/${dialed_extension}@${domain_name}"/>
>> ? ?</condition>
>> ?</extension>
>>
>> In the features.xml file I have extension attented_xfer, that is using
>> the att_xfer application :
>>
>> ?<extension name="attented_xfer">
>> ? ? ?<condition field="destination_number" expression="^attented_xfer$">
>> ? ? ? ?<action application="set" data="continue_on_fail=true"/>
>> ? ? ? ?<action application="read" data="3 4 ivr/ivr-enter_ext.wav
>> attxfer_callthis 30000 #"/>
>> ? ? ? ?<action application="set" data="origination_cancel_key=#"/>
>> ? ? ? ?<action application="att_xfer"
>> data="user/${attxfer_callthis}@${domain_name}"/>
>> ? ? ?</condition>
>> ? ?</extension>
>>
>> Also I have set the the origination cancel key will be # - so if B
>> wants to cancel the call to C and to go back to A - just uses the #
>> key :)
>>
>> Again - thank you very much :) :)
>>
>> Anatoliy Kounitskiy
>>
>> On Thu, Aug 27, 2009 at 7:48 PM, Anthony
>> Minessale<anthony.minessale at gmail.com> wrote:
>> > please retest with latest SVN trunk
>> >
>> > On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy
>> > <anatoliy at kounitskiy.com> wrote:
>> >>
>> >> Just for information - the idea :
>> >> A---calls---> B ---att_xfer---> C ? ?, B hangs C up and goes back to A
>> >> ( I`m sorry C is not answering :) )
>> >>
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anatoliy Kounitskiy
>> -------------------------
>> E-mail: anatoliy at kounitskiy.com
>> Mobile: +359898913540
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540


From chad at apartmentlines.com  Fri Aug 28 04:58:59 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Fri, 28 Aug 2009 07:58:59 -0400
Subject: [Freeswitch-users] Who is "callie"?
In-Reply-To: <4A96E8F1.8010106@nowthor.com>
References: <4A96AEA8.3030308@nowthor.com>	<87f2f3b90908270918u66439bd2wd046a4dc485caeb8@mail.gmail.com>
	<D206B5F9-2D91-4652-A944-09CAD425ABBD@jerris.com>
	<4A96E8F1.8010106@nowthor.com>
Message-ID: <BAD6BA9A-2192-4676-A03E-4B5B02AA2063@apartmentlines.com>

i've also got a list of voicemail-related prompts that i'm going to  
use callie for.  i'm willing to throw my stuff into the pot, too, if  
we can have this turned around in two weeks.  otherwise i'll probably  
have to press on, as i have a deadline to make.  lemme know...


On Aug 27, 2009, at 4:13 PM, Carlos S. Antunes wrote:

> Mike,
>
> Sure! I am planning on doing a session "soon", maybe in a couple of  
> weeks or so. My only question is whether GM Voices license will  
> allow this kind of thing. Do you or anyone else know?
>
> Carlos
>
> Michael Jerris wrote:
>>
>> Also, note, we have a few sound files we would like to add to our  
>> default set.  If you are doing a session anyways and are willing  
>> please let me know if we can throw a few other prompts in your list  
>> to be recorded.
>>
>> Mike
>>
>> On Aug 27, 2009, at 12:18 PM, Michael Collins wrote:
>>
>>>
>>>
>>> On Thu, Aug 27, 2009 at 9:04 AM, Carlos S. Antunes  
>>> <csa at nowthor.com> wrote:
>>> Hi!
>>>
>>> I searched the wiki but couldn't find the answer. Is "callie" a real
>>> woman one may ask to record additional sounds?
>>>
>>> "Callie" is one of the voices from GM Voices. She is definitely  
>>> available for custom work. Visit www.gmvoices.com for more info.  
>>> Tell them that the FreeSWITCH project sent you. :)
>>> -MC
>>>
>>>
>>> Thanks!
>>>
>>> Carlos
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Fri Aug 28 06:10:59 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 28 Aug 2009 08:10:59 -0500
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A976782.5000101@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>	<700500D9-4508-4848-A198-76692574AF15@jerris.com>
	<DEBF02AF-E36C-47FA-88C7-363BCC91282F@freeswitch.org>
	<4A976782.5000101@nowthor.com>
Message-ID: <C8F7DF8A-C219-4745-893D-48D39ACA040D@freeswitch.org>

Thats still debatable.... The RFC's on these things read like chinese  
stereo instructions.

/b


On Aug 28, 2009, at 12:13 AM, Carlos S. Antunes wrote:

> Brian,
>
> You've been vindicated. Callcentric is now advertising zero weighted  
> SRV records! :)
>
> I've re-enabled SRV lookups for the Callcentric profile and will  
> monitor to see if I get any errors.
>
> Carlos



From nagalenoj at gmail.com  Thu Aug 27 23:44:31 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Thu, 27 Aug 2009 23:44:31 -0700 (PDT)
Subject: [Freeswitch-users]  How can it be done with freeswitch?
Message-ID: <25184845.post@talk.nabble.com>


Dear friends,
    I am using freeswitch for the last one month and trying to write perl
scripts to do some tasks. 
I am trying a new thing, which is something that I don't how to do it and I
don't whether it can be done with freeswitch.   my requirement is as
follows,
             A calls B[This extension has to execute a script].
             The script has to do the following,
              * Cut the call and callback to the number.
              * Get a phone number[DTMF] from A.
              * Dial the number(C) and bridge A and C.   
              * When C cuts the call, the control has to comeback to script.
              * Ask another number to A to dial out, get another phone
number[DTMF] from A.
              * Dial the number(D) and bridge A and D.

Please, help me to do this. Is there any other documentation to read about
the perl freeswitch functions.
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From math.parent at gmail.com  Fri Aug 28 02:15:04 2009
From: math.parent at gmail.com (Mathieu Parent)
Date: Fri, 28 Aug 2009 11:15:04 +0200
Subject: [Freeswitch-users] mod_fax not working
Message-ID: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com>

Hi,

I just started testing Freeswitch, this works well. This is powerfull
and easy to administrate.

I just have one problem and one request. The request is BOUNTY-5: IMAP
integration of voicemail (I will try to propose a patch if this is
within my competencies)

The problem is with mod_fax not receiving any fax.

Relevant extension is:
    <extension name="fax2mail">
        <condition field="destination_number" expression="^(9998)$">
            <action application="set" data="fax_verbose=true"/>
            <action application="set" data="fax_disable_v17=false"/>
            <action application="set" data="fax_use_ecm=true"/>
            <action application="answer" />
            <action application="playback" data="silence_stream://2000"/>
            <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/>
            <action application="hangup"/>
        </condition>
    </extension>

Test case is a 1-page fax coming from a Minolta printer connected to a
Cisco VG248 (via RJ11). The VG248 port is registred on a Cisco Call
Manager communicating with FS via a SIP trunk.

Log is attached:
- first try from 11:04:05 to 11:04:31
- 2nd try from 11:04:57 to 11:05:22
- 3rd try from 11:05:48 to 11:06:08

Do you need more information?

Regards

Mathieu Parent
-------------- next part --------------
2009-08-28 11:04:05.852705 [NOTICE] switch_channel.c:602 New Channel sofia/external/7629 at 85.0.30.241 [bd163f2c-93b1-11de-ae8e-27b1291d9722]
2009-08-28 11:04:05.852705 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_NEW
2009-08-28 11:04:05.852705 [DEBUG] switch_core_state_machine.c:404 (sofia/external/7629 at 85.0.30.241) State NEW
2009-08-28 11:04:05.852705 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [received][100]
2009-08-28 11:04:05.852705 [DEBUG] sofia.c:3296 Remote SDP:
v=0
o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 85.0.30.241
s=SIP Call
c=IN IP4 85.0.30.240
t=0 0
m=audio 28086 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2009-08-28 11:04:05.852705 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2009-08-28 11:04:05.852705 [DEBUG] sofia_glue.c:2029 Set Codec sofia/external/7629 at 85.0.30.241 PCMU/8000 20 ms 160 samples
2009-08-28 11:04:05.852705 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 101
2009-08-28 11:04:05.852705 [DEBUG] sofia.c:3455 (sofia/external/7629 at 85.0.30.241) State Change CS_NEW -> CS_INIT
2009-08-28 11:04:05.852705 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_INIT
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:481 (sofia/external/7629 at 85.0.30.241) State INIT
2009-08-28 11:04:05.856715 [DEBUG] mod_sofia.c:83 sofia/external/7629 at 85.0.30.241 SOFIA INIT
2009-08-28 11:04:05.856715 [DEBUG] mod_sofia.c:111 (sofia/external/7629 at 85.0.30.241) State Change CS_INIT -> CS_ROUTING
2009-08-28 11:04:05.856715 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:481 (sofia/external/7629 at 85.0.30.241) State INIT going to sleep
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_ROUTING
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:484 (sofia/external/7629 at 85.0.30.241) State ROUTING
2009-08-28 11:04:05.856715 [DEBUG] mod_sofia.c:130 sofia/external/7629 at 85.0.30.241 SOFIA ROUTING
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:78 sofia/external/7629 at 85.0.30.241 Standard ROUTING
2009-08-28 11:04:05.856715 [INFO] mod_dialplan_xml.c:315 Processing FAX D.I.C. 7629->9998 in context public
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->unloop] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->outside_call] continue=true
Dialplan: sofia/external/7629 at 85.0.30.241 Absolute Condition [outside_call]
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(outside_call=true)
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->call_debug] continue=true
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->public_extensions] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [public_extensions] destination_number(9998) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->vmain] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [vmain] destination_number(9998) =~ /^(9999)$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->fax2mail] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (PASS) [fax2mail] destination_number(9998) =~ /^(9998)$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_verbose=true)
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_disable_v17=false)
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_use_ecm=true)
Dialplan: sofia/external/7629 at 85.0.30.241 Action answer()
Dialplan: sofia/external/7629 at 85.0.30.241 Action playback(silence_stream://2000)
Dialplan: sofia/external/7629 at 85.0.30.241 Action rxfax(/tmp/${uuid}.rxfax.tiff)
Dialplan: sofia/external/7629 at 85.0.30.241 Action hangup()
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:114 (sofia/external/7629 at 85.0.30.241) State Change CS_ROUTING -> CS_EXECUTE
2009-08-28 11:04:05.856715 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:484 (sofia/external/7629 at 85.0.30.241) State ROUTING going to sleep
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_EXECUTE
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:491 (sofia/external/7629 at 85.0.30.241) State EXECUTE
2009-08-28 11:04:05.856715 [DEBUG] mod_sofia.c:173 sofia/external/7629 at 85.0.30.241 SOFIA EXECUTE
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:151 sofia/external/7629 at 85.0.30.241 Standard EXECUTE
EXECUTE sofia/external/7629 at 85.0.30.241 set(outside_call=true)
2009-08-28 11:04:05.856715 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [outside_call]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_verbose=true)
2009-08-28 11:04:05.856715 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_verbose]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_disable_v17=false)
2009-08-28 11:04:05.856715 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_disable_v17]=[false]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_use_ecm=true)
2009-08-28 11:04:05.856715 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_use_ecm]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 answer()
2009-08-28 11:04:05.856715 [DEBUG] mod_dptools.c:649 sofia/external/7629 at 85.0.30.241 receive message [ANSWER]
2009-08-28 11:04:05.856715 [DEBUG] sofia_glue.c:2263 AUDIO RTP [sofia/external/7629 at 85.0.30.241] 10.30.15.137 port 31604 -> 85.0.30.240 port 28086 codec: 0 ms: 20
2009-08-28 11:04:05.856715 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms
2009-08-28 11:04:05.860708 [DEBUG] mod_sofia.c:552 Local SDP sofia/external/7629 at 85.0.30.241:
v=0
o=FreeSWITCH 1251418641 1251418642 IN IP4 10.30.15.137
s=FreeSWITCH
c=IN IP4 10.30.15.137
t=0 0
m=audio 31604 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-08-28 11:04:05.860708 [DEBUG] switch_core_session.c:630 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:05.860708 [NOTICE] mod_dptools.c:649 Channel [sofia/external/7629 at 85.0.30.241] has been answered
2009-08-28 11:04:05.860708 [DEBUG] switch_channel.c:182 sofia/external/7629 at 85.0.30.241 receive message [AUDIO_SYNC]
EXECUTE sofia/external/7629 at 85.0.30.241 playback(silence_stream://2000)
2009-08-28 11:04:05.860708 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 8000hz 1 channels 20ms
2009-08-28 11:04:05.860708 [DEBUG] switch_core_io.c:649 sofia/external/7629 at 85.0.30.241 receive message [TRANSCODING_NECESSARY]
2009-08-28 11:04:05.860708 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [completed][200]
2009-08-28 11:04:05.872721 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [ready][200]
2009-08-28 11:04:07.868730 [DEBUG] switch_ivr_play_say.c:1391 done playing file
EXECUTE sofia/external/7629 at 85.0.30.241 rxfax(/tmp/bd163f2c-93b1-11de-ae8e-27b1291d9722.rxfax.tiff)
2009-08-28 11:04:07.868730 [DEBUG] mod_fax.c:591 Raw read codec activation Success L16 20000
2009-08-28 11:04:07.868730 [DEBUG] mod_fax.c:607 Raw write codec activation Success L16
2009-08-28 11:04:07.868730 [DEBUG] switch_channel.c:182 sofia/external/7629 at 85.0.30.241 receive message [AUDIO_SYNC]
2009-08-28 11:04:07.908721 [DEBUG] switch_core_io.c:232 sofia/external/7629 at 85.0.30.241 receive message [TRANSCODING_NECESSARY]
2009-08-28 11:04:08.468727 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2009-08-28 11:04:08.488736 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1
2009-08-28 11:04:10.68454 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2009-08-28 11:04:10.568485 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_A_CED, state 1
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Starting answer mode
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_A_CED to T30_PHASE_B_TX
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 1 to 17
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  CSI without final frame tag
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30 DIS:
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Can receive fax: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= 2-D coding: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Polled sub-address: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= T.43 coding: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Plane interleave: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Selective polling: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Sub-addressing: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Password: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Character mode: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Digital network capability: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Duplex capability: Half only
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= JPEG coding: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Full colour mode: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Custom illuminant: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Custom gamut range: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DIS with final frame tag
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2009-08-28 11:04:12.988723 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:04:13.68696 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:04:13.68696 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2009-08-28 11:04:13.68696 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 4
2009-08-28 11:04:13.68696 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:04:13.68696 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4
2009-08-28 11:04:13.88698 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:13.108696 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:13.128697 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:13.148696 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:13.148696 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:13.168483 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:15.228722 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:15.728724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Retry number 1
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  CSI without final frame tag
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30 DIS:
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Can receive fax: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= 2-D coding: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Polled sub-address: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= T.43 coding: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Plane interleave: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Selective polling: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Sub-addressing: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Password: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Character mode: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Digital network capability: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Duplex capability: Half only
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= JPEG coding: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Full colour mode: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Custom illuminant: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Custom gamut range: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DIS with final frame tag
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2009-08-28 11:04:18.588474 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:04:18.668474 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:04:18.668474 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2009-08-28 11:04:18.668474 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 4
2009-08-28 11:04:18.668474 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:04:18.668474 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4
2009-08-28 11:04:18.688478 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:18.708475 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:18.728726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:18.748726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:18.968729 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:19.188723 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:19.208726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:19.508723 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 17
2009-08-28 11:04:19.508723 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4A
2009-08-28 11:04:20.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:04:20.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 7
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T4A (13920 remaining)
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  DCS with final frame tag
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 In state 17
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 DCS:
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Receive fax: Set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 00..= Selected data signalling rate: V.17 14400bps
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= 2-D coding: Not set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..00= Recording width: 215mm +- 1%
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Minimum scan line time: 0ms
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Frame size: 256 octets
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Selected compression 3
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Get document at 14400bps, modem 7
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 17 to 7
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-08-28 11:04:20.848726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:04:20.848726 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_C_NON_ECM_RX
2009-08-28 11:04:20.848726 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:04:20.848726 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 7
2009-08-28 11:04:20.848726 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:04:20.928727 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:04:20.928727 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:04:21.68725 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:04:21.128725 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2009-08-28 11:04:21.128725 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:04:21.128725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:04:21.128725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:04:21.248725 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training in progress (-3) in state 7
2009-08-28 11:04:21.748711 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:04:22.168711 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:04:22.508710 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:04:22.808710 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:04:22.868712 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:04:23.648711 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:04:24.48712 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2009-08-28 11:04:24.68714 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:04:24.68714 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:04:27.108725 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:04:27.108725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:04:27.268725 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:04:27.468727 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 7
2009-08-28 11:04:27.468727 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2A
2009-08-28 11:04:28.648725 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:04:28.648725 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 3
2009-08-28 11:04:28.648725 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T2A (14720 remaining)
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Stop none (0 remaining)
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  DCS with final frame tag
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 In state 7
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected DCS received in state 7
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 7 to 3
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DCN with final frame tag
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 fa
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW FAX Switching from V.17 + V.21 to V.21 (-14.65dBm0)
2009-08-28 11:04:28.988725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 3
2009-08-28 11:04:28.988725 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_C_NON_ECM_RX to T30_PHASE_D_TX
2009-08-28 11:04:28.988725 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:04:28.988725 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:04:30.48729 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2009-08-28 11:04:30.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2009-08-28 11:04:30.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Disconnecting
2009-08-28 11:04:30.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E
2009-08-28 11:04:30.128724 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:04:30.128724 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 1
2009-08-28 11:04:30.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 3 to 2
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_E, state 2
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:167 ==============================================================================
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:180 Fax processing not successful - result (13) Unexpected message received.
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:185 Remote station id: 
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:186 Local station id:  SpanDSP Fax Ident
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:187 Pages transferred: 0
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:189 Total fax pages:   0
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:190 Image resolution:  8031x7700
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:191 Transfer Rate:     14400
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:193 ECM status         on
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:194 remote country:   
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:195 remote vendor:    
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:196 remote model:     
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:198 ==============================================================================
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 2 to 32
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 8
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 8
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete
2009-08-28 11:04:31.148726 [DEBUG] switch_core_codec.c:122 Restore original codec.
EXECUTE sofia/external/7629 at 85.0.30.241 hangup()
2009-08-28 11:04:31.148726 [NOTICE] mod_dptools.c:633 Hangup sofia/external/7629 at 85.0.30.241 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-28 11:04:31.148726 [DEBUG] switch_channel.c:1683 Send signal sofia/external/7629 at 85.0.30.241 [KILL]
2009-08-28 11:04:31.148726 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:491 (sofia/external/7629 at 85.0.30.241) State EXECUTE going to sleep
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_HANGUP
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:434 (sofia/external/7629 at 85.0.30.241) State HANGUP
2009-08-28 11:04:31.148726 [DEBUG] mod_sofia.c:338 Channel sofia/external/7629 at 85.0.30.241 hanging up, cause: NORMAL_CLEARING
2009-08-28 11:04:31.148726 [DEBUG] mod_sofia.c:396 Sending BYE to sofia/external/7629 at 85.0.30.241
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:46 sofia/external/7629 at 85.0.30.241 Standard HANGUP, cause: NORMAL_CLEARING
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:434 (sofia/external/7629 at 85.0.30.241) State HANGUP going to sleep
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:476 (sofia/external/7629 at 85.0.30.241) State Change CS_HANGUP -> CS_REPORTING
2009-08-28 11:04:31.148726 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_REPORTING
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:612 (sofia/external/7629 at 85.0.30.241) State REPORTING
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:53 sofia/external/7629 at 85.0.30.241 Standard REPORTING, cause: NORMAL_CLEARING
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:612 (sofia/external/7629 at 85.0.30.241) State REPORTING going to sleep
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:411 (sofia/external/7629 at 85.0.30.241) State Change CS_REPORTING -> CS_DESTROY
2009-08-28 11:04:31.148726 [DEBUG] switch_core_session.c:1068 Session 24 (sofia/external/7629 at 85.0.30.241) Locked, Waiting on external entities
2009-08-28 11:04:31.148726 [NOTICE] switch_core_session.c:1086 Session 24 (sofia/external/7629 at 85.0.30.241) Ended
2009-08-28 11:04:31.148726 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/7629 at 85.0.30.241 [CS_DESTROY]
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:564 (sofia/external/7629 at 85.0.30.241) State DESTROY
2009-08-28 11:04:31.148726 [DEBUG] mod_sofia.c:255 sofia/external/7629 at 85.0.30.241 SOFIA DESTROY
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:60 sofia/external/7629 at 85.0.30.241 Standard DESTROY
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:564 (sofia/external/7629 at 85.0.30.241) State DESTROY going to sleep
2009-08-28 11:04:57.372710 [NOTICE] switch_channel.c:602 New Channel sofia/external/7629 at 85.0.30.241 [dbcb8bfc-93b1-11de-ae8e-27b1291d9722]
2009-08-28 11:04:57.372710 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_NEW
2009-08-28 11:04:57.372710 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [received][100]
2009-08-28 11:04:57.372710 [DEBUG] sofia.c:3296 Remote SDP:
v=0
o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 85.0.30.241
s=SIP Call
c=IN IP4 85.0.30.240
t=0 0
m=audio 28090 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2009-08-28 11:04:57.372710 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2009-08-28 11:04:57.372710 [DEBUG] sofia_glue.c:2029 Set Codec sofia/external/7629 at 85.0.30.241 PCMU/8000 20 ms 160 samples
2009-08-28 11:04:57.372710 [DEBUG] switch_core_state_machine.c:404 (sofia/external/7629 at 85.0.30.241) State NEW
2009-08-28 11:04:57.372710 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 101
2009-08-28 11:04:57.372710 [DEBUG] sofia.c:3455 (sofia/external/7629 at 85.0.30.241) State Change CS_NEW -> CS_INIT
2009-08-28 11:04:57.372710 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_INIT
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:481 (sofia/external/7629 at 85.0.30.241) State INIT
2009-08-28 11:04:57.376739 [DEBUG] mod_sofia.c:83 sofia/external/7629 at 85.0.30.241 SOFIA INIT
2009-08-28 11:04:57.376739 [DEBUG] mod_sofia.c:111 (sofia/external/7629 at 85.0.30.241) State Change CS_INIT -> CS_ROUTING
2009-08-28 11:04:57.376739 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:481 (sofia/external/7629 at 85.0.30.241) State INIT going to sleep
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_ROUTING
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:484 (sofia/external/7629 at 85.0.30.241) State ROUTING
2009-08-28 11:04:57.376739 [DEBUG] mod_sofia.c:130 sofia/external/7629 at 85.0.30.241 SOFIA ROUTING
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:78 sofia/external/7629 at 85.0.30.241 Standard ROUTING
2009-08-28 11:04:57.376739 [INFO] mod_dialplan_xml.c:315 Processing FAX D.I.C. 7629->9998 in context public
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->unloop] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->outside_call] continue=true
Dialplan: sofia/external/7629 at 85.0.30.241 Absolute Condition [outside_call]
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(outside_call=true)
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->call_debug] continue=true
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->public_extensions] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [public_extensions] destination_number(9998) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->vmain] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [vmain] destination_number(9998) =~ /^(9999)$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->fax2mail] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (PASS) [fax2mail] destination_number(9998) =~ /^(9998)$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_verbose=true)
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_disable_v17=false)
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_use_ecm=true)
Dialplan: sofia/external/7629 at 85.0.30.241 Action answer()
Dialplan: sofia/external/7629 at 85.0.30.241 Action playback(silence_stream://2000)
Dialplan: sofia/external/7629 at 85.0.30.241 Action rxfax(/tmp/${uuid}.rxfax.tiff)
Dialplan: sofia/external/7629 at 85.0.30.241 Action hangup()
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:114 (sofia/external/7629 at 85.0.30.241) State Change CS_ROUTING -> CS_EXECUTE
2009-08-28 11:04:57.376739 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:484 (sofia/external/7629 at 85.0.30.241) State ROUTING going to sleep
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_EXECUTE
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:491 (sofia/external/7629 at 85.0.30.241) State EXECUTE
2009-08-28 11:04:57.376739 [DEBUG] mod_sofia.c:173 sofia/external/7629 at 85.0.30.241 SOFIA EXECUTE
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:151 sofia/external/7629 at 85.0.30.241 Standard EXECUTE
EXECUTE sofia/external/7629 at 85.0.30.241 set(outside_call=true)
2009-08-28 11:04:57.376739 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [outside_call]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_verbose=true)
2009-08-28 11:04:57.376739 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_verbose]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_disable_v17=false)
2009-08-28 11:04:57.376739 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_disable_v17]=[false]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_use_ecm=true)
2009-08-28 11:04:57.376739 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_use_ecm]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 answer()
2009-08-28 11:04:57.376739 [DEBUG] mod_dptools.c:649 sofia/external/7629 at 85.0.30.241 receive message [ANSWER]
2009-08-28 11:04:57.376739 [DEBUG] sofia_glue.c:2263 AUDIO RTP [sofia/external/7629 at 85.0.30.241] 10.30.15.137 port 32446 -> 85.0.30.240 port 28090 codec: 0 ms: 20
2009-08-28 11:04:57.376739 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms
2009-08-28 11:04:57.380744 [DEBUG] mod_sofia.c:552 Local SDP sofia/external/7629 at 85.0.30.241:
v=0
o=FreeSWITCH 1251417851 1251417852 IN IP4 10.30.15.137
s=FreeSWITCH
c=IN IP4 10.30.15.137
t=0 0
m=audio 32446 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-08-28 11:04:57.380744 [DEBUG] switch_core_session.c:630 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:57.380744 [NOTICE] mod_dptools.c:649 Channel [sofia/external/7629 at 85.0.30.241] has been answered
2009-08-28 11:04:57.380744 [DEBUG] switch_channel.c:182 sofia/external/7629 at 85.0.30.241 receive message [AUDIO_SYNC]
2009-08-28 11:04:57.380744 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [completed][200]
EXECUTE sofia/external/7629 at 85.0.30.241 playback(silence_stream://2000)
2009-08-28 11:04:57.380744 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 8000hz 1 channels 20ms
2009-08-28 11:04:57.380744 [DEBUG] switch_core_io.c:649 sofia/external/7629 at 85.0.30.241 receive message [TRANSCODING_NECESSARY]
2009-08-28 11:04:57.388729 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [ready][200]
2009-08-28 11:04:59.388741 [DEBUG] switch_ivr_play_say.c:1391 done playing file
EXECUTE sofia/external/7629 at 85.0.30.241 rxfax(/tmp/dbcb8bfc-93b1-11de-ae8e-27b1291d9722.rxfax.tiff)
2009-08-28 11:04:59.388741 [DEBUG] mod_fax.c:591 Raw read codec activation Success L16 20000
2009-08-28 11:04:59.388741 [DEBUG] mod_fax.c:607 Raw write codec activation Success L16
2009-08-28 11:04:59.388741 [DEBUG] switch_channel.c:182 sofia/external/7629 at 85.0.30.241 receive message [AUDIO_SYNC]
2009-08-28 11:04:59.428463 [DEBUG] switch_core_io.c:232 sofia/external/7629 at 85.0.30.241 receive message [TRANSCODING_NECESSARY]
2009-08-28 11:04:59.988724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2009-08-28 11:05:00.8724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1
2009-08-28 11:05:01.568724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2009-08-28 11:05:02.68697 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_A_CED, state 1
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Starting answer mode
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_A_CED to T30_PHASE_B_TX
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 1 to 17
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  CSI without final frame tag
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30 DIS:
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Can receive fax: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= 2-D coding: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Polled sub-address: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= T.43 coding: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Plane interleave: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Selective polling: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Sub-addressing: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Password: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Character mode: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Digital network capability: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Duplex capability: Half only
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= JPEG coding: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Full colour mode: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Custom illuminant: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Custom gamut range: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DIS with final frame tag
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2009-08-28 11:05:04.508695 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:04.588710 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:04.588710 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2009-08-28 11:05:04.588710 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 4
2009-08-28 11:05:04.588710 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:05:04.588710 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4
2009-08-28 11:05:04.608711 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:04.628710 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:04.648711 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:04.648711 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:04.668726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:04.688724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:06.748695 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:07.248745 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW T.30 Retry number 1
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  CSI without final frame tag
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30 DIS:
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Can receive fax: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= 2-D coding: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Polled sub-address: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= T.43 coding: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Plane interleave: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Selective polling: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Sub-addressing: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Password: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Character mode: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Digital network capability: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Duplex capability: Half only
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= JPEG coding: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Full colour mode: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Custom illuminant: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Custom gamut range: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DIS with final frame tag
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2009-08-28 11:05:10.108472 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:10.188469 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:10.188469 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2009-08-28 11:05:10.188469 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 4
2009-08-28 11:05:10.188469 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:05:10.188469 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4
2009-08-28 11:05:10.208465 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:10.228473 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:10.228473 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:10.248465 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:10.268468 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:10.268468 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:10.288466 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:10.288466 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:10.468464 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:10.688461 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 17
2009-08-28 11:05:10.688461 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4A
2009-08-28 11:05:10.788458 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:05:10.788458 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 7
2009-08-28 11:05:10.888470 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:05:10.888470 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 7
2009-08-28 11:05:12.28461 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:05:12.28461 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 7
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T4A (11840 remaining)
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  DCS with final frame tag
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 In state 17
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 DCS:
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Receive fax: Set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 00..= Selected data signalling rate: V.17 14400bps
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= 2-D coding: Not set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..00= Recording width: 215mm +- 1%
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Minimum scan line time: 0ms
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Frame size: 256 octets
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 Selected compression 3
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 Get document at 14400bps, modem 7
2009-08-28 11:05:12.320684 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 17 to 7
2009-08-28 11:05:12.320684 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-08-28 11:05:12.368695 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:05:12.368695 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_C_NON_ECM_RX
2009-08-28 11:05:12.368695 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:12.368695 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 7
2009-08-28 11:05:12.368695 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:05:12.428696 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:05:12.428696 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:05:12.588696 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:05:12.628696 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2009-08-28 11:05:12.628696 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:05:12.628696 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:05:12.628696 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:05:12.748696 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training in progress (-3) in state 7
2009-08-28 11:05:13.268481 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:05:14.8720 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:05:14.328710 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:05:14.648695 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:05:15.568732 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2009-08-28 11:05:15.568732 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:05:18.608475 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:05:18.608475 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:05:18.748726 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:05:19.8726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 7
2009-08-28 11:05:19.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2A
2009-08-28 11:05:20.168726 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:05:20.168726 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 3
2009-08-28 11:05:20.168726 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T2A (14880 remaining)
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Stop none (0 remaining)
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  DCS with final frame tag
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 In state 7
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected DCS received in state 7
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 7 to 3
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DCN with final frame tag
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 fa
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW FAX Switching from V.17 + V.21 to V.21 (-14.62dBm0)
2009-08-28 11:05:20.488724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 3
2009-08-28 11:05:20.488724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_C_NON_ECM_RX to T30_PHASE_D_TX
2009-08-28 11:05:20.488724 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:20.488724 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:05:21.548725 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2009-08-28 11:05:21.628731 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2009-08-28 11:05:21.628731 [DEBUG] mod_fax.c:137 FLOW T.30 Disconnecting
2009-08-28 11:05:21.628731 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E
2009-08-28 11:05:21.628731 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:21.628731 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 1
2009-08-28 11:05:21.628731 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 3 to 2
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_E, state 2
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:167 ==============================================================================
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:180 Fax processing not successful - result (13) Unexpected message received.
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:185 Remote station id: 
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:186 Local station id:  SpanDSP Fax Ident
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:187 Pages transferred: 0
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:189 Total fax pages:   0
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:190 Image resolution:  8031x7700
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:191 Transfer Rate:     14400
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:193 ECM status         on
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:194 remote country:   
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:195 remote vendor:    
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:196 remote model:     
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:198 ==============================================================================
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 2 to 32
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 8
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 8
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete
2009-08-28 11:05:22.648728 [DEBUG] switch_core_codec.c:122 Restore original codec.
EXECUTE sofia/external/7629 at 85.0.30.241 hangup()
2009-08-28 11:05:22.648728 [NOTICE] mod_dptools.c:633 Hangup sofia/external/7629 at 85.0.30.241 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-28 11:05:22.648728 [DEBUG] switch_channel.c:1683 Send signal sofia/external/7629 at 85.0.30.241 [KILL]
2009-08-28 11:05:22.648728 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:491 (sofia/external/7629 at 85.0.30.241) State EXECUTE going to sleep
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_HANGUP
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:434 (sofia/external/7629 at 85.0.30.241) State HANGUP
2009-08-28 11:05:22.648728 [DEBUG] mod_sofia.c:338 Channel sofia/external/7629 at 85.0.30.241 hanging up, cause: NORMAL_CLEARING
2009-08-28 11:05:22.648728 [DEBUG] mod_sofia.c:396 Sending BYE to sofia/external/7629 at 85.0.30.241
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:46 sofia/external/7629 at 85.0.30.241 Standard HANGUP, cause: NORMAL_CLEARING
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:434 (sofia/external/7629 at 85.0.30.241) State HANGUP going to sleep
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:476 (sofia/external/7629 at 85.0.30.241) State Change CS_HANGUP -> CS_REPORTING
2009-08-28 11:05:22.648728 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_REPORTING
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:612 (sofia/external/7629 at 85.0.30.241) State REPORTING
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:53 sofia/external/7629 at 85.0.30.241 Standard REPORTING, cause: NORMAL_CLEARING
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:612 (sofia/external/7629 at 85.0.30.241) State REPORTING going to sleep
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:411 (sofia/external/7629 at 85.0.30.241) State Change CS_REPORTING -> CS_DESTROY
2009-08-28 11:05:22.648728 [DEBUG] switch_core_session.c:1068 Session 25 (sofia/external/7629 at 85.0.30.241) Locked, Waiting on external entities
2009-08-28 11:05:22.648728 [NOTICE] switch_core_session.c:1086 Session 25 (sofia/external/7629 at 85.0.30.241) Ended
2009-08-28 11:05:22.648728 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/7629 at 85.0.30.241 [CS_DESTROY]
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:564 (sofia/external/7629 at 85.0.30.241) State DESTROY
2009-08-28 11:05:22.648728 [DEBUG] mod_sofia.c:255 sofia/external/7629 at 85.0.30.241 SOFIA DESTROY
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:60 sofia/external/7629 at 85.0.30.241 Standard DESTROY
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:564 (sofia/external/7629 at 85.0.30.241) State DESTROY going to sleep
2009-08-28 11:05:48.912723 [NOTICE] switch_channel.c:602 New Channel sofia/external/7629 at 85.0.30.241 [fa83c460-93b1-11de-ae8e-27b1291d9722]
2009-08-28 11:05:48.912723 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_NEW
2009-08-28 11:05:48.912723 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [received][100]
2009-08-28 11:05:48.912723 [DEBUG] sofia.c:3296 Remote SDP:
v=0
o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 85.0.30.241
s=SIP Call
c=IN IP4 85.0.30.240
t=0 0
m=audio 28094 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2009-08-28 11:05:48.912723 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2009-08-28 11:05:48.912723 [DEBUG] sofia_glue.c:2029 Set Codec sofia/external/7629 at 85.0.30.241 PCMU/8000 20 ms 160 samples
2009-08-28 11:05:48.912723 [DEBUG] switch_core_state_machine.c:404 (sofia/external/7629 at 85.0.30.241) State NEW
2009-08-28 11:05:48.912723 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 101
2009-08-28 11:05:48.912723 [DEBUG] sofia.c:3455 (sofia/external/7629 at 85.0.30.241) State Change CS_NEW -> CS_INIT
2009-08-28 11:05:48.912723 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_INIT
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:481 (sofia/external/7629 at 85.0.30.241) State INIT
2009-08-28 11:05:48.916860 [DEBUG] mod_sofia.c:83 sofia/external/7629 at 85.0.30.241 SOFIA INIT
2009-08-28 11:05:48.916860 [DEBUG] mod_sofia.c:111 (sofia/external/7629 at 85.0.30.241) State Change CS_INIT -> CS_ROUTING
2009-08-28 11:05:48.916860 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:481 (sofia/external/7629 at 85.0.30.241) State INIT going to sleep
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_ROUTING
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:484 (sofia/external/7629 at 85.0.30.241) State ROUTING
2009-08-28 11:05:48.916860 [DEBUG] mod_sofia.c:130 sofia/external/7629 at 85.0.30.241 SOFIA ROUTING
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:78 sofia/external/7629 at 85.0.30.241 Standard ROUTING
2009-08-28 11:05:48.916860 [INFO] mod_dialplan_xml.c:315 Processing FAX D.I.C. 7629->9998 in context public
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->unloop] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->outside_call] continue=true
Dialplan: sofia/external/7629 at 85.0.30.241 Absolute Condition [outside_call]
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(outside_call=true)
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->call_debug] continue=true
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->public_extensions] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [public_extensions] destination_number(9998) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->vmain] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [vmain] destination_number(9998) =~ /^(9999)$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->fax2mail] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (PASS) [fax2mail] destination_number(9998) =~ /^(9998)$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_verbose=true)
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_disable_v17=false)
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_use_ecm=true)
Dialplan: sofia/external/7629 at 85.0.30.241 Action answer()
Dialplan: sofia/external/7629 at 85.0.30.241 Action playback(silence_stream://2000)
Dialplan: sofia/external/7629 at 85.0.30.241 Action rxfax(/tmp/${uuid}.rxfax.tiff)
Dialplan: sofia/external/7629 at 85.0.30.241 Action hangup()
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:114 (sofia/external/7629 at 85.0.30.241) State Change CS_ROUTING -> CS_EXECUTE
2009-08-28 11:05:48.916860 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:484 (sofia/external/7629 at 85.0.30.241) State ROUTING going to sleep
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_EXECUTE
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:491 (sofia/external/7629 at 85.0.30.241) State EXECUTE
2009-08-28 11:05:48.916860 [DEBUG] mod_sofia.c:173 sofia/external/7629 at 85.0.30.241 SOFIA EXECUTE
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:151 sofia/external/7629 at 85.0.30.241 Standard EXECUTE
EXECUTE sofia/external/7629 at 85.0.30.241 set(outside_call=true)
2009-08-28 11:05:48.916860 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [outside_call]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_verbose=true)
2009-08-28 11:05:48.916860 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_verbose]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_disable_v17=false)
2009-08-28 11:05:48.916860 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_disable_v17]=[false]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_use_ecm=true)
2009-08-28 11:05:48.916860 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_use_ecm]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 answer()
2009-08-28 11:05:48.916860 [DEBUG] mod_dptools.c:649 sofia/external/7629 at 85.0.30.241 receive message [ANSWER]
2009-08-28 11:05:48.916860 [DEBUG] sofia_glue.c:2263 AUDIO RTP [sofia/external/7629 at 85.0.30.241] 10.30.15.137 port 24154 -> 85.0.30.240 port 28094 codec: 0 ms: 20
2009-08-28 11:05:48.916860 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms
2009-08-28 11:05:48.920709 [DEBUG] mod_sofia.c:552 Local SDP sofia/external/7629 at 85.0.30.241:
v=0
o=FreeSWITCH 1251426194 1251426195 IN IP4 10.30.15.137
s=FreeSWITCH
c=IN IP4 10.30.15.137
t=0 0
m=audio 24154 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-08-28 11:05:48.920709 [DEBUG] switch_core_session.c:630 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:05:48.920709 [NOTICE] mod_dptools.c:649 Channel [sofia/external/7629 at 85.0.30.241] has been answered
2009-08-28 11:05:48.920709 [DEBUG] switch_channel.c:182 sofia/external/7629 at 85.0.30.241 receive message [AUDIO_SYNC]
2009-08-28 11:05:48.920709 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [completed][200]
EXECUTE sofia/external/7629 at 85.0.30.241 playback(silence_stream://2000)
2009-08-28 11:05:48.920709 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 8000hz 1 channels 20ms
2009-08-28 11:05:48.920709 [DEBUG] switch_core_io.c:649 sofia/external/7629 at 85.0.30.241 receive message [TRANSCODING_NECESSARY]
2009-08-28 11:05:48.928705 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [ready][200]
2009-08-28 11:05:50.928734 [DEBUG] switch_ivr_play_say.c:1391 done playing file
EXECUTE sofia/external/7629 at 85.0.30.241 rxfax(/tmp/fa83c460-93b1-11de-ae8e-27b1291d9722.rxfax.tiff)
2009-08-28 11:05:50.928734 [DEBUG] mod_fax.c:591 Raw read codec activation Success L16 20000
2009-08-28 11:05:50.928734 [DEBUG] mod_fax.c:607 Raw write codec activation Success L16
2009-08-28 11:05:50.928734 [DEBUG] switch_channel.c:182 sofia/external/7629 at 85.0.30.241 receive message [AUDIO_SYNC]
2009-08-28 11:05:50.968739 [DEBUG] switch_core_io.c:232 sofia/external/7629 at 85.0.30.241 receive message [TRANSCODING_NECESSARY]
2009-08-28 11:05:51.548728 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2009-08-28 11:05:51.548728 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1
2009-08-28 11:05:53.128723 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2009-08-28 11:05:53.628727 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_A_CED, state 1
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Starting answer mode
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_A_CED to T30_PHASE_B_TX
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 1 to 17
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  CSI without final frame tag
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30 DIS:
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Can receive fax: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= 2-D coding: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Polled sub-address: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= T.43 coding: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Plane interleave: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Selective polling: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Sub-addressing: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Password: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Character mode: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Digital network capability: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Duplex capability: Half only
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= JPEG coding: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Full colour mode: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Custom illuminant: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Custom gamut range: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DIS with final frame tag
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2009-08-28 11:05:56.48725 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:56.128732 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:56.128732 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2009-08-28 11:05:56.128732 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 4
2009-08-28 11:05:56.128732 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:05:56.128732 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4
2009-08-28 11:05:56.148725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:56.148725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:56.148725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:56.168727 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:56.168727 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:56.208724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:56.228725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:56.228725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:56.248726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:56.268724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:56.448723 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:56.668724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 17
2009-08-28 11:05:56.668724 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4A
2009-08-28 11:05:56.748724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 17
2009-08-28 11:05:56.928724 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:05:56.928724 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 7
2009-08-28 11:05:58.8714 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:05:58.8714 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 7
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T4A (11840 remaining)
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  DCS with final frame tag
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 In state 17
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 DCS:
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Receive fax: Set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 00..= Selected data signalling rate: V.17 14400bps
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= 2-D coding: Not set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..00= Recording width: 215mm +- 1%
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Minimum scan line time: 0ms
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Frame size: 256 octets
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 Selected compression 3
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 Get document at 14400bps, modem 7
2009-08-28 11:05:58.300704 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 17 to 7
2009-08-28 11:05:58.300704 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-08-28 11:05:58.348715 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:05:58.348715 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_C_NON_ECM_RX
2009-08-28 11:05:58.348715 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:58.348715 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 7
2009-08-28 11:05:58.348715 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:05:58.408715 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:05:58.408715 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:05:58.568726 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:05:58.608717 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2009-08-28 11:05:58.608717 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:05:58.608717 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:05:58.608717 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:05:58.748714 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training in progress (-3) in state 7
2009-08-28 11:05:59.248723 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:05:59.648725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:06:00.48734 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:06:00.308724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:06:01.548728 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2009-08-28 11:06:01.548728 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:06:04.588723 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:06:04.588723 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:06:04.728724 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:06:04.828724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 7
2009-08-28 11:06:04.828724 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2A
2009-08-28 11:06:06.148723 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:06:06.148723 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 3
2009-08-28 11:06:06.148723 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T2A (13760 remaining)
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Stop none (0 remaining)
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  DCS with final frame tag
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 In state 7
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected DCS received in state 7
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 7 to 3
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DCN with final frame tag
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 fa
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW FAX Switching from V.17 + V.21 to V.21 (-14.51dBm0)
2009-08-28 11:06:06.468724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 3
2009-08-28 11:06:06.468724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_C_NON_ECM_RX to T30_PHASE_D_TX
2009-08-28 11:06:06.468724 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:06:06.468724 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:06:07.548713 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2009-08-28 11:06:07.628704 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2009-08-28 11:06:07.628704 [DEBUG] mod_fax.c:137 FLOW T.30 Disconnecting
2009-08-28 11:06:07.628704 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E
2009-08-28 11:06:07.628704 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:06:07.628704 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 1
2009-08-28 11:06:07.628704 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 3 to 2
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_E, state 2
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:167 ==============================================================================
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:180 Fax processing not successful - result (13) Unexpected message received.
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:185 Remote station id: 
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:186 Local station id:  SpanDSP Fax Ident
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:187 Pages transferred: 0
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:189 Total fax pages:   0
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:190 Image resolution:  8031x7700
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:191 Transfer Rate:     14400
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:193 ECM status         on
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:194 remote country:   
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:195 remote vendor:    
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:196 remote model:     
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:198 ==============================================================================
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 2 to 32
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 8
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 8
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete
2009-08-28 11:06:08.648545 [DEBUG] switch_core_codec.c:122 Restore original codec.
EXECUTE sofia/external/7629 at 85.0.30.241 hangup()
2009-08-28 11:06:08.648545 [NOTICE] mod_dptools.c:633 Hangup sofia/external/7629 at 85.0.30.241 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-28 11:06:08.648545 [DEBUG] switch_channel.c:1683 Send signal sofia/external/7629 at 85.0.30.241 [KILL]
2009-08-28 11:06:08.648545 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:06:08.648545 [DEBUG] switch_core_state_machine.c:491 (sofia/external/7629 at 85.0.30.241) State EXECUTE going to sleep
2009-08-28 11:06:08.648545 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_HANGUP
2009-08-28 11:06:08.648545 [DEBUG] switch_core_state_machine.c:434 (sofia/external/7629 at 85.0.30.241) State HANGUP
2009-08-28 11:06:08.648545 [DEBUG] mod_sofia.c:338 Channel sofia/external/7629 at 85.0.30.241 hanging up, cause: NORMAL_CLEARING
2009-08-28 11:06:08.648545 [DEBUG] mod_sofia.c:396 Sending BYE to sofia/external/7629 at 85.0.30.241
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:46 sofia/external/7629 at 85.0.30.241 Standard HANGUP, cause: NORMAL_CLEARING
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:434 (sofia/external/7629 at 85.0.30.241) State HANGUP going to sleep
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:476 (sofia/external/7629 at 85.0.30.241) State Change CS_HANGUP -> CS_REPORTING
2009-08-28 11:06:08.652458 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_REPORTING
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:612 (sofia/external/7629 at 85.0.30.241) State REPORTING
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:53 sofia/external/7629 at 85.0.30.241 Standard REPORTING, cause: NORMAL_CLEARING
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:612 (sofia/external/7629 at 85.0.30.241) State REPORTING going to sleep
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:411 (sofia/external/7629 at 85.0.30.241) State Change CS_REPORTING -> CS_DESTROY
2009-08-28 11:06:08.652458 [DEBUG] switch_core_session.c:1068 Session 26 (sofia/external/7629 at 85.0.30.241) Locked, Waiting on external entities
2009-08-28 11:06:08.652458 [NOTICE] switch_core_session.c:1086 Session 26 (sofia/external/7629 at 85.0.30.241) Ended
2009-08-28 11:06:08.652458 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/7629 at 85.0.30.241 [CS_DESTROY]
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:564 (sofia/external/7629 at 85.0.30.241) State DESTROY
2009-08-28 11:06:08.652458 [DEBUG] mod_sofia.c:255 sofia/external/7629 at 85.0.30.241 SOFIA DESTROY
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:60 sofia/external/7629 at 85.0.30.241 Standard DESTROY
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:564 (sofia/external/7629 at 85.0.30.241) State DESTROY going to sleep

From brian at freeswitch.org  Fri Aug 28 06:29:05 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 28 Aug 2009 08:29:05 -0500
Subject: [Freeswitch-users] Who is "callie"?
In-Reply-To: <4A96E8F1.8010106@nowthor.com>
References: <4A96AEA8.3030308@nowthor.com>	<87f2f3b90908270918u66439bd2wd046a4dc485caeb8@mail.gmail.com>
	<D206B5F9-2D91-4652-A944-09CAD425ABBD@jerris.com>
	<4A96E8F1.8010106@nowthor.com>
Message-ID: <EDADECDD-5644-4854-A8A2-E3DAF9A2DC19@freeswitch.org>

Yes you own the recordings.  Its work for hire.  Do you want to  
include the recordings in our sound repo?

/b

On Aug 27, 2009, at 3:13 PM, Carlos S. Antunes wrote:

> Mike,
>
> Sure! I am planning on doing a session "soon", maybe in a couple of  
> weeks or so. My only question is whether GM Voices license will  
> allow this kind of thing. Do you or anyone else know?
>
> Carlos



From anthony.minessale at gmail.com  Fri Aug 28 07:57:47 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 09:57:47 -0500
Subject: [Freeswitch-users] timeout while waiting for CF_BRIDGED flag
In-Reply-To: <4A97B8AD.3030009@gmail.com>
References: <4A97B8AD.3030009@gmail.com>
Message-ID: <191c3a030908280757n398077e8n9889ac58fde24b4@mail.gmail.com>

10 seconds is a pretty long time for the 2nd half of the bridge to start?
You can try increasing it but you probably have a more profound problem if
you are hitting that code.

When you start the bridge the code has 2 session pointers, it sets up state
handler on the b leg and changes state to
get it to execute the audio bridge thread then it executes the same thread
on the a leg so you end up with
2 sessions running audio_bridge_thread on each other.

Each leg thread uses that timeout you mentioned to make sure it's not
waiting forever for the bridge to start in case something
happens to the other one.

You should follow what each leg is doing and find out why the other one did
not make it into audio bridge thread.

It may be not answered or some other pre-conditon for thread. (are you
uuid_bridging 2 channels where neither one is answered yet?)


On Fri, Aug 28, 2009 at 5:59 AM, Tamas Cseke <cstomi.levlist at gmail.com>wrote:

> Hello,
>
> we have a strange problem, sometimes calls are hung up with
> DESTINATION_OUT_OF_ORDER cause.
> I added some extra log lines, and it turned out that
>
> switch_ivr_bridge.c:140 a timeout occurs while waiting for CF_BRIDGED flag
> switch_channel_wait_for_flag(chan_b, CF_BRIDGED, SWITCH_TRUE, 10000,
> chan_a); returns SWITCH_STATUS_TIMEOUT
>
> As far as I understand the flag on B channel is set in another thread,
> so maybe it happens slow sometimes.
> We are using r14144. I know it is quiet old, but maybe this code hasn't
> changed since them
>
> Do you have any idea, what could cause this?
> Do you think it is a bug?
>
> Thanks in advance,
> Tamas
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Fri Aug 28 08:49:32 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 08:49:32 -0700
Subject: [Freeswitch-users] freeswitch odbc problem
In-Reply-To: <27c25bc40908272351g345ad1e2w31dc945e8f9a0331@mail.gmail.com>
References: <27c25bc40908272351g345ad1e2w31dc945e8f9a0331@mail.gmail.com>
Message-ID: <87f2f3b90908280849w34bda66dt4d6ac3227b1fddf3@mail.gmail.com>

Juan,

Is this a new OS install? Make sure that you have all of the prerequisites
installed. Hopefully you're on CentOS and can just do this:

yum -y install unixODBC-devel gcc-c++ compat-gcc-32 compat-gcc-32-c++
autoconf automake db4-devel gdbm-devel ncurses-devel wget

Let us know how it goes.
-MC

On Thu, Aug 27, 2009 at 11:51 PM, Juan Backson <juanbackson at gmail.com>wrote:

> Hi,
> In a new environment, I am getting the following error when building the
> latest freeswitch from svn.  Does anyone know how to resolve it?
>
> gcc -I/usr/src/freeswitch-snapshot/src/include
> -I/usr/src/freeswitch-snapshot/libs/libteletone/src -fPIC -Werror
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
> -g -O2 -Wall -std=c99 -pedantic -o .libs/freeswitch freeswitch-switch.o  -lm
> ./.libs/libfreeswitch.so
> -L/usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib
> /usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib/.libs/libexpat.a
> /usr/src/freeswitch-snapshot/libs/apr/.libs/libapr-1.a
> -L/usr/src/freeswitch-snapshot/libs/srtp -L/usr/kerberos/lib64
> libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt -lpthread
> libs/libedit/src/.libs/libedit.a -lssl -lcrypto -ldl -lz -lncurses
> -Wl,--rpath -Wl,/usr/local/freeswitch/lib
> ./.libs/libfreeswitch.so: undefined reference to `SQLGetData'
> ./.libs/libfreeswitch.so: undefined reference to `SQLDisconnect'
> ./.libs/libfreeswitch.so: undefined reference to `SQLSetConnectAttr'
> ./.libs/libfreeswitch.so: undefined reference to `SQLGetDiagRec'
> ./.libs/libfreeswitch.so: undefined reference to `SQLConnect'
> ./.libs/libfreeswitch.so: undefined reference to `SQLExecute'
> ./.libs/libfreeswitch.so: undefined reference to `SQLNumResultCols'
> ./.libs/libfreeswitch.so: undefined reference to `SQLFreeHandle'
> ./.libs/libfreeswitch.so: undefined reference to `SQLGetInfo'
> ./.libs/libfreeswitch.so: undefined reference to `SQLAllocHandle'
> ./.libs/libfreeswitch.so: undefined reference to `SQLDriverConnect'
> ./.libs/libfreeswitch.so: undefined reference to `SQLPrepare'
> ./.libs/libfreeswitch.so: undefined reference to `SQLError'
> ./.libs/libfreeswitch.so: undefined reference to `SQLRowCount'
> ./.libs/libfreeswitch.so: undefined reference to `SQLDescribeCol'
> ./.libs/libfreeswitch.so: undefined reference to `SQLFetch'
> ./.libs/libfreeswitch.so: undefined reference to `SQLSetEnvAttr'
> collect2: ld returned 1 exit status
> make[2]: *** [freeswitch] Error 1
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Aug 28 09:07:06 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 09:07:06 -0700
Subject: [Freeswitch-users] How can it be done with freeswitch?
In-Reply-To: <25184845.post@talk.nabble.com>
References: <25184845.post@talk.nabble.com>
Message-ID: <87f2f3b90908280907w4812a8dbr7a97385f9afdd376@mail.gmail.com>

Nagalenoj,

This is definitely possible to do with FreeSWITCH. However, it is not
exactly simple. You have some learning to do. My recommendation is to learn
about event socket. Most likely you'll need to use outbound event socket.
(Read about event socket and ESL, the event socket library, on the wiki.)

I also recommend that you look at some of the example scripts in
src/libs/esl/perl/ which show some of the ways to handle calls with a
script. You've already done a good job of breaking down the steps, now
experiment with handling each of these steps with a script. We don't have
extensive Perl script examples which means we would love to see your code
when you have it all figured out! :)

-MC

On Thu, Aug 27, 2009 at 11:44 PM, Nagalenoj <nagalenoj at gmail.com> wrote:

>
> Dear friends,
>    I am using freeswitch for the last one month and trying to write perl
> scripts to do some tasks.
> I am trying a new thing, which is something that I don't how to do it and I
> don't whether it can be done with freeswitch.   my requirement is as
> follows,
>             A calls B[This extension has to execute a script].
>             The script has to do the following,
>              * Cut the call and callback to the number.
>              * Get a phone number[DTMF] from A.
>              * Dial the number(C) and bridge A and C.
>              * When C cuts the call, the control has to comeback to script.
>              * Ask another number to A to dial out, get another phone
> number[DTMF] from A.
>              * Dial the number(D) and bridge A and D.
>
> Please, help me to do this. Is there any other documentation to read about
> the perl freeswitch functions.
> --
> View this message in context:
> http://www.nabble.com/How-can-it-be-done-with-freeswitch--tp25184845p25184845.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Aug 28 09:10:22 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 09:10:22 -0700
Subject: [Freeswitch-users] mod_fax not working
In-Reply-To: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com>
References: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com>
Message-ID: <87f2f3b90908280910i3f0566f3tcae8fd7e8cff5d9d@mail.gmail.com>

On Fri, Aug 28, 2009 at 2:15 AM, Mathieu Parent <math.parent at gmail.com>wrote:

> Hi,
>
> I just started testing Freeswitch, this works well. This is powerfull
> and easy to administrate.
>
> I just have one problem and one request. The request is BOUNTY-5: IMAP
> integration of voicemail (I will try to propose a patch if this is
> within my competencies)
>
> The problem is with mod_fax not receiving any fax.
>
> Relevant extension is:
>    <extension name="fax2mail">
>        <condition field="destination_number" expression="^(9998)$">
>            <action application="set" data="fax_verbose=true"/>
>            <action application="set" data="fax_disable_v17=false"/>
>            <action application="set" data="fax_use_ecm=true"/>
>            <action application="answer" />
>            <action application="playback" data="silence_stream://2000"/>
>            <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/>
>            <action application="hangup"/>
>        </condition>
>    </extension>
>
> Test case is a 1-page fax coming from a Minolta printer connected to a
> Cisco VG248 (via RJ11). The VG248 port is registred on a Cisco Call
> Manager communicating with FS via a SIP trunk.
>
> Log is attached:
> - first try from 11:04:05 to 11:04:31
> - 2nd try from 11:04:57 to 11:05:22
> - 3rd try from 11:05:48 to 11:06:08
>
> Do you need more information?
>

Put your FS console in debug mode (press F6 or type "console loglevel 7")
and then make another test fax call, capturing the output. Look for obvious
clues. NOTE: The debug logs are extensive so don't fret if you feel like
you're buried in information. Pastebin your log output (
pastebin.freeswitch.org) and put the link in this email thread. The experts
can then help you figure it out from there...
-MC
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From msc at freeswitch.org  Fri Aug 28 09:11:05 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 09:11:05 -0700
Subject: [Freeswitch-users] mod_fax not working
In-Reply-To: <87f2f3b90908280910i3f0566f3tcae8fd7e8cff5d9d@mail.gmail.com>
References: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com>
	<87f2f3b90908280910i3f0566f3tcae8fd7e8cff5d9d@mail.gmail.com>
Message-ID: <87f2f3b90908280911k3d7ab400l7858f69e0bd70a4a@mail.gmail.com>

Doh! I just noticed you already attached the log. My bad. Let the experts
check it out.
-MC

On Fri, Aug 28, 2009 at 9:10 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Aug 28, 2009 at 2:15 AM, Mathieu Parent <math.parent at gmail.com>wrote:
>
>> Hi,
>>
>> I just started testing Freeswitch, this works well. This is powerfull
>> and easy to administrate.
>>
>> I just have one problem and one request. The request is BOUNTY-5: IMAP
>> integration of voicemail (I will try to propose a patch if this is
>> within my competencies)
>>
>> The problem is with mod_fax not receiving any fax.
>>
>> Relevant extension is:
>>    <extension name="fax2mail">
>>        <condition field="destination_number" expression="^(9998)$">
>>            <action application="set" data="fax_verbose=true"/>
>>            <action application="set" data="fax_disable_v17=false"/>
>>            <action application="set" data="fax_use_ecm=true"/>
>>            <action application="answer" />
>>            <action application="playback" data="silence_stream://2000"/>
>>            <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/>
>>            <action application="hangup"/>
>>        </condition>
>>    </extension>
>>
>> Test case is a 1-page fax coming from a Minolta printer connected to a
>> Cisco VG248 (via RJ11). The VG248 port is registred on a Cisco Call
>> Manager communicating with FS via a SIP trunk.
>>
>> Log is attached:
>> - first try from 11:04:05 to 11:04:31
>> - 2nd try from 11:04:57 to 11:05:22
>> - 3rd try from 11:05:48 to 11:06:08
>>
>> Do you need more information?
>>
>
> Put your FS console in debug mode (press F6 or type "console loglevel 7")
> and then make another test fax call, capturing the output. Look for obvious
> clues. NOTE: The debug logs are extensive so don't fret if you feel like
> you're buried in information. Pastebin your log output (
> pastebin.freeswitch.org) and put the link in this email thread. The
> experts can then help you figure it out from there...
> -MC
>
>
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From msc at freeswitch.org  Fri Aug 28 09:18:17 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 09:18:17 -0700
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <1cd828b60908280413v6e5726cck72a5dcaa94ce3705@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
	<1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
	<191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>
	<1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>
	<87f2f3b90908271130v4bb7deads45d5f1c190ea0661@mail.gmail.com>
	<1cd828b60908280413v6e5726cck72a5dcaa94ce3705@mail.gmail.com>
Message-ID: <87f2f3b90908280918w77653b85jd7d4823635e1b5a4@mail.gmail.com>

Thanks! I made very minor changes. We appreciate you helping out with
documenting new features.
-MC

On Fri, Aug 28, 2009 at 4:13 AM, Anatoliy Kounitskiy <
anatoliy at kounitskiy.com> wrote:

> The wiki page is updated (
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer ). Hope
> it is understandable :)
>
> Anatoliy Kounitskiy
>
> On Thu, Aug 27, 2009 at 9:30 PM, Michael Collins<msc at freeswitch.org>
> wrote:
> > Thanks! I'll be in touch to help you if you need any wiki editing
> > assistance.
> > -MC
> >
> > On Thu, Aug 27, 2009 at 11:18 AM, Anatoliy Kounitskiy
> > <anatoliy at kounitskiy.com> wrote:
> >>
> >> Thank you all for the great job :)
> >> Now it works as I wanted!!
> >>
> >> Tomorrow I'll  will try to update the wiki :)
> >>
> >> But with few words ( I tried to strip the extensions to minimum for
> >> more easy read for users).
> >> In the dialplan I have created the extension showed below:
> >>
> >>  <extension name="local_number">
> >>    <condition field="destination_number" expression="^(\d{3})$">
> >>      <action application="set" data="dialed_extension=$1"/>
> >>      <action application="export" data="dialed_extension=$1"/>
> >>      <action application="bind_meta_app" data="1 b s
> >> execute_extension::attented_xfer XML features"/>
> >>      <action application="set" data="transfer_ringback=$${hold_music}"/>
> >>      <action application="set" data="call_timeout=10"/>
> >>      <action application="set" data="hangup_after_bridge=true"/>
> >>      <action application="bridge"
> >> data="user/${dialed_extension}@${domain_name}"/>
> >>    </condition>
> >>  </extension>
> >>
> >> In the features.xml file I have extension attented_xfer, that is using
> >> the att_xfer application :
> >>
> >>  <extension name="attented_xfer">
> >>      <condition field="destination_number" expression="^attented_xfer$">
> >>        <action application="set" data="continue_on_fail=true"/>
> >>        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
> >> attxfer_callthis 30000 #"/>
> >>        <action application="set" data="origination_cancel_key=#"/>
> >>        <action application="att_xfer"
> >> data="user/${attxfer_callthis}@${domain_name}"/>
> >>      </condition>
> >>    </extension>
> >>
> >> Also I have set the the origination cancel key will be # - so if B
> >> wants to cancel the call to C and to go back to A - just uses the #
> >> key :)
> >>
> >> Again - thank you very much :) :)
> >>
> >> Anatoliy Kounitskiy
> >>
> >> On Thu, Aug 27, 2009 at 7:48 PM, Anthony
> >> Minessale<anthony.minessale at gmail.com> wrote:
> >> > please retest with latest SVN trunk
> >> >
> >> > On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy
> >> > <anatoliy at kounitskiy.com> wrote:
> >> >>
> >> >> Just for information - the idea :
> >> >> A---calls---> B ---att_xfer---> C    , B hangs C up and goes back to
> A
> >> >> ( I`m sorry C is not answering :) )
> >> >>
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:213-799-1400
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anatoliy Kounitskiy
> >> -------------------------
> >> E-mail: anatoliy at kounitskiy.com
> >> Mobile: +359898913540
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anatoliy Kounitskiy
> -------------------------
> E-mail: anatoliy at kounitskiy.com
> Mobile: +359898913540
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From foxb at abv.bg  Fri Aug 28 09:42:28 2009
From: foxb at abv.bg (Hristo Benev)
Date: Fri, 28 Aug 2009 19:42:28 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>

 Hello Rod,

I did the change.

Here is extract of console:
-----------------------------------------------------------------------------------------
variable_continue_on_fail: [true]
variable_sip_h_X-ROUTE: [LOOKUP]
variable_export_vars: [sip_h_X-ROUTE]
variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20]
variable_sip_redirect_contact_0: [sip:France at PEER_01]
variable_sip_redirected_to: [sip:France at PEER_01]
variable_sip_redirect_contact_user_0: [France] <-------------------------------------------
variable_sip_redirect_contact_host_0: [PEER_01]
variable_sip_redirect_dialstring_0: [sofia/internal/sip:France at PEER_01]
variable_sip_redirect_dialstring: [sofia/internal/sip:France at PEER_01]
variable_proto_specific_hangup_cause: [sip:503]
variable_sip_hangup_phrase: [DNS Error]
variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE]
variable_ROUTE_GW: [France]
variable_AREA: [France]
variable_current_application: [info]
------------------------------------------------------------------------------------------
I have different value it is actually the description field as shown here:

--------------------------------
 /opt/kamailio/sbin/kamctl cr show
cr carrier names
+----+---------+
| id | carrier |
+----+---------+
|  1 | default |
+----+---------+
cr domain names
+----+---------+
| id | domain  |
+----+---------+
|  1 | default |
+----+---------+
cr routes
+----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
| id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description |
+----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
|  1 |       1 |      1 | 1000        |     0 |    0 |    1 |     0 | PEER_01      |                |                | France      |
+----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
-----------------------------------------

And here is what I have in kamailio:
-------------------------------------------------------------------------------------
####### Routing Logic ########


# main request routing logic

route{

        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                exit;
        }

        t_check_trans();

        if ($rU==NULL) {
                # request with no Username in RURI
                sl_send_reply("484","Address Incomplete");
                exit;
        }

        # LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP
        if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){
                if(!cr_route("default", "default", "$rU", "$rU", "call_id","$avp(s:route_desc)")){
                     #xlog("ROUTING FAILED: no route found for $rU");
                     sl_send_reply("604", "Unable to route this call");
                     exit;
                } else {
                     xlog("LOOKUP FOUND: $rd $avp(s:route_desc)");
                     avp_pushto("$ru/username", "$avp(s:route_desc)");
                     sl_send_reply("302", "$rd");
                     exit;
                }
        }
}
----------------------------------------------------------------------------------

Another question...
In that part of FreeSwitch dialplan.xml

-----------------------------------
  <context name="ROUTING">

    <extension name="PEER_01">
      <condition field="${sip_h_X-ROUTE}" expression="PEER_01">
        <action application="set" data="hangup_after_bridge=true"/>

-----------------------------------

X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the regex is true.


As for thanks - for sure by default they are also for the developers of both apps.
I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning so it will be nice to have something working to start with.



 >-------- ?????????? ????? --------
 >??:  rod 
 >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >??: freeswitch-users at lists.freeswitch.org
 >????????? ??: ?????, 2009, ?????? 28 09:54:00 EEST

 >Hello,
 >
 >the trace seems good.
 >If you check the answer from Kamailio, you'll see that Kamailio answers 
 >with "302 PEER_01".
 >
 >As Michael Collins stated before, you can get the variable containing 
 >"PEER_01", then this variable is stored in a custom variable.
 >
 >In your dialplan, may you please add:
 >
 >, just before the transfer line, eg:
 >
 > 
 >         
 >         
 >         
 >         
 >         
 >         
 >	 
 >         
 >      
 >
 >
 >Using application Info, you'll see on the console (or CDR) the list of 
 >variables used for this call.  You should see the content of 
 >"${sip_redirect_contact_user_0}" that should contain  the value 
 >"PEER_01". Please check this and let me know.
 >
 >For the new configuration file, no problem for sharing. But as I wrote 
 >on the wiki page, I worked on this setup cause LCR module was not 
 >available when I start working on FS, nor I'm a good programmer to write 
 >a server side HTTP script (used by xml_curl) that could scale to my needs.
 >
 >I enhanced a bit this configuration with support for fallback routing, 
 >almost realtime graph (every minutes using www.cacti.net and some 
 >functions in FS like limit_hash) of number of concurrent calls per AREA, 
 >PEER...
 >
 >Thanks for "the good tutorial", but don't forget the dev team who did 
 >this great product ;-)
 >
 >rod.
 >
 >
 >Hristo Benev a ?crit :
 >>  I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).
 >>
 >> If you need additional info I'll provide it.
 >>
 >> Here is trace:
 >>
 >> ngrep -d any -nn -i '1000' port 5062 -W byline
 >> interface: any
 >> filter: (ip or ip6) and ( port 5062 )
 >> match: 1000
 >> #
 >> U 10.10.10.10:5090 -> 127.0.0.1:5062
 >> INVITE sip:1000 at 127.0.0.1:5062 SIP/2.0.
 >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
 >> Max-Forwards: 69.
 >> From: "Extension 1001" ;tag=978g69jZaFpBD.
 >> To: .
 >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
 >> CSeq: 119574771 INVITE.
 >> Contact: .
 >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
 >> Supported: timer, precondition, path, replaces.
 >> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
 >> Content-Type: application/sdp.
 >> Content-Disposition: session.
 >> Content-Length: 429.
 >> X-ROUTE: LOOKUP.
 >> Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off.
 >> .
 >> v=0.
 >> o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
 >> s=FreeSWITCH.
 >> c=IN IP4 10.10.10.10.
 >> t=0 0.
 >> m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
 >> a=rtpmap:0 PCMU/8000.
 >> a=rtpmap:115 G7221/32000.
 >> a=fmtp:115 bitrate=48000.
 >> a=rtpmap:107 G7221/16000.
 >> a=fmtp:107 bitrate=32000.
 >> a=rtpmap:9 G722/8000.
 >> a=rtpmap:8 PCMA/8000.
 >> a=rtpmap:3 GSM/8000.
 >> a=rtpmap:101 telephone-event/8000.
 >> a=fmtp:101 0-16.
 >> a=rtpmap:13 CN/8000.
 >> a=ptime:20.
 >>
 >> #
 >> U 127.0.0.1:5062 -> 10.10.10.10:5090
 >> SIP/2.0 302 PEER_01.
 >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
 >> From: "Extension 1001" ;tag=978g69jZaFpBD.
 >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
 >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
 >> CSeq: 119574771 INVITE.
 >> Contact: sip:France at PEER_01.
 >> Server: Kamailio (1.5.2-notls (i386/linux)).
 >> Content-Length: 0.
 >> .
 >>
 >> #
 >> U 10.10.10.10:5090 -> 127.0.0.1:5062
 >> ACK sip:1000 at 127.0.0.1:5062 SIP/2.0.
 >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
 >> Max-Forwards: 69.
 >> From: "Extension 1001" ;tag=978g69jZaFpBD.
 >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
 >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
 >> CSeq: 119574771 ACK.
 >> Content-Length: 0.
 >> .
 >>
 >>
 >>
 >>
 >>
 >>  >-------- ?????????? ????? --------
 >>  >??:  Hristo Benev 
 >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >??: freeswitch-users at lists.freeswitch.org
 >>  >????????? ??: ?????????, 2009, ?????? 27 14:58:53 EEST
 >>
 >>  > 
 >>  >Bojnour,
 >>  >
 >>  >I'll send a trace ASAP.
 >>  >
 >>  >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
 >>  >
 >>  >I'm really interested to see the updates of the project.
 >>  >
 >>  >Thank you for the good tutorial.
 >>  >
 >>  >Hristo
 >>  >
 >>  >
 >>  > >-------- ?????????? ????? --------
 >>  > >??:  rod 
 >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  > >??: freeswitch-users at lists.freeswitch.org
 >>  > >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST
 >>  >
 >>  > >Hi Hristo,
 >>  > >
 >>  > >I'm the author of this setup and wiki page. I did a lot of modifications 
 >>  > >on this setup (alternative routing if failure essentially) but don't 
 >>  > >have too much time to update the wiki.
 >>  > >
 >>  > >May you please send me an ngrep trace when you call 1000:
 >>  > >
 >>  > >ngrep -d any -nn -i '1000' port 5060 -W byline 
 >>  > >
 >>  > >I will check what's happening.
 >>  > >Do you have an entry for 1000 in your mysql database ?
 >>  > >
 >>  > >regards,
 >>  > >rod
 >>  > >
 >>  > >Hristo Benev a ?crit :
 >>  > >>  
 >>  > >> It seems that the problem is on kamailio configuration.
 >>  > >> Will ask on their list.
 >>  > >>
 >>  > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
 >>  > >>
 >>  > >> I have it setup as friend in asterisk, but still ???
 >>  > >>
 >>  > >> Any ideas?
 >>  > >>
 >>  > >> Thanks,
 >>  > >>
 >>  > >>
 >>  > >>  >-------- ?????????? ????? --------
 >>  > >>  >??:  Hristo Benev 
 >>  > >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  > >>  >??: freeswitch-users at lists.freeswitch.org
 >>  > >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
 >>  > >>
 >>  > >>  > I think that the problem is here:
 >>  > >>  >-------------------------
 >>  > >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
 >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
 >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
 >>  > >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
 >>  > >>  >--------------------------
 >>  > >>  >
 >>  > >>  >Actually Regex FAIL
 >>  > >>  >
 >>  > >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
 >>  > >>  >Here is my  default.xml:
 >>  > >>  >----------------
 >>  > >>  >
 >>  > >>  >
 >>  > >>  >
 >>  > >>  >  
 >>  > >>  >
 >>  > >>  >     
 >>  > >>  >      
 >>  > >>  >         
 >>  > >>  >         
 >>  > >>  >         
 >>  > >>  >         
 >>  > >>  >         
 >>  > >>  >         
 >>  > >>  >         
 >>  > >>  >      
 >>  > >>  >    
 >>  > >>  >
 >>  > >>  >   
 >>  > >>  >
 >>  > >>  >  
 >>  > >>  >
 >>  > >>  >    
 >>  > >>  >      
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >      
 >>  > >>  >    
 >>  > >>  >
 >>  > >>  >  
 >>  > >>  >
 >>  > >>  >
 >>  > >>  >
 >>  > >>  >--------------------------
 >>  > >>  >
 >>  > >>  >
 >>  > >>  >
 >>  > >>  > >-------- ?????????? ????? --------
 >>  > >>  > >??:  Brian West 
 >>  > >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  > >>  > >??: freeswitch-users at lists.freeswitch.org
 >>  > >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
 >>  > >>  >
 >>  > >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
 >>  > >>  >You have to process all 302's in the dialplan. 
 >>  > >>  >Set this on your sofia profile
 >>  > >>  >You can set these variables sip_redirect_profile,
 >>  > >>  >sip_redirect_context,
 >>  > >>  >sip_redirect_dialplan,
 >>  > >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
 >>  > >>  >sip_redirected_to,
 >>  > >>  >sip_redirect_contact_user_%d,
 >>  > >>  >sip_redirect_contact_host_%d,
 >>  > >>  >sip_redirect_contact_params_%d,
 >>  > >>  >sip_redirect_dialstring_%d,
 >>  > >>  >sip_redirect_dialstring,
 >>  > >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
 >>  > >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
 >>  > >>  > >
 >>  > >>  >
 >>  > >>  >_______________________________________________
 >>  > >>  >FreeSWITCH-users mailing list
 >>  > >>  >FreeSWITCH-users at lists.freeswitch.org
 >>  > >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  > >>  >http://www.freeswitch.org
 >>  > >>  >
 >>  > >>
 >>  > >> _______________________________________________
 >>  > >> FreeSWITCH-users mailing list
 >>  > >> FreeSWITCH-users at lists.freeswitch.org
 >>  > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  > >> http://www.freeswitch.org
 >>  > >>   
 >>  > >
 >>  > >_______________________________________________
 >>  > >FreeSWITCH-users mailing list
 >>  > >FreeSWITCH-users at lists.freeswitch.org
 >>  > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  > >http://www.freeswitch.org
 >>  > >
 >>  >
 >>  >_______________________________________________
 >>  >FreeSWITCH-users mailing list
 >>  >FreeSWITCH-users at lists.freeswitch.org
 >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >http://www.freeswitch.org
 >>  >
 >>
 >> _______________________________________________
 >> FreeSWITCH-users mailing list
 >> FreeSWITCH-users at lists.freeswitch.org
 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >> http://www.freeswitch.org
 >>   
 >
 >_______________________________________________
 >FreeSWITCH-users mailing list
 >FreeSWITCH-users at lists.freeswitch.org
 >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >http://www.freeswitch.org
 >


From steveu at coppice.org  Fri Aug 28 10:01:12 2009
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 29 Aug 2009 01:01:12 +0800
Subject: [Freeswitch-users] mod_fax not working
In-Reply-To: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com>
References: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com>
Message-ID: <4A980D58.5020808@coppice.org>

Hi Mathieu,

On 08/28/2009 05:15 PM, Mathieu Parent wrote:
> Hi,
>
> I just started testing Freeswitch, this works well. This is powerfull
> and easy to administrate.
>
> I just have one problem and one request. The request is BOUNTY-5: IMAP
> integration of voicemail (I will try to propose a patch if this is
> within my competencies)
>
> The problem is with mod_fax not receiving any fax.
>
> Relevant extension is:
>      <extension name="fax2mail">
>          <condition field="destination_number" expression="^(9998)$">
>              <action application="set" data="fax_verbose=true"/>
>              <action application="set" data="fax_disable_v17=false"/>
>              <action application="set" data="fax_use_ecm=true"/>
>              <action application="answer" />
>              <action application="playback" data="silence_stream://2000"/>
>              <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/>
>              <action application="hangup"/>
>          </condition>
>      </extension>
>
> Test case is a 1-page fax coming from a Minolta printer connected to a
> Cisco VG248 (via RJ11). The VG248 port is registred on a Cisco Call
> Manager communicating with FS via a SIP trunk.
>
> Log is attached:
> - first try from 11:04:05 to 11:04:31
> - 2nd try from 11:04:57 to 11:05:22
> - 3rd try from 11:05:48 to 11:06:08
>
> Do you need more information?
>
> Regards
>
> Mathieu Parent
>    
The log shows the same thing happening every time. A bad CRC from the 
far end, followed by a good DCS frame followed by what seems to be 
rubbish. I think I'd need an audio log from one of these calls to figure 
out any more.

Steve



From msc at freeswitch.org  Fri Aug 28 10:24:27 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 10:24:27 -0700
Subject: [Freeswitch-users] newbie alert. Help with dialplan
In-Reply-To: <f9c510070908271116n33c3f35di44eea1790f62bfc8@mail.gmail.com>
References: <f9c510070908271116n33c3f35di44eea1790f62bfc8@mail.gmail.com>
Message-ID: <87f2f3b90908281024i1ecc6238o3bcab80e9464797e@mail.gmail.com>

Julian,

It might be better if you gave us the run-down on what your current Asterisk
extension does. Most likely there is an elegant way of doing it in
FreeSWITCH. Part of the unlearing of the Asterisk way is coming to grips
with the fact that the FS dialplan is not a programming language in and of
itself. While it's convenient to have GotoIf's all over the place, in the
long run that's inefficient.

Let's start from the top-down instead of the bottom-up. Discussion your
application in general terms, i.e., non-Asterisk-specific language. Tell us
what it does, not how it does it, and then we can think it through using FS
concepts.

-MC

On Thu, Aug 27, 2009 at 11:16 AM, Julian Lyndon-Smith <asterisk at dotr.com>wrote:

> I am a long-time asterisk user (2005), so I come from an unfortunate
> position of having to unlearn an awful lot of stuff in order to make
> freeswitch do the things I want ;(
>
> I *think* i've got my head around using mod_curl_xml (?) to read all
> the config stuff from my webserver. I *think* I've got my head around
> setting up the sip clients etc etc ...
>
> However, where I am really struggling is the dialplan. For the life of
> me I simply cannot seem to grasp the fs way - that's no disrespect to
> the fs way, but perhaps the failure of this old brain to change !
>
>  have pastebinned an example of show 1234546 at inboundq at
> http://www.pastebin.ca/1544890. If possible, would someone be able to
> show me how to convert this dialplan to the fs way ? If I could be
> given a little foot-up I'm sure that I would be able to convert the
> rest of the dialplan !
>
> Thanks in advance, and please go easy !
>
> Julian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Aug 28 10:43:35 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 28 Aug 2009 12:43:35 -0500
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
In-Reply-To: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>
References: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>
Message-ID: <86BDA956-125F-423E-931F-7FFC0F679C59@freeswitch.org>

Now you have to actually catch this in your dialplan on FreeSWITCH and  
execute the bridge application to the dialstring provided

/b

On Aug 28, 2009, at 11:42 AM, Hristo Benev wrote:

> variable_sip_redirect_dialstring_0: [sofia/internal/ 
> sip:France at PEER_01]
> variable_sip_redirect_dialstring: [sofia/internal/sip:France at PEER_01]



From paveraware at hotmail.com  Fri Aug 28 11:14:57 2009
From: paveraware at hotmail.com (Christensen Tom)
Date: Fri, 28 Aug 2009 18:14:57 +0000
Subject: [Freeswitch-users] Hello, and stuff.
Message-ID: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>


As a background, I ran an asterisk consulting company for about 3 years that I gave up on 2 years ago after repeatedly failing to achieve any sort of stability on any sort install over about 30 phones, I gave up.

Maybe that was wrong, I am open to the possibility that I just didn't know enough and I was building things wrong, but I worked inside the asterisk code (which I feel is a hopeless mess), I implemented a few small custom features, anyway... 

I'm coming back into the VoIP space now, and I'm wondering what sort of issues can I expect in trying to pick up and learn freeswitch?  From what I've read on the website, it appears to have a much more sane architecture.  I've used Cisco, Broadsoft, and asterisk in the past.  By far the least stable and worst general call quality was asterisk.  I constantly contended with strange call quality issues in asterisk, lots of echo (even with hardware echo cancellation cards), lots of jitter, lots of call break up (even on small systems with 10-20 users, using QoS on the network, and in general doing everything I could to prioritize voice over anything else).

When I used Cisco call manager and broadsoft, the voice quality issues were basically non-existant, as long as the network was running QoS echo, stutter, calls breaking up, just didn't happen.  So, I guess my question is, does freeswitch show a marked improvement over asterisk in this department?  As long as you configure QoS and have hardware echo cancellation does it actually work reliably?

Thanks for any additional information about freeswitch you can provide as well.  I am a software developer primarily by trade, but I do lots of consulting type work in the SME space and I've had a couple projects thrown to me that require some integration with a phone system, and I just can't in good conscience recommend asterisk anymore.


-Tom
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From msc at freeswitch.org  Fri Aug 28 11:30:06 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 11:30:06 -0700
Subject: [Freeswitch-users] Group Call
In-Reply-To: <5ff3e3a70908251015ue88b8b9p7a5e091d15a6b5ae@mail.gmail.com>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
	<5ff3e3a70908251015ue88b8b9p7a5e091d15a6b5ae@mail.gmail.com>
Message-ID: <87f2f3b90908281130g1e33668fs44b817f9c622641c@mail.gmail.com>

On Tue, Aug 25, 2009 at 10:15 AM, Shawn L. Djernes <sdjernes at gmail.com>wrote:

> Woops,
> That line was an attempt to try something from an older diaplan example.
>  Copied and pasted from a file on my computer not the server.  The actual
> line is:
>
> <action application="bridge"
> data="{ignore_early_media=true}${group_call(sdjernes+A@${domain_name})}"/>
>
> The +A is to ring all phones in the group at the same time.
>
>
Can you confirm what you see on the CLI when you do this:
group_call sdjernes+A at ewr.djernes.net <sdjernes%2BA at ewr.djernes.net>

Let's make sure the group is properly defined, etc. before doing anything
else.
-MC
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From rs at runsolutions.com  Fri Aug 28 11:52:25 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Fri, 28 Aug 2009 20:52:25 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Message-ID: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>

Hello List,

I have read the current thread about scalability and I would need some  
advice about a callcenter setup:

First where I come from:
I have lot's of problems with an asterisk solution. I have regular  
crash's and lock-ups, with downgrading and other stuff i got it  
somewhat stable, but have nevertheless regular hickups. I am desperate  
and want to get rid of asterisk and I hope that freeSwitch will  
provide me with a more stable solution.


Our Setup (really nothing special):
* 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3  
GIG of RAM, Xircom analog switch connected to mobile stations for 4  
different providers, Digium 4port cards TP400<something>
* 8 queues
* ~60 agents (which logon, logoff, pause, unpause), not more than 40  
concurrently online
* ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
* Music on Hold in the call-queues
* No special announcement
* Transfers between calls in queues and different agents as well as  
non agents (i mention this because we have transfer related chrashes  
in asterisk)

The current Problems:
* Lockups with different causes (ranging from calls not terminated to  
heavy thread locking through the AMI interface)
* Crashes and library aborts (pthread, libc, crashes related to music  
on hold, app_queue, transfers)

We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are  
now back to 1.4.21.2 (stock debian) as anything beyond that is for  
whatever reason highly unstable for our szenario. Maybe we should have  
been segmenting the box into one asterisk dedicted to talking to the  
hardware, one especial for queue/sip handling, i do not know. (all  
issues are well documented in issues.asterisk.org, but it seems to be  
very, very difficult to get to the bottom of them as they exist since  
1.4.23 as it seems and are open until know and not fixable since month.)


Now, I really would appreciate some success-stories on how you guys  
managed to get a stable pbx system with freeSWITCH in regard of HA and  
scalability:

* How to segment freeSWITCH? Or is it stable enough to handle all in  
one for such a szenario as outlined above?
* What would be the best strategie for High Availability / Failover?
	-> I read in the WIKI (featurelist) that Livemigration of calls from  
one box to another should be possible?
	-> I was thinking about using memcached for storing all state  
information so another freeswitch box can takeover calls from the  
first box if it dies, is this possible? If so, how?
	-> Is there anotherway to somehow configure freeSWITCH that in the  
event of a crash i do not loose the current established calls?

Basically I just want a stable PBX where I do not have to fear every  
day it will core-dump or abort or Lock up. Is freeSWITCH mature enough  
so i can sleep at night for at least 3 month without a crash?



Thank you for your Time and help in advance, and I am more than  
willing to take all the information gathered here and create a wiki  
page to help other people with the same questions/problems.

best
Ray


-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares



From msc at freeswitch.org  Fri Aug 28 11:54:25 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 11:54:25 -0700
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
Message-ID: <87f2f3b90908281154s2a1496b4l2e3aa06545649633@mail.gmail.com>

Tom,
Welcome! Sadly, your experience is not unique...

On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom <paveraware at hotmail.com>wrote:

>  As a background, I ran an asterisk consulting company for about 3 years
> that I gave up on 2 years ago after repeatedly failing to achieve any sort
> of stability on any sort install over about 30 phones, I gave up.
>

The consensus I've seen is that the larger the install, the more likely one
is to have inexplicable issues.

>
>
> Maybe that was wrong, I am open to the possibility that I just didn't know
> enough and I was building things wrong, but I worked inside the asterisk
> code (which I feel is a hopeless mess), I implemented a few small custom
> features, anyway...
>

Any software that openly admits that a function is "pure nastiness" but
doesn't change it from version 1.0, 1.2, 1.4, or 1.6 has questionable
leadership IMHO. (grep the Asterisk source tree for "nastiness" and you'll
see it.)

>
>
> I'm coming back into the VoIP space now, and I'm wondering what sort of
> issues can I expect in trying to pick up and learn freeswitch?  From what
> I've read on the website, it appears to have a much more sane architecture.
> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
> stable and worst general call quality was asterisk.  I constantly contended
> with strange call quality issues in asterisk, lots of echo (even with
> hardware echo cancellation cards), lots of jitter, lots of call break up
> (even on small systems with 10-20 users, using QoS on the network, and in
> general doing everything I could to prioritize voice over anything else).
>

Again, your experience isn't unique...

>
> When I used Cisco call manager and broadsoft, the voice quality issues were
> basically non-existant, as long as the network was running QoS echo,
> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
> does freeswitch show a marked improvement over asterisk in this department?
> As long as you configure QoS and have hardware echo cancellation does it
> actually work reliably?
>

We receive lots of reports that FreeSWITCH is a vast improvement over not
only Asterisk but proprietary solutions as well. The FS architecture is, as
you mentioned, not insane. It is well thought out and therefore highly
flexible, extensible, and scalable. I'm not aware of anything - OSS or
proprietary - that can match FS in these three areas.

>
> Thanks for any additional information about freeswitch you can provide as
> well.  I am a software developer primarily by trade, but I do lots of
> consulting type work in the SME space and I've had a couple projects thrown
> to me that require some integration with a phone system, and I just can't in
> good conscience recommend asterisk anymore.
>

Are you comfortable with the lack of a super slick GUI? :) Some GUIs are in
development but the power users are quite happy with doing the emacs (or
vim) shuffle with the XML config files. Furthermore, the ways that FS allows
you to connect and control are fantastic: mod_xml_curl for dynamic
configurations, event-socket for external control (think of it like AMI not
sucking and being turbo-charged), mod_xml_rpc for RPC goodness... Anyway,
the list is impressive.

I can honestly say that every week we get new people looking at FreeSWITCH
and saying, "Wow, this is incredible." I can definitely, in good conscience,
recommend you investigate FS more deeply. I'm confident you'll be happy with
the return on your investment.

Hope it all works out for you! Join us in #freeswitch on irc.freenode.net if
you want to chat in real-time.
-Michael
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From gmaruzz at celliax.org  Fri Aug 28 11:55:40 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 20:55:40 +0200
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
Message-ID: <7b197bef0908281155w92eff9dp1ffcb379ae4c8438@mail.gmail.com>

Hi Tom,

best is to try it for yourself, you cannot expect from the FS mailing
list an answer like: "you know, fs is nor a marked improvement on
anything, we just like to spend time together" :-)

-gm


On Fri, Aug 28, 2009 at 8:14 PM, Christensen Tom<paveraware at hotmail.com> wrote:
> As a background, I ran an asterisk consulting company for about 3 years that
> I gave up on 2 years ago after repeatedly failing to achieve any sort of
> stability on any sort install over about 30 phones, I gave up.
>
> Maybe that was wrong, I am open to the possibility that I just didn't know
> enough and I was building things wrong, but I worked inside the asterisk
> code (which I feel is a hopeless mess), I implemented a few small custom
> features, anyway...
>
> I'm coming back into the VoIP space now, and I'm wondering what sort of
> issues can I expect in trying to pick up and learn freeswitch?? From what
> I've read on the website, it appears to have a much more sane architecture.
> I've used Cisco, Broadsoft, and asterisk in the past.? By far the least
> stable and worst general call quality was asterisk.? I constantly contended
> with strange call quality issues in asterisk, lots of echo (even with
> hardware echo cancellation cards), lots of jitter, lots of call break up
> (even on small systems with 10-20 users, using QoS on the network, and in
> general doing everything I could to prioritize voice over anything else).
>
> When I used Cisco call manager and broadsoft, the voice quality issues were
> basically non-existant, as long as the network was running QoS echo,
> stutter, calls breaking up, just didn't happen.? So, I guess my question is,
> does freeswitch show a marked improvement over asterisk in this department?
> As long as you configure QoS and have hardware echo cancellation does it
> actually work reliably?
>
> Thanks for any additional information about freeswitch you can provide as
> well.? I am a software developer primarily by trade, but I do lots of
> consulting type work in the SME space and I've had a couple projects thrown
> to me that require some integration with a phone system, and I just can't in
> good conscience recommend asterisk anymore.
>
>
> -Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Fri Aug 28 12:04:22 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 12:04:22 -0700
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <7b197bef0908281155w92eff9dp1ffcb379ae4c8438@mail.gmail.com>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
	<7b197bef0908281155w92eff9dp1ffcb379ae4c8438@mail.gmail.com>
Message-ID: <87f2f3b90908281204g47e9f3cw3b5fa9c2c96d0e5e@mail.gmail.com>

On Fri, Aug 28, 2009 at 11:55 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> Hi Tom,
>
> best is to try it for yourself, you cannot expect from the FS mailing
> list an answer like: "you know, fs is nor a marked improvement on
> anything, we just like to spend time together" :-)
>

Although, in reality, some of us do actually enjoy spending time together.
:P
-MC
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From gmaruzz at celliax.org  Fri Aug 28 12:09:05 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 21:09:05 +0200
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <87f2f3b90908281204g47e9f3cw3b5fa9c2c96d0e5e@mail.gmail.com>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
	<7b197bef0908281155w92eff9dp1ffcb379ae4c8438@mail.gmail.com> 
	<87f2f3b90908281204g47e9f3cw3b5fa9c2c96d0e5e@mail.gmail.com>
Message-ID: <7b197bef0908281209w63b05fefhb85926f2eecbbdee@mail.gmail.com>

On Fri, Aug 28, 2009 at 9:04 PM, Michael Collins<msc at freeswitch.org> wrote:
>
>
> On Fri, Aug 28, 2009 at 11:55 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> Hi Tom,
>>
>> best is to try it for yourself, you cannot expect from the FS mailing
>> list an answer like: "you know, fs is nor a marked improvement on
>> anything, we just like to spend time together" :-)
>
> Although, in reality, some of us do actually enjoy spending time together.
> :P
> -MC

meh too! ;-)


From anthony.minessale at gmail.com  Fri Aug 28 12:11:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 14:11:04 -0500
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <87f2f3b90908281154s2a1496b4l2e3aa06545649633@mail.gmail.com>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
	<87f2f3b90908281154s2a1496b4l2e3aa06545649633@mail.gmail.com>
Message-ID: <191c3a030908281211x3af65360q236d7b6a6564c055@mail.gmail.com>

http://www.freeswitch.org/node/117

That's essentially the story of why I wrote FS.


On Fri, Aug 28, 2009 at 1:54 PM, Michael Collins <msc at freeswitch.org> wrote:

> Tom,
> Welcome! Sadly, your experience is not unique...
>
> On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom <paveraware at hotmail.com>wrote:
>
>>  As a background, I ran an asterisk consulting company for about 3 years
>> that I gave up on 2 years ago after repeatedly failing to achieve any sort
>> of stability on any sort install over about 30 phones, I gave up.
>>
>
> The consensus I've seen is that the larger the install, the more likely one
> is to have inexplicable issues.
>
>>
>>
>> Maybe that was wrong, I am open to the possibility that I just didn't know
>> enough and I was building things wrong, but I worked inside the asterisk
>> code (which I feel is a hopeless mess), I implemented a few small custom
>> features, anyway...
>>
>
> Any software that openly admits that a function is "pure nastiness" but
> doesn't change it from version 1.0, 1.2, 1.4, or 1.6 has questionable
> leadership IMHO. (grep the Asterisk source tree for "nastiness" and you'll
> see it.)
>
>>
>>
>> I'm coming back into the VoIP space now, and I'm wondering what sort of
>> issues can I expect in trying to pick up and learn freeswitch?  From what
>> I've read on the website, it appears to have a much more sane architecture.
>> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
>> stable and worst general call quality was asterisk.  I constantly contended
>> with strange call quality issues in asterisk, lots of echo (even with
>> hardware echo cancellation cards), lots of jitter, lots of call break up
>> (even on small systems with 10-20 users, using QoS on the network, and in
>> general doing everything I could to prioritize voice over anything else).
>>
>
> Again, your experience isn't unique...
>
>>
>> When I used Cisco call manager and broadsoft, the voice quality issues
>> were basically non-existant, as long as the network was running QoS echo,
>> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
>> does freeswitch show a marked improvement over asterisk in this department?
>> As long as you configure QoS and have hardware echo cancellation does it
>> actually work reliably?
>>
>
> We receive lots of reports that FreeSWITCH is a vast improvement over not
> only Asterisk but proprietary solutions as well. The FS architecture is, as
> you mentioned, not insane. It is well thought out and therefore highly
> flexible, extensible, and scalable. I'm not aware of anything - OSS or
> proprietary - that can match FS in these three areas.
>
>>
>> Thanks for any additional information about freeswitch you can provide as
>> well.  I am a software developer primarily by trade, but I do lots of
>> consulting type work in the SME space and I've had a couple projects thrown
>> to me that require some integration with a phone system, and I just can't in
>> good conscience recommend asterisk anymore.
>>
>
> Are you comfortable with the lack of a super slick GUI? :) Some GUIs are in
> development but the power users are quite happy with doing the emacs (or
> vim) shuffle with the XML config files. Furthermore, the ways that FS allows
> you to connect and control are fantastic: mod_xml_curl for dynamic
> configurations, event-socket for external control (think of it like AMI not
> sucking and being turbo-charged), mod_xml_rpc for RPC goodness... Anyway,
> the list is impressive.
>
> I can honestly say that every week we get new people looking at FreeSWITCH
> and saying, "Wow, this is incredible." I can definitely, in good conscience,
> recommend you investigate FS more deeply. I'm confident you'll be happy with
> the return on your investment.
>
> Hope it all works out for you! Join us in #freeswitch on irc.freenode.netif you want to chat in real-time.
> -Michael
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gmaruzz at celliax.org  Fri Aug 28 12:11:38 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 21:11:38 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
Message-ID: <7b197bef0908281211u51c1ce43q34f321378816b40@mail.gmail.com>

Usually you don't need to worry about stability issues with FS.

For scalability, peoples tend to use openser or some other sip
loadbalancer in fron of fs, but you probably would not need that.

Live migration of calls is not yet possible, tough.

-giovanni



On Fri, Aug 28, 2009 at 8:52 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
> Hello List,
>
> I have read the current thread about scalability and I would need some
> advice about a callcenter setup:
>
> First where I come from:
> I have lot's of problems with an asterisk solution. I have regular
> crash's and lock-ups, with downgrading and other stuff i got it
> somewhat stable, but have nevertheless regular hickups. I am desperate
> and want to get rid of asterisk and I hope that freeSwitch will
> provide me with a more stable solution.
>
>
> Our Setup (really nothing special):
> * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3
> GIG of RAM, Xircom analog switch connected to mobile stations for 4
> different providers, Digium 4port cards TP400<something>
> * 8 queues
> * ~60 agents (which logon, logoff, pause, unpause), not more than 40
> concurrently online
> * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
> * Music on Hold in the call-queues
> * No special announcement
> * Transfers between calls in queues and different agents as well as
> non agents (i mention this because we have transfer related chrashes
> in asterisk)
>
> The current Problems:
> * Lockups with different causes (ranging from calls not terminated to
> heavy thread locking through the AMI interface)
> * Crashes and library aborts (pthread, libc, crashes related to music
> on hold, app_queue, transfers)
>
> We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are
> now back to 1.4.21.2 (stock debian) as anything beyond that is for
> whatever reason highly unstable for our szenario. Maybe we should have
> been segmenting the box into one asterisk dedicted to talking to the
> hardware, one especial for queue/sip handling, i do not know. (all
> issues are well documented in issues.asterisk.org, but it seems to be
> very, very difficult to get to the bottom of them as they exist since
> 1.4.23 as it seems and are open until know and not fixable since month.)
>
>
> Now, I really would appreciate some success-stories on how you guys
> managed to get a stable pbx system with freeSWITCH in regard of HA and
> scalability:
>
> * How to segment freeSWITCH? Or is it stable enough to handle all in
> one for such a szenario as outlined above?
> * What would be the best strategie for High Availability / Failover?
> ? ? ? ?-> I read in the WIKI (featurelist) that Livemigration of calls from
> one box to another should be possible?
> ? ? ? ?-> I was thinking about using memcached for storing all state
> information so another freeswitch box can takeover calls from the
> first box if it dies, is this possible? If so, how?
> ? ? ? ?-> Is there anotherway to somehow configure freeSWITCH that in the
> event of a crash i do not loose the current established calls?
>
> Basically I just want a stable PBX where I do not have to fear every
> day it will core-dump or abort or Lock up. Is freeSWITCH mature enough
> so i can sleep at night for at least 3 month without a crash?
>
>
>
> Thank you for your Time and help in advance, and I am more than
> willing to take all the information gathered here and create a wiki
> page to help other people with the same questions/problems.
>
> best
> Ray
>
>
> --
> Raimund Sacherer
> -
> RunSolutions
> ? ? Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 - ?Palma de Mallorca
> Baleares
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From tayeb.meftah at gmail.com  Fri Aug 28 13:29:38 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Fri, 28 Aug 2009 20:29:38 +0000
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
 robustness patch
In-Reply-To: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
Message-ID: <4A983E32.2050003@gmail.com>

hello gmaruzz,
thank you for your good works!
Skypiax work Wel now, except for a small bug, but i'm not sur if you see 
it  or no:
if i load the module manualy, for the first load crach and don't detect 
the active runing skype client
but if i do load mod_skypiax aguin will load and work perfectly
thanks
Giovanni Maruzzelli wrote:
> Hi FreeSWITCHers,
>
> all the users of mod_skypiax are kindly requested to test the svn trunk 14519.
>
> It contains a lot of changes meant to add stability and robustness,
> toward a production environment.
>
> Let me know how your feelings, and please add to the Jira any possible
> bug/issue/etc.
>
> Thanks to you all,
>
> -giovanni
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
>
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>   



__________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com




From testa at voicetechnology.com.br  Fri Aug 28 12:33:51 2009
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Fri, 28 Aug 2009 16:33:51 -0300
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
Message-ID: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>

Hi folks,What linux distro would you suggest to deploy FS with django:
Centos 5.3 or Ubuntu LTS?

Thanks
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From anthony.minessale at gmail.com  Fri Aug 28 12:37:08 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 14:37:08 -0500
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
Message-ID: <191c3a030908281237l29570d6fr6e591402d4a22f95@mail.gmail.com>

please use CentOS 5


On Fri, Aug 28, 2009 at 2:33 PM, Fernando Testa <
testa at voicetechnology.com.br> wrote:

> Hi folks,What linux distro would you suggest to deploy FS with django:
> Centos 5.3 or Ubuntu LTS?
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Fri Aug 28 12:37:17 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 12:37:17 -0700
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <7b197bef0908281211u51c1ce43q34f321378816b40@mail.gmail.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<7b197bef0908281211u51c1ce43q34f321378816b40@mail.gmail.com>
Message-ID: <87f2f3b90908281237ye0571e0m9116da183092b154@mail.gmail.com>

On Fri, Aug 28, 2009 at 12:11 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> Usually you don't need to worry about stability issues with FS.
>
> For scalability, peoples tend to use openser or some other sip
> loadbalancer in fron of fs, but you probably would not need that.
>
> Live migration of calls is not yet possible, tough.
>

Brian West has done some testing with live migrations but I don't know where
he left off. Brian, were you using OpenVZ? I forget... In any case, FS
allows you to try to do this with the hope that it will actually work in a
production environment.

As for the other things - yes, FS can work with the TDM card and the queues,
etc. If you are in a position to install FS on a sandbox machine for testing
then that would be your best bet. I recommend diving in, which is probably
what you did when you first started learning Asterisk...

Have fun!
-MC
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From msc at freeswitch.org  Fri Aug 28 12:37:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 12:37:54 -0700
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
Message-ID: <87f2f3b90908281237k2fb599c2ue36d20b8807cf31d@mail.gmail.com>

On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa <
testa at voicetechnology.com.br> wrote:

> Hi folks,What linux distro would you suggest to deploy FS with django:
> Centos 5.3 or Ubuntu LTS?
>
CentOS 5.3 FTW!
-MC
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From gmaruzz at celliax.org  Fri Aug 28 12:41:05 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 21:41:05 +0200
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness 	patch
In-Reply-To: <4A983E32.2050003@gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com> 
	<4A983E32.2050003@gmail.com>
Message-ID: <7b197bef0908281241v4d7bb74an437d6bb177dcc655@mail.gmail.com>

On Fri, Aug 28, 2009 at 10:29 PM, Meftah Tayeb<tayeb.meftah at gmail.com> wrote:
> if i load the module manualy, for the first load crach and don't detect
> the active runing skype client
> but if i do load mod_skypiax aguin will load and work perfectly

Meftah,

I've not seen that bug, it works first time for me.

Anyway, I'll double check it, and I open A jira for you ;-)

-giovanni



> thanks
> Giovanni Maruzzelli wrote:
>> Hi FreeSWITCHers,
>>
>> all the users of mod_skypiax are kindly requested to test the svn trunk 14519.
>>
>> It contains a lot of changes meant to add stability and robustness,
>> toward a production environment.
>>
>> Let me know how your feelings, and please add to the Jira any possible
>> bug/issue/etc.
>>
>> Thanks to you all,
>>
>> -giovanni
>>
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>>
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From gmaruzz at celliax.org  Fri Aug 28 12:44:32 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 21:44:32 +0200
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness 	patch
In-Reply-To: <7b197bef0908281241v4d7bb74an437d6bb177dcc655@mail.gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com> 
	<4A983E32.2050003@gmail.com>
	<7b197bef0908281241v4d7bb74an437d6bb177dcc655@mail.gmail.com>
Message-ID: <7b197bef0908281244s4a6bff51m4f4634715f136c6c@mail.gmail.com>

http://jira.freeswitch.org/browse/MODSKYPIAX-49


On Fri, Aug 28, 2009 at 9:41 PM, Giovanni Maruzzelli<gmaruzz at celliax.org> wrote:
> On Fri, Aug 28, 2009 at 10:29 PM, Meftah Tayeb<tayeb.meftah at gmail.com> wrote:
>> if i load the module manualy, for the first load crach and don't detect
>> the active runing skype client
>> but if i do load mod_skypiax aguin will load and work perfectly
>
> Meftah,
>
> I've not seen that bug, it works first time for me.
>
> Anyway, I'll double check it, and I open A jira for you ;-)
>
> -giovanni
>
>
>
>> thanks
>> Giovanni Maruzzelli wrote:
>>> Hi FreeSWITCHers,
>>>
>>> all the users of mod_skypiax are kindly requested to test the svn trunk 14519.
>>>
>>> It contains a lot of changes meant to add stability and robustness,
>>> toward a production environment.
>>>
>>> Let me know how your feelings, and please add to the Jira any possible
>>> bug/issue/etc.
>>>
>>> Thanks to you all,
>>>
>>> -giovanni
>>>
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>>
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________
>>>
>>> The message was checked by ESET NOD32 Antivirus.
>>>
>>> http://www.eset.com
>>>
>>>
>>>
>>
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>


From aep.lists at it46.se  Fri Aug 28 12:50:00 2009
From: aep.lists at it46.se (Alberto Escudero)
Date: Fri, 28 Aug 2009 21:50:00 +0200
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <87f2f3b90908281237k2fb599c2ue36d20b8807cf31d@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
	<87f2f3b90908281237k2fb599c2ue36d20b8807cf31d@mail.gmail.com>
Message-ID: <d36c422fa99228a711fb7eba47c7e94f.squirrel@correo.nodo50.org>

Debian :)
-- 
Stopping junk mailers is good for the environment

> On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa <
> testa at voicetechnology.com.br> wrote:
>
>> Hi folks,What linux distro would you suggest to deploy FS with django:
>> Centos 5.3 or Ubuntu LTS?
>>
> CentOS 5.3 FTW!
> -MC
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From paveraware at hotmail.com  Fri Aug 28 12:54:43 2009
From: paveraware at hotmail.com (Christensen Tom)
Date: Fri, 28 Aug 2009 19:54:43 +0000
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <87f2f3b90908281154s2a1496b4l2e3aa06545649633@mail.gmail.com>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
	<87f2f3b90908281154s2a1496b4l2e3aa06545649633@mail.gmail.com>
Message-ID: <BAY120-W71D36BC5F54F0DBEAE097DFF50@phx.gbl>


I am totally fine without a slick GUI interface.  The first 2 years of asterisk stuff I did was all in on the CLI in <editor of your choice> (I use vim most of the time, but not for religious reasons...).
Anyway, thanks for the info, I'll be setting up a freeswitch system this weekend expect to see me on IRC and here..
Thanks!
-Tom
Date: Fri, 28 Aug 2009 11:54:25 -0700
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Hello, and stuff.

Tom,
Welcome! Sadly, your experience is not unique...

On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom <paveraware at hotmail.com> wrote:







As a background, I ran an asterisk consulting company for about 3 years that I gave up on 2 years ago after repeatedly failing to achieve any sort of stability on any sort install over about 30 phones, I gave up.


The consensus I've seen is that the larger the install, the more likely one is to have inexplicable issues. 




Maybe that was wrong, I am open to the possibility that I just didn't know enough and I was building things wrong, but I worked inside the asterisk code (which I feel is a hopeless mess), I implemented a few small custom features, anyway... 


Any software that openly admits that a function is "pure nastiness" but doesn't change it from version 1.0, 1.2, 1.4, or 1.6 has questionable leadership IMHO. (grep the Asterisk source tree for "nastiness" and you'll see it.) 




I'm coming back into the VoIP space now, and I'm wondering what sort of issues can I expect in trying to pick up and learn freeswitch?  From what I've read on the website, it appears to have a much more sane architecture.  I've used Cisco, Broadsoft, and asterisk in the past.  By far the least stable and worst general call quality was asterisk.  I constantly contended with strange call quality issues in asterisk, lots of echo (even with hardware echo cancellation cards), lots of jitter, lots of call break up (even on small systems with 10-20 users, using QoS on the network, and in general doing everything I could to prioritize voice over anything else).



Again, your experience isn't unique... 

When I used Cisco call manager and broadsoft, the voice quality issues were basically non-existant, as long as the network was running QoS echo, stutter, calls breaking up, just didn't happen.  So, I guess my question is, does freeswitch show a marked improvement over asterisk in this department?  As long as you configure QoS and have hardware echo cancellation does it actually work reliably?



We receive lots of reports that FreeSWITCH is a vast improvement over not only Asterisk but proprietary solutions as well. The FS architecture is, as you mentioned, not insane. It is well thought out and therefore highly flexible, extensible, and scalable. I'm not aware of anything - OSS or proprietary - that can match FS in these three areas. 



Thanks for any additional information about freeswitch you can provide as well.  I am a software developer primarily by trade, but I do lots of consulting type work in the SME space and I've had a couple projects thrown to me that require some integration with a phone system, and I just can't in good conscience recommend asterisk anymore.



Are you comfortable with the lack of a super slick GUI? :) Some GUIs are in development but the power users are quite happy with doing the emacs (or vim) shuffle with the XML config files. Furthermore, the ways that FS allows you to connect and control are fantastic: mod_xml_curl for dynamic configurations, event-socket for external control (think of it like AMI not sucking and being turbo-charged), mod_xml_rpc for RPC goodness... Anyway, the list is impressive.



I can honestly say that every week we get new people looking at FreeSWITCH and saying, "Wow, this is incredible." I can definitely, in good conscience, recommend you investigate FS more deeply. I'm confident you'll be happy with the return on your investment.



Hope it all works out for you! Join us in #freeswitch on irc.freenode.net if you want to chat in real-time.
-Michael


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From gmaruzz at celliax.org  Fri Aug 28 12:58:42 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 21:58:42 +0200
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <BAY120-W71D36BC5F54F0DBEAE097DFF50@phx.gbl>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
	<87f2f3b90908281154s2a1496b4l2e3aa06545649633@mail.gmail.com> 
	<BAY120-W71D36BC5F54F0DBEAE097DFF50@phx.gbl>
Message-ID: <7b197bef0908281258i2b71b724hd0bb19ca6b69bdac@mail.gmail.com>

Welcome on board Tom! And sorry for being witty before ;-)
-gm


On Fri, Aug 28, 2009 at 9:54 PM, Christensen Tom<paveraware at hotmail.com> wrote:
> I am totally fine without a slick GUI interface. ?The first 2 years of
> asterisk stuff I did was all in on the CLI in <editor of your choice> (I use
> vim most of the time, but not for religious reasons...).
> Anyway, thanks for the info, I'll be setting up a freeswitch system this
> weekend expect to see me on IRC and here..
> Thanks!
> -Tom
> ________________________________
> Date: Fri, 28 Aug 2009 11:54:25 -0700
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Hello, and stuff.
>
> Tom,
> Welcome! Sadly, your experience is not unique...
>
> On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom <paveraware at hotmail.com>
> wrote:
>
> As a background, I ran an asterisk consulting company for about 3 years that
> I gave up on 2 years ago after repeatedly failing to achieve any sort of
> stability on any sort install over about 30 phones, I gave up.
>
> The consensus I've seen is that the larger the install, the more likely one
> is to have inexplicable issues.
>
>
> Maybe that was wrong, I am open to the possibility that I just didn't know
> enough and I was building things wrong, but I worked inside the asterisk
> code (which I feel is a hopeless mess), I implemented a few small custom
> features, anyway...
>
> Any software that openly admits that a function is "pure nastiness" but
> doesn't change it from version 1.0, 1.2, 1.4, or 1.6 has questionable
> leadership IMHO. (grep the Asterisk source tree for "nastiness" and you'll
> see it.)
>
>
> I'm coming back into the VoIP space now, and I'm wondering what sort of
> issues can I expect in trying to pick up and learn freeswitch?? From what
> I've read on the website, it appears to have a much more sane architecture.
> I've used Cisco, Broadsoft, and asterisk in the past.? By far the least
> stable and worst general call quality was asterisk.? I constantly contended
> with strange call quality issues in asterisk, lots of echo (even with
> hardware echo cancellation cards), lots of jitter, lots of call break up
> (even on small systems with 10-20 users, using QoS on the network, and in
> general doing everything I could to prioritize voice over anything else).
>
> Again, your experience isn't unique...
>
> When I used Cisco call manager and broadsoft, the voice quality issues were
> basically non-existant, as long as the network was running QoS echo,
> stutter, calls breaking up, just didn't happen.? So, I guess my question is,
> does freeswitch show a marked improvement over asterisk in this department?
> As long as you configure QoS and have hardware echo cancellation does it
> actually work reliably?
>
> We receive lots of reports that FreeSWITCH is a vast improvement over not
> only Asterisk but proprietary solutions as well. The FS architecture is, as
> you mentioned, not insane. It is well thought out and therefore highly
> flexible, extensible, and scalable. I'm not aware of anything - OSS or
> proprietary - that can match FS in these three areas.
>
> Thanks for any additional information about freeswitch you can provide as
> well.? I am a software developer primarily by trade, but I do lots of
> consulting type work in the SME space and I've had a couple projects thrown
> to me that require some integration with a phone system, and I just can't in
> good conscience recommend asterisk anymore.
>
> Are you comfortable with the lack of a super slick GUI? :) Some GUIs are in
> development but the power users are quite happy with doing the emacs (or
> vim) shuffle with the XML config files. Furthermore, the ways that FS allows
> you to connect and control are fantastic: mod_xml_curl for dynamic
> configurations, event-socket for external control (think of it like AMI not
> sucking and being turbo-charged), mod_xml_rpc for RPC goodness... Anyway,
> the list is impressive.
>
> I can honestly say that every week we get new people looking at FreeSWITCH
> and saying, "Wow, this is incredible." I can definitely, in good conscience,
> recommend you investigate FS more deeply. I'm confident you'll be happy with
> the return on your investment.
>
> Hope it all works out for you! Join us in #freeswitch on irc.freenode.net if
> you want to chat in real-time.
> -Michael
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From michal.bielicki at halo2.pl  Fri Aug 28 13:00:44 2009
From: michal.bielicki at halo2.pl (Michal Bielicki)
Date: Fri, 28 Aug 2009 22:00:44 +0200
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <d36c422fa99228a711fb7eba47c7e94f.squirrel@correo.nodo50.org>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
	<87f2f3b90908281237k2fb599c2ue36d20b8807cf31d@mail.gmail.com>
	<d36c422fa99228a711fb7eba47c7e94f.squirrel@correo.nodo50.org>
Message-ID: <CFCEE02A-A1EE-41E3-AC3D-C57F35990111@halo2.pl>

OpenSolaris :P

Am 28.08.2009 um 21:50 schrieb Alberto Escudero:

> Debian :)
> --  
> Stopping junk mailers is good for the environment
>
>> On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa <
>> testa at voicetechnology.com.br> wrote:
>>
>>> Hi folks,What linux distro would you suggest to deploy FS with  
>>> django:
>>> Centos 5.3 or Ubuntu LTS?
>>>
>> CentOS 5.3 FTW!
>> -MC
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. info at halokwadrat.de
e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
Hauptgesch?ftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 0000153539

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From asterisk at dotr.com  Fri Aug 28 13:02:29 2009
From: asterisk at dotr.com (Julian Lyndon-Smith)
Date: Fri, 28 Aug 2009 21:02:29 +0100
Subject: [Freeswitch-users] newbie alert. Help with dialplan
In-Reply-To: <87f2f3b90908281024i1ecc6238o3bcab80e9464797e@mail.gmail.com>
References: <f9c510070908271116n33c3f35di44eea1790f62bfc8@mail.gmail.com>
	<87f2f3b90908281024i1ecc6238o3bcab80e9464797e@mail.gmail.com>
Message-ID: <f9c510070908281302h6870a515h8294b04fb5ea1607@mail.gmail.com>

Thanks Michael. What the extension currently does is :

A) Add 1 to the current list of Queue calls (we have a blended system
where I need to dynamically allocate agents from outbound to inbound)
This counter is maintained in the phone system
B) call my application via a web call (CURL) to tell it that a new
Queue call has come in. This web call returns various bits of
information regarding the queue (message to play to the caller,
message to play to the agent, is the queue open etc)
C) if this fails (bad queue) then drop out
D) start recording, play the welcome message to the caller
E) Ask them if they want to participate in an automated questionnaire
at the end of the call.
F) If they agree, play a thank you message
G) If the call is out of hours, go to voicemail
H) Send a message to the application that a call is about to enter the queue
I) Join the queue
J) if the call is dropped out of the queue because no-one answered,
record that status
K) Stop recording

If they have answered yes to the questionnaire, then redirect the call
to an IVR when the agent hangs up

When the agent hangs up, or when the caller hangs up, then a jabber
message is sent to the agent, and a web service is called to indicate
the end of the call


2009/8/28 Michael Collins <msc at freeswitch.org>:
> Julian,
>
> It might be better if you gave us the run-down on what your current Asterisk
> extension does. Most likely there is an elegant way of doing it in
> FreeSWITCH. Part of the unlearing of the Asterisk way is coming to grips
> with the fact that the FS dialplan is not a programming language in and of
> itself. While it's convenient to have GotoIf's all over the place, in the
> long run that's inefficient.
>
> Let's start from the top-down instead of the bottom-up. Discussion your
> application in general terms, i.e., non-Asterisk-specific language. Tell us
> what it does, not how it does it, and then we can think it through using FS
> concepts.
>
> -MC
>
> On Thu, Aug 27, 2009 at 11:16 AM, Julian Lyndon-Smith <asterisk at dotr.com>
> wrote:
>>
>> I am a long-time asterisk user (2005), so I come from an unfortunate
>> position of having to unlearn an awful lot of stuff in order to make
>> freeswitch do the things I want ;(
>>
>> I *think* i've got my head around using mod_curl_xml (?) to read all
>> the config stuff from my webserver. I *think* I've got my head around
>> setting up the sip clients etc etc ...
>>
>> However, where I am really struggling is the dialplan. For the life of
>> me I simply cannot seem to grasp the fs way - that's no disrespect to
>> the fs way, but perhaps the failure of this old brain to change !
>>
>> ?have pastebinned an example of show 1234546 at inboundq at
>> http://www.pastebin.ca/1544890. If possible, would someone be able to
>> show me how to convert this dialplan to the fs way ? If I could be
>> given a little foot-up I'm sure that I would be able to convert the
>> rest of the dialplan !
>>
>> Thanks in advance, and please go easy !
>>
>> Julian
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From gmaruzz at celliax.org  Fri Aug 28 13:02:15 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 22:02:15 +0200
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <CFCEE02A-A1EE-41E3-AC3D-C57F35990111@halo2.pl>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com> 
	<87f2f3b90908281237k2fb599c2ue36d20b8807cf31d@mail.gmail.com> 
	<d36c422fa99228a711fb7eba47c7e94f.squirrel@correo.nodo50.org> 
	<CFCEE02A-A1EE-41E3-AC3D-C57F35990111@halo2.pl>
Message-ID: <7b197bef0908281302n43fa2c8aoc6d713f2681fedd5@mail.gmail.com>

Vista!


On Fri, Aug 28, 2009 at 10:00 PM, Michal
Bielicki<michal.bielicki at halo2.pl> wrote:
> OpenSolaris :P
>
> Am 28.08.2009 um 21:50 schrieb Alberto Escudero:
>
>> Debian :)
>> -- Stopping junk mailers is good for the environment
>>
>>> On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa <
>>> testa at voicetechnology.com.br> wrote:
>>>
>>>> Hi folks,What linux distro would you suggest to deploy FS with django:
>>>> Centos 5.3 or Ubuntu LTS?
>>>>
>>> CentOS 5.3 FTW!
>>> -MC
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> Michal Bielicki
> Leiter der Niederlassung
> HaloKwadrat Sp. z o.o.
> Niederlassung Kleinmachnow
> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
> Ust.Id.: DE261885536
> Geschaeftsfuehrer: Aleksander Wiercinski
> Meiereifeld 2b, 14532 Kleinmachnow
> t. +49 33203 263220
> f. +49 33203 263229 sip. info at halokwadrat.de
> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
> Hauptgesch?ftsstelle:
> Halo Kwadrat Sp. z o.o.
> ul. Polna 46/14
> 00-644 Warszawa, Polen
> EIngetragen im HRB Warszawa, KRS 0000153539
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From edpimentl at gmail.com  Fri Aug 28 13:07:00 2009
From: edpimentl at gmail.com (EdPimentl)
Date: Fri, 28 Aug 2009 16:07:00 -0400
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
Message-ID: <9dc4a1670908281307o30723ddr210b5c6efc87230c@mail.gmail.com>

Centos 5.3 / Solaris ..... of course
And I am looking at http://moblin.org/  for tiny embedded system.

Best regards,
-E
Gpro.ws
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From lfurrea at gmail.com  Fri Aug 28 13:09:37 2009
From: lfurrea at gmail.com (Luis F Urrea)
Date: Fri, 28 Aug 2009 14:09:37 -0600
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <9dc4a1670908281307o30723ddr210b5c6efc87230c@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
	<9dc4a1670908281307o30723ddr210b5c6efc87230c@mail.gmail.com>
Message-ID: <f120f4c80908281309o28c3eca3h5e4f7efc821e1adf@mail.gmail.com>

Gladly using FreeBSD 64 bit for a >35 ext PBX

On Fri, Aug 28, 2009 at 2:07 PM, EdPimentl <edpimentl at gmail.com> wrote:

> Centos 5.3 / Solaris ..... of course
> And I am looking at http://moblin.org/  for tiny embedded system.
>
> Best regards,
> -E
> Gpro.ws
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Aug 28 13:09:06 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 15:09:06 -0500
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <9dc4a1670908281307o30723ddr210b5c6efc87230c@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
	<9dc4a1670908281307o30723ddr210b5c6efc87230c@mail.gmail.com>
Message-ID: <191c3a030908281309ma63a591tbc77294f98424a77@mail.gmail.com>

no seriously, use CentOS
None of these other guys have to support FS ;)

If you plan on doing it all without asking for help, then you are welcome to
any OS that FS compiles on. =D



On Fri, Aug 28, 2009 at 3:07 PM, EdPimentl <edpimentl at gmail.com> wrote:

> Centos 5.3 / Solaris ..... of course
> And I am looking at http://moblin.org/  for tiny embedded system.
>
> Best regards,
> -E
> Gpro.ws
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From pjintheusa at gmail.com  Fri Aug 28 13:13:12 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 28 Aug 2009 16:13:12 -0400
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
	<191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
Message-ID: <367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>

thanks for the reply.

Tried this and exactly the same result. The prompt is slowed - as if
the person is drunk.

I do notice that the originating call SDP shows:

a=rtpmap:0 PCMU/8000

but the play tries to use:

EXECUTE sofia/external/6093693828
playback(prompts/press-1-to-accept-call-from.wav)
2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
Activated L16 at 8000hz 1 channels 20ms


Should the codec be PCMU at 8000hz as opposed to L16 at 8000hz? If so of
there a way of forcing this codec for play.

Thanks


Phillip


Full code & logs
http://pastebin.freeswitch.org/10147
http://pastebin.freeswitch.org/10150



On Thu, Aug 27, 2009 at 4:30 PM, Anthony
Minessale<anthony.minessale at gmail.com> wrote:
> try wedging {ignore_early_media=true} before the first [ in your dial string
> and eliminate the code waiting for answer.
>
>
> On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
>>
>> Hi there,
>>
>> I know there are other ways of doing this. I am just trying get to
>> know have fun with the FreeSWITCH API.
>>
>> I am using originate and park a call:
>>
>> fsApi.Execute("originate",
>>
>> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
>>
>> That works great, the phone rings. I want to play something to this
>> called party when they pick up.
>>
>> ?while (!blegSession.answered())
>> ?{
>> ? ? ? ?Log.WriteLine(LogLevel.Alert,
>> "Inside::CallReturns:!Session.answered::Loop");
>> ? ? ? ?blegSession.sleep(500, 1);
>> ?}
>>
>> string promptFile = "prompts/whisper.wav";
>> blegSession.StreamFile(promptFile, 0);
>>
>>
>> This works - but the audio is choppy and slow.
>>
>> Is there something I need to do to that parked call before streaming that
>> file?
>>
>>
>> Thanks for any input.
>>
>> Phil
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From anthony.minessale at gmail.com  Fri Aug 28 13:33:47 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 15:33:47 -0500
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
	<191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
	<367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>
Message-ID: <191c3a030908281333w39f001bcx7b9914f7bdda4f8f@mail.gmail.com>

the L16 codec is used because the wav files is raw PCM audio
which is being encoded to PCMU

Have you tried doing this to a locally registered phone?



On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> thanks for the reply.
>
> Tried this and exactly the same result. The prompt is slowed - as if
> the person is drunk.
>
> I do notice that the originating call SDP shows:
>
> a=rtpmap:0 PCMU/8000
>
> but the play tries to use:
>
> EXECUTE sofia/external/6093693828
> playback(prompts/press-1-to-accept-call-from.wav)
> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
>
> Should the codec be PCMU at 8000hz as opposed to L16 at 8000hz? If so of
> there a way of forcing this codec for play.
>
> Thanks
>
>
> Phillip
>
>
> Full code & logs
> http://pastebin.freeswitch.org/10147
> http://pastebin.freeswitch.org/10150
>
>
>
> On Thu, Aug 27, 2009 at 4:30 PM, Anthony
> Minessale<anthony.minessale at gmail.com> wrote:
> > try wedging {ignore_early_media=true} before the first [ in your dial
> string
> > and eliminate the code waiting for answer.
> >
> >
> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <pjintheusa at gmail.com>
> wrote:
> >>
> >> Hi there,
> >>
> >> I know there are other ways of doing this. I am just trying get to
> >> know have fun with the FreeSWITCH API.
> >>
> >> I am using originate and park a call:
> >>
> >> fsApi.Execute("originate",
> >>
> >>
> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
> >> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
> >>
> >> That works great, the phone rings. I want to play something to this
> >> called party when they pick up.
> >>
> >>  while (!blegSession.answered())
> >>  {
> >>        Log.WriteLine(LogLevel.Alert,
> >> "Inside::CallReturns:!Session.answered::Loop");
> >>        blegSession.sleep(500, 1);
> >>  }
> >>
> >> string promptFile = "prompts/whisper.wav";
> >> blegSession.StreamFile(promptFile, 0);
> >>
> >>
> >> This works - but the audio is choppy and slow.
> >>
> >> Is there something I need to do to that parked call before streaming
> that
> >> file?
> >>
> >>
> >> Thanks for any input.
> >>
> >> Phil
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From diego.viola at gmail.com  Fri Aug 28 13:34:41 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 28 Aug 2009 20:34:41 +0000
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
Message-ID: <86a32abc0908281334v4033e714jd695d19d6957eb28@mail.gmail.com>

Hi Christensen,

Welcome, you made a good choice on FreeSWITCH, and FS is much better at
those things than Asterisk.

Good luck, we are here to help you and tell us your experience later :).

Diego

On Fri, Aug 28, 2009 at 6:14 PM, Christensen Tom <paveraware at hotmail.com>wrote:

>  As a background, I ran an asterisk consulting company for about 3 years
> that I gave up on 2 years ago after repeatedly failing to achieve any sort
> of stability on any sort install over about 30 phones, I gave up.
>
> Maybe that was wrong, I am open to the possibility that I just didn't know
> enough and I was building things wrong, but I worked inside the asterisk
> code (which I feel is a hopeless mess), I implemented a few small custom
> features, anyway...
>
> I'm coming back into the VoIP space now, and I'm wondering what sort of
> issues can I expect in trying to pick up and learn freeswitch?  From what
> I've read on the website, it appears to have a much more sane architecture.
> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
> stable and worst general call quality was asterisk.  I constantly contended
> with strange call quality issues in asterisk, lots of echo (even with
> hardware echo cancellation cards), lots of jitter, lots of call break up
> (even on small systems with 10-20 users, using QoS on the network, and in
> general doing everything I could to prioritize voice over anything else).
>
> When I used Cisco call manager and broadsoft, the voice quality issues were
> basically non-existant, as long as the network was running QoS echo,
> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
> does freeswitch show a marked improvement over asterisk in this department?
> As long as you configure QoS and have hardware echo cancellation does it
> actually work reliably?
>
> Thanks for any additional information about freeswitch you can provide as
> well.  I am a software developer primarily by trade, but I do lots of
> consulting type work in the SME space and I've had a couple projects thrown
> to me that require some integration with a phone system, and I just can't in
> good conscience recommend asterisk anymore.
>
>
> -Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Aug 28 13:40:40 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 15:40:40 -0500
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <191c3a030908281333w39f001bcx7b9914f7bdda4f8f@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
	<191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
	<367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>
	<191c3a030908281333w39f001bcx7b9914f7bdda4f8f@mail.gmail.com>
Message-ID: <191c3a030908281340m83d3c55q96f1257341250b86@mail.gmail.com>

nevermind,

I didn't look closely at your code.

how did you end up with blegSession ?
Are you running this script from the CLI or via the application interface.







On Fri, Aug 28, 2009 at 3:33 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> the L16 codec is used because the wav files is raw PCM audio
> which is being encoded to PCMU
>
> Have you tried doing this to a locally registered phone?
>
>
>
>
> On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> thanks for the reply.
>>
>> Tried this and exactly the same result. The prompt is slowed - as if
>> the person is drunk.
>>
>> I do notice that the originating call SDP shows:
>>
>> a=rtpmap:0 PCMU/8000
>>
>> but the play tries to use:
>>
>> EXECUTE sofia/external/6093693828
>> playback(prompts/press-1-to-accept-call-from.wav)
>> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>>
>>
>> Should the codec be PCMU at 8000hz as opposed to L16 at 8000hz? If so of
>> there a way of forcing this codec for play.
>>
>> Thanks
>>
>>
>> Phillip
>>
>>
>> Full code & logs
>> http://pastebin.freeswitch.org/10147
>> http://pastebin.freeswitch.org/10150
>>
>>
>>
>> On Thu, Aug 27, 2009 at 4:30 PM, Anthony
>> Minessale<anthony.minessale at gmail.com> wrote:
>> > try wedging {ignore_early_media=true} before the first [ in your dial
>> string
>> > and eliminate the code waiting for answer.
>> >
>> >
>> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <pjintheusa at gmail.com>
>> wrote:
>> >>
>> >> Hi there,
>> >>
>> >> I know there are other ways of doing this. I am just trying get to
>> >> know have fun with the FreeSWITCH API.
>> >>
>> >> I am using originate and park a call:
>> >>
>> >> fsApi.Execute("originate",
>> >>
>> >>
>> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>> >> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
>> >>
>> >> That works great, the phone rings. I want to play something to this
>> >> called party when they pick up.
>> >>
>> >>  while (!blegSession.answered())
>> >>  {
>> >>        Log.WriteLine(LogLevel.Alert,
>> >> "Inside::CallReturns:!Session.answered::Loop");
>> >>        blegSession.sleep(500, 1);
>> >>  }
>> >>
>> >> string promptFile = "prompts/whisper.wav";
>> >> blegSession.StreamFile(promptFile, 0);
>> >>
>> >>
>> >> This works - but the audio is choppy and slow.
>> >>
>> >> Is there something I need to do to that parked call before streaming
>> that
>> >> file?
>> >>
>> >>
>> >> Thanks for any input.
>> >>
>> >> Phil
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From pjintheusa at gmail.com  Fri Aug 28 13:53:16 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 28 Aug 2009 16:53:16 -0400
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <191c3a030908281340m83d3c55q96f1257341250b86@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
	<191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
	<367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>
	<191c3a030908281333w39f001bcx7b9914f7bdda4f8f@mail.gmail.com>
	<191c3a030908281340m83d3c55q96f1257341250b86@mail.gmail.com>
Message-ID: <367751820908281353l69b26639va13b3d63321b89e5@mail.gmail.com>

The dialplan is is kicking off the C# script:

<condition field="destination_number" expression="^(12125553666)$">
    * <action application="managed" data="MyFSProjects.ProcessCall"/>*
</condition>

I launch a thread and specify the uuid to originate:

fsApi.Execute("originate",
string.Format("{{ignore_early_media=true,absolute_codec_string='PCMU'}}[*
origination_uuid={0}*,origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
&park", *call.Uuid*, call.OutgoingCallerID, call.NumberToDial));

in the callback i try and play the wav file:

FreeSWITCH.Native.ManagedSession blegSession = new
FreeSWITCH.Native.ManagedSession(callID);

if (blegSession.Ready() && blegSession.mediaReady())
{
      blegSession.Answer();
      blegSession.Execute("playback",
"prompts/press-1-to-accept-call-from.wav");
}

I go on to bridge the leg a and b - that bits works fine.



On Fri, Aug 28, 2009 at 4:40 PM, Anthony Minessale<
anthony.minessale at gmail.com> wrote:
> nevermind,
>
> I didn't look closely at your code.
>
> how did you end up with blegSession ?
> Are you running this script from the CLI or via the application interface.
>
>
>
>
>
>
>
> On Fri, Aug 28, 2009 at 3:33 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> the L16 codec is used because the wav files is raw PCM audio
>> which is being encoded to PCMU
>>
>> Have you tried doing this to a locally registered phone?
>>
>>
>>
>> On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones <pjintheusa at gmail.com>
>> wrote:
>>>
>>> thanks for the reply.
>>>
>>> Tried this and exactly the same result. The prompt is slowed - as if
>>> the person is drunk.
>>>
>>> I do notice that the originating call SDP shows:
>>>
>>> a=rtpmap:0 PCMU/8000
>>>
>>> but the play tries to use:
>>>
>>> EXECUTE sofia/external/6093693828
>>> playback(prompts/press-1-to-accept-call-from.wav)
>>> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
>>> Activated L16 at 8000hz 1 channels 20ms
>>>
>>>
>>> Should the codec be PCMU at 8000hz as opposed to L16 at 8000hz? If so of
>>> there a way of forcing this codec for play.
>>>
>>> Thanks
>>>
>>>
>>> Phillip
>>>
>>>
>>> Full code & logs
>>> http://pastebin.freeswitch.org/10147
>>> http://pastebin.freeswitch.org/10150
>>>
>>>
>>>
>>> On Thu, Aug 27, 2009 at 4:30 PM, Anthony
>>> Minessale<anthony.minessale at gmail.com> wrote:
>>> > try wedging {ignore_early_media=true} before the first [ in your dial
>>> > string
>>> > and eliminate the code waiting for answer.
>>> >
>>> >
>>> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <pjintheusa at gmail.com>
>>> > wrote:
>>> >>
>>> >> Hi there,
>>> >>
>>> >> I know there are other ways of doing this. I am just trying get to
>>> >> know have fun with the FreeSWITCH API.
>>> >>
>>> >> I am using originate and park a call:
>>> >>
>>> >> fsApi.Execute("originate",
>>> >>
>>> >>
>>> >>
string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>>> >> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
>>> >>
>>> >> That works great, the phone rings. I want to play something to this
>>> >> called party when they pick up.
>>> >>
>>> >>  while (!blegSession.answered())
>>> >>  {
>>> >>        Log.WriteLine(LogLevel.Alert,
>>> >> "Inside::CallReturns:!Session.answered::Loop");
>>> >>        blegSession.sleep(500, 1);
>>> >>  }
>>> >>
>>> >> string promptFile = "prompts/whisper.wav";
>>> >> blegSession.StreamFile(promptFile, 0);
>>> >>
>>> >>
>>> >> This works - but the audio is choppy and slow.
>>> >>
>>> >> Is there something I need to do to that parked call before streaming
>>> >> that
>>> >> file?
>>> >>
>>> >>
>>> >> Thanks for any input.
>>> >>
>>> >> Phil
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> > pstn:213-799-1400
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dave at 3c.co.uk  Fri Aug 28 14:01:53 2009
From: dave at 3c.co.uk (David Knell)
Date: Sat, 29 Aug 2009 00:01:53 +0300
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
Message-ID: <1251493313.4438.8.camel@dk-d820>

Hi Raimund,

One FreeSWITCH box will be quite enough to handle the call volumes that
you're talking about, and it ought to be much more stable than the
Asterisk solution which you've outlined below.

It's probably best to forget about live failover without calls dropping
- this isn't something that's supported, and there'd be a lot of work to
do to develop code to keep two boxed in sync.  Once you get used to a
stable solution - i.e. something which doesn't crash - then live
failover, HA, etc., will seem somewhat irrelevant.

I recently did some work on an FS-based call center solution - drop me a
note if I could be of any help with yours.

Cheers --

Dave

> Hello List,
> 
> I have read the current thread about scalability and I would need some  
> advice about a callcenter setup:
> 
> First where I come from:
> I have lot's of problems with an asterisk solution. I have regular  
> crash's and lock-ups, with downgrading and other stuff i got it  
> somewhat stable, but have nevertheless regular hickups. I am desperate  
> and want to get rid of asterisk and I hope that freeSwitch will  
> provide me with a more stable solution.
> 
> 
> Our Setup (really nothing special):
> * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3  
> GIG of RAM, Xircom analog switch connected to mobile stations for 4  
> different providers, Digium 4port cards TP400<something>
> * 8 queues
> * ~60 agents (which logon, logoff, pause, unpause), not more than 40  
> concurrently online
> * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
> * Music on Hold in the call-queues
> * No special announcement
> * Transfers between calls in queues and different agents as well as  
> non agents (i mention this because we have transfer related chrashes  
> in asterisk)
> 
> The current Problems:
> * Lockups with different causes (ranging from calls not terminated to  
> heavy thread locking through the AMI interface)
> * Crashes and library aborts (pthread, libc, crashes related to music  
> on hold, app_queue, transfers)
> 
> We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are  
> now back to 1.4.21.2 (stock debian) as anything beyond that is for  
> whatever reason highly unstable for our szenario. Maybe we should have  
> been segmenting the box into one asterisk dedicted to talking to the  
> hardware, one especial for queue/sip handling, i do not know. (all  
> issues are well documented in issues.asterisk.org, but it seems to be  
> very, very difficult to get to the bottom of them as they exist since  
> 1.4.23 as it seems and are open until know and not fixable since month.)
> 
> 
> Now, I really would appreciate some success-stories on how you guys  
> managed to get a stable pbx system with freeSWITCH in regard of HA and  
> scalability:
> 
> * How to segment freeSWITCH? Or is it stable enough to handle all in  
> one for such a szenario as outlined above?
> * What would be the best strategie for High Availability / Failover?
> 	-> I read in the WIKI (featurelist) that Livemigration of calls from  
> one box to another should be possible?
> 	-> I was thinking about using memcached for storing all state  
> information so another freeswitch box can takeover calls from the  
> first box if it dies, is this possible? If so, how?
> 	-> Is there anotherway to somehow configure freeSWITCH that in the  
> event of a crash i do not loose the current established calls?
> 
> Basically I just want a stable PBX where I do not have to fear every  
> day it will core-dump or abort or Lock up. Is freeSWITCH mature enough  
> so i can sleep at night for at least 3 month without a crash?
> 
> 
> 
> Thank you for your Time and help in advance, and I am more than  
> willing to take all the information gathered here and create a wiki  
> page to help other people with the same questions/problems.
> 
> best
> Ray
> 
> 
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From anthony.minessale at gmail.com  Fri Aug 28 14:12:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 16:12:14 -0500
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <367751820908281353l69b26639va13b3d63321b89e5@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
	<191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
	<367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>
	<191c3a030908281333w39f001bcx7b9914f7bdda4f8f@mail.gmail.com>
	<191c3a030908281340m83d3c55q96f1257341250b86@mail.gmail.com>
	<367751820908281353l69b26639va13b3d63321b89e5@mail.gmail.com>
Message-ID: <191c3a030908281412h51b75088s8c40bd0a53688928@mail.gmail.com>

Taking over the session while it's parked like that from your code and
asking it to play a file is making it do 2 things at once.
The session's thread is already busy in the park loop so you would have to
change it's state to something passive like soft_execute so
it's thread was not doing anything or send it a message so the park loop
would pick it up by using the uuid_broadcast app::arg

there are variables you can set that begin group_confirm_* that you can set
to do this all without even using a script.

http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_variables




On Fri, Aug 28, 2009 at 3:53 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> The dialplan is is kicking off the C# script:
>
> <condition field="destination_number" expression="^(12125553666)$">
>     * <action application="managed" data="MyFSProjects.ProcessCall"/>*
> </condition>
>
> I launch a thread and specify the uuid to originate:
>
> fsApi.Execute("originate",
> string.Format("{{ignore_early_media=true,absolute_codec_string='PCMU'}}[*
> origination_uuid={0}*,origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
> &park", *call.Uuid*, call.OutgoingCallerID, call.NumberToDial));
>
> in the callback i try and play the wav file:
>
> FreeSWITCH.Native.ManagedSession blegSession = new
> FreeSWITCH.Native.ManagedSession(callID);
>
> if (blegSession.Ready() && blegSession.mediaReady())
> {
>       blegSession.Answer();
>       blegSession.Execute("playback",
> "prompts/press-1-to-accept-call-from.wav");
> }
>
> I go on to bridge the leg a and b - that bits works fine.
>
>
>
> On Fri, Aug 28, 2009 at 4:40 PM, Anthony Minessale<
> anthony.minessale at gmail.com> wrote:
> > nevermind,
> >
> > I didn't look closely at your code.
> >
> > how did you end up with blegSession ?
> > Are you running this script from the CLI or via the application
> interface.
> >
> >
> >
> >
> >
> >
> >
> > On Fri, Aug 28, 2009 at 3:33 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> the L16 codec is used because the wav files is raw PCM audio
> >> which is being encoded to PCMU
> >>
> >> Have you tried doing this to a locally registered phone?
> >>
> >>
> >>
> >> On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones <pjintheusa at gmail.com>
> >> wrote:
> >>>
> >>> thanks for the reply.
> >>>
> >>> Tried this and exactly the same result. The prompt is slowed - as if
> >>> the person is drunk.
> >>>
> >>> I do notice that the originating call SDP shows:
> >>>
> >>> a=rtpmap:0 PCMU/8000
> >>>
> >>> but the play tries to use:
> >>>
> >>> EXECUTE sofia/external/6093693828
> >>> playback(prompts/press-1-to-accept-call-from.wav)
> >>> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
> >>> Activated L16 at 8000hz 1 channels 20ms
> >>>
> >>>
> >>> Should the codec be PCMU at 8000hz as opposed to L16 at 8000hz? If so of
> >>> there a way of forcing this codec for play.
> >>>
> >>> Thanks
> >>>
> >>>
> >>> Phillip
> >>>
> >>>
> >>> Full code & logs
> >>> http://pastebin.freeswitch.org/10147
> >>> http://pastebin.freeswitch.org/10150
> >>>
> >>>
> >>>
> >>> On Thu, Aug 27, 2009 at 4:30 PM, Anthony
> >>> Minessale<anthony.minessale at gmail.com> wrote:
> >>> > try wedging {ignore_early_media=true} before the first [ in your dial
> >>> > string
> >>> > and eliminate the code waiting for answer.
> >>> >
> >>> >
> >>> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <pjintheusa at gmail.com
> >
> >>> > wrote:
> >>> >>
> >>> >> Hi there,
> >>> >>
> >>> >> I know there are other ways of doing this. I am just trying get to
> >>> >> know have fun with the FreeSWITCH API.
> >>> >>
> >>> >> I am using originate and park a call:
> >>> >>
> >>> >> fsApi.Execute("originate",
> >>> >>
> >>> >>
> >>> >>
> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
> >>> >> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
> >>> >>
> >>> >> That works great, the phone rings. I want to play something to this
> >>> >> called party when they pick up.
> >>> >>
> >>> >>  while (!blegSession.answered())
> >>> >>  {
> >>> >>        Log.WriteLine(LogLevel.Alert,
> >>> >> "Inside::CallReturns:!Session.answered::Loop");
> >>> >>        blegSession.sleep(500, 1);
> >>> >>  }
> >>> >>
> >>> >> string promptFile = "prompts/whisper.wav";
> >>> >> blegSession.StreamFile(promptFile, 0);
> >>> >>
> >>> >>
> >>> >> This works - but the audio is choppy and slow.
> >>> >>
> >>> >> Is there something I need to do to that parked call before streaming
> >>> >> that
> >>> >> file?
> >>> >>
> >>> >>
> >>> >> Thanks for any input.
> >>> >>
> >>> >> Phil
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> >
> >>> >
> >>> > --
> >>> > Anthony Minessale II
> >>> >
> >>> > FreeSWITCH http://www.freeswitch.org/
> >>> > ClueCon http://www.cluecon.com/
> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >
> >>> > AIM: anthm
> >>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> > IRC: irc.freenode.net #freeswitch
> >>> >
> >>> > FreeSWITCH Developer Conference
> >>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> > iax:guest at conference.freeswitch.org/888
> >>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> > pstn:213-799-1400
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:213-799-1400
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From oseslija at gmail.com  Fri Aug 28 15:03:12 2009
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sat, 29 Aug 2009 00:03:12 +0200
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
Message-ID: <4468a6770908281503gd7c003ewf44fd353f78c21e4@mail.gmail.com>

I currently use FS for a managed PBX services for many companies. It's so
configurable and extensible, I don't think I would change it for anything. I
have Cisco CME and SCCP phones in my office all interconnected to FS doing
logic and thinking. I also have Asterisks as PSTN PRI gateways. FS does all
the job for me in the world, the other two are just used as protocol
gateways. I wouldn't change FS for anything else now, after a year and the
half of playing with it. I fully recommend it.

Regards,
Ognjen

P.S: To be fair, the echo on the hw ec cards is not really Asterisk's fault.



On Fri, Aug 28, 2009 at 8:14 PM, Christensen Tom <paveraware at hotmail.com>wrote:

>  As a background, I ran an asterisk consulting company for about 3 years
> that I gave up on 2 years ago after repeatedly failing to achieve any sort
> of stability on any sort install over about 30 phones, I gave up.
>
> Maybe that was wrong, I am open to the possibility that I just didn't know
> enough and I was building things wrong, but I worked inside the asterisk
> code (which I feel is a hopeless mess), I implemented a few small custom
> features, anyway...
>
> I'm coming back into the VoIP space now, and I'm wondering what sort of
> issues can I expect in trying to pick up and learn freeswitch?  From what
> I've read on the website, it appears to have a much more sane architecture.
> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
> stable and worst general call quality was asterisk.  I constantly contended
> with strange call quality issues in asterisk, lots of echo (even with
> hardware echo cancellation cards), lots of jitter, lots of call break up
> (even on small systems with 10-20 users, using QoS on the network, and in
> general doing everything I could to prioritize voice over anything else).
>
> When I used Cisco call manager and broadsoft, the voice quality issues were
> basically non-existant, as long as the network was running QoS echo,
> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
> does freeswitch show a marked improvement over asterisk in this department?
> As long as you configure QoS and have hardware echo cancellation does it
> actually work reliably?
>
> Thanks for any additional information about freeswitch you can provide as
> well.  I am a software developer primarily by trade, but I do lots of
> consulting type work in the SME space and I've had a couple projects thrown
> to me that require some integration with a phone system, and I just can't in
> good conscience recommend asterisk anymore.
>
>
> -Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From asannucci at gmail.com  Fri Aug 28 15:05:17 2009
From: asannucci at gmail.com (bakko)
Date: Sat, 29 Aug 2009 00:05:17 +0200
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <7b197bef0908281302n43fa2c8aoc6d713f2681fedd5@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
	<87f2f3b90908281237k2fb599c2ue36d20b8807cf31d@mail.gmail.com>
	<d36c422fa99228a711fb7eba47c7e94f.squirrel@correo.nodo50.org>
	<CFCEE02A-A1EE-41E3-AC3D-C57F35990111@halo2.pl>
	<7b197bef0908281302n43fa2c8aoc6d713f2681fedd5@mail.gmail.com>
Message-ID: <5D224FBE6FC943AC826B02B5DB4E2166@voztovoice>

windows 95 :P

----- Original Message ----- 
From: "Giovanni Maruzzelli" <gmaruzz at celliax.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, August 28, 2009 10:02 PM
Subject: Re: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS


Vista!


On Fri, Aug 28, 2009 at 10:00 PM, Michal
Bielicki<michal.bielicki at halo2.pl> wrote:
> OpenSolaris :P
>
> Am 28.08.2009 um 21:50 schrieb Alberto Escudero:
>
>> Debian :)
>> -- Stopping junk mailers is good for the environment
>>
>>> On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa <
>>> testa at voicetechnology.com.br> wrote:
>>>
>>>> Hi folks,What linux distro would you suggest to deploy FS with django:
>>>> Centos 5.3 or Ubuntu LTS?
>>>>
>>> CentOS 5.3 FTW!
>>> -MC
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> Michal Bielicki
> Leiter der Niederlassung
> HaloKwadrat Sp. z o.o.
> Niederlassung Kleinmachnow
> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
> Ust.Id.: DE261885536
> Geschaeftsfuehrer: Aleksander Wiercinski
> Meiereifeld 2b, 14532 Kleinmachnow
> t. +49 33203 263220
> f. +49 33203 263229 sip. info at halokwadrat.de
> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
> Hauptgesch?ftsstelle:
> Halo Kwadrat Sp. z o.o.
> ul. Polna 46/14
> 00-644 Warszawa, Polen
> EIngetragen im HRB Warszawa, KRS 0000153539
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 



From diego.viola at gmail.com  Fri Aug 28 15:16:56 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 28 Aug 2009 22:16:56 +0000
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <4468a6770908281503gd7c003ewf44fd353f78c21e4@mail.gmail.com>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
	<4468a6770908281503gd7c003ewf44fd353f78c21e4@mail.gmail.com>
Message-ID: <86a32abc0908281516kef890fg681dd0f916c101d@mail.gmail.com>

You guys should put your experiences here :).

http://wiki.freeswitch.org/wiki/Testimonials

Diego

On Fri, Aug 28, 2009 at 10:03 PM, Ognjen Seslija <oseslija at gmail.com> wrote:

> I currently use FS for a managed PBX services for many companies. It's so
> configurable and extensible, I don't think I would change it for anything. I
> have Cisco CME and SCCP phones in my office all interconnected to FS doing
> logic and thinking. I also have Asterisks as PSTN PRI gateways. FS does all
> the job for me in the world, the other two are just used as protocol
> gateways. I wouldn't change FS for anything else now, after a year and the
> half of playing with it. I fully recommend it.
>
> Regards,
> Ognjen
>
> P.S: To be fair, the echo on the hw ec cards is not really Asterisk's
> fault.
>
>
>
> On Fri, Aug 28, 2009 at 8:14 PM, Christensen Tom <paveraware at hotmail.com>wrote:
>
>>  As a background, I ran an asterisk consulting company for about 3 years
>> that I gave up on 2 years ago after repeatedly failing to achieve any sort
>> of stability on any sort install over about 30 phones, I gave up.
>>
>> Maybe that was wrong, I am open to the possibility that I just didn't know
>> enough and I was building things wrong, but I worked inside the asterisk
>> code (which I feel is a hopeless mess), I implemented a few small custom
>> features, anyway...
>>
>> I'm coming back into the VoIP space now, and I'm wondering what sort of
>> issues can I expect in trying to pick up and learn freeswitch?  >From what
>> I've read on the website, it appears to have a much more sane architecture.
>> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
>> stable and worst general call quality was asterisk.  I constantly contended
>> with strange call quality issues in asterisk, lots of echo (even with
>> hardware echo cancellation cards), lots of jitter, lots of call break up
>> (even on small systems with 10-20 users, using QoS on the network, and in
>> general doing everything I could to prioritize voice over anything else).
>>
>> When I used Cisco call manager and broadsoft, the voice quality issues
>> were basically non-existant, as long as the network was running QoS echo,
>> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
>> does freeswitch show a marked improvement over asterisk in this department?
>> As long as you configure QoS and have hardware echo cancellation does it
>> actually work reliably?
>>
>> Thanks for any additional information about freeswitch you can provide as
>> well.  I am a software developer primarily by trade, but I do lots of
>> consulting type work in the SME space and I've had a couple projects thrown
>> to me that require some integration with a phone system, and I just can't in
>> good conscience recommend asterisk anymore.
>>
>>
>> -Tom
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tayeb.meftah at gmail.com  Fri Aug 28 17:17:19 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 29 Aug 2009 00:17:19 +0000
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
Message-ID: <4A98738F.2000001@gmail.com>

hello,
freeswitch is Awesome and perfect!
if you know XML, freeswitch use XML for there configuration file, no 
same to asterisk that trying to create a configuration language!
this unic featur let you integrate freeswitch with any Application that 
support and Parce XML including Web 2.0
freeswitch also support protocol transcoding, including SIP, H.323, 
Iax2, Skype and GTalk!
Real Time Feature: freeswitch support XML_Curl that let you configure 
freeswitch remotly from a web server that fetch data from a DB and 
return it to freeswitch in XML format
XMLRPC let you execute remote api command using your web browser or your 
XMLRPC application
about Event Socket, this let you build any type of application that talk 
to freeswitch or freeswitch talk to it in inbound and outbound mode
good luk!
Christensen Tom wrote:
> As a background, I ran an asterisk consulting company for about 3 
> years that I gave up on 2 years ago after repeatedly failing to 
> achieve any sort of stability on any sort install over about 30 
> phones, I gave up.
>
> Maybe that was wrong, I am open to the possibility that I just didn't 
> know enough and I was building things wrong, but I worked inside the 
> asterisk code (which I feel is a hopeless mess), I implemented a few 
> small custom features, anyway...
>
> I'm coming back into the VoIP space now, and I'm wondering what sort 
> of issues can I expect in trying to pick up and learn freeswitch?  
> From what I've read on the website, it appears to have a much more 
> sane architecture.  I've used Cisco, Broadsoft, and asterisk in the 
> past.  By far the least stable and worst general call quality was 
> asterisk.  I constantly contended with strange call quality issues in 
> asterisk, lots of echo (even with hardware echo cancellation cards), 
> lots of jitter, lots of call break up (even on small systems with 
> 10-20 users, using QoS on the network, and in general doing everything 
> I could to prioritize voice over anything else).
>
> When I used Cisco call manager and broadsoft, the voice quality issues 
> were basically non-existant, as long as the network was running QoS 
> echo, stutter, calls breaking up, just didn't happen.  So, I guess my 
> question is, does freeswitch show a marked improvement over asterisk 
> in this department?  As long as you configure QoS and have hardware 
> echo cancellation does it actually work reliably?
>
> Thanks for any additional information about freeswitch you can provide 
> as well.  I am a software developer primarily by trade, but I do lots 
> of consulting type work in the SME space and I've had a couple 
> projects thrown to me that require some integration with a phone 
> system, and I just can't in good conscience recommend asterisk anymore.
>
>
> -Tom
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus 
> signature database 4378 (20090828) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   





__________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From tomabroad at gmail.com  Fri Aug 28 16:35:46 2009
From: tomabroad at gmail.com (tom)
Date: Fri, 28 Aug 2009 19:35:46 -0400
Subject: [Freeswitch-users] alternative to java-asterisk
Message-ID: <6f7c60c40908281635i2618a23p65a879302ff113f4@mail.gmail.com>

hi,

with the java-asterisk bridge its possible to to AMI-stuff. is thi as well
possible with freeswitch? if so, are there any tutorials, docu etc. out
there?
thx
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From msc at freeswitch.org  Fri Aug 28 16:53:25 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 16:53:25 -0700
Subject: [Freeswitch-users] alternative to java-asterisk
In-Reply-To: <6f7c60c40908281635i2618a23p65a879302ff113f4@mail.gmail.com>
References: <6f7c60c40908281635i2618a23p65a879302ff113f4@mail.gmail.com>
Message-ID: <87f2f3b90908281653s6d7c8f72i43feb8eb739f46db@mail.gmail.com>

On Fri, Aug 28, 2009 at 4:35 PM, tom <tomabroad at gmail.com> wrote:

> hi,
>
> with the java-asterisk bridge its possible to to AMI-stuff. is thi as well
> possible with freeswitch? if so, are there any tutorials, docu etc. out
> there?
> thx


The AMI in Asterisk is analogous to the event socket in FreeSWITCH. (The
event socket is extremely powerful.) The FS developers have created ESL -
the event socket library, which abstracts away some of the mundane parts of
creating a socket connection to FS. I believe they added provisional support
for Java. We don't have a tutorial per se on the ESL, especially for Java.
However, we sure could use an adventurous Java programmer to roll up his
sleeves and help test & document all this stuff. :)

Check out src/libs/esl in the FS source tree and the event socket stuff on
the wiki:
http://wiki.freeswitch.org/wiki/Event_Socket
http://wiki.freeswitch.org/wiki/Event_Socket_Library

Have fun!
-MC
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From sdjernes at gmail.com  Fri Aug 28 18:35:02 2009
From: sdjernes at gmail.com (Shawn L. Djernes)
Date: Fri, 28 Aug 2009 20:35:02 -0500
Subject: [Freeswitch-users] Group Call
In-Reply-To: <87f2f3b90908281130g1e33668fs44b817f9c622641c@mail.gmail.com>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
	<5ff3e3a70908251015ue88b8b9p7a5e091d15a6b5ae@mail.gmail.com>
	<87f2f3b90908281130g1e33668fs44b817f9c622641c@mail.gmail.com>
Message-ID: <5ff3e3a70908281835g3983c7d0ma8716629b45b5f78@mail.gmail.com>

Here is a copy of the data from the console at log 7
freeswitch at internal> group_call
sdjernes+A at ewr.djernes.net<sdjernes%2BA at ewr.djernes.net>
-ERR no reply

freeswitch at internal> group_call sdjernes at ewr.djernes.net
error/NO_ROUTE_DESTINATION
freeswitch at internal> group_call
sdjernes+F at ewr.djernes.net<sdjernes%2BF at ewr.djernes.net>
-ERR no reply

freeswitch at internal> group_call
sdjernes+A at freeswitch.djernes.net<sdjernes%2BA at freeswitch.djernes.net>
-ERR no reply

freeswitch at internal>

On Fri, Aug 28, 2009 at 1:30 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Aug 25, 2009 at 10:15 AM, Shawn L. Djernes <sdjernes at gmail.com>wrote:
>
>> Woops,
>> That line was an attempt to try something from an older diaplan example.
>>  Copied and pasted from a file on my computer not the server.  The actual
>> line is:
>>
>> <action application="bridge"
>> data="{ignore_early_media=true}${group_call(sdjernes+A@
>> ${domain_name})}"/>
>>
>> The +A is to ring all phones in the group at the same time.
>>
>>
> Can you confirm what you see on the CLI when you do this:
> group_call sdjernes+A at ewr.djernes.net <sdjernes%2BA at ewr.djernes.net>
>
> Let's make sure the group is properly defined, etc. before doing anything
> else.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Shawn L. Djernes
SD Consulting
shawn at djernes.org | sdjernes at gmail.com
MSN: wizardwlf at hotmail.com
402.345.7734 | 402.350.6973 Cell
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From diego.viola at gmail.com  Fri Aug 28 19:34:32 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 29 Aug 2009 02:34:32 +0000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
Message-ID: <86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>

Yes, FreeSWITCH is a system that you can trust 100%. I have switched my
Asterisk servers to FreeSWITCH and have peace now.

If I were you I would get rid of Asterisk and use FreeSWITCH, FS will handle
all what you want very well.

And I agree with David, fail-over is kinda irrelevant since the FS doesn't
crash like Asterisk does.

Diego


On Fri, Aug 28, 2009 at 6:52 PM, Raimund Sacherer <rs at runsolutions.com>wrote:

> Hello List,
>
> I have read the current thread about scalability and I would need some
> advice about a callcenter setup:
>
> First where I come from:
> I have lot's of problems with an asterisk solution. I have regular
> crash's and lock-ups, with downgrading and other stuff i got it
> somewhat stable, but have nevertheless regular hickups. I am desperate
> and want to get rid of asterisk and I hope that freeSwitch will
> provide me with a more stable solution.
>
>
> Our Setup (really nothing special):
> * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3
> GIG of RAM, Xircom analog switch connected to mobile stations for 4
> different providers, Digium 4port cards TP400<something>
> * 8 queues
> * ~60 agents (which logon, logoff, pause, unpause), not more than 40
> concurrently online
> * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
> * Music on Hold in the call-queues
> * No special announcement
> * Transfers between calls in queues and different agents as well as
> non agents (i mention this because we have transfer related chrashes
> in asterisk)
>
> The current Problems:
> * Lockups with different causes (ranging from calls not terminated to
> heavy thread locking through the AMI interface)
> * Crashes and library aborts (pthread, libc, crashes related to music
> on hold, app_queue, transfers)
>
> We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are
> now back to 1.4.21.2 (stock debian) as anything beyond that is for
> whatever reason highly unstable for our szenario. Maybe we should have
> been segmenting the box into one asterisk dedicted to talking to the
> hardware, one especial for queue/sip handling, i do not know. (all
> issues are well documented in issues.asterisk.org, but it seems to be
> very, very difficult to get to the bottom of them as they exist since
> 1.4.23 as it seems and are open until know and not fixable since month.)
>
>
> Now, I really would appreciate some success-stories on how you guys
> managed to get a stable pbx system with freeSWITCH in regard of HA and
> scalability:
>
> * How to segment freeSWITCH? Or is it stable enough to handle all in
> one for such a szenario as outlined above?
> * What would be the best strategie for High Availability / Failover?
>        -> I read in the WIKI (featurelist) that Livemigration of calls from
> one box to another should be possible?
>        -> I was thinking about using memcached for storing all state
> information so another freeswitch box can takeover calls from the
> first box if it dies, is this possible? If so, how?
>        -> Is there anotherway to somehow configure freeSWITCH that in the
> event of a crash i do not loose the current established calls?
>
> Basically I just want a stable PBX where I do not have to fear every
> day it will core-dump or abort or Lock up. Is freeSWITCH mature enough
> so i can sleep at night for at least 3 month without a crash?
>
>
>
> Thank you for your Time and help in advance, and I am more than
> willing to take all the information gathered here and create a wiki
> page to help other people with the same questions/problems.
>
> best
> Ray
>
>
> --
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Fri Aug 28 19:37:12 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 29 Aug 2009 02:37:12 +0000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <1251493313.4438.8.camel@dk-d820>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<1251493313.4438.8.camel@dk-d820>
Message-ID: <86a32abc0908281937g4eea9d83x667bd3455f7a45bc@mail.gmail.com>

Hi David,

What have you used on FS for call center, mod_fifo?

Can you describe your experience with that, I'm currently interested in call
center + FS scenario.

Diego

On Fri, Aug 28, 2009 at 9:01 PM, David Knell <dave at 3c.co.uk> wrote:

> Hi Raimund,
>
> One FreeSWITCH box will be quite enough to handle the call volumes that
> you're talking about, and it ought to be much more stable than the
> Asterisk solution which you've outlined below.
>
> It's probably best to forget about live failover without calls dropping
> - this isn't something that's supported, and there'd be a lot of work to
> do to develop code to keep two boxed in sync.  Once you get used to a
> stable solution - i.e. something which doesn't crash - then live
> failover, HA, etc., will seem somewhat irrelevant.
>
> I recently did some work on an FS-based call center solution - drop me a
> note if I could be of any help with yours.
>
> Cheers --
>
> Dave
>
> > Hello List,
> >
> > I have read the current thread about scalability and I would need some
> > advice about a callcenter setup:
> >
> > First where I come from:
> > I have lot's of problems with an asterisk solution. I have regular
> > crash's and lock-ups, with downgrading and other stuff i got it
> > somewhat stable, but have nevertheless regular hickups. I am desperate
> > and want to get rid of asterisk and I hope that freeSwitch will
> > provide me with a more stable solution.
> >
> >
> > Our Setup (really nothing special):
> > * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3
> > GIG of RAM, Xircom analog switch connected to mobile stations for 4
> > different providers, Digium 4port cards TP400<something>
> > * 8 queues
> > * ~60 agents (which logon, logoff, pause, unpause), not more than 40
> > concurrently online
> > * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
> > * Music on Hold in the call-queues
> > * No special announcement
> > * Transfers between calls in queues and different agents as well as
> > non agents (i mention this because we have transfer related chrashes
> > in asterisk)
> >
> > The current Problems:
> > * Lockups with different causes (ranging from calls not terminated to
> > heavy thread locking through the AMI interface)
> > * Crashes and library aborts (pthread, libc, crashes related to music
> > on hold, app_queue, transfers)
> >
> > We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are
> > now back to 1.4.21.2 (stock debian) as anything beyond that is for
> > whatever reason highly unstable for our szenario. Maybe we should have
> > been segmenting the box into one asterisk dedicted to talking to the
> > hardware, one especial for queue/sip handling, i do not know. (all
> > issues are well documented in issues.asterisk.org, but it seems to be
> > very, very difficult to get to the bottom of them as they exist since
> > 1.4.23 as it seems and are open until know and not fixable since month.)
> >
> >
> > Now, I really would appreciate some success-stories on how you guys
> > managed to get a stable pbx system with freeSWITCH in regard of HA and
> > scalability:
> >
> > * How to segment freeSWITCH? Or is it stable enough to handle all in
> > one for such a szenario as outlined above?
> > * What would be the best strategie for High Availability / Failover?
> >       -> I read in the WIKI (featurelist) that Livemigration of calls
> from
> > one box to another should be possible?
> >       -> I was thinking about using memcached for storing all state
> > information so another freeswitch box can takeover calls from the
> > first box if it dies, is this possible? If so, how?
> >       -> Is there anotherway to somehow configure freeSWITCH that in the
> > event of a crash i do not loose the current established calls?
> >
> > Basically I just want a stable PBX where I do not have to fear every
> > day it will core-dump or abort or Lock up. Is freeSWITCH mature enough
> > so i can sleep at night for at least 3 month without a crash?
> >
> >
> >
> > Thank you for your Time and help in advance, and I am more than
> > willing to take all the information gathered here and create a wiki
> > page to help other people with the same questions/problems.
> >
> > best
> > Ray
> >
> >
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dave at 3c.co.uk  Fri Aug 28 20:49:17 2009
From: dave at 3c.co.uk (David Knell)
Date: Sat, 29 Aug 2009 06:49:17 +0300
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <86a32abc0908281937g4eea9d83x667bd3455f7a45bc@mail.gmail.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<1251493313.4438.8.camel@dk-d820>
	<86a32abc0908281937g4eea9d83x667bd3455f7a45bc@mail.gmail.com>
Message-ID: <1251517757.4438.25.camel@dk-d820>

Hi Diego,

We didn't use mod_fifo; we built our own queues using an application
hanging off the event socket.  This was partly down to typical
programmer hubris, and partly to allow us to do things that mod_fifo
might not without either (a) trawling through mod_fifo, or (b) pestering
Anthony.

As a brief outline:
- operators each live in their own conference
- supervisors can add themselves to these conferences for monitoring or
coaching
- callers sit in queues for the various skills that the operators have.

When an operator is free, look at each queue for the skills that the
operator has, and choose the caller who's been waiting longest.

I used a one-thread-per-call model, so there's a little bit of thought
needed to get the access to shared structures right but, essentially,
given the low frequency of access to them and the fact that nothing was
time-critical at the sub-second level, I just wrapped all of the
accesses in a single global mutex, which was easy and is pretty
foolproof.  The latter is important, given that it was me writing the
stuff ;-)

Cheers --

Dave

> Hi David,
> 
> What have you used on FS for call center, mod_fifo?
> 
> Can you describe your experience with that, I'm currently interested
> in call center + FS scenario.
> 
> Diego
> 
> On Fri, Aug 28, 2009 at 9:01 PM, David Knell <dave at 3c.co.uk> wrote:
>         Hi Raimund,
>         
>         One FreeSWITCH box will be quite enough to handle the call
>         volumes that
>         you're talking about, and it ought to be much more stable than
>         the
>         Asterisk solution which you've outlined below.
>         
>         It's probably best to forget about live failover without calls
>         dropping
>         - this isn't something that's supported, and there'd be a lot
>         of work to
>         do to develop code to keep two boxed in sync.  Once you get
>         used to a
>         stable solution - i.e. something which doesn't crash - then
>         live
>         failover, HA, etc., will seem somewhat irrelevant.
>         
>         I recently did some work on an FS-based call center solution -
>         drop me a
>         note if I could be of any help with yours.
>         
>         Cheers --
>         
>         Dave
>         
>         
>         > Hello List,
>         >
>         > I have read the current thread about scalability and I would
>         need some
>         > advice about a callcenter setup:
>         >
>         > First where I come from:
>         > I have lot's of problems with an asterisk solution. I have
>         regular
>         > crash's and lock-ups, with downgrading and other stuff i got
>         it
>         > somewhat stable, but have nevertheless regular hickups. I am
>         desperate
>         > and want to get rid of asterisk and I hope that freeSwitch
>         will
>         > provide me with a more stable solution.
>         >
>         >
>         > Our Setup (really nothing special):
>         > * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack
>         server, 3
>         > GIG of RAM, Xircom analog switch connected to mobile
>         stations for 4
>         > different providers, Digium 4port cards TP400<something>
>         > * 8 queues
>         > * ~60 agents (which logon, logoff, pause, unpause), not more
>         than 40
>         > concurrently online
>         > * ~ 7K - 9K calls (well, CDR entries) a day (not that much
>         for a bpx)
>         > * Music on Hold in the call-queues
>         > * No special announcement
>         > * Transfers between calls in queues and different agents as
>         well as
>         > non agents (i mention this because we have transfer related
>         chrashes
>         > in asterisk)
>         >
>         > The current Problems:
>         > * Lockups with different causes (ranging from calls not
>         terminated to
>         > heavy thread locking through the AMI interface)
>         > * Crashes and library aborts (pthread, libc, crashes related
>         to music
>         > on hold, app_queue, transfers)
>         >
>         > We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5,
>         1.4.26 and are
>         > now back to 1.4.21.2 (stock debian) as anything beyond that
>         is for
>         > whatever reason highly unstable for our szenario. Maybe we
>         should have
>         > been segmenting the box into one asterisk dedicted to
>         talking to the
>         > hardware, one especial for queue/sip handling, i do not
>         know. (all
>         > issues are well documented in issues.asterisk.org, but it
>         seems to be
>         > very, very difficult to get to the bottom of them as they
>         exist since
>         > 1.4.23 as it seems and are open until know and not fixable
>         since month.)
>         >
>         >
>         > Now, I really would appreciate some success-stories on how
>         you guys
>         > managed to get a stable pbx system with freeSWITCH in regard
>         of HA and
>         > scalability:
>         >
>         > * How to segment freeSWITCH? Or is it stable enough to
>         handle all in
>         > one for such a szenario as outlined above?
>         > * What would be the best strategie for High Availability /
>         Failover?
>         >       -> I read in the WIKI (featurelist) that Livemigration
>         of calls from
>         > one box to another should be possible?
>         >       -> I was thinking about using memcached for storing
>         all state
>         > information so another freeswitch box can takeover calls
>         from the
>         > first box if it dies, is this possible? If so, how?
>         >       -> Is there anotherway to somehow configure freeSWITCH
>         that in the
>         > event of a crash i do not loose the current established
>         calls?
>         >
>         > Basically I just want a stable PBX where I do not have to
>         fear every
>         > day it will core-dump or abort or Lock up. Is freeSWITCH
>         mature enough
>         > so i can sleep at night for at least 3 month without a
>         crash?
>         >
>         >
>         >
>         > Thank you for your Time and help in advance, and I am more
>         than
>         > willing to take all the information gathered here and create
>         a wiki
>         > page to help other people with the same questions/problems.
>         >
>         > best
>         > Ray
>         >
>         >
>         --
>         
>         David Knell, Director, 3C Limited
>         T: +44 20 3298 2000
>         E: dave at 3c.co.uk
>         W: http://www.3c.co.uk
>         
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From nagalenoj at gmail.com  Fri Aug 28 21:26:44 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Fri, 28 Aug 2009 21:26:44 -0700 (PDT)
Subject: [Freeswitch-users] How can it be done with freeswitch?
In-Reply-To: <87f2f3b90908280907w4812a8dbr7a97385f9afdd376@mail.gmail.com>
References: <25184845.post@talk.nabble.com>
	<87f2f3b90908280907w4812a8dbr7a97385f9afdd376@mail.gmail.com>
Message-ID: <25199956.post@talk.nabble.com>


Thanks for your help. Is it the only possible way.?
Else, will it be done with the functions listed in the below url,
http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes

I was trying to do with these classes and functions.


mercutioviz wrote:
> 
> Nagalenoj,
> 
> This is definitely possible to do with FreeSWITCH. However, it is not
> exactly simple. You have some learning to do. My recommendation is to
> learn
> about event socket. Most likely you'll need to use outbound event socket.
> (Read about event socket and ESL, the event socket library, on the wiki.)
> 
> I also recommend that you look at some of the example scripts in
> src/libs/esl/perl/ which show some of the ways to handle calls with a
> script. You've already done a good job of breaking down the steps, now
> experiment with handling each of these steps with a script. We don't have
> extensive Perl script examples which means we would love to see your code
> when you have it all figured out! :)
> 
> -MC
> 
> On Thu, Aug 27, 2009 at 11:44 PM, Nagalenoj <nagalenoj at gmail.com> wrote:
> 
>>
>> Dear friends,
>>    I am using freeswitch for the last one month and trying to write perl
>> scripts to do some tasks.
>> I am trying a new thing, which is something that I don't how to do it and
>> I
>> don't whether it can be done with freeswitch.   my requirement is as
>> follows,
>>             A calls B[This extension has to execute a script].
>>             The script has to do the following,
>>              * Cut the call and callback to the number.
>>              * Get a phone number[DTMF] from A.
>>              * Dial the number(C) and bridge A and C.
>>              * When C cuts the call, the control has to comeback to
>> script.
>>              * Ask another number to A to dial out, get another phone
>> number[DTMF] from A.
>>              * Dial the number(D) and bridge A and D.
>>
>> Please, help me to do this. Is there any other documentation to read
>> about
>> the perl freeswitch functions.
>> --
>> View this message in context:
>> http://www.nabble.com/How-can-it-be-done-with-freeswitch--tp25184845p25184845.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/How-can-it-be-done-with-freeswitch--tp25184845p25199956.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From kawarod at laposte.net  Fri Aug 28 22:43:24 2009
From: kawarod at laposte.net (rod)
Date: Sat, 29 Aug 2009 09:43:24 +0400
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
In-Reply-To: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>
References: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>
Message-ID: <4A98BFFC.9060003@laposte.net>

Ok,

I found what's happening. I probably did some change on the wiki to 
start reflect the new configuration, without having time enough to check 
the configuration. There is a mistake in your dialplan configuration. 
You should put this instead:

wrong line:

<action application="set" data="ROUTE_GW=${sip_redirect_contact_user_0}"/>

please correct with:
<action application="set" data="ROUTE_GW=${sip_redirect_contact_host_0}"/>


This should match PEER_01 in dialplan instead of trying matching 
France at PEER_01.

Let me know if this is right now.

rod

Hristo Benev a ?crit :
>  Hello Rod,
>
> I did the change.
>
> Here is extract of console:
> -----------------------------------------------------------------------------------------
> variable_continue_on_fail: [true]
> variable_sip_h_X-ROUTE: [LOOKUP]
> variable_export_vars: [sip_h_X-ROUTE]
> variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20]
> variable_sip_redirect_contact_0: [sip:France at PEER_01]
> variable_sip_redirected_to: [sip:France at PEER_01]
> variable_sip_redirect_contact_user_0: [France] <-------------------------------------------
> variable_sip_redirect_contact_host_0: [PEER_01]
> variable_sip_redirect_dialstring_0: [sofia/internal/sip:France at PEER_01]
> variable_sip_redirect_dialstring: [sofia/internal/sip:France at PEER_01]
> variable_proto_specific_hangup_cause: [sip:503]
> variable_sip_hangup_phrase: [DNS Error]
> variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE]
> variable_ROUTE_GW: [France]
> variable_AREA: [France]
> variable_current_application: [info]
> ------------------------------------------------------------------------------------------
> I have different value it is actually the description field as shown here:
>
> --------------------------------
>  /opt/kamailio/sbin/kamctl cr show
> cr carrier names
> +----+---------+
> | id | carrier |
> +----+---------+
> |  1 | default |
> +----+---------+
> cr domain names
> +----+---------+
> | id | domain  |
> +----+---------+
> |  1 | default |
> +----+---------+
> cr routes
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description |
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> |  1 |       1 |      1 | 1000        |     0 |    0 |    1 |     0 | PEER_01      |                |                | France      |
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> -----------------------------------------
>
> And here is what I have in kamailio:
> -------------------------------------------------------------------------------------
> ####### Routing Logic ########
>
>
> # main request routing logic
>
> route{
>
>         if (!mf_process_maxfwd_header("10")) {
>                 sl_send_reply("483","Too Many Hops");
>                 exit;
>         }
>
>         t_check_trans();
>
>         if ($rU==NULL) {
>                 # request with no Username in RURI
>                 sl_send_reply("484","Address Incomplete");
>                 exit;
>         }
>
>         # LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP
>         if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){
>                 if(!cr_route("default", "default", "$rU", "$rU", "call_id","$avp(s:route_desc)")){
>                      #xlog("ROUTING FAILED: no route found for $rU");
>                      sl_send_reply("604", "Unable to route this call");
>                      exit;
>                 } else {
>                      xlog("LOOKUP FOUND: $rd $avp(s:route_desc)");
>                      avp_pushto("$ru/username", "$avp(s:route_desc)");
>                      sl_send_reply("302", "$rd");
>                      exit;
>                 }
>         }
> }
> ----------------------------------------------------------------------------------
>
> Another question...
> In that part of FreeSwitch dialplan.xml
>
> -----------------------------------
>   <context name="ROUTING">
>
>     <extension name="PEER_01">
>       <condition field="${sip_h_X-ROUTE}" expression="PEER_01">
>         <action application="set" data="hangup_after_bridge=true"/>
>
> -----------------------------------
>
> X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the regex is true.
>
>
> As for thanks - for sure by default they are also for the developers of both apps.
> I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning so it will be nice to have something working to start with.
>
>
>
>  >-------- ?????????? ????? --------
>  >??:  rod 
>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >??: freeswitch-users at lists.freeswitch.org
>  >????????? ??: ?????, 2009, ?????? 28 09:54:00 EEST
>
>  >Hello,
>  >
>  >the trace seems good.
>  >If you check the answer from Kamailio, you'll see that Kamailio answers 
>  >with "302 PEER_01".
>  >
>  >As Michael Collins stated before, you can get the variable containing 
>  >"PEER_01", then this variable is stored in a custom variable.
>  >
>  >In your dialplan, may you please add:
>  >
>  >, just before the transfer line, eg:
>  >
>  > 
>  >         
>  >         
>  >         
>  >         
>  >         
>  >         
>  >	 
>  >         
>  >      
>  >
>  >
>  >Using application Info, you'll see on the console (or CDR) the list of 
>  >variables used for this call.  You should see the content of 
>  >"${sip_redirect_contact_user_0}" that should contain  the value 
>  >"PEER_01". Please check this and let me know.
>  >
>  >For the new configuration file, no problem for sharing. But as I wrote 
>  >on the wiki page, I worked on this setup cause LCR module was not 
>  >available when I start working on FS, nor I'm a good programmer to write 
>  >a server side HTTP script (used by xml_curl) that could scale to my needs.
>  >
>  >I enhanced a bit this configuration with support for fallback routing, 
>  >almost realtime graph (every minutes using www.cacti.net and some 
>  >functions in FS like limit_hash) of number of concurrent calls per AREA, 
>  >PEER...
>  >
>  >Thanks for "the good tutorial", but don't forget the dev team who did 
>  >this great product ;-)
>  >
>  >rod.
>  >
>  >
>  >Hristo Benev a ?crit :
>  >>  I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).
>  >>
>  >> If you need additional info I'll provide it.
>  >>
>  >> Here is trace:
>  >>
>  >> ngrep -d any -nn -i '1000' port 5062 -W byline
>  >> interface: any
>  >> filter: (ip or ip6) and ( port 5062 )
>  >> match: 1000
>  >> #
>  >> U 10.10.10.10:5090 -> 127.0.0.1:5062
>  >> INVITE sip:1000 at 127.0.0.1:5062 SIP/2.0.
>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
>  >> Max-Forwards: 69.
>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
>  >> To: .
>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
>  >> CSeq: 119574771 INVITE.
>  >> Contact: .
>  >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
>  >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
>  >> Supported: timer, precondition, path, replaces.
>  >> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
>  >> Content-Type: application/sdp.
>  >> Content-Disposition: session.
>  >> Content-Length: 429.
>  >> X-ROUTE: LOOKUP.
>  >> Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off.
>  >> .
>  >> v=0.
>  >> o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
>  >> s=FreeSWITCH.
>  >> c=IN IP4 10.10.10.10.
>  >> t=0 0.
>  >> m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
>  >> a=rtpmap:0 PCMU/8000.
>  >> a=rtpmap:115 G7221/32000.
>  >> a=fmtp:115 bitrate=48000.
>  >> a=rtpmap:107 G7221/16000.
>  >> a=fmtp:107 bitrate=32000.
>  >> a=rtpmap:9 G722/8000.
>  >> a=rtpmap:8 PCMA/8000.
>  >> a=rtpmap:3 GSM/8000.
>  >> a=rtpmap:101 telephone-event/8000.
>  >> a=fmtp:101 0-16.
>  >> a=rtpmap:13 CN/8000.
>  >> a=ptime:20.
>  >>
>  >> #
>  >> U 127.0.0.1:5062 -> 10.10.10.10:5090
>  >> SIP/2.0 302 PEER_01.
>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
>  >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
>  >> CSeq: 119574771 INVITE.
>  >> Contact: sip:France at PEER_01.
>  >> Server: Kamailio (1.5.2-notls (i386/linux)).
>  >> Content-Length: 0.
>  >> .
>  >>
>  >> #
>  >> U 10.10.10.10:5090 -> 127.0.0.1:5062
>  >> ACK sip:1000 at 127.0.0.1:5062 SIP/2.0.
>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
>  >> Max-Forwards: 69.
>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
>  >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
>  >> CSeq: 119574771 ACK.
>  >> Content-Length: 0.
>  >> .
>  >>
>  >>
>  >>
>  >>
>  >>
>  >>  >-------- ?????????? ????? --------
>  >>  >??:  Hristo Benev 
>  >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  >??: freeswitch-users at lists.freeswitch.org
>  >>  >????????? ??: ?????????, 2009, ?????? 27 14:58:53 EEST
>  >>
>  >>  > 
>  >>  >Bojnour,
>  >>  >
>  >>  >I'll send a trace ASAP.
>  >>  >
>  >>  >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
>  >>  >
>  >>  >I'm really interested to see the updates of the project.
>  >>  >
>  >>  >Thank you for the good tutorial.
>  >>  >
>  >>  >Hristo
>  >>  >
>  >>  >
>  >>  > >-------- ?????????? ????? --------
>  >>  > >??:  rod 
>  >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  > >??: freeswitch-users at lists.freeswitch.org
>  >>  > >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST
>  >>  >
>  >>  > >Hi Hristo,
>  >>  > >
>  >>  > >I'm the author of this setup and wiki page. I did a lot of modifications 
>  >>  > >on this setup (alternative routing if failure essentially) but don't 
>  >>  > >have too much time to update the wiki.
>  >>  > >
>  >>  > >May you please send me an ngrep trace when you call 1000:
>  >>  > >
>  >>  > >ngrep -d any -nn -i '1000' port 5060 -W byline 
>  >>  > >
>  >>  > >I will check what's happening.
>  >>  > >Do you have an entry for 1000 in your mysql database ?
>  >>  > >
>  >>  > >regards,
>  >>  > >rod
>  >>  > >
>  >>  > >Hristo Benev a ?crit :
>  >>  > >>  
>  >>  > >> It seems that the problem is on kamailio configuration.
>  >>  > >> Will ask on their list.
>  >>  > >>
>  >>  > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
>  >>  > >>
>  >>  > >> I have it setup as friend in asterisk, but still ???
>  >>  > >>
>  >>  > >> Any ideas?
>  >>  > >>
>  >>  > >> Thanks,
>  >>  > >>
>  >>  > >>
>  >>  > >>  >-------- ?????????? ????? --------
>  >>  > >>  >??:  Hristo Benev 
>  >>  > >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  > >>  >??: freeswitch-users at lists.freeswitch.org
>  >>  > >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
>  >>  > >>
>  >>  > >>  > I think that the problem is here:
>  >>  > >>  >-------------------------
>  >>  > >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
>  >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
>  >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
>  >>  > >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
>  >>  > >>  >--------------------------
>  >>  > >>  >
>  >>  > >>  >Actually Regex FAIL
>  >>  > >>  >
>  >>  > >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
>  >>  > >>  >Here is my  default.xml:
>  >>  > >>  >----------------
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >  
>  >>  > >>  >
>  >>  > >>  >     
>  >>  > >>  >      
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >      
>  >>  > >>  >    
>  >>  > >>  >
>  >>  > >>  >   
>  >>  > >>  >
>  >>  > >>  >  
>  >>  > >>  >
>  >>  > >>  >    
>  >>  > >>  >      
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >      
>  >>  > >>  >    
>  >>  > >>  >
>  >>  > >>  >  
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >--------------------------
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  > >-------- ?????????? ????? --------
>  >>  > >>  > >??:  Brian West 
>  >>  > >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  > >>  > >??: freeswitch-users at lists.freeswitch.org
>  >>  > >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
>  >>  > >>  >
>  >>  > >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
>  >>  > >>  >You have to process all 302's in the dialplan. 
>  >>  > >>  >Set this on your sofia profile
>  >>  > >>  >You can set these variables sip_redirect_profile,
>  >>  > >>  >sip_redirect_context,
>  >>  > >>  >sip_redirect_dialplan,
>  >>  > >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
>  >>  > >>  >sip_redirected_to,
>  >>  > >>  >sip_redirect_contact_user_%d,
>  >>  > >>  >sip_redirect_contact_host_%d,
>  >>  > >>  >sip_redirect_contact_params_%d,
>  >>  > >>  >sip_redirect_dialstring_%d,
>  >>  > >>  >sip_redirect_dialstring,
>  >>  > >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
>  >>  > >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
>  >>  > >>  > >
>  >>  > >>  >
>  >>  > >>  >_______________________________________________
>  >>  > >>  >FreeSWITCH-users mailing list
>  >>  > >>  >FreeSWITCH-users at lists.freeswitch.org
>  >>  > >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  > >>  >http://www.freeswitch.org
>  >>  > >>  >
>  >>  > >>
>  >>  > >> _______________________________________________
>  >>  > >> FreeSWITCH-users mailing list
>  >>  > >> FreeSWITCH-users at lists.freeswitch.org
>  >>  > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  > >> http://www.freeswitch.org
>  >>  > >>   
>  >>  > >
>  >>  > >_______________________________________________
>  >>  > >FreeSWITCH-users mailing list
>  >>  > >FreeSWITCH-users at lists.freeswitch.org
>  >>  > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  > >http://www.freeswitch.org
>  >>  > >
>  >>  >
>  >>  >_______________________________________________
>  >>  >FreeSWITCH-users mailing list
>  >>  >FreeSWITCH-users at lists.freeswitch.org
>  >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  >http://www.freeswitch.org
>  >>  >
>  >>
>  >> _______________________________________________
>  >> FreeSWITCH-users mailing list
>  >> FreeSWITCH-users at lists.freeswitch.org
>  >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >> http://www.freeswitch.org
>  >>   
>  >
>  >_______________________________________________
>  >FreeSWITCH-users mailing list
>  >FreeSWITCH-users at lists.freeswitch.org
>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >http://www.freeswitch.org
>  >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From kawarod at laposte.net  Fri Aug 28 22:56:22 2009
From: kawarod at laposte.net (rod)
Date: Sat, 29 Aug 2009 09:56:22 +0400
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
In-Reply-To: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>
References: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>
Message-ID: <4A98C306.1060801@laposte.net>

you are right for the regex. This is part of an old setup, correct with:

<extension name="PEER_01">
      <condition field="${ROUTE_GW}" expression="PEER_01">
	<action application="set" data="hangup_after_bridge=true"/>



Hristo Benev a ?crit :
>  Hello Rod,
>
> I did the change.
>
> Here is extract of console:
> -----------------------------------------------------------------------------------------
> variable_continue_on_fail: [true]
> variable_sip_h_X-ROUTE: [LOOKUP]
> variable_export_vars: [sip_h_X-ROUTE]
> variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20]
> variable_sip_redirect_contact_0: [sip:France at PEER_01]
> variable_sip_redirected_to: [sip:France at PEER_01]
> variable_sip_redirect_contact_user_0: [France] <-------------------------------------------
> variable_sip_redirect_contact_host_0: [PEER_01]
> variable_sip_redirect_dialstring_0: [sofia/internal/sip:France at PEER_01]
> variable_sip_redirect_dialstring: [sofia/internal/sip:France at PEER_01]
> variable_proto_specific_hangup_cause: [sip:503]
> variable_sip_hangup_phrase: [DNS Error]
> variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE]
> variable_ROUTE_GW: [France]
> variable_AREA: [France]
> variable_current_application: [info]
> ------------------------------------------------------------------------------------------
> I have different value it is actually the description field as shown here:
>
> --------------------------------
>  /opt/kamailio/sbin/kamctl cr show
> cr carrier names
> +----+---------+
> | id | carrier |
> +----+---------+
> |  1 | default |
> +----+---------+
> cr domain names
> +----+---------+
> | id | domain  |
> +----+---------+
> |  1 | default |
> +----+---------+
> cr routes
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description |
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> |  1 |       1 |      1 | 1000        |     0 |    0 |    1 |     0 | PEER_01      |                |                | France      |
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> -----------------------------------------
>
> And here is what I have in kamailio:
> -------------------------------------------------------------------------------------
> ####### Routing Logic ########
>
>
> # main request routing logic
>
> route{
>
>         if (!mf_process_maxfwd_header("10")) {
>                 sl_send_reply("483","Too Many Hops");
>                 exit;
>         }
>
>         t_check_trans();
>
>         if ($rU==NULL) {
>                 # request with no Username in RURI
>                 sl_send_reply("484","Address Incomplete");
>                 exit;
>         }
>
>         # LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP
>         if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){
>                 if(!cr_route("default", "default", "$rU", "$rU", "call_id","$avp(s:route_desc)")){
>                      #xlog("ROUTING FAILED: no route found for $rU");
>                      sl_send_reply("604", "Unable to route this call");
>                      exit;
>                 } else {
>                      xlog("LOOKUP FOUND: $rd $avp(s:route_desc)");
>                      avp_pushto("$ru/username", "$avp(s:route_desc)");
>                      sl_send_reply("302", "$rd");
>                      exit;
>                 }
>         }
> }
> ----------------------------------------------------------------------------------
>
> Another question...
> In that part of FreeSwitch dialplan.xml
>
> -----------------------------------
>   <context name="ROUTING">
>
>     <extension name="PEER_01">
>       <condition field="${sip_h_X-ROUTE}" expression="PEER_01">
>         <action application="set" data="hangup_after_bridge=true"/>
>
> -----------------------------------
>
> X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the regex is true.
>
>
> As for thanks - for sure by default they are also for the developers of both apps.
> I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning so it will be nice to have something working to start with.
>
>
>
>  >-------- ?????????? ????? --------
>  >??:  rod 
>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >??: freeswitch-users at lists.freeswitch.org
>  >????????? ??: ?????, 2009, ?????? 28 09:54:00 EEST
>
>  >Hello,
>  >
>  >the trace seems good.
>  >If you check the answer from Kamailio, you'll see that Kamailio answers 
>  >with "302 PEER_01".
>  >
>  >As Michael Collins stated before, you can get the variable containing 
>  >"PEER_01", then this variable is stored in a custom variable.
>  >
>  >In your dialplan, may you please add:
>  >
>  >, just before the transfer line, eg:
>  >
>  > 
>  >         
>  >         
>  >         
>  >         
>  >         
>  >         
>  >	 
>  >         
>  >      
>  >
>  >
>  >Using application Info, you'll see on the console (or CDR) the list of 
>  >variables used for this call.  You should see the content of 
>  >"${sip_redirect_contact_user_0}" that should contain  the value 
>  >"PEER_01". Please check this and let me know.
>  >
>  >For the new configuration file, no problem for sharing. But as I wrote 
>  >on the wiki page, I worked on this setup cause LCR module was not 
>  >available when I start working on FS, nor I'm a good programmer to write 
>  >a server side HTTP script (used by xml_curl) that could scale to my needs.
>  >
>  >I enhanced a bit this configuration with support for fallback routing, 
>  >almost realtime graph (every minutes using www.cacti.net and some 
>  >functions in FS like limit_hash) of number of concurrent calls per AREA, 
>  >PEER...
>  >
>  >Thanks for "the good tutorial", but don't forget the dev team who did 
>  >this great product ;-)
>  >
>  >rod.
>  >
>  >
>  >Hristo Benev a ?crit :
>  >>  I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).
>  >>
>  >> If you need additional info I'll provide it.
>  >>
>  >> Here is trace:
>  >>
>  >> ngrep -d any -nn -i '1000' port 5062 -W byline
>  >> interface: any
>  >> filter: (ip or ip6) and ( port 5062 )
>  >> match: 1000
>  >> #
>  >> U 10.10.10.10:5090 -> 127.0.0.1:5062
>  >> INVITE sip:1000 at 127.0.0.1:5062 SIP/2.0.
>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
>  >> Max-Forwards: 69.
>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
>  >> To: .
>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
>  >> CSeq: 119574771 INVITE.
>  >> Contact: .
>  >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
>  >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
>  >> Supported: timer, precondition, path, replaces.
>  >> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
>  >> Content-Type: application/sdp.
>  >> Content-Disposition: session.
>  >> Content-Length: 429.
>  >> X-ROUTE: LOOKUP.
>  >> Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off.
>  >> .
>  >> v=0.
>  >> o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
>  >> s=FreeSWITCH.
>  >> c=IN IP4 10.10.10.10.
>  >> t=0 0.
>  >> m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
>  >> a=rtpmap:0 PCMU/8000.
>  >> a=rtpmap:115 G7221/32000.
>  >> a=fmtp:115 bitrate=48000.
>  >> a=rtpmap:107 G7221/16000.
>  >> a=fmtp:107 bitrate=32000.
>  >> a=rtpmap:9 G722/8000.
>  >> a=rtpmap:8 PCMA/8000.
>  >> a=rtpmap:3 GSM/8000.
>  >> a=rtpmap:101 telephone-event/8000.
>  >> a=fmtp:101 0-16.
>  >> a=rtpmap:13 CN/8000.
>  >> a=ptime:20.
>  >>
>  >> #
>  >> U 127.0.0.1:5062 -> 10.10.10.10:5090
>  >> SIP/2.0 302 PEER_01.
>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
>  >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
>  >> CSeq: 119574771 INVITE.
>  >> Contact: sip:France at PEER_01.
>  >> Server: Kamailio (1.5.2-notls (i386/linux)).
>  >> Content-Length: 0.
>  >> .
>  >>
>  >> #
>  >> U 10.10.10.10:5090 -> 127.0.0.1:5062
>  >> ACK sip:1000 at 127.0.0.1:5062 SIP/2.0.
>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
>  >> Max-Forwards: 69.
>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
>  >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
>  >> CSeq: 119574771 ACK.
>  >> Content-Length: 0.
>  >> .
>  >>
>  >>
>  >>
>  >>
>  >>
>  >>  >-------- ?????????? ????? --------
>  >>  >??:  Hristo Benev 
>  >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  >??: freeswitch-users at lists.freeswitch.org
>  >>  >????????? ??: ?????????, 2009, ?????? 27 14:58:53 EEST
>  >>
>  >>  > 
>  >>  >Bojnour,
>  >>  >
>  >>  >I'll send a trace ASAP.
>  >>  >
>  >>  >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
>  >>  >
>  >>  >I'm really interested to see the updates of the project.
>  >>  >
>  >>  >Thank you for the good tutorial.
>  >>  >
>  >>  >Hristo
>  >>  >
>  >>  >
>  >>  > >-------- ?????????? ????? --------
>  >>  > >??:  rod 
>  >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  > >??: freeswitch-users at lists.freeswitch.org
>  >>  > >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST
>  >>  >
>  >>  > >Hi Hristo,
>  >>  > >
>  >>  > >I'm the author of this setup and wiki page. I did a lot of modifications 
>  >>  > >on this setup (alternative routing if failure essentially) but don't 
>  >>  > >have too much time to update the wiki.
>  >>  > >
>  >>  > >May you please send me an ngrep trace when you call 1000:
>  >>  > >
>  >>  > >ngrep -d any -nn -i '1000' port 5060 -W byline 
>  >>  > >
>  >>  > >I will check what's happening.
>  >>  > >Do you have an entry for 1000 in your mysql database ?
>  >>  > >
>  >>  > >regards,
>  >>  > >rod
>  >>  > >
>  >>  > >Hristo Benev a ?crit :
>  >>  > >>  
>  >>  > >> It seems that the problem is on kamailio configuration.
>  >>  > >> Will ask on their list.
>  >>  > >>
>  >>  > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
>  >>  > >>
>  >>  > >> I have it setup as friend in asterisk, but still ???
>  >>  > >>
>  >>  > >> Any ideas?
>  >>  > >>
>  >>  > >> Thanks,
>  >>  > >>
>  >>  > >>
>  >>  > >>  >-------- ?????????? ????? --------
>  >>  > >>  >??:  Hristo Benev 
>  >>  > >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  > >>  >??: freeswitch-users at lists.freeswitch.org
>  >>  > >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
>  >>  > >>
>  >>  > >>  > I think that the problem is here:
>  >>  > >>  >-------------------------
>  >>  > >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
>  >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
>  >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
>  >>  > >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
>  >>  > >>  >--------------------------
>  >>  > >>  >
>  >>  > >>  >Actually Regex FAIL
>  >>  > >>  >
>  >>  > >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
>  >>  > >>  >Here is my  default.xml:
>  >>  > >>  >----------------
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >  
>  >>  > >>  >
>  >>  > >>  >     
>  >>  > >>  >      
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >      
>  >>  > >>  >    
>  >>  > >>  >
>  >>  > >>  >   
>  >>  > >>  >
>  >>  > >>  >  
>  >>  > >>  >
>  >>  > >>  >    
>  >>  > >>  >      
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >      
>  >>  > >>  >    
>  >>  > >>  >
>  >>  > >>  >  
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >--------------------------
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  > >-------- ?????????? ????? --------
>  >>  > >>  > >??:  Brian West 
>  >>  > >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  > >>  > >??: freeswitch-users at lists.freeswitch.org
>  >>  > >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
>  >>  > >>  >
>  >>  > >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
>  >>  > >>  >You have to process all 302's in the dialplan. 
>  >>  > >>  >Set this on your sofia profile
>  >>  > >>  >You can set these variables sip_redirect_profile,
>  >>  > >>  >sip_redirect_context,
>  >>  > >>  >sip_redirect_dialplan,
>  >>  > >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
>  >>  > >>  >sip_redirected_to,
>  >>  > >>  >sip_redirect_contact_user_%d,
>  >>  > >>  >sip_redirect_contact_host_%d,
>  >>  > >>  >sip_redirect_contact_params_%d,
>  >>  > >>  >sip_redirect_dialstring_%d,
>  >>  > >>  >sip_redirect_dialstring,
>  >>  > >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
>  >>  > >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
>  >>  > >>  > >
>  >>  > >>  >
>  >>  > >>  >_______________________________________________
>  >>  > >>  >FreeSWITCH-users mailing list
>  >>  > >>  >FreeSWITCH-users at lists.freeswitch.org
>  >>  > >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  > >>  >http://www.freeswitch.org
>  >>  > >>  >
>  >>  > >>
>  >>  > >> _______________________________________________
>  >>  > >> FreeSWITCH-users mailing list
>  >>  > >> FreeSWITCH-users at lists.freeswitch.org
>  >>  > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  > >> http://www.freeswitch.org
>  >>  > >>   
>  >>  > >
>  >>  > >_______________________________________________
>  >>  > >FreeSWITCH-users mailing list
>  >>  > >FreeSWITCH-users at lists.freeswitch.org
>  >>  > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  > >http://www.freeswitch.org
>  >>  > >
>  >>  >
>  >>  >_______________________________________________
>  >>  >FreeSWITCH-users mailing list
>  >>  >FreeSWITCH-users at lists.freeswitch.org
>  >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  >http://www.freeswitch.org
>  >>  >
>  >>
>  >> _______________________________________________
>  >> FreeSWITCH-users mailing list
>  >> FreeSWITCH-users at lists.freeswitch.org
>  >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >> http://www.freeswitch.org
>  >>   
>  >
>  >_______________________________________________
>  >FreeSWITCH-users mailing list
>  >FreeSWITCH-users at lists.freeswitch.org
>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >http://www.freeswitch.org
>  >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   


From thangappan143 at gmail.com  Fri Aug 28 23:45:28 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Sat, 29 Aug 2009 12:15:28 +0530
Subject: [Freeswitch-users] FreeSWITCH environmental variables
Message-ID: <7aa29e790908282345r511dcc82kd767063b93d370dd@mail.gmail.com>

Dear all,

        In the case of asterisk PBX. I can get all the information about the
call from the environmental variable itself like Caller id ,called id,
channel no. But in freeswitch where all the environmental variables are
resides? How can I access all the variables?

Please Help me?

-- 
Regards,
Thangappan.M
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From jaybinks at gmail.com  Sat Aug 29 01:17:31 2009
From: jaybinks at gmail.com (Jay Binks)
Date: Sat, 29 Aug 2009 18:17:31 +1000
Subject: [Freeswitch-users] FreeSWITCH environmental variables
In-Reply-To: <7aa29e790908282345r511dcc82kd767063b93d370dd@mail.gmail.com>
References: <7aa29e790908282345r511dcc82kd767063b93d370dd@mail.gmail.com>
Message-ID: <82795EF6-1698-4DA5-A3F3-23A6CF47CDAC@gmail.com>

Check out INFO
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info

Throw that in your dialplan the look at your logs...  You should find  
what your after..

Jay



On 29/08/2009, at 16:45, "Thangappan.M" <thangappan143 at gmail.com> wrote:

> Dear all,
>
>         In the case of asterisk PBX. I can get all the information  
> about the call from the environmental variable itself like Caller  
> id ,called id, channel no. But in freeswitch where all the  
> environmental variables are resides? How can I access all the  
> variables?
>
> Please Help me?
>
> -- 
> Regards,
> Thangappan.M
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From m.krivushin at imarto.net  Sat Aug 29 01:21:16 2009
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Sat, 29 Aug 2009 15:21:16 +0700
Subject: [Freeswitch-users] FreeSWITCH environmental variables
In-Reply-To: <7aa29e790908282345r511dcc82kd767063b93d370dd@mail.gmail.com>
References: <7aa29e790908282345r511dcc82kd767063b93d370dd@mail.gmail.com>
Message-ID: <5be734a50908290121ya04c76n2a0bb3c50083edda@mail.gmail.com>

Try to read docuemntation first:

http://wiki.freeswitch.org/wiki/Channel_Variables

2009/8/29 Thangappan.M <thangappan143 at gmail.com>

> Dear all,
>
>         In the case of asterisk PBX. I can get all the information about
> the call from the environmental variable itself like Caller id ,called id,
> channel no. But in freeswitch where all the environmental variables are
> resides? How can I access all the variables?
>
> Please Help me?
>
> --
> Regards,
> Thangappan.M
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From paveraware at hotmail.com  Sat Aug 29 02:28:23 2009
From: paveraware at hotmail.com (Christensen Tom)
Date: Sat, 29 Aug 2009 09:28:23 +0000
Subject: [Freeswitch-users] Success! and an issue.
Message-ID: <BAY120-W6EC265B637176ED41128BDFF40@phx.gbl>


well, just made my first extension to extension call using freeswitch, so that was pretty painless.  Huge fan of the config layout at least (having the extensions defined each in their own file is going to be so much nicer than asterisks huge jumble).

However, I did run into one issue, not sure if its somehow related to the phones I'm using (softphone of unknown quality (weephone) on iphones)... I'm using the default config, when I dial 4000 for voicemail, it works, however at apparently random times during the menu navigation I get call waiting beeps and I have an incoming call from 4000.  If I decline or leave this call unanswered, my call into voicemail gets disconnected.  If I "accept" this call, it apparently disconnects immediately (IE I only have 1 active call), however I can then continue navigating the menu, etc... Very strange to me, I will be setting up a couple of polycom 501s on this system tomorrow and will report if I see similar issues.
Anyway, so far I'm pleased with freeswitch, I've got a list of features I need to support before I can deploy it, so I'll be bugging people (sorry), but so far thanks!
-Tom


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From steve.kurzeja at gmail.com  Sat Aug 29 02:58:37 2009
From: steve.kurzeja at gmail.com (Steve Kurzeja)
Date: Sat, 29 Aug 2009 21:58:37 +1200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
Message-ID: <5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>

On Sat, Aug 29, 2009 at 2:34 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Yes, FreeSWITCH is a system that you can trust 100%. I have switched my
> Asterisk servers to FreeSWITCH and have peace now.
>
> If I were you I would get rid of Asterisk and use FreeSWITCH, FS will
> handle all what you want very well.
>
> And I agree with David, fail-over is kinda irrelevant since the FS doesn't
> crash like Asterisk does.
>
>
>
You still have hardware failures and fail-over is also useful for hit-less
maintenance on boxes.

I'd be interested to know how Brian West was approaching his live migration
work.

Steve
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From rs at runsolutions.com  Sat Aug 29 06:17:11 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Sat, 29 Aug 2009 15:17:11 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
Message-ID: <71FD2763-32C5-49CA-8E4D-7C0E301D2187@runsolutions.com>

Oh yeah, that would be so helpfull for my situation, as my client  
*demands* now a solution where he can press a big red button and all  
fails over to another box. Hi es totally scared because of the Lockups  
in Asterisk which under specific situations including AMI, Automated  
Call Setup, and murphy led to a lockup of the entire machine, no  
console was working anymore, only cold-reset could do it.

So, IF there is the possibility for life-takeover, / failover etc. I  
would love to here how has been done.

I am very experienced with openvz and use for about two years now only  
openvz virtualization servers for anything because of live-migration  
etc. But as I am new in this company we could not adopt this until now.

So Please Ken, if you can, describe what need's to be done to get a  
failover / takeover working (an outline would be enough)

Thanks in Advance

-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 29, 2009, at 11:58 AM, Steve Kurzeja wrote:

> On Sat, Aug 29, 2009 at 2:34 PM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> Yes, FreeSWITCH is a system that you can trust 100%. I have switched  
> my Asterisk servers to FreeSWITCH and have peace now.
>
> If I were you I would get rid of Asterisk and use FreeSWITCH, FS  
> will handle all what you want very well.
>
> And I agree with David, fail-over is kinda irrelevant since the FS  
> doesn't crash like Asterisk does.
>
>
>
> You still have hardware failures and fail-over is also useful for  
> hit-less maintenance on boxes.
>
> I'd be interested to know how Brian West was approaching his live  
> migration work.
>
> Steve
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From tomabroad at gmail.com  Sat Aug 29 06:46:27 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 09:46:27 -0400
Subject: [Freeswitch-users] newbie questions
Message-ID: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>

hi, totally new to freeswitch, but not to asterisk. just installed it, made
samples but do not have 1000-1019 as extensions. so i thought i create:
<include>
  <user id="1000" mailbox="1000">
    <params>
      <param name="password" value="1234"/>
      <param name="vm-password" value="1000"/>
    </params>
    <variables>
      <variable name="accountcode" value="1000"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1000"/>
      <variable name="effective_caller_id_number" value="1000"/>
    </variables>
  </user>
</include>
 and save it as 1000.xml in the internal-sip folder. cli tells me now:
2009-08-29 09:43:55.858101 [WARNING] sofia_reg.c:1771 Can't find user [
1000 at 192.168.1.4]
You must define a domain called '192.168.1.4' in your directory and add a
user with the id="1000" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.

what do i have to do now to get it
a) internally running
b) talk to an external cllient?


thx
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From rupa at rupa.com  Sat Aug 29 07:07:15 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 29 Aug 2009 09:07:15 -0500
Subject: [Freeswitch-users] mod_limit and memcache
In-Reply-To: <b39273530908270742i1e018ed2x953203e4eee7bb2f@mail.gmail.com>
References: <a44688a0908270649y6cde3c26pb739c7d144e1b86b@mail.gmail.com>
	<b39273530908270742i1e018ed2x953203e4eee7bb2f@mail.gmail.com>
Message-ID: <b39273530908290707y625223y6a993d54cb8ce88b@mail.gmail.com>

Ok, it was a bit more work than expected, but I have a first cut at
mod_limit using memcache checked in.  Test if you can.  I've not put
docs on the wiki 'cause some of the implementation / setup details are
going to change with regards to mod_memcache.   Oh, and mod_limit's
api may be changing to make it more pluggable.

Basically,

Setup mod_memcache (on wiki)

It uses the same user facing api as limit_hash and friends except it
is limit_memcache and limit_memcache_usage.  so follow mod_limit's
docs with suitable substitutions.


On Thu, Aug 27, 2009 at 9:42 AM, Rupa Schomaker<rupa at rupa.com> wrote:
> limit and memcache haven't been introduced to each other yet -- it is
> on my (semi-long) list of things to do.
>
> If you want it you can:
>
> 1) do it yourself and submit the patches
> 2) open a jira and hope someone does it
> 3) open a jira + bounty and someone will probably do it
>
> It will get done eventually, just hasn't been a itch for ME to scratch yet.
>
>
> To do it:
>
> 1) I need to make it possible to call inc/dec methods of mod_memcache
> to support an expiration time.
>
> 2) mod_limit.c - use the hash limit as a guide
>
> Initial pitfalls:
>
> hash limits concurrent access/modification of the hash and by
> implication limit_hash_item_t (hash data) by using a mutex. ?We can't
> mutex across FS instances.
>
> So perhaps split up limit_has_item_t and spread it across multiple
> keys. ?So instead of one key marked as realm_id, we could have
> realm_id_total_usage realm_id_rate_usage and realm_id_last_check.
>
> This does mean that rate_usage and total_usage can inc/dec independent
> of each other, but I think the logic will still be "ok" *IF* we
> remember to decrement earlier incremented items in the event a later
> item is failed. ? (so, say we increment rate but fail on total we need
> to remember to decrement rate so that we have no net effect on the
> counters)
>
> Alternatively, we could use CAS support and pull the limit_hash_item_t
> item from memcache, twiddle it and then try to put it back only if the
> check info is the same (no one else has changed the entry). ?If the
> entry has changed, pull the new version, do the limit logic, and try
> again. ?Loop that a few times until you succeed or give up. ?Problem
> is that CAS needs to be explicitly turned on in memcache (some distros
> compile with it off), is relatively new in memcache (hint: may have
> issues) and has some performance/memory downsides though by how much
> I'm not sure.
>
> Thoughts?
>
> On Thu, Aug 27, 2009 at 8:49 AM, Woody Dickson<woodydickson at gmail.com> wrote:
>> Hello,
>>
>> I read something that talks about using memcache for mod_limit before.?? Is
>> it something that is available now?
>>
>> If I have multiple instances of freeswitch that need to share the same limit
>> status, it there any existing solution?
>>
>> If no existing solution is available, what is the best way to go about
>> modifying mod_limit to accomplish limiting for multiple freeswitch servers
>> together?
>>
>> Thanks,
>> Woody
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> -Rupa
>



-- 
-Rupa


From max.bridgewater at gmail.com  Sat Aug 29 07:07:57 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sat, 29 Aug 2009 10:07:57 -0400
Subject: [Freeswitch-users] Disabling Core Dumping
Message-ID: <c55d46950908290707u16fc5d40m3c23f9a9ca450738@mail.gmail.com>

Hi,

Whenever i stop freeswich, it creates a core dump. How can i disable that?

thanks,
Max.
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From rs at runsolutions.com  Sat Aug 29 07:33:15 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Sat, 29 Aug 2009 16:33:15 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <71FD2763-32C5-49CA-8E4D-7C0E301D2187@runsolutions.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
	<71FD2763-32C5-49CA-8E4D-7C0E301D2187@runsolutions.com>
Message-ID: <136791EF-26D2-4DC7-91F6-2D468122F06C@runsolutions.com>

Thinking about it, maybe we can create a solution, if some of us work  
together:

My strength are in virtualization, linux, development, databases,  
integration, etc.
What I do not now much about is how SIP (and everything else for that  
matter in the Voice world) works under the hood, and how it's  
implemented in FS.

I know that the state information for a call has to be stored and  
retrieved somewhere and somehow, only I do not know that part. What I  
know is that it hast to be do-able to store all the stream information  
(ip's, port's, current state's, etc.) in a very fast database (e.g. my  
idea would be memcached) so another FS could just take this  
information and take over the call, maybe you loose a second of voice,  
maybe you loose the recorded call file or a part of it, but that  
should be it. (SipFoundry has a boxed opensource PBX, which, of course  
is not flexible like FreeSWITCH or Asterisk, but has Call Live  
Migration and Call Live Failover integrated!).

What I want is for my company to be able to sell a 99.99 uptime PBX  
(we do mostly call-center related stuff), which can scale well, and  
can grow with the company without lot's of hassles, my Dream would be:

To begin with:

One Hardware Node with the essential hardware (digium cards for  
example).
On this node are OpenVZ virtualized containers:
[VirtCnt1: FS which only talks to the Hardware and forwards  
everything] = Could be replaced with hardware media gateway, etc.
[VirtCnt2: FS which handles the PBX] \___ Loadbalanced, with odbc or  
xml, Failover, Livetakeover
[VirtCnt3: FS which handles the PBX] /
[VirtCnt4: Database for state information] (maybe something as  
resource-friendly as memcached? ressource heavvy database?)

With this we can achieve all this:

Problem with VirtCnt2 (e.g. crash, lock, ...)
* VirtCnt3 can take over.
-> You are free without stress to investigate the problem, you can  
debug and analyze whyle the machine is still running
-> you can also create a machine-state-dump of the virtual container,  
dump the container as well, copy the data to your lab and restore the  
machine up the state which it was running with the problem, so you can  
liveinvestigate it in the lab (some prerequirements given, but easy  
doable)
-> just think about the possibility of better bugreports because  
someone can take the time to read out all the data with GDB to  
investigate the proper cause of a machine Lock!

You want to upgrade to a new FreeSWITCH version?
* Take VirtCnt2 out of the LoadBalancing Scheme,
* Stop it, Clone it,
* Upgrade FreeSWITCH in the cloned Container
* Start the cloned container
* if there's something wrong, stop it and restart the original VirtCnt2
-> No problem at all, you can Test on the Live Hardware, with part of  
the Live users (maybe a low-volume queue) to be sure everything works  
out fine before you activate the full loadbalance

Server on it's own can't handle the load
* Buy new machine
* Setup Hardware Node
* Livemigrate VirtCnt3 (no downtime)

Now the first Server with the VrtCnt1 and VirtCnt2 as well has to much  
load
* Buy new machine
* Setup Hardware Node
* Livemigrate VirtCnt2 (no downtime)
-> Now you have a 3 server solution (1 mediaprox, 2 loadbalanced /  
failover PBXes) out of the first box you bought, without headaches,  
because the system was built for it from the beginning!

The Database drains to much?
* Buy new machine
* Setup Hardware Node
* Livemigrate database VirtCnt4 (no downtime)

You want to upgrade Hardware/Kernel in Hardware node 1?
* Livemigrate VirtCnt2 to a hotstandby machine, or to the other PBX  
machine, upgrade the hardware, Re-Livemigrate the containers. (no  
downtime)
* OR just break the loadbalancing, wait until all current calls are  
teared down correctly, upgrade machine, reenable the loadbalancer

You want an exact copy of the first server for Hardware HA?
* Buy new machine
* Setup Hardware node
* Buy hardware PRI switchover box
* Clone VirtCnt1 - VirtCnt4 to the new machine
* Make basic failover configuration


-> the sky's the limit, as the saying goes ...


So, I can do all the openvz stuff and the integration with database /  
memcached / heartbeat / whatever is needed here, someone there to be  
willing to work with me on this on the FreeSWITCH side? or at least  
provide me with the necessary information about what's needed / how to  
talk / what states from FreeSWITCH?

I know this seems very ambitious but if this could be made in a rather  
relativly easy to setup package, with good documentation, it would be  
a boost for FreeSWITCH, i am sure, because after all this is what  
everyone is grown accustomed to from good old phone companys and the  
good old pbx's: carrier grade uptimes ...

Thanks for everyone reading up until here,
all the best,

Ray



-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 29, 2009, at 3:17 PM, Raimund Sacherer wrote:

> Oh yeah, that would be so helpfull for my situation, as my client  
> *demands* now a solution where he can press a big red button and all  
> fails over to another box. Hi es totally scared because of the  
> Lockups in Asterisk which under specific situations including AMI,  
> Automated Call Setup, and murphy led to a lockup of the entire  
> machine, no console was working anymore, only cold-reset could do it.
>
> So, IF there is the possibility for life-takeover, / failover etc. I  
> would love to here how has been done.
>
> I am very experienced with openvz and use for about two years now  
> only openvz virtualization servers for anything because of live- 
> migration etc. But as I am new in this company we could not adopt  
> this until now.
>
> So Please Ken, if you can, describe what need's to be done to get a  
> failover / takeover working (an outline would be enough)
>
> Thanks in Advance
>
> -- 
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
> On Aug 29, 2009, at 11:58 AM, Steve Kurzeja wrote:
>
>> On Sat, Aug 29, 2009 at 2:34 PM, Diego Viola  
>> <diego.viola at gmail.com> wrote:
>> Yes, FreeSWITCH is a system that you can trust 100%. I have  
>> switched my Asterisk servers to FreeSWITCH and have peace now.
>>
>> If I were you I would get rid of Asterisk and use FreeSWITCH, FS  
>> will handle all what you want very well.
>>
>> And I agree with David, fail-over is kinda irrelevant since the FS  
>> doesn't crash like Asterisk does.
>>
>>
>>
>> You still have hardware failures and fail-over is also useful for  
>> hit-less maintenance on boxes.
>>
>> I'd be interested to know how Brian West was approaching his live  
>> migration work.
>>
>> Steve
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sat Aug 29 08:09:26 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:09:26 -0500
Subject: [Freeswitch-users] Disabling Core Dumping
In-Reply-To: <c55d46950908290707u16fc5d40m3c23f9a9ca450738@mail.gmail.com>
References: <c55d46950908290707u16fc5d40m3c23f9a9ca450738@mail.gmail.com>
Message-ID: <6EA6BE3E-AE31-4628-A204-76B385F6FC20@freeswitch.org>

report the issue on jira.... If its dumping a core something is wrong  
and we need to fix it.

/b

On Aug 29, 2009, at 9:07 AM, Max Bridgewater wrote:

> Hi,
>
> Whenever i stop freeswich, it creates a core dump. How can i disable  
> that?
>
> thanks,
> Max.
> ________



From brian at freeswitch.org  Sat Aug 29 08:12:35 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:12:35 -0500
Subject: [Freeswitch-users] Group Call
In-Reply-To: <ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
Message-ID: <81148697-292E-4E9C-A26A-D1536E9D8A8E@freeswitch.org>

Don't confuse this with the actual group/ endpoint.

/b

On Aug 25, 2009, at 11:41 AM, Jo?o Mesquita wrote:

> According to wiki:
> group,[insert|delete|call]:<group name>:<url>,group [insert|delete| 
> call
>



From brian at freeswitch.org  Sat Aug 29 08:14:37 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:14:37 -0500
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
Message-ID: <6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>

they don't go in sip_profiles... its in conf/directory/default/ and  
they are all there already.  Maybe your configs were not properly  
installed?

/b

On Aug 29, 2009, at 8:46 AM, tom wrote:

>
> what do i have to do now to get it
> a) internally running
> b) talk to an external cllient?



From brian at freeswitch.org  Sat Aug 29 08:15:35 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:15:35 -0500
Subject: [Freeswitch-users] Success! and an issue.
In-Reply-To: <BAY120-W6EC265B637176ED41128BDFF40@phx.gbl>
References: <BAY120-W6EC265B637176ED41128BDFF40@phx.gbl>
Message-ID: <92598075-4C54-4EED-80D0-3263DE495C68@freeswitch.org>

Can you get a sip trace and console trace because this is highly  
unlikely.

/b

On Aug 29, 2009, at 4:28 AM, Christensen Tom wrote:

> However, I did run into one issue, not sure if its somehow related  
> to the phones I'm using (softphone of unknown quality (weephone) on  
> iphones)... I'm using the default config, when I dial 4000 for  
> voicemail, it works, however at apparently random times during the  
> menu navigation I get call waiting beeps and I have an incoming call  
> from 4000.  If I decline or leave this call unanswered, my call into  
> voicemail gets disconnected.  If I "accept" this call, it apparently  
> disconnects immediately (IE I only have 1 active call), however I  
> can then continue navigating the menu, etc... Very strange to me, I  
> will be setting up a couple of polycom 501s on this system tomorrow  
> and will report if I see similar issues.

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From brian at freeswitch.org  Sat Aug 29 08:16:13 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:16:13 -0500
Subject: [Freeswitch-users] FreeSWITCH environmental variables
In-Reply-To: <82795EF6-1698-4DA5-A3F3-23A6CF47CDAC@gmail.com>
References: <7aa29e790908282345r511dcc82kd767063b93d370dd@mail.gmail.com>
	<82795EF6-1698-4DA5-A3F3-23A6CF47CDAC@gmail.com>
Message-ID: <F94A3DA5-7481-43B1-9011-076FCC8188A0@freeswitch.org>

You can also uuid_dump the uuid to see the info also.

/b

On Aug 29, 2009, at 3:17 AM, Jay Binks wrote:

> Check out INFO
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info
>
> Throw that in your dialplan the look at your logs...  You should find
> what your after..
>
> Jay

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From tomabroad at gmail.com  Sat Aug 29 08:19:03 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 11:19:03 -0400
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
Message-ID: <6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>

yes u are right, i was mistaken. everything in there by default...still the
same domain error msg. what am i supposed to do?
tjx
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From brian at freeswitch.org  Sat Aug 29 08:19:32 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:19:32 -0500
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
Message-ID: <A9DC8F8B-B0D3-4AC3-A67B-88F5CFCFDB3A@freeswitch.org>

I was able to do this using OpenVZ, You can get away with it on  
smaller instances... like if you're doing one instance per company but  
don't expect live migration to work as well on large instances with  
thousands of calls up at once. You need a fast network, fast disks and  
to follow the howto on the wiki.

/b

On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote:

> You still have hardware failures and fail-over is also useful for  
> hit-less maintenance on boxes.
>
> I'd be interested to know how Brian West was approaching his live  
> migration work.
>
> Steve



From brian at freeswitch.org  Sat Aug 29 08:22:52 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:22:52 -0500
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
Message-ID: <153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>

Are you on the latest configs?  This message is unlikely in the  
default configs.  Please double check that you have the internal.xml  
from the default install it forces the register domain to match the IP  
so you won't get this error.

/b

On Aug 29, 2009, at 10:19 AM, tom wrote:

> yes u are right, i was mistaken. everything in there by  
> default...still the same domain error msg. what am i supposed to do?
> tjx
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From tomabroad at gmail.com  Sat Aug 29 08:28:32 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 11:28:32 -0400
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
	<153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
Message-ID: <6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>

mh, a search for "domain=" doesnt brin anything up ininternal.xml... im
total new to this...sorry
do i ahev to add this?

//yes im using the 1.0.4 tar




On Sat, Aug 29, 2009 at 11:22 AM, Brian West <brian at freeswitch.org> wrote:

> Are you on the latest configs?  This message is unlikely in the
> default configs.  Please double check that you have the internal.xml
> from the default install it forces the register domain to match the IP
> so you won't get this error.
>
> /b
>
> On Aug 29, 2009, at 10:19 AM, tom wrote:
>
> > yes u are right, i was mistaken. everything in there by
> > default...still the same domain error msg. what am i supposed to do?
> > tjx
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sat Aug 29 08:33:28 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:33:28 -0500
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
	<153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
	<6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>
Message-ID: <98E01B85-5134-48FA-9200-C01D96E48AA5@freeswitch.org>

If you don't have these

     <!--all inbound reg will look in this domain for the users -->
     <param name="force-register-domain" value="$${domain}"/>
     <!--all inbound reg will stored in the db using this domain -->
     <param name="force-register-db-domain" value="$${domain}"/>

Then you need to wipe your conf dir and make samples again please.

/b

On Aug 29, 2009, at 10:28 AM, tom wrote:

>
> mh, a search for "domain=" doesnt brin anything up ininternal.xml...  
> im total new to this...sorry
> do i ahev to add this?
>
> //yes im using the 1.0.4 tar
>
>
>
>
> On Sat, Aug 29, 2009 at 11:22 AM, Brian West <brian at freeswitch.org>  
> wrote:
> Are you on the latest configs?  This message is unlikely in the
> default configs.  Please double check that you have the internal.xml
> from the default install it forces the register domain to match the IP
> so you won't get this error.
>
> /b
>
> On Aug 29, 2009, at 10:19 AM, tom wrote:
>
> > yes u are right, i was mistaken. everything in there by
> > default...still the same domain error msg. what am i supposed to do?
> > tjx
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tomabroad at gmail.com  Sat Aug 29 08:40:04 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 11:40:04 -0400
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <98E01B85-5134-48FA-9200-C01D96E48AA5@freeswitch.org>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
	<153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
	<6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>
	<98E01B85-5134-48FA-9200-C01D96E48AA5@freeswitch.org>
Message-ID: <6f7c60c40908290840w466c37d6v28dcc0f38f632cce@mail.gmail.com>

ok, what u sent is fine, i have that. do i have to do somethign in that
section:?
  <domains>
    <!-- indicator to parse the directory for domains with parse="true" to
get gateways-->
    <domain name="$${domain}" parse="true"/>
    <!-- indicator to parse the directory for domains with parse="true" to
get gateways and alias every domain to this profile -->
    <!--<domain name="all" alias="true" parse="true"/>-->
    <domain name="all" alias="true" parse="false"/>
  </domains>


?thx


On Sat, Aug 29, 2009 at 11:33 AM, Brian West <brian at freeswitch.org> wrote:

> If you don't have these
>     <!--all inbound reg will look in this domain for the users -->
>
>
>     <param name="force-register-domain" value="$${domain}"/>
>
>
>     <!--all inbound reg will stored in the db using this domain -->
>
>
>     <param name="force-register-db-domain" value="$${domain}"/>
>
> Then you need to wipe your conf dir and make samples again please.
>
> /b
>
> On Aug 29, 2009, at 10:28 AM, tom wrote:
>
>
> mh, a search for "domain=" doesnt brin anything up ininternal.xml... im
> total new to this...sorry
> do i ahev to add this?
>
> //yes im using the 1.0.4 tar
>
>
>
>
> On Sat, Aug 29, 2009 at 11:22 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Are you on the latest configs?  This message is unlikely in the
>> default configs.  Please double check that you have the internal.xml
>> from the default install it forces the register domain to match the IP
>> so you won't get this error.
>>
>> /b
>>
>> On Aug 29, 2009, at 10:19 AM, tom wrote:
>>
>> > yes u are right, i was mistaken. everything in there by
>> > default...still the same domain error msg. what am i supposed to do?
>> > tjx
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Aug 29 08:56:13 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 29 Aug 2009 08:56:13 -0700
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <6f7c60c40908290840w466c37d6v28dcc0f38f632cce@mail.gmail.com>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
	<153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
	<6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>
	<98E01B85-5134-48FA-9200-C01D96E48AA5@freeswitch.org>
	<6f7c60c40908290840w466c37d6v28dcc0f38f632cce@mail.gmail.com>
Message-ID: <B549B159-D098-4155-A0A7-F5CCDF4BFE0A@freeswitch.org>

Tom,

You should not have to do anything other than make samples. If you are  
having these kinds of problems then something went very wrong during  
the install. Try starting from scratch just to be sure. Also, what  
distro are you running?
-MC

Sent from my iPhone

On Aug 29, 2009, at 8:40 AM, tom <tomabroad at gmail.com> wrote:

>
> ok, what u sent is fine, i have that. do i have to do somethign in  
> that section:?
>   <domains>
>     <!-- indicator to parse the directory for domains with  
> parse="true" to get gateways-->
>     <domain name="$${domain}" parse="true"/>
>     <!-- indicator to parse the directory for domains with  
> parse="true" to get gateways and alias every domain to this profile  
> -->
>     <!--<domain name="all" alias="true" parse="true"/>-->
>     <domain name="all" alias="true" parse="false"/>
>   </domains>
>
>
> ?thx
>
>
> On Sat, Aug 29, 2009 at 11:33 AM, Brian West <brian at freeswitch.org>  
> wrote:
> If you don't have these
>
>     <!--all inbound reg will look in this domain for the users -->
>     <param name="force-register-domain" value="$${domain}"/>
>     <!--all inbound reg will stored in the db using this domain -->
>     <param name="force-register-db-domain" value="$${domain}"/>
>
> Then you need to wipe your conf dir and make samples again please.
>
> /b
>
> On Aug 29, 2009, at 10:28 AM, tom wrote:
>
>>
>> mh, a search for "domain=" doesnt brin anything up  
>> ininternal.xml... im total new to this...sorry
>> do i ahev to add this?
>>
>> //yes im using the 1.0.4 tar
>>
>>
>>
>>
>> On Sat, Aug 29, 2009 at 11:22 AM, Brian West <brian at freeswitch.org>  
>> wrote:
>> Are you on the latest configs?  This message is unlikely in the
>> default configs.  Please double check that you have the internal.xml
>> from the default install it forces the register domain to match the  
>> IP
>> so you won't get this error.
>>
>> /b
>>
>> On Aug 29, 2009, at 10:19 AM, tom wrote:
>>
>> > yes u are right, i was mistaken. everything in there by
>> > default...still the same domain error msg. what am i supposed to  
>> do?
>> > tjx
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Sat Aug 29 09:01:46 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 29 Aug 2009 11:01:46 -0500
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <A9DC8F8B-B0D3-4AC3-A67B-88F5CFCFDB3A@freeswitch.org>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
	<A9DC8F8B-B0D3-4AC3-A67B-88F5CFCFDB3A@freeswitch.org>
Message-ID: <191c3a030908290901w56b4fd26h467d75ec58dbc380@mail.gmail.com>

We have previously estimated the development of live fail over (after a box
dies where live migration is no longer possible) to exceed 100k in
development costs.

It requires several additions to the sofia sip library, freeswitch and a
dependancy on some other code we would have to implement to manage it.

It may or may not be worth it to raise that kind of funding just to avoid an
occasional disaster.

Then there is a matter of securing the time of the developers necessary to
carry out the implementation.

On Aug 29, 2009 10:19 AM, "Brian West" <brian at freeswitch.org> wrote:

I was able to do this using OpenVZ, You can get away with it on
smaller instances... like if you're doing one instance per company but
don't expect live migration to work as well on large instances with
thousands of calls up at once. You need a fast network, fast disks and
to follow the howto on the wiki.

/b

On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: > You still have hardware
failures and fail-over...

_______________________________________________ FreeSWITCH-users mailing
list FreeSWITCH-users at lists...
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From e.schmidbauer at gmail.com  Sat Aug 29 09:05:42 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Sat, 29 Aug 2009 12:05:42 -0400
Subject: [Freeswitch-users] external sip profile
Message-ID: <2cef777b0908290905jea80cech40b8b75895f28218@mail.gmail.com>

I am using a fresh install of freeswitch trunk and I am unable to
register a phone on port 5080. I set the the registrar of the phone to
server.host.com:5080 but when i do a sofia status profile external,
there are no registrations. when i do a sofia status profile internal,
it shows the phone registered to port 5060. Is this a bug in
freeswitch or is there some configuration I need to change?


From brian at freeswitch.org  Sat Aug 29 09:14:45 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 11:14:45 -0500
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <6f7c60c40908290840w466c37d6v28dcc0f38f632cce@mail.gmail.com>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
	<153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
	<6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>
	<98E01B85-5134-48FA-9200-C01D96E48AA5@freeswitch.org>
	<6f7c60c40908290840w466c37d6v28dcc0f38f632cce@mail.gmail.com>
Message-ID: <7400B5BC-EC99-464E-B97F-D3DCD3CE2180@freeswitch.org>

No you shouldn't have to touch ANYTHING in the default configs at all  
to get 1000 with password 1234 to register... I recommend you:

cd /usr/local/freeswitch
mv conf conf.old
then do make samples again.

/b

On Aug 29, 2009, at 10:40 AM, tom wrote:

>
> ok, what u sent is fine, i have that. do i have to do somethign in  
> that section:?
>   <domains>
>     <!-- indicator to parse the directory for domains with  
> parse="true" to get gateways-->
>     <domain name="$${domain}" parse="true"/>
>     <!-- indicator to parse the directory for domains with  
> parse="true" to get gateways and alias every domain to this profile  
> -->
>     <!--<domain name="all" alias="true" parse="true"/>-->
>     <domain name="all" alias="true" parse="false"/>
>   </domains>
>
>
> ?thx



From brian at freeswitch.org  Sat Aug 29 09:16:05 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 11:16:05 -0500
Subject: [Freeswitch-users] external sip profile
In-Reply-To: <2cef777b0908290905jea80cech40b8b75895f28218@mail.gmail.com>
References: <2cef777b0908290905jea80cech40b8b75895f28218@mail.gmail.com>
Message-ID: <53656C07-1D94-45EB-88D0-81A6C59058A9@freeswitch.org>

No, enable sip trace I suspect your config is wrong in the end point.

/b

On Aug 29, 2009, at 11:05 AM, e schmidbauer wrote:

> I am using a fresh install of freeswitch trunk and I am unable to
> register a phone on port 5080. I set the the registrar of the phone to
> server.host.com:5080 but when i do a sofia status profile external,
> there are no registrations. when i do a sofia status profile internal,
> it shows the phone registered to port 5060. Is this a bug in
> freeswitch or is there some configuration I need to change?



From e.schmidbauer at gmail.com  Sat Aug 29 10:20:42 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Sat, 29 Aug 2009 13:20:42 -0400
Subject: [Freeswitch-users] external sip profile
In-Reply-To: <53656C07-1D94-45EB-88D0-81A6C59058A9@freeswitch.org>
References: <2cef777b0908290905jea80cech40b8b75895f28218@mail.gmail.com>
	<53656C07-1D94-45EB-88D0-81A6C59058A9@freeswitch.org>
Message-ID: <2cef777b0908291020jdbeac97q3160606ffc6d6a72@mail.gmail.com>

You were correct. The client (ekiga) i was using forces the connection
on port 5060.

On Sat, Aug 29, 2009 at 12:16 PM, Brian West<brian at freeswitch.org> wrote:
> No, enable sip trace I suspect your config is wrong in the end point.
>
> /b
>
> On Aug 29, 2009, at 11:05 AM, e schmidbauer wrote:
>
>> I am using a fresh install of freeswitch trunk and I am unable to
>> register a phone on port 5080. I set the the registrar of the phone to
>> server.host.com:5080 but when i do a sofia status profile external,
>> there are no registrations. when i do a sofia status profile internal,
>> it shows the phone registered to port 5060. Is this a bug in
>> freeswitch or is there some configuration I need to change?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>


From mike at jerris.com  Sat Aug 29 10:45:08 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 29 Aug 2009 13:45:08 -0400
Subject: [Freeswitch-users] Success! and an issue.
In-Reply-To: <92598075-4C54-4EED-80D0-3263DE495C68@freeswitch.org>
References: <BAY120-W6EC265B637176ED41128BDFF40@phx.gbl>
	<92598075-4C54-4EED-80D0-3263DE495C68@freeswitch.org>
Message-ID: <2FF2B6E7-7E01-449B-87A4-F06CD65B4AF7@jerris.com>

This sounds to me like a phone that doesn't know that a re-invite is  
not a new call and session timers are turned on.  A sip trace will  
show you if this is the case, and if it is, that phone is indeed badly  
broken.

Mike


On Aug 29, 2009, at 11:15 AM, Brian West wrote:

> Can you get a sip trace and console trace because this is highly  
> unlikely.
>
> /b
>
> On Aug 29, 2009, at 4:28 AM, Christensen Tom wrote:
>
>> However, I did run into one issue, not sure if its somehow related  
>> to the phones I'm using (softphone of unknown quality (weephone) on  
>> iphones)... I'm using the default config, when I dial 4000 for  
>> voicemail, it works, however at apparently random times during the  
>> menu navigation I get call waiting beeps and I have an incoming  
>> call from 4000.  If I decline or leave this call unanswered, my  
>> call into voicemail gets disconnected.  If I "accept" this call, it  
>> apparently disconnects immediately (IE I only have 1 active call),  
>> however I can then continue navigating the menu, etc... Very  
>> strange to me, I will be setting up a couple of polycom 501s on  
>> this system tomorrow and will report if I see similar issues.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From dave at 3c.co.uk  Sat Aug 29 11:05:57 2009
From: dave at 3c.co.uk (David Knell)
Date: Sat, 29 Aug 2009 21:05:57 +0300
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <136791EF-26D2-4DC7-91F6-2D468122F06C@runsolutions.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
	<71FD2763-32C5-49CA-8E4D-7C0E301D2187@runsolutions.com>
	<136791EF-26D2-4DC7-91F6-2D468122F06C@runsolutions.com>
Message-ID: <1251569157.5146.22.camel@dk-d820>

Hi Raimund,

The difficult bit in all of this is having calls seamlessly transferred
from one box to another when the first box dies.  There's a lot of state
associated with a call, not all of which is easy to replicate across.

99.99% uptime implies an average of no more than 15 seconds downtime
during a 40 hour business week (or about one unscheduled reboot every
couple of months), which is easily achievable using FreeSWITCH (and, in
my experience, Asterisk) on standard hardware.

People agonize about their four or five nines too much, in my opinion.
Folk are used to their phones crashing, needing rebooting and dropping
calls - we've cellphones to thank for that - and a half-decent VoIP
solution will knock the spots off your favourite mobile carrier for
reliability.  Plus there's all the external factors - 99.999% uptime on
your PRI means that someone can only drive a digger through the cable
once every 273 years, assuming that it takes a day to fix, and no
telco's going to give you that in an SLA.  And your power can't go off.
And so on.

Lastly, I'm afraid that virtualization and VoIP don't play well
together, at least not if you want to achieve a sensible density.  The
large number of small packets being moved around the network interfaces
- both physical and virtual - will quickly chew up your CPU.

Cheers --

Dave


> Thinking about it, maybe we can create a solution, if some of us work
> together:
> 
> 
> My strength are in virtualization, linux, development, databases,
> integration, etc.
> What I do not now much about is how SIP (and everything else for that
> matter in the Voice world) works under the hood, and how it's
> implemented in FS.
> 
> 
> I know that the state information for a call has to be stored and
> retrieved somewhere and somehow, only I do not know that part. What I
> know is that it hast to be do-able to store all the stream information
> (ip's, port's, current state's, etc.) in a very fast database (e.g. my
> idea would be memcached) so another FS could just take this
> information and take over the call, maybe you loose a second of voice,
> maybe you loose the recorded call file or a part of it, but that
> should be it. (SipFoundry has a boxed opensource PBX, which, of course
> is not flexible like FreeSWITCH or Asterisk, but has Call Live
> Migration and Call Live Failover integrated!).
> 
> 
> What I want is for my company to be able to sell a 99.99 uptime PBX
> (we do mostly call-center related stuff), which can scale well, and
> can grow with the company without lot's of hassles, my Dream would be:
> 
> 
> To begin with: 
> 
> 
> One Hardware Node with the essential hardware (digium cards for
> example).
> On this node are OpenVZ virtualized containers:
> [VirtCnt1: FS which only talks to the Hardware and forwards
> everything] = Could be replaced with hardware media gateway, etc.
> [VirtCnt2: FS which handles the PBX] \___ Loadbalanced, with odbc or
> xml, Failover, Livetakeover
> [VirtCnt3: FS which handles the PBX] /
> [VirtCnt4: Database for state information] (maybe something as
> resource-friendly as memcached? ressource heavvy database?)
> 
> 
> With this we can achieve all this:
> 
> 
> Problem with VirtCnt2 (e.g. crash, lock, ...)
> * VirtCnt3 can take over. 
> -> You are free without stress to investigate the problem, you can
> debug and analyze whyle the machine is still running
> -> you can also create a machine-state-dump of the virtual container,
> dump the container as well, copy the data to your lab and restore the
> machine up the state which it was running with the problem, so you can
> liveinvestigate it in the lab (some prerequirements given, but easy
> doable)
> -> just think about the possibility of better bugreports because
> someone can take the time to read out all the data with GDB to
> investigate the proper cause of a machine Lock!
> 
> 
> You want to upgrade to a new FreeSWITCH version?
> * Take VirtCnt2 out of the LoadBalancing Scheme,
> * Stop it, Clone it, 
> * Upgrade FreeSWITCH in the cloned Container
> * Start the cloned container
> * if there's something wrong, stop it and restart the original
> VirtCnt2
> -> No problem at all, you can Test on the Live Hardware, with part of
> the Live users (maybe a low-volume queue) to be sure everything works
> out fine before you activate the full loadbalance
> 
> 
> Server on it's own can't handle the load
> * Buy new machine
> * Setup Hardware Node
> * Livemigrate VirtCnt3 (no downtime) 
> 
> 
> Now the first Server with the VrtCnt1 and VirtCnt2 as well has to much
> load
> * Buy new machine
> * Setup Hardware Node
> * Livemigrate VirtCnt2 (no downtime)
> -> Now you have a 3 server solution (1 mediaprox, 2 loadbalanced /
> failover PBXes) out of the first box you bought, without headaches,
> because the system was built for it from the beginning!
> 
> 
> The Database drains to much?
> * Buy new machine
> * Setup Hardware Node
> * Livemigrate database VirtCnt4 (no downtime)
> 
> 
> You want to upgrade Hardware/Kernel in Hardware node 1?
> * Livemigrate VirtCnt2 to a hotstandby machine, or to the other PBX
> machine, upgrade the hardware, Re-Livemigrate the containers. (no
> downtime)
> * OR just break the loadbalancing, wait until all current calls are
> teared down correctly, upgrade machine, reenable the loadbalancer
> 
> 
> You want an exact copy of the first server for Hardware HA?
> * Buy new machine
> * Setup Hardware node
> * Buy hardware PRI switchover box
> * Clone VirtCnt1 - VirtCnt4 to the new machine
> * Make basic failover configuration 
> 
> 
> 
> 
> -> the sky's the limit, as the saying goes ...
> 
> 
> 
> 
> So, I can do all the openvz stuff and the integration with database /
> memcached / heartbeat / whatever is needed here, someone there to be
> willing to work with me on this on the FreeSWITCH side? or at least
> provide me with the necessary information about what's needed / how to
> talk / what states from FreeSWITCH?
> 
> 
> I know this seems very ambitious but if this could be made in a rather
> relativly easy to setup package, with good documentation, it would be
> a boost for FreeSWITCH, i am sure, because after all this is what
> everyone is grown accustomed to from good old phone companys and the
> good old pbx's: carrier grade uptimes ...
> 
> 
> Thanks for everyone reading up until here,
> all the best,
> 
> 
> Ray
> 
> 
> 
> 
> 
> 
> -- 
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
> 
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
> 
> On Aug 29, 2009, at 3:17 PM, Raimund Sacherer wrote:
> 
> > Oh yeah, that would be so helpfull for my situation, as my client
> > *demands* now a solution where he can press a big red button and all
> > fails over to another box. Hi es totally scared because of the
> > Lockups in Asterisk which under specific situations including AMI,
> > Automated Call Setup, and murphy led to a lockup of the entire
> > machine, no console was working anymore, only cold-reset could do
> > it.
> > 
> > 
> > So, IF there is the possibility for life-takeover, / failover etc. I
> > would love to here how has been done.
> > 
> > 
> > I am very experienced with openvz and use for about two years now
> > only openvz virtualization servers for anything because of
> > live-migration etc. But as I am new in this company we could not
> > adopt this until now.
> > 
> > 
> > So Please Ken, if you can, describe what need's to be done to get a
> > failover / takeover working (an outline would be enough)
> > 
> > 
> > Thanks in Advance
> > 
> > -- 
> > Raimund Sacherer
> > -
> > RunSolutions
> >     Open Source It Consulting
> > -
> > 
> > Parc Bit - Centro Empresarial Son Espanyol
> > Edificio Estel - Local 3D
> > 07121 -  Palma de Mallorca
> > Baleares
> > 
> > On Aug 29, 2009, at 11:58 AM, Steve Kurzeja wrote:
> > 
> > > On Sat, Aug 29, 2009 at 2:34 PM, Diego Viola
> > > <diego.viola at gmail.com> wrote:
> > >         Yes, FreeSWITCH is a system that you can trust 100%. I
> > >         have switched my Asterisk servers to FreeSWITCH and have
> > >         peace now.
> > >         
> > >         If I were you I would get rid of Asterisk and use
> > >         FreeSWITCH, FS will handle all what you want very well.
> > >         
> > >         And I agree with David, fail-over is kinda irrelevant
> > >         since the FS doesn't crash like Asterisk does.
> > >         
> > >         
> > >  
> > > You still have hardware failures and fail-over is also useful for
> > > hit-less maintenance on boxes.
> > > 
> > > I'd be interested to know how Brian West was approaching his live
> > > migration work. 
> > > 
> > > Steve
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From erik at erikwickstrom.com  Sat Aug 29 11:21:03 2009
From: erik at erikwickstrom.com (Erik Wickstrom)
Date: Sat, 29 Aug 2009 11:21:03 -0700
Subject: [Freeswitch-users] Transferring to another extension is VERY slow.
Message-ID: <3d381e170908291121k3bd64ecnacf4f7941962e2ed@mail.gmail.com>

Hi,

I'm having some trouble debugging why the transfer command is so slow
in my case.  I have a lua IVR setup which responds to user selections
by transferring the call to another dialplan ext.  Pretty basic stuff.

The problem is, it takes 30-40 seconds for the the transfer to
actually start after the key press.  After about 30 seconds of
silence, "hold music" finally starts and the other extension finally
answers shortly thereafter.

I've tried transferring to endpoint extensions, the 5000 sample IVR,
9999 hold music -- in all cases there is this long pause.

Does anyone know what could be causing this?

Thanks!
Erik


From rs at runsolutions.com  Sat Aug 29 11:33:21 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Sat, 29 Aug 2009 20:33:21 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <191c3a030908290901w56b4fd26h467d75ec58dbc380@mail.gmail.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
	<A9DC8F8B-B0D3-4AC3-A67B-88F5CFCFDB3A@freeswitch.org>
	<191c3a030908290901w56b4fd26h467d75ec58dbc380@mail.gmail.com>
Message-ID: <261B13B1-CBE5-4AE4-9ACB-3AE19D556B8A@runsolutions.com>

Hmm, so basically 100 interested companys which each chip in 1000  
bucks :-)

sounds like lots of manpower, but on the other hand, I do not know the  
issues regarding the SIP protocoll, but, basically, isn't it *just* to  
tell another FS box to listen on port x for voicetraffic, forward it  
to ip on port y?

ok, i understand there's a lot going on under the hood, i guess it  
would mean to setup a call, but take care to not really set up the  
call, just the internal state ...

hmm, could it theoretically be done with the event system? ok, i guess  
I have to dive further into the internals to fully understand the scope.


But a live migration, where the box is available, be possible right  
now? Would be a step i would like to implement just to be able to do  
work on a hardware node if necesary without interrupting the service ...


-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 29, 2009, at 6:01 PM, Anthony Minessale wrote:

> We have previously estimated the development of live fail over  
> (after a box dies where live migration is no longer possible) to  
> exceed 100k in development costs.
>
> It requires several additions to the sofia sip library, freeswitch  
> and a dependancy on some other code we would have to implement to  
> manage it.
>
> It may or may not be worth it to raise that kind of funding just to  
> avoid an occasional disaster.
>
> Then there is a matter of securing the time of the developers  
> necessary to carry out the implementation.
>
>
>> On Aug 29, 2009 10:19 AM, "Brian West" <brian at freeswitch.org> wrote:
>>
>> I was able to do this using OpenVZ, You can get away with it on
>> smaller instances... like if you're doing one instance per company  
>> but
>> don't expect live migration to work as well on large instances with
>> thousands of calls up at once. You need a fast network, fast disks  
>> and
>> to follow the howto on the wiki.
>>
>> /b
>> On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: > You still have  
>> hardware failures and fail-over...
>>
>> _______________________________________________ FreeSWITCH-users  
>> mailing list FreeSWITCH-users at lists...
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mitul at enterux.com  Sat Aug 29 11:36:24 2009
From: mitul at enterux.com (Mitul Limbani)
Date: Sun, 30 Aug 2009 00:06:24 +0530
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <1251569157.5146.22.camel@dk-d820>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
	<71FD2763-32C5-49CA-8E4D-7C0E301D2187@runsolutions.com>
	<136791EF-26D2-4DC7-91F6-2D468122F06C@runsolutions.com>
	<1251569157.5146.22.camel@dk-d820>
Message-ID: <416B910B-6058-4387-9563-3491B091D8B4@enterux.com>

On PRI circuits failover with ha is possible using a red tone  
appliance, but I m sure it's going to be almost impossible to transfer  
the running calls during the event of fs box going down.

Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com

On 29-Aug-2009, at 11:35 PM, David Knell <dave at 3c.co.uk> wrote:

> Hi Raimund,
>
> The difficult bit in all of this is having calls seamlessly  
> transferred
> from one box to another when the first box dies.  There's a lot of  
> state
> associated with a call, not all of which is easy to replicate across.
>
> 99.99% uptime implies an average of no more than 15 seconds downtime
> during a 40 hour business week (or about one unscheduled reboot every
> couple of months), which is easily achievable using FreeSWITCH (and,  
> in
> my experience, Asterisk) on standard hardware.
>
> People agonize about their four or five nines too much, in my opinion.
> Folk are used to their phones crashing, needing rebooting and dropping
> calls - we've cellphones to thank for that - and a half-decent VoIP
> solution will knock the spots off your favourite mobile carrier for
> reliability.  Plus there's all the external factors - 99.999% uptime  
> on
> your PRI means that someone can only drive a digger through the cable
> once every 273 years, assuming that it takes a day to fix, and no
> telco's going to give you that in an SLA.  And your power can't go  
> off.
> And so on.
>
> Lastly, I'm afraid that virtualization and VoIP don't play well
> together, at least not if you want to achieve a sensible density.  The
> large number of small packets being moved around the network  
> interfaces
> - both physical and virtual - will quickly chew up your CPU.
>
> Cheers --
>
> Dave
>
>
>> Thinking about it, maybe we can create a solution, if some of us work
>> together:
>>
>>
>> My strength are in virtualization, linux, development, databases,
>> integration, etc.
>> What I do not now much about is how SIP (and everything else for that
>> matter in the Voice world) works under the hood, and how it's
>> implemented in FS.
>>
>>
>> I know that the state information for a call has to be stored and
>> retrieved somewhere and somehow, only I do not know that part. What I
>> know is that it hast to be do-able to store all the stream  
>> information
>> (ip's, port's, current state's, etc.) in a very fast database (e.g.  
>> my
>> idea would be memcached) so another FS could just take this
>> information and take over the call, maybe you loose a second of  
>> voice,
>> maybe you loose the recorded call file or a part of it, but that
>> should be it. (SipFoundry has a boxed opensource PBX, which, of  
>> course
>> is not flexible like FreeSWITCH or Asterisk, but has Call Live
>> Migration and Call Live Failover integrated!).
>>
>>
>> What I want is for my company to be able to sell a 99.99 uptime PBX
>> (we do mostly call-center related stuff), which can scale well, and
>> can grow with the company without lot's of hassles, my Dream would  
>> be:
>>
>>
>> To begin with:
>>
>>
>> One Hardware Node with the essential hardware (digium cards for
>> example).
>> On this node are OpenVZ virtualized containers:
>> [VirtCnt1: FS which only talks to the Hardware and forwards
>> everything] = Could be replaced with hardware media gateway, etc.
>> [VirtCnt2: FS which handles the PBX] \___ Loadbalanced, with odbc or
>> xml, Failover, Livetakeover
>> [VirtCnt3: FS which handles the PBX] /
>> [VirtCnt4: Database for state information] (maybe something as
>> resource-friendly as memcached? ressource heavvy database?)
>>
>>
>> With this we can achieve all this:
>>
>>
>> Problem with VirtCnt2 (e.g. crash, lock, ...)
>> * VirtCnt3 can take over.
>> -> You are free without stress to investigate the problem, you can
>> debug and analyze whyle the machine is still running
>> -> you can also create a machine-state-dump of the virtual container,
>> dump the container as well, copy the data to your lab and restore the
>> machine up the state which it was running with the problem, so you  
>> can
>> liveinvestigate it in the lab (some prerequirements given, but easy
>> doable)
>> -> just think about the possibility of better bugreports because
>> someone can take the time to read out all the data with GDB to
>> investigate the proper cause of a machine Lock!
>>
>>
>> You want to upgrade to a new FreeSWITCH version?
>> * Take VirtCnt2 out of the LoadBalancing Scheme,
>> * Stop it, Clone it,
>> * Upgrade FreeSWITCH in the cloned Container
>> * Start the cloned container
>> * if there's something wrong, stop it and restart the original
>> VirtCnt2
>> -> No problem at all, you can Test on the Live Hardware, with part of
>> the Live users (maybe a low-volume queue) to be sure everything works
>> out fine before you activate the full loadbalance
>>
>>
>> Server on it's own can't handle the load
>> * Buy new machine
>> * Setup Hardware Node
>> * Livemigrate VirtCnt3 (no downtime)
>>
>>
>> Now the first Server with the VrtCnt1 and VirtCnt2 as well has to  
>> much
>> load
>> * Buy new machine
>> * Setup Hardware Node
>> * Livemigrate VirtCnt2 (no downtime)
>> -> Now you have a 3 server solution (1 mediaprox, 2 loadbalanced /
>> failover PBXes) out of the first box you bought, without headaches,
>> because the system was built for it from the beginning!
>>
>>
>> The Database drains to much?
>> * Buy new machine
>> * Setup Hardware Node
>> * Livemigrate database VirtCnt4 (no downtime)
>>
>>
>> You want to upgrade Hardware/Kernel in Hardware node 1?
>> * Livemigrate VirtCnt2 to a hotstandby machine, or to the other PBX
>> machine, upgrade the hardware, Re-Livemigrate the containers. (no
>> downtime)
>> * OR just break the loadbalancing, wait until all current calls are
>> teared down correctly, upgrade machine, reenable the loadbalancer
>>
>>
>> You want an exact copy of the first server for Hardware HA?
>> * Buy new machine
>> * Setup Hardware node
>> * Buy hardware PRI switchover box
>> * Clone VirtCnt1 - VirtCnt4 to the new machine
>> * Make basic failover configuration
>>
>>
>>
>>
>> -> the sky's the limit, as the saying goes ...
>>
>>
>>
>>
>> So, I can do all the openvz stuff and the integration with database /
>> memcached / heartbeat / whatever is needed here, someone there to be
>> willing to work with me on this on the FreeSWITCH side? or at least
>> provide me with the necessary information about what's needed / how  
>> to
>> talk / what states from FreeSWITCH?
>>
>>
>> I know this seems very ambitious but if this could be made in a  
>> rather
>> relativly easy to setup package, with good documentation, it would be
>> a boost for FreeSWITCH, i am sure, because after all this is what
>> everyone is grown accustomed to from good old phone companys and the
>> good old pbx's: carrier grade uptimes ...
>>
>>
>> Thanks for everyone reading up until here,
>> all the best,
>>
>>
>> Ray
>>
>>
>>
>>
>>
>>
>> -- 
>> Raimund Sacherer
>> -
>> RunSolutions
>>    Open Source It Consulting
>> -
>>
>> Parc Bit - Centro Empresarial Son Espanyol
>> Edificio Estel - Local 3D
>> 07121 -  Palma de Mallorca
>> Baleares
>>
>> On Aug 29, 2009, at 3:17 PM, Raimund Sacherer wrote:
>>
>>> Oh yeah, that would be so helpfull for my situation, as my client
>>> *demands* now a solution where he can press a big red button and all
>>> fails over to another box. Hi es totally scared because of the
>>> Lockups in Asterisk which under specific situations including AMI,
>>> Automated Call Setup, and murphy led to a lockup of the entire
>>> machine, no console was working anymore, only cold-reset could do
>>> it.
>>>
>>>
>>> So, IF there is the possibility for life-takeover, / failover etc. I
>>> would love to here how has been done.
>>>
>>>
>>> I am very experienced with openvz and use for about two years now
>>> only openvz virtualization servers for anything because of
>>> live-migration etc. But as I am new in this company we could not
>>> adopt this until now.
>>>
>>>
>>> So Please Ken, if you can, describe what need's to be done to get a
>>> failover / takeover working (an outline would be enough)
>>>
>>>
>>> Thanks in Advance
>>>
>>> -- 
>>> Raimund Sacherer
>>> -
>>> RunSolutions
>>>    Open Source It Consulting
>>> -
>>>
>>> Parc Bit - Centro Empresarial Son Espanyol
>>> Edificio Estel - Local 3D
>>> 07121 -  Palma de Mallorca
>>> Baleares
>>>
>>> On Aug 29, 2009, at 11:58 AM, Steve Kurzeja wrote:
>>>
>>>> On Sat, Aug 29, 2009 at 2:34 PM, Diego Viola
>>>> <diego.viola at gmail.com> wrote:
>>>>        Yes, FreeSWITCH is a system that you can trust 100%. I
>>>>        have switched my Asterisk servers to FreeSWITCH and have
>>>>        peace now.
>>>>
>>>>        If I were you I would get rid of Asterisk and use
>>>>        FreeSWITCH, FS will handle all what you want very well.
>>>>
>>>>        And I agree with David, fail-over is kinda irrelevant
>>>>        since the FS doesn't crash like Asterisk does.
>>>>
>>>>
>>>>
>>>> You still have hardware failures and fail-over is also useful for
>>>> hit-less maintenance on boxes.
>>>>
>>>> I'd be interested to know how Brian West was approaching his live
>>>> migration work.
>>>>
>>>> Steve
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
> -- 
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org


From krice at freeswitch.org  Sat Aug 29 11:44:26 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 29 Aug 2009 13:44:26 -0500
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <261B13B1-CBE5-4AE4-9ACB-3AE19D556B8A@runsolutions.com>
Message-ID: <C6BEE13A.5E372%krice@freeswitch.org>

It is not possible to do a live migration at all right now... There is no
way to move a call and have all the states required for that call to
magically re-appear on a different instance. This will require a fair bit of
work to get there. 

It is possible to configure fs via some back end DB magic to share
configurations ie: n+1 or ?warm standby? style fault tolerance but you are
going to loose the calls that are up when the failure occurs.


From: Raimund Sacherer <rs at runsolutions.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Sat, 29 Aug 2009 20:33:21 +0200
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

Hmm, so basically 100 interested companys which each chip in 1000 bucks :-)

sounds like lots of manpower, but on the other hand, I do not know the
issues regarding the SIP protocoll, but, basically, isn't it *just* to tell
another FS box to listen on port x for voicetraffic, forward it to ip on
port y?

ok, i understand there's a lot going on under the hood, i guess it would
mean to setup a call, but take care to not really set up the call, just the
internal state ...

hmm, could it theoretically be done with the event system? ok, i guess I
have to dive further into the internals to fully understand the scope.


But a live migration, where the box is available, be possible right now?
Would be a step i would like to implement just to be able to do work on a
hardware node if necesary without interrupting the service ...


-- 
Raimund Sacherer
-
RunSolutions
    Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 29, 2009, at 6:01 PM, Anthony Minessale wrote:

> 
> We have previously estimated the development of live fail over (after a box
> dies where live migration is no longer possible) to exceed 100k in development
> costs.
> 
> It requires several additions to the sofia sip library, freeswitch and a
> dependancy on some other code we would have to implement to manage it.
> 
> It may or may not be worth it to raise that kind of funding just to avoid an
> occasional disaster.
> 
> Then there is a matter of securing the time of the developers necessary to
> carry out the implementation.
> 
> 
>> On Aug 29, 2009 10:19 AM, "Brian West" <brian at freeswitch.org> wrote:
>> 
>> I was able to do this using OpenVZ, You can get away with it on
>> smaller instances... like if you're doing one instance per company but
>> don't expect live migration to work as well on large instances with
>> thousands of calls up at once. You need a fast network, fast disks and
>> to follow the howto on the wiki.
>> 
>> /b
>> 
>> On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: > You still have hardware
>> failures and fail-over...
>> 
>> _______________________________________________ FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists...
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org

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From mgg at giagnocavo.net  Sat Aug 29 12:39:52 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Sat, 29 Aug 2009 15:39:52 -0400
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <261B13B1-CBE5-4AE4-9ACB-3AE19D556B8A@runsolutions.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
	<A9DC8F8B-B0D3-4AC3-A67B-88F5CFCFDB3A@freeswitch.org>
	<191c3a030908290901w56b4fd26h467d75ec58dbc380@mail.gmail.com>
	<261B13B1-CBE5-4AE4-9ACB-3AE19D556B8A@runsolutions.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702D7DF5127@mse17be1.mse17.exchange.ms>

Also please remember this means every _application_ also must be "failover" aware. So, if you're using a little calling card or billing app to control your routing, that app will have to explicitly enroll its state as well as restore itself.

Live migration is possible via OpenVZ. OVZ does not have the same overhead of other virtualization techniques such as Hyper-V, ESX, Xen, etc. So you can do media without much worry.  OVZ can work with other virtualization, too; you can spin it up inside of a VMware machine.

But this is *live* migration of the container. It's "just" pausing your container, copying the whole memory, and resuming it on another node. This is not "dead" migration or failover after calls break. However, you could try running a very frequent sync in OVZ, so that if there was a node failure, you could resume from a previously saved state. But, I highly doubt performance would be good enough for this to work successfully on any serious deployment.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raimund Sacherer
Sent: Saturday, August 29, 2009 12:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

Hmm, so basically 100 interested companys which each chip in 1000 bucks :-)

sounds like lots of manpower, but on the other hand, I do not know the issues regarding the SIP protocoll, but, basically, isn't it *just* to tell another FS box to listen on port x for voicetraffic, forward it to ip on port y?

ok, i understand there's a lot going on under the hood, i guess it would mean to setup a call, but take care to not really set up the call, just the internal state ...

hmm, could it theoretically be done with the event system? ok, i guess I have to dive further into the internals to fully understand the scope.


But a live migration, where the box is available, be possible right now? Would be a step i would like to implement just to be able to do work on a hardware node if necesary without interrupting the service ...


--
Raimund Sacherer
-
RunSolutions
    Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 29, 2009, at 6:01 PM, Anthony Minessale wrote:



We have previously estimated the development of live fail over (after a box dies where live migration is no longer possible) to exceed 100k in development costs.

It requires several additions to the sofia sip library, freeswitch and a dependancy on some other code we would have to implement to manage it.

It may or may not be worth it to raise that kind of funding just to avoid an occasional disaster.

Then there is a matter of securing the time of the developers necessary to carry out the implementation.

On Aug 29, 2009 10:19 AM, "Brian West" <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:

I was able to do this using OpenVZ, You can get away with it on
smaller instances... like if you're doing one instance per company but
don't expect live migration to work as well on large instances with
thousands of calls up at once. You need a fast network, fast disks and
to follow the howto on the wiki.

/b

On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: > You still have hardware failures and fail-over...

_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From mitul at enterux.com  Sat Aug 29 12:56:16 2009
From: mitul at enterux.com (Mitul Limbani)
Date: Sun, 30 Aug 2009 01:26:16 +0530
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <C6BEE13A.5E372%krice@freeswitch.org>
References: <C6BEE13A.5E372%krice@freeswitch.org>
Message-ID: <A6E1A4F9-F0D5-4645-A72D-26365CF6FC7B@enterux.com>

Is itnpossible to have a db cluster know the state of each and every  
call and then use Heartbeat on this db +
fs cluster so that clients see only one ip where as internally all fs  
boxes refer db for call states, db again is under replication.

This in the thioery can be written, but I am sure if we think bit more  
on this direction the problem seem to be getting addressed.

Other guys also chip in their 2 cents, we just need 50 of em to make a  
full dollar.

Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com

On 30-Aug-2009, at 12:14 AM, Ken Rice <krice at freeswitch.org> wrote:

> It is not possible to do a live migration at all right now... There  
> is no way to move a call and have all the states required for that  
> call to magically re-appear on a different instance. This will  
> require a fair bit of work to get there.
>
> It is possible to configure fs via some back end DB magic to share  
> configurations ie: n+1 or ?warm standby? style fault tolerance  
> but you are going to loose the calls that are up when the failure oc 
> curs.
>
> From: Raimund Sacherer <rs at runsolutions.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 29 Aug 2009 20:33:21 +0200
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
>
> Hmm, so basically 100 interested companys which each chip in 1000  
> bucks :-)
>
> sounds like lots of manpower, but on the other hand, I do not know  
> the issues regarding the SIP protocoll, but, basically, isn't it  
> *just* to tell another FS box to listen on port x for voicetraffic,  
> forward it to ip on port y?
>
> ok, i understand there's a lot going on under the hood, i guess it  
> would mean to setup a call, but take care to not really set up the  
> call, just the internal state ...
>
> hmm, could it theoretically be done with the event system? ok, i  
> guess I have to dive further into the internals to fully understand  
> the scope.
>
>
> But a live migration, where the box is available, be possible right  
> now? Would be a step i would like to implement just to be able to do  
> work on a hardware node if necesary without interrupting the  
> service ...
>
>
> -- 
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
> On Aug 29, 2009, at 6:01 PM, Anthony Minessale wrote:
>
>
> We have previously estimated the development of live fail over  
> (after a box dies where live migration is no longer possible) to  
> exceed 100k in development costs.
>
> It requires several additions to the sofia sip library, freeswitch  
> and a dependancy on some other code we would have to implement to  
> manage it.
>
> It may or may not be worth it to raise that kind of funding just to  
> avoid an occasional disaster.
>
> Then there is a matter of securing the time of the developers  
> necessary to carry out the implementation.
>
>
> On Aug 29, 2009 10:19 AM, "Brian West" <brian at freeswitch.org> wrote:
>
> I was able to do this using OpenVZ, You can get away with it on
> smaller instances... like if you're doing one instance per company but
> don't expect live migration to work as well on large instances with
> thousands of calls up at once. You need a fast network, fast disks and
> to follow the howto on the wiki.
>
> /b
>
> On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: > You still have  
> hardware failures and fail-over...
>
> _______________________________________________ FreeSWITCH-users  
> mailing list FreeSWITCH-users at lists...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From brian at freeswitch.org  Sat Aug 29 13:04:25 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 15:04:25 -0500
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <A6E1A4F9-F0D5-4645-A72D-26365CF6FC7B@enterux.com>
References: <C6BEE13A.5E372%krice@freeswitch.org>
	<A6E1A4F9-F0D5-4645-A72D-26365CF6FC7B@enterux.com>
Message-ID: <B55A1F03-DC5E-4F32-99C8-3AB9D11B2C8A@freeswitch.org>

The sip stack needs to be modified to spin that data up into the state  
machine so that it can take over calls once the fail over takes  
place... its not an easy task.

/b

On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:

> Is itnpossible to have a db cluster know the state of each and every  
> call and then use Heartbeat on this db +
> fs cluster so that clients see only one ip where as internally all  
> fs boxes refer db for call states, db again is under replication.
>
> This in the thioery can be written, but I am sure if we think bit  
> more on this direction the problem seem to be getting addressed.
>
> Other guys also chip in their 2 cents, we just need 50 of em to make  
> a full dollar.
>
> Thanks & Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions Pvt. Ltd.,
> The Enterprise Linux Company (r),
> http://www.enterux.com
> http://www.entVoice.com

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From pete at privateconnect.com  Sat Aug 29 14:25:31 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Sat, 29 Aug 2009 14:25:31 -0700
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Message-ID: <20090829142531.2ad02225396a31c9de30536f2e338977.911e38f47a.wbe@email04.secureserver.net>

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From pete at privateconnect.com  Sat Aug 29 14:52:29 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Sat, 29 Aug 2009 14:52:29 -0700
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Message-ID: <20090829145229.2ad02225396a31c9de30536f2e338977.c53fcdc891.wbe@email04.secureserver.net>

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From sdjernes at gmail.com  Sat Aug 29 14:55:41 2009
From: sdjernes at gmail.com (Shawn L. Djernes)
Date: Sat, 29 Aug 2009 16:55:41 -0500
Subject: [Freeswitch-users] Group Call
In-Reply-To: <81148697-292E-4E9C-A26A-D1536E9D8A8E@freeswitch.org>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
	<81148697-292E-4E9C-A26A-D1536E9D8A8E@freeswitch.org>
Message-ID: <5ff3e3a70908291455n61f34479t29f83b60528aadc0@mail.gmail.com>

This seems to be a problem in the 1.0.4 tarball release.  A friend with SVN
14305 the group_call from fs_cli works.
Shawn

On Sat, Aug 29, 2009 at 10:12 AM, Brian West <brian at freeswitch.org> wrote:

> Don't confuse this with the actual group/ endpoint.
>
> /b
>
> On Aug 25, 2009, at 11:41 AM, Jo?o Mesquita wrote:
>
> > According to wiki:
> > group,[insert|delete|call]:<group name>:<url>,group [insert|delete|
> > call
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shawn L. Djernes
SD Consulting
shawn at djernes.org | sdjernes at gmail.com
MSN: wizardwlf at hotmail.com
402.345.7734 | 402.350.6973 Cell
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From brian at freeswitch.org  Sat Aug 29 17:14:08 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 19:14:08 -0500
Subject: [Freeswitch-users] Group Call
In-Reply-To: <5ff3e3a70908291455n61f34479t29f83b60528aadc0@mail.gmail.com>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
	<81148697-292E-4E9C-A26A-D1536E9D8A8E@freeswitch.org>
	<5ff3e3a70908291455n61f34479t29f83b60528aadc0@mail.gmail.com>
Message-ID: <DE1C323C-ADE0-4575-993F-6D6BB6649AA6@freeswitch.org>

Group call works for me on mine... I suspect your config is not  
correct... tar up your directory and mail it to me off list if you can  
please.

freeswitch at default> version
FreeSWITCH Version 1.0.trunk (14665M)

freeswitch at default> group_call default
[presence_id=1000 at example.com]sofia/internal/sip:1000 at 192.168.1.31,[presence_id=1006 at example.com 
]sofia/internal/sip:gw+11 at 192.168.1.112:6060;transport=udp,[presence_id=1007 at example.com 
]sofia/internal/sip:1007 at 192.168.1.22, 
[presence_id=1015 at example.com]sofia/internal/sip:1015 at 192.168.1.101:5061;fs_nat=yes;fs_path=sip%3A1015%4080.93.247.26%3A1100
freeswitch at default> group_call sales
[presence_id=1000 at example.com]sofia/internal/sip:1000 at 192.168.1.31,[presence_id=1004 at example.com 
]sofia/internal/sip:gw+15 at 12.237.205.134:6060;transport=udp
freeswitch at default> group_call billing
[presence_id=1006 at example.com]sofia/internal/sip:gw+11 at 192.168.1.112:6060;transport=udp 
,[presence_id=1007 at example.com]sofia/internal/sip:1007 at 192.168.1.22
freeswitch at default>


Thanks,
/b

On Aug 29, 2009, at 4:55 PM, Shawn L. Djernes wrote:

> This seems to be a problem in the 1.0.4 tarball release.  A friend  
> with SVN 14305 the group_call from fs_cli works.
>
> Shawn



From tomabroad at gmail.com  Sat Aug 29 17:33:22 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 20:33:22 -0400
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <7400B5BC-EC99-464E-B97F-D3DCD3CE2180@freeswitch.org>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
	<153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
	<6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>
	<98E01B85-5134-48FA-9200-C01D96E48AA5@freeswitch.org>
	<6f7c60c40908290840w466c37d6v28dcc0f38f632cce@mail.gmail.com>
	<7400B5BC-EC99-464E-B97F-D3DCD3CE2180@freeswitch.org>
Message-ID: <6f7c60c40908291733t527068e4y92310ba83569e3fc@mail.gmail.com>

hi and thx for all ur help! i basically re-svned & installed it & internally
is fine now. playtime ;-)
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From tomabroad at gmail.com  Sat Aug 29 17:34:09 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 20:34:09 -0400
Subject: [Freeswitch-users] sipdroid
Message-ID: <6f7c60c40908291734k555b06f5hf321e6e2a514f5c2@mail.gmail.com>

hi,

sipdroid was working with asterisk, anyone running it with FS?

thx
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From tomabroad at gmail.com  Sat Aug 29 17:49:52 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 20:49:52 -0400
Subject: [Freeswitch-users] FS - external users
Message-ID: <6f7c60c40908291749sbb3e63do3011f8e251e2332a@mail.gmail.com>

hi,

as newbie here a simple one:

based on the std-installation, what do i need to do to have a user connected
from the outside. lets assume i want to leave 1000-1020 as they are, i make
a copy of 1020, call it 1021 and then?
should i leave it where it is? do i have to move the file? and what about he
conetxt?

thx
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From max.bridgewater at gmail.com  Sat Aug 29 18:30:10 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sat, 29 Aug 2009 21:30:10 -0400
Subject: [Freeswitch-users] What is the difference between preAnswer and
	ring_ready?
Message-ID: <c55d46950908291830w6b83a6a7h17a6faf67732d58a@mail.gmail.com>

Hi,
Assuming an inbound call, I have trouble understanding what the supposed
difference between the following two set of instructions is:

session.execute("ring_ready");
session.set("set", "ringback=/home/ring.wav");

and

session.preAnswer();
session.streamFile("/home/ring.wav")

In practice, however, the first scenario doesn't work for me although the
example here suggests otherwise:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready
Any idea?

Thanks,
Max.
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From jason at jasonjgw.net  Sat Aug 29 18:50:30 2009
From: jason at jasonjgw.net (Jason White)
Date: Sun, 30 Aug 2009 11:50:30 +1000
Subject: [Freeswitch-users] What is the difference between preAnswer and
 ring_ready?
In-Reply-To: <c55d46950908291830w6b83a6a7h17a6faf67732d58a@mail.gmail.com>
References: <c55d46950908291830w6b83a6a7h17a6faf67732d58a@mail.gmail.com>
Message-ID: <20090830015030.GA3163@jdc.jasonjgw.net>

Max Bridgewater <max.bridgewater at gmail.com> wrote:
> Hi,
> Assuming an inbound call, I have trouble understanding what the supposed
> difference between the following two set of instructions is:
> 
> session.execute("ring_ready");
> session.set("set", "ringback=/home/ring.wav");

Ring_ready sends a SIP "ringing" message and will use your device's default
ringback sound - at least, that has been my experience. I should remember the
message number in the SIP protocol, but I don't (the SIP experts here will
undoubtedly have the details at the ready).
> 
> and
> 
> session.preAnswer();
> session.streamFile("/home/ring.wav")

I think that will play the file as early media.



From tomabroad at gmail.com  Sat Aug 29 19:37:47 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 22:37:47 -0400
Subject: [Freeswitch-users] liverpie as gui-proxy?
Message-ID: <6f7c60c40908291937h57c632d5l84b6ec5ac1286c48@mail.gmail.com>

hi,

i was actually looking for something completely else, but reading wiki  led
me to the tought:
if mod_event can basically do all kinds of FS-things, why not having a
standalone ruby-on-rails as as gui which interacts with the evnet-socket via
liverpie?

im a total newbie to FS, just writing down my thoughts...


thx for ur replies
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From tomabroad at gmail.com  Sat Aug 29 19:55:14 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 22:55:14 -0400
Subject: [Freeswitch-users] FS: possible to create conference + password on
	the fly?
Message-ID: <6f7c60c40908291955i6417d51emfbade3aefc9cdf52@mail.gmail.com>

at best, from an external application, language is not important. as far i
have seen, could liverpie (aka the mod_event) do that? thx
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From sdjernes at gmail.com  Sat Aug 29 19:57:32 2009
From: sdjernes at gmail.com (Shawn L. Djernes)
Date: Sat, 29 Aug 2009 21:57:32 -0500
Subject: [Freeswitch-users] Group Call
In-Reply-To: <DE1C323C-ADE0-4575-993F-6D6BB6649AA6@freeswitch.org>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
	<81148697-292E-4E9C-A26A-D1536E9D8A8E@freeswitch.org>
	<5ff3e3a70908291455n61f34479t29f83b60528aadc0@mail.gmail.com>
	<DE1C323C-ADE0-4575-993F-6D6BB6649AA6@freeswitch.org>
Message-ID: <5ff3e3a70908291957u3b63721bjf60a37b939fced9b@mail.gmail.com>

The two systems in question are Debian with Freeswitch build using the
dpkg-buildpackage.

One of the systems has been customized but the other has a plain build and
install on it, so it is not the customization.

Is there a debian / ubuntu user on the list that can test and tell me if
this works for you.  The 14305 system is Fedora build with RPM so something
may be broken in the build system.

Thanks,
Shawn

On Sat, Aug 29, 2009 at 7:14 PM, Brian West <brian at freeswitch.org> wrote:

> Group call works for me on mine... I suspect your config is not
> correct... tar up your directory and mail it to me off list if you can
> please.
>
> freeswitch at default> version
> FreeSWITCH Version 1.0.trunk (14665M)
>
> freeswitch at default> group_call default
> [presence_id=1000 at example.com]sofia/internal/sip:1000 at 192.168.1.31<sip%3A1000 at 192.168.1.31>
> ,[presence_id=1006 at example.com
> ]sofia/internal/sip:gw+11 at 192.168.1.112:6060;transport=udp,[presence_id=
> 1007 at example.com
> ]sofia/internal/sip:1007 at 192.168.1.22 <sip%3A1007 at 192.168.1.22>,
> [presence_id=1015 at example.com]sofia/internal/sip:1015 at 192.168.1.101:5061
> ;fs_nat=yes;fs_path=sip%3A1015%4080.93.247.26%3A1100
> freeswitch at default> group_call sales
> [presence_id=1000 at example.com]sofia/internal/sip:1000 at 192.168.1.31<sip%3A1000 at 192.168.1.31>
> ,[presence_id=1004 at example.com
> ]sofia/internal/sip:gw+15 at 12.237.205.134:6060;transport=udp
> freeswitch at default> group_call billing
> [presence_id=1006 at example.com]sofia/internal/sip:gw+11 at 192.168.1.112:6060
> ;transport=udp
> ,[presence_id=1007 at example.com]sofia/internal/sip:1007 at 192.168.1.22<sip%3A1007 at 192.168.1.22>
> freeswitch at default>
>
>
> Thanks,
> /b
>
> On Aug 29, 2009, at 4:55 PM, Shawn L. Djernes wrote:
>
> > This seems to be a problem in the 1.0.4 tarball release.  A friend
> > with SVN 14305 the group_call from fs_cli works.
> >
> > Shawn
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shawn L. Djernes
SD Consulting
shawn at djernes.org | sdjernes at gmail.com
MSN: wizardwlf at hotmail.com
402.345.7734 | 402.350.6973 Cell
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From steveu at coppice.org  Sat Aug 29 20:17:01 2009
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 30 Aug 2009 11:17:01 +0800
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <20090829145229.2ad02225396a31c9de30536f2e338977.c53fcdc891.wbe@email04.secureserver.net>
References: <20090829145229.2ad02225396a31c9de30536f2e338977.c53fcdc891.wbe@email04.secureserver.net>
Message-ID: <4A99EF2D.3010704@coppice.org>

This sounds like so many "redundancy" projects that will probably offer 
nothing in the real world.

On 08/30/2009 05:52 AM, Pete Mueller wrote:
> I guess I should also mention that Xen is a side-project.
>
> When considering this issue for an existing production systems, we 
> chose to put as much HA into hardware as we can.  We are not concerned 
> with FS crashing, as so far we've never seen that happen (except when 
> our module caused it :)  So for each of our systems:
> - We have dual NIC cards (onboad NIC + PCI card) both bridged together 
> in case one fails
NICs hardly ever fail. Its the wiring which is the vulnerable area. How 
independent can you make the two wiring paths, when they come from the 
same box?
> - We have redundant power supplies.
Even with a good UPS, power fails more often than a high quality power 
supply. Just how independent are the two power sources feeding your two 
power supplies? Do you have two completely independent UPS sets? Do you 
have spacially diverse wiring from them?
> - We use Mirrored Solid State Disks for local storage (far better MTBF 
> than HDD, a lot faster too)
My experience so far is that SSD reliability is very poor, with entire 
drives disappearing, rather than just getting the odd bad sector. I 
guess to balance this, hard disk drive reliability seems to have 
plummeted in the last year or so, after several good years.
> - All but OS and speed-critical data is stored on a NAS device
NAS == more wiring. More wiring == more vulnerabilities. Are you sure 
your setup is a win? NAS tends to help keep the data secure, but it 
isn't good for reliable access to that data.
> - We have redundant DBs with Memcache infront for speed
>
> At the same time we chose to use COTS hardware (SuperMicro 
> chassis/MoBo) rather than the big-boys like IBM or Dell.  This kept 
> the overall cost per machine low.  Initially some were concerned that 
> not having a name like IBM on our servers would be concerning to some 
> potential clients.  The solution was to pay a company to deisgn and 
> build a custom face plate for the SuperMicro boxes.  Which oddly looks 
> more impressive to clients that a rack full of IBM faceplates.  It was 
> suprisingly low cost for the faceplates too.
Some years ago we made an entire custom chassis for off the shelf 
boards. The quotes for fabricating that in small numbers were all over 
the place, but we ended with a good quality chassis at low cost. Most 
off the shelf rack mount enclosures are really pricy, so it isn't that 
hard to match their price with a custom build. We ended up with  a 
better design (at least for our purposes) that cost us no more. It can 
really make your stuff stand out.

A simple respray of the front panel can achieve a distinctive look at 
low cost too. :-)
>
> For scalability, OpenSIPS was our choice.  There's a very nice 
> tutorial on their website on how to configure Load Balancing.

Regards,
Steve



From marc at kasteris.com  Sat Aug 29 20:24:46 2009
From: marc at kasteris.com (Marc Orenberg)
Date: Sat, 29 Aug 2009 20:24:46 -0700 (PDT)
Subject: [Freeswitch-users] prompt playback volume using mod_shout
Message-ID: <158584.63541.qm@web50801.mail.re2.yahoo.com>

Hi, 

I'm finding that when I record a caller's voice using mod_shout in MP3 format, the recording is too soft when I play it back.
I see there's a "volume" setting in shout.conf.xml, but it says, "Dont change these unless you are insane".
When I insanely try to increase this value, I don't here any difference in volume.
Is this the right knob, or is there some other way to do this?

Thanks,
Marc
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From mrene_lists at avgs.ca  Sat Aug 29 20:54:29 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sat, 29 Aug 2009 23:54:29 -0400
Subject: [Freeswitch-users] What is the difference between preAnswer and
	ring_ready?
In-Reply-To: <20090830015030.GA3163@jdc.jasonjgw.net>
References: <c55d46950908291830w6b83a6a7h17a6faf67732d58a@mail.gmail.com>
	<20090830015030.GA3163@jdc.jasonjgw.net>
Message-ID: <6B5963B0-C53A-4005-A14F-DD3140F29560@avgs.ca>

Hi,

In SIP, ring_ready will send a 180 ringing, pre answer will send a 183  
with a session description, so that some audio can be exchanged while  
the call isnt answered yet (used a lot for inband error messages etc).

Regards,

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 29-Aug-09, at 9:50 PM, Jason White wrote:

> Max Bridgewater <max.bridgewater at gmail.com> wrote:
>> Hi,
>> Assuming an inbound call, I have trouble understanding what the  
>> supposed
>> difference between the following two set of instructions is:
>>
>> session.execute("ring_ready");
>> session.set("set", "ringback=/home/ring.wav");
>
> Ring_ready sends a SIP "ringing" message and will use your device's  
> default
> ringback sound - at least, that has been my experience. I should  
> remember the
> message number in the SIP protocol, but I don't (the SIP experts  
> here will
> undoubtedly have the details at the ready).
>>
>> and
>>
>> session.preAnswer();
>> session.streamFile("/home/ring.wav")
>
> I think that will play the file as early media.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From dave at 3c.co.uk  Sat Aug 29 21:07:43 2009
From: dave at 3c.co.uk (David Knell)
Date: Sun, 30 Aug 2009 07:07:43 +0300
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <4A99EF2D.3010704@coppice.org>
References: <20090829145229.2ad02225396a31c9de30536f2e338977.c53fcdc891.wbe@email04.secureserver.net>
	<4A99EF2D.3010704@coppice.org>
Message-ID: <1251605263.5146.40.camel@dk-d820>

On Sun, 2009-08-30 at 11:17 +0800, Steve Underwood wrote:
> This sounds like so many "redundancy" projects that will probably offer 
> nothing in the real world.
> 
> On 08/30/2009 05:52 AM, Pete Mueller wrote:
> > I guess I should also mention that Xen is a side-project.
> >
> > When considering this issue for an existing production systems, we 
> > chose to put as much HA into hardware as we can.  We are not concerned 
> > with FS crashing, as so far we've never seen that happen (except when 
> > our module caused it :)  So for each of our systems:
> > - We have dual NIC cards (onboad NIC + PCI card) both bridged together 
> > in case one fails
> NICs hardly ever fail. Its the wiring which is the vulnerable area. How 
> independent can you make the two wiring paths, when they come from the 
> same box?

This is one area where you can do quite well.  A simple setup:
two machines (1, 2), two NICs (A, B) in each, two switches (S1, S2)
- wire up 1A <-> S1 <-> 2A, 1B <-> S2 <-> 2B
- run OSPF across the links
allows you to unplug any cable or any switch without interrupting
communications for more than a second or two if the OSPF timers are
suitably set.

This generalises nicely - we used to run two machines as web servers,
each advertising the same IP address via OSPF to the routers via a setup
like the one above.  Unplug any one thing, and the whole still worked.

Three complete power outages in the data center we were in in 18 months,
one of which took out a number of power supplies, neatly illustrated
Steve's point: our real-world reliability was determined elsewhere.

--Dave

-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From pete at privateconnect.com  Sat Aug 29 22:06:20 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Sat, 29 Aug 2009 22:06:20 -0700
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Message-ID: <20090829220620.2ad02225396a31c9de30536f2e338977.7a28828e7b.wbe@email04.secureserver.net>

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From m.krivushin at imarto.net  Sat Aug 29 22:06:41 2009
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Sun, 30 Aug 2009 12:06:41 +0700
Subject: [Freeswitch-users] FS: possible to create conference + password
	on the fly?
In-Reply-To: <6f7c60c40908291955i6417d51emfbade3aefc9cdf52@mail.gmail.com>
References: <6f7c60c40908291955i6417d51emfbade3aefc9cdf52@mail.gmail.com>
Message-ID: <5be734a50908292206m3a6cafa0l6c02b17bba3aff01@mail.gmail.com>

I think you need to read documentation around a week, try to do something
with FS. And then most of our questions go out.

Yes, you can do that - dynamic conference with pass.

2009/8/30 tom <tomabroad at gmail.com>

> at best, from an external application, language is not important. as far i
> have seen, could liverpie (aka the mod_event) do that? thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jason at jasonjgw.net  Sat Aug 29 22:34:45 2009
From: jason at jasonjgw.net (Jason White)
Date: Sun, 30 Aug 2009 15:34:45 +1000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <20090829220620.2ad02225396a31c9de30536f2e338977.7a28828e7b.wbe@email04.secureserver.net>
References: <20090829220620.2ad02225396a31c9de30536f2e338977.7a28828e7b.wbe@email04.secureserver.net>
Message-ID: <20090830053445.GA15076@jdc.jasonjgw.net>

Pete Mueller <pete at privateconnect.com> wrote:
>    There is a need for ensuring that calls do not drop, but we must balance
>    that with the cost of making the system redundant.? We took some small,
>    inexpensive measures, to improve our odds, but we could spend a lot more,
>    for basically nothing more than giving some client a warm fuzzy.

I think this is one area where, as indicated earlier in the thread, a lot of
development effort would be needed to obtain that extra degree of reliability.

>From a broader perspective, the question is whether, over the next decade or
two, VoIP can compete with the PSTN in reliability. My (limited) understanding
is that PSTN equipment typically achieves 99.99999% uptime, and if VoIP
systems are going to play in that arena, it would be desirable for
free/open-source software to do so.

If FreeSWITCH itself is working correctly, all you need is a hardware failure
or a kernel panic or a network outage to drop that up-time substantially, not
to mention dropping the calls as well, which I've never experienced as a user
of the PSTN due to equipment at the telephone exchange.

I have, however, experienced some rather low-quality PSTN calls over
international lines, which have the added disadvantage of being expensive to
use.



From jan.kubr at gmail.com  Sat Aug 29 23:28:43 2009
From: jan.kubr at gmail.com (Jan Kubr)
Date: Sun, 30 Aug 2009 08:28:43 +0200
Subject: [Freeswitch-users] liverpie as gui-proxy?
In-Reply-To: <6f7c60c40908291937h57c632d5l84b6ec5ac1286c48@mail.gmail.com>
References: <6f7c60c40908291937h57c632d5l84b6ec5ac1286c48@mail.gmail.com>
Message-ID: <698401620908292328w22ce398do3936d09beb1a1fb2@mail.gmail.com>

You can use this setup to control calls, but I believe not to change
configuration. Dunno what you're after.

On Sunday, August 30, 2009, tom <tomabroad at gmail.com> wrote:
> hi,
>
> i was actually looking for something completely else, but reading wiki? led me to the tought:
> if mod_event can basically do all kinds of FS-things, why not having a standalone ruby-on-rails as as gui which interacts with the evnet-socket via liverpie?
>
> im a total newbie to FS, just writing down my thoughts...
>
>
> thx for ur replies
>


From rs at runsolutions.com  Sun Aug 30 06:19:54 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Sun, 30 Aug 2009 15:19:54 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <B55A1F03-DC5E-4F32-99C8-3AB9D11B2C8A@freeswitch.org>
References: <C6BEE13A.5E372%krice@freeswitch.org>
	<A6E1A4F9-F0D5-4645-A72D-26365CF6FC7B@enterux.com>
	<B55A1F03-DC5E-4F32-99C8-3AB9D11B2C8A@freeswitch.org>
Message-ID: <FC48B8FE-9919-40EC-A878-5F06410186F2@runsolutions.com>

So, the first way would be to have a look on the sip stack, which is,  
in fact, sofia ..., well, sound's like a nice, fun, long-going hobby  
project to me :-)

I will definitly at least look at the sip code to check if it is  
something i would myself willingly give over to ...

But I *really* do want  a setup where it does not matter if one  
specific FS instance is UP or NOT ...


-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 29, 2009, at 10:04 PM, Brian West wrote:

> The sip stack needs to be modified to spin that data up into the  
> state machine so that it can take over calls once the fail over  
> takes place... its not an easy task.
>
> /b
>
> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>
>> Is itnpossible to have a db cluster know the state of each and  
>> every call and then use Heartbeat on this db +
>> fs cluster so that clients see only one ip where as internally all  
>> fs boxes refer db for call states, db again is under replication.
>>
>> This in the thioery can be written, but I am sure if we think bit  
>> more on this direction the problem seem to be getting addressed.
>>
>> Other guys also chip in their 2 cents, we just need 50 of em to  
>> make a full dollar.
>>
>> Thanks & Regards,
>> Mitul Limbani,
>> Founder & CEO,
>> Enterux Solutions Pvt. Ltd.,
>> The Enterprise Linux Company (r),
>> http://www.enterux.com
>> http://www.entVoice.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From max.bridgewater at gmail.com  Sun Aug 30 08:25:45 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 30 Aug 2009 11:25:45 -0400
Subject: [Freeswitch-users] Caller Authentication
Message-ID: <c55d46950908300825v6a298150y406931f286891ed@mail.gmail.com>

Guys,

This question might be a bit off-topic, so please bear with me as i could
hardly find a better place to solicit some help.
I suspect caller authentication can be challenging in the broader context of
VoIP/SIP. Now, if we restrict the set of calls to those coming from the GSM
network, is there a way to tell if the Caller Id is coming from the right
source?  It doesn't need to be a bulletproofed solution. For instance, if i
can already say that a given call is certainly coming from a GSM network
(and that the call metadata weren't tampered with), i could maybe rely on
the GSM authentication mechanism.

Any idea?
Max.
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From erwin.davis at gmail.com  Sat Aug 29 15:01:17 2009
From: erwin.davis at gmail.com (Erwin Huang)
Date: Sat, 29 Aug 2009 18:01:17 -0400
Subject: [Freeswitch-users] freeswitch compile error
Message-ID: <ec9a78d50908291501k111ee428la67ab73345bb1a9f@mail.gmail.com>

Hi, I am new to freeswitch. I downloaded 1.04 source tarball and I got error
during the compiling.
Below is the error message, please advise what was going wrong. Thanks,

mkdir .libs
Compiling mod_cepstral.c ...
mod_cepstral.c:41:19: error: swift.h: No such file or directory
mod_cepstral.c:53: error: expected '=', ',', ';', 'asm' or '__attribute__'
before '*' token
mod_cepstral.c:57: error: expected specifier-qualifier-list before
'swift_background_t'
mod_cepstral.c:69: error: expected '=', ',', ';', 'asm' or '__attribute__'
before 'write_audio'
mod_cepstral.c: In function 'cepstral_speech_open':
mod_cepstral.c:122: error: 'cepstral_t' has no member named 'audio_buffer'
mod_cepstral.c:128: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c:131: error: 'cepstral_t' has no member named 'params'
mod_cepstral.c:132: error: 'cepstral_t' has no member named 'params'
mod_cepstral.c:134: error: 'cepstral_t' has no member named 'params'
mod_cepstral.c:138: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:138: error: 'engine' undeclared (first use in this function)
mod_cepstral.c:138: error: (Each undeclared identifier is reported only once
mod_cepstral.c:138: error: for each function it appears in.)
mod_cepstral.c:138: error: 'cepstral_t' has no member named 'params'
mod_cepstral.c:144: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:151: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:151: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:157: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:157: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:162: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:169: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:169: error: 'write_audio' undeclared (first use in this
function)
mod_cepstral.c:169: error: 'SWIFT_EVENT_AUDIO' undeclared (first use in this
function)
mod_cepstral.c: In function 'cepstral_speech_close':
mod_cepstral.c:187: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:188: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:189: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:189: error: 'SWIFT_ASYNC_ANY' undeclared (first use in this
function)
mod_cepstral.c:189: error: 'SWIFT_EVENT_NOW' undeclared (first use in this
function)
mod_cepstral.c:191: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:192: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:195: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:198: error: 'cepstral_t' has no member named 'audio_buffer'
mod_cepstral.c: In function 'cepstral_speech_feed_tts':
mod_cepstral.c:213: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:214: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:216: error: 'cepstral_t' has no member named 'tts_stream'
mod_cepstral.c:226: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:226: error: 'cepstral_t' has no member named 'tts_stream'
mod_cepstral.c:234: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:234: error: 'cepstral_t' has no member named 'tts_stream'
mod_cepstral.c: In function 'cepstral_speech_flush_tts':
mod_cepstral.c:249: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:250: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:251: error: 'cepstral_t' has no member named 'audio_buffer'
mod_cepstral.c:252: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c:253: error: 'cepstral_t' has no member named 'audio_buffer'
mod_cepstral.c:254: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c:256: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:256: error: 'SWIFT_ASYNC_ANY' undeclared (first use in this
function)
mod_cepstral.c:256: error: 'SWIFT_EVENT_NOW' undeclared (first use in this
function)
mod_cepstral.c: In function 'cepstral_speech_read_tts':
mod_cepstral.c:270: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:271: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:272: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:272: error: 'cepstral_t' has no member named 'tts_stream'
mod_cepstral.c:274: error: 'SWIFT_STATUS_RUNNING' undeclared (first use in
this function)
mod_cepstral.c:275: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:279: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c:280: error: 'cepstral_t' has no member named 'audio_buffer'
mod_cepstral.c:281: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c:284: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:292: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:306: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c:307: error: 'cepstral_t' has no member named 'audio_buffer'
mod_cepstral.c:318: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c: In function 'cepstral_text_param_tts':
mod_cepstral.c:337: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:337: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:338: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:338: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:339: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:339: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:342: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:347: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:355: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:355: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:357: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:357: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:362: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:362: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:364: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:364: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:368: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:378: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c: In function 'cepstral_numeric_param_tts':
mod_cepstral.c:388: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c: In function 'cepstral_float_param_tts':
mod_cepstral.c:400: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c: In function 'mod_cepstral_load':
mod_cepstral.c:409: error: 'engine' undeclared (first use in this function)
make[5]: *** [mod_cepstral.lo] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_cepstral-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
[root at proxy1 freeswitch]#
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From nandy1925 at gmail.com  Sat Aug 29 18:53:22 2009
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sun, 30 Aug 2009 09:53:22 +0800
Subject: [Freeswitch-users] FS - external users
In-Reply-To: <6f7c60c40908291749sbb3e63do3011f8e251e2332a@mail.gmail.com>
References: <6f7c60c40908291749sbb3e63do3011f8e251e2332a@mail.gmail.com>
Message-ID: <7d0bfd8c0908291853q564bcf87tb566046988deb0d2@mail.gmail.com>

no need to modify anything on the directory entries - unless you add more
extension numbers. FS has a uPNP feature already. if your router has uPNP,
the external phone only needs the ff:
1. public IP of the FS (or hostname)
2. username
3. password

Items 2 & 3 are found in the directory entries.


On Sun, Aug 30, 2009 at 8:49 AM, tom <tomabroad at gmail.com> wrote:

> hi,
>
> as newbie here a simple one:
>
> based on the std-installation, what do i need to do to have a user
> connected from the outside. lets assume i want to leave 1000-1020 as they
> are, i make a copy of 1020, call it 1021 and then?
> should i leave it where it is? do i have to move the file? and what about
> he conetxt?
>
> thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sun Aug 30 09:13:14 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 11:13:14 -0500
Subject: [Freeswitch-users] freeswitch compile error
In-Reply-To: <ec9a78d50908291501k111ee428la67ab73345bb1a9f@mail.gmail.com>
References: <ec9a78d50908291501k111ee428la67ab73345bb1a9f@mail.gmail.com>
Message-ID: <61A2778C-A1CA-4FD9-A2E5-6766C76CBCCF@freeswitch.org>

Unless you have installed the Cepstral SDK you can't compile  
mod_cepstral.  Please visit www.cepstral.com to purchase voices and  
the SDK is included.

/b

On Aug 29, 2009, at 5:01 PM, Erwin Huang wrote:

> mkdir .libs
> Compiling mod_cepstral.c ...
> mod_cepstral.c:41:19: error: swift.h: No such file or directory



From dave at 3c.co.uk  Sun Aug 30 09:25:17 2009
From: dave at 3c.co.uk (David Knell)
Date: Sun, 30 Aug 2009 19:25:17 +0300
Subject: [Freeswitch-users] Caller Authentication
In-Reply-To: <c55d46950908300825v6a298150y406931f286891ed@mail.gmail.com>
References: <c55d46950908300825v6a298150y406931f286891ed@mail.gmail.com>
Message-ID: <1251649517.6737.10.camel@dk-d820>

Hi Max,

I think that the answer is generally no - for example, I'm trivially
easily able to place a call to my GSM phone with any CLI that I choose.
If you can work around that, then consider the case where I divert my
GSM phone to you - the call to you will come from the GSM network, but I
can still set an arbitrary CLI.

Cheers --

Dave

> Guys,
> 
> This question might be a bit off-topic, so please bear with me as i
> could hardly find a better place to solicit some help.
> I suspect caller authentication can be challenging in the broader
> context of VoIP/SIP. Now, if we restrict the set of calls to those
> coming from the GSM network, is there a way to tell if the Caller Id
> is coming from the right source?  It doesn't need to be a
> bulletproofed solution. For instance, if i can already say that a
> given call is certainly coming from a GSM network (and that the call
> metadata weren't tampered with), i could maybe rely on the GSM
> authentication mechanism. 
> 
> Any idea?
> Max.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From tomabroad at gmail.com  Sun Aug 30 09:25:47 2009
From: tomabroad at gmail.com (tom)
Date: Sun, 30 Aug 2009 12:25:47 -0400
Subject: [Freeswitch-users] FS - external users
In-Reply-To: <7d0bfd8c0908291853q564bcf87tb566046988deb0d2@mail.gmail.com>
References: <6f7c60c40908291749sbb3e63do3011f8e251e2332a@mail.gmail.com>
	<7d0bfd8c0908291853q564bcf87tb566046988deb0d2@mail.gmail.com>
Message-ID: <6f7c60c40908300925p5c4e62abk898c3025eea3725a@mail.gmail.com>

thx, all working now!

On Sat, Aug 29, 2009 at 9:53 PM, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> no need to modify anything on the directory entries - unless you add more
> extension numbers. FS has a uPNP feature already. if your router has uPNP,
> the external phone only needs the ff:
> 1. public IP of the FS (or hostname)
> 2. username
> 3. password
>
> Items 2 & 3 are found in the directory entries.
>
>
> On Sun, Aug 30, 2009 at 8:49 AM, tom <tomabroad at gmail.com> wrote:
>
>> hi,
>>
>> as newbie here a simple one:
>>
>> based on the std-installation, what do i need to do to have a user
>> connected from the outside. lets assume i want to leave 1000-1020 as they
>> are, i make a copy of 1020, call it 1021 and then?
>> should i leave it where it is? do i have to move the file? and what about
>> he conetxt?
>>
>> thx
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From aep.lists at it46.se  Sun Aug 30 09:59:17 2009
From: aep.lists at it46.se (Alberto Escudero)
Date: Sun, 30 Aug 2009 18:59:17 +0200
Subject: [Freeswitch-users] freeswitch compile error
In-Reply-To: <61A2778C-A1CA-4FD9-A2E5-6766C76CBCCF@freeswitch.org>
References: <ec9a78d50908291501k111ee428la67ab73345bb1a9f@mail.gmail.com>
	<61A2778C-A1CA-4FD9-A2E5-6766C76CBCCF@freeswitch.org>
Message-ID: <1c14317d05b7457e797b3e94b0db02dc.squirrel@correo.nodo50.org>

You can always download it and test it for a while!. It will insert some
nice commercials when running TTS. I ended up buying a license, Allison
convinced me :) And if you plan to use TTS via PSTN or GSM, purchase a 8
Khz voice.

/aep
-- 
Stopping junk mailers is good for the environment

> Unless you have installed the Cepstral SDK you can't compile
> mod_cepstral.  Please visit www.cepstral.com to purchase voices and
> the SDK is included.
>
> /b
>
> On Aug 29, 2009, at 5:01 PM, Erwin Huang wrote:
>
>> mkdir .libs
>> Compiling mod_cepstral.c ...
>> mod_cepstral.c:41:19: error: swift.h: No such file or directory
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>




From max.bridgewater at gmail.com  Sun Aug 30 10:47:58 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 30 Aug 2009 13:47:58 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
Message-ID: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>

Hi,

I was trying to build freeswitch with cepstral enabled. Here is the procdure
i am following:
http://wiki.freeswitch.org/wiki/Mod_cepstral#Can_I_use_a_16khz_.22desktop_voice.22.3F

Unfortunately, the build fails as show in the log message hereunder.  In a
previous thread, Brian says that the SDK is needed. Yet on the above wiki
page, it is said that the SDK is not needed.
Any suggestion?

Max.

Compiling mod_cepstral.c ...
Creating mod_cepstral.so...
/usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when searching
for -lswift
/usr/bin/ld: cannot find -lswift
collect2: ld returned 1 exit status
gcc -w -I/home/installs/freeswitch/src/include
-I/home/installs/freeswitch/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -I/include
-I/opt/swift/include -shared -o .libs/mod_cepstral.so -shared -Wl,-x
.libs/mod_cepstral.o  -L/lib -L/opt/swift/lib -lswift -lm
/home/installs/freeswitch/.libs/libfreeswitch.so
-L/home/installs/freeswitch/libs/apr-util/xml/expat/lib
/home/installs/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/home/installs/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt
-lpthread -L/home/installs/freeswitch/libs/srtp -L/usr/kerberos/lib64 -lssl
-lcrypto -ldl -lz -lncurses   -Wl,--rpath -Wl,/usr/local/freeswitch/lib
-Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_cepstral.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_cepstral-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
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From brian at freeswitch.org  Sun Aug 30 10:58:16 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 12:58:16 -0500
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
Message-ID: <77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>

You still have to have the voices installed... the SDK comes with each  
voice for Linux ... for windows you have to have the SDK...

/b

On Aug 30, 2009, at 12:47 PM, Max Bridgewater wrote:

> Hi,
>
> I was trying to build freeswitch with cepstral enabled. Here is the  
> procdure i am following:
> http://wiki.freeswitch.org/wiki/Mod_cepstral#Can_I_use_a_16khz_.22desktop_voice.22.3F
>
> Unfortunately, the build fails as show in the log message  
> hereunder.  In a previous thread, Brian says that the SDK is needed.  
> Yet on the above wiki page, it is said that the SDK is not needed.
> Any suggestion?
>
> Max.

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From max.bridgewater at gmail.com  Sun Aug 30 11:09:31 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 30 Aug 2009 14:09:31 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
	<77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
Message-ID: <c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>

OK, it's bad and good news. It's good news because it's reassuring that i
don't need a separate SDK. Its bad news because i already installed a voice
(Jean-Pierre) following the procedure describe on the wiki. I will now
delete it and install the Allison voice to see if there is any difference.
Will let ya know.

On Sun, Aug 30, 2009 at 1:58 PM, Brian West <brian at freeswitch.org> wrote:

> You still have to have the voices installed... the SDK comes with each
> voice for Linux ... for windows you have to have the SDK...
> /b
>
> On Aug 30, 2009, at 12:47 PM, Max Bridgewater wrote:
>
> Hi,
>
> I was trying to build freeswitch with cepstral enabled. Here is the
> procdure i am following:
>
> http://wiki.freeswitch.org/wiki/Mod_cepstral#Can_I_use_a_16khz_.22desktop_voice.22.3F
>
> Unfortunately, the build fails as show in the log message hereunder.  In a
> previous thread, Brian says that the SDK is needed. Yet on the above wiki
> page, it is said that the SDK is not needed.
> Any suggestion?
>
> Max.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From max.bridgewater at gmail.com  Sun Aug 30 11:33:24 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 30 Aug 2009 14:33:24 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
	<77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
	<c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
Message-ID: <c55d46950908301133r2888ebffq8bfb93bbaf9e24a4@mail.gmail.com>

Even more bad news. I downloaded the Allison voice and have the same error
message. Any idea what this means?


mkdir .libs
Compiling mod_cepstral.c ...
Creating mod_cepstral.so...
/usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when searching
for -lswift
/usr/bin/ld: cannot find -lswift
collect2: ld returned 1 exit status
gcc -w -I/home/installs/freeswitch/src/include
-I/home/installs/freeswitch/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -I/opt/swift/include
-I/opt/swift/include -shared -o .libs/mod_cepstral.so -shared -Wl,-x
.libs/mod_cepstral.o  -L/opt/swift/lib -lswift -lm
/home/installs/freeswitch/.libs/libfreeswitch.so
-L/home/installs/freeswitch/libs/apr-util/xml/expat/lib
/home/installs/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/home/installs/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt
-lpthread -L/home/installs/freeswitch/libs/srtp -L/usr/kerberos/lib64 -lssl
-lcrypto -ldl -lz -lncurses   -Wl,--rpath -Wl,/usr/local/freeswitch/lib
-Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_cepstral.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_cepstral-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
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From brian at freeswitch.org  Sun Aug 30 11:36:38 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 13:36:38 -0500
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
	<77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
	<c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
Message-ID: <2ABAA784-4B60-4AB0-932C-616BE3102904@freeswitch.org>

You also failed to read this:

http://wiki.freeswitch.org/wiki/Mod_cepstral#libswift.so. 
4:_cannot_open_shared_object_file:_No_such_file_or_directory

You have to add the directory to your ld.so.conf so it can find -lswift.

/b

On Aug 30, 2009, at 1:09 PM, Max Bridgewater wrote:

> OK, it's bad and good news. It's good news because it's reassuring  
> that i don't need a separate SDK. Its bad news because i already  
> installed a voice (Jean-Pierre) following the procedure describe on  
> the wiki. I will now delete it and install the Allison voice to see  
> if there is any difference. Will let ya know.
>
> On Sun, Aug 30, 2009 at 1:58 PM, Brian West <brian at freeswitch.org>  
> wrote:
> You still have to have the voices installed... the SDK comes with  
> each voice for Linux ... for windows you have to have the SDK...
>
> /b
>
> On Aug 30, 2009, at 12:47 PM, Max Bridgewater wrote:
>
>> Hi,
>>
>> I was trying to build freeswitch with cepstral enabled. Here is the  
>> procdure i am following:
>> http://wiki.freeswitch.org/wiki/Mod_cepstral#Can_I_use_a_16khz_.22desktop_voice.22.3F
>>
>> Unfortunately, the build fails as show in the log message  
>> hereunder.  In a previous thread, Brian says that the SDK is  
>> needed. Yet on the above wiki page, it is said that the SDK is not  
>> needed.
>> Any suggestion?
>>
>> Max.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sun Aug 30 11:46:56 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 13:46:56 -0500
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <c55d46950908301133r2888ebffq8bfb93bbaf9e24a4@mail.gmail.com>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
	<77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
	<c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
	<c55d46950908301133r2888ebffq8bfb93bbaf9e24a4@mail.gmail.com>
Message-ID: <42B40E01-ACBC-4303-8D91-53D2DD1C97A0@freeswitch.org>

Yes please add /opt/swift/lib to /etc/ld.so.conf as per the FAQ.

/b

On Aug 30, 2009, at 1:33 PM, Max Bridgewater wrote:

> /usr/bin/ld: cannot find -lswift



From max.bridgewater at gmail.com  Sun Aug 30 11:59:36 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 30 Aug 2009 14:59:36 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <2ABAA784-4B60-4AB0-932C-616BE3102904@freeswitch.org>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
	<77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
	<c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
	<2ABAA784-4B60-4AB0-932C-616BE3102904@freeswitch.org>
Message-ID: <c55d46950908301159j46d06904k2f2a97ecb9915650@mail.gmail.com>

Please bear with me. Here is the content of my  /etc/ld.so.conf:

include ld.so.conf.d/*.conf
/opt/swift/lib

And when i do "ldconfig -p | grep swift" , i get the following:

   libswift.so.5 (libc6) => /opt/swift/lib/libswift.so.5
   libswift.so (libc6) => /opt/swift/lib/libswift.so
   libceplex_us.so.5 (libc6) => /opt/swift/lib/libceplex_us.so.5
   libceplex_us.so (libc6) => /opt/swift/lib/libceplex_us.so
   libceplang_en.so.5 (libc6) => /opt/swift/lib/libceplang_en.so.5
   libceplang_en.so (libc6) => /opt/swift/lib/libceplang_en.so

So, can it be that Freeswitch is looking for libswift.so.4 instead of
libswift.so.5 (as the first line of the wiki page is actually saying)? If
so, where do i get this 4.x version? I can't find it in on cepstral site.

Max.
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From pbd at suspiria.net  Sun Aug 30 12:00:38 2009
From: pbd at suspiria.net (Public Dump)
Date: Sun, 30 Aug 2009 21:00:38 +0200
Subject: [Freeswitch-users] delay buildup in conference
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
	<191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com> 
Message-ID: <13C421883438EB42B9E2C30069FD4AB77004E8C0F7@crushinator.central.local>

I have tested the suggested setting:

<settings>

      <param name="rtp-autoflush-during-bridge" value="true"/>



But the problem remains. L



Von: Public Dump
Gesendet: Mittwoch, 26. August 2009 23:37
An: 'freeswitch-users at lists.freeswitch.org'
Betreff: AW: [Freeswitch-users] delay buildup in conference



I am quite sure that it's not a problem with the network connection, since the delay builds up gradually over time, while the host is reachable without any problems.



I have added the suggested config lines and will check the results.



By the way, can I remap the hash key, so that is does not end the conference ? I've tried remapping it, but freeswitch seems to ignore the setting (the console shows: hangup mapped to '*' ... but # still ends the call).

It is irritating for users, since most users try to enter the conference pin the #-key at the end.



Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 22:45
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] delay buildup in conference



There is no bug, it's all dependent on your network conditions.
I spend literally 8-12 hours a day on a conference and there is no delay.

The important param is
<param name="rtp-autoflush-during-bridge" value="true"/>

in the sofia profile in question.

if you have delay with that in place, then it's probably not FS

On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd at suspiria.net<mailto:pbd at suspiria.net>> wrote:

Hi,



I am on a quite recent version (i assume):



FreeSWITCH Version 1.0.trunk (14461)



Should the bug be fixed in this revision ? What config settings would a have to check to limit delay (even at the cost of reduction in quality).



Thanks



Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 20:57
An: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] delay buildup in conference



which revision are you on?
The defaults on the latest code and examples should be configured to minimize delay.
Some of the older revisions built up some delay issues from udp buffering when timers were not synced.

On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net<mailto:pbd at suspiria.net>> wrote:

When running conferences with users dialed in from a PSTN gateway (SIP) and directly from remote SIP endpoints there is an ever longer buildup in delay, reaching up to multiple seconds. Is there any way to limit the delay ?



I am not 100% sure whether the delays is caused by the SIP jitter buffer of freeswitch or directly by the conference module.



Any advice?



Thanks


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400

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From brian at freeswitch.org  Sun Aug 30 12:10:40 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 14:10:40 -0500
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <c55d46950908301159j46d06904k2f2a97ecb9915650@mail.gmail.com>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
	<77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
	<c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
	<2ABAA784-4B60-4AB0-932C-616BE3102904@freeswitch.org>
	<c55d46950908301159j46d06904k2f2a97ecb9915650@mail.gmail.com>
Message-ID: <83278230-3C0C-40B1-85A6-1CF2ED19DBF6@freeswitch.org>

make sure you run ldconfig after you add the line then try to compile  
it again.


/b

On Aug 30, 2009, at 1:59 PM, Max Bridgewater wrote:

> Please bear with me. Here is the content of my  /etc/ld.so.conf:
>
> include ld.so.conf.d/*.conf
> /opt/swift/lib



From pete at privateconnect.com  Sun Aug 30 12:33:32 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Sun, 30 Aug 2009 12:33:32 -0700
Subject: [Freeswitch-users]
	=?utf-8?q?Troubles_build_with_mod=5Fcepstral?=
Message-ID: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>

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From mrene_lists at avgs.ca  Sun Aug 30 12:35:41 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 30 Aug 2009 15:35:41 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
References: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
Message-ID: <7CF0A496-7F1A-4CC7-A1F5-88C17BD990DF@avgs.ca>

... skipping incompatible /opt/swift/lib/libswift.so when searching  
for -lswift thats a sign you're trying to link a 64 bits binary with a  
32 bits one.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 30-Aug-09, at 3:33 PM, Pete Mueller wrote:

>  skipping incompatible /opt/swift/lib/libswift.so when searching for  
> -lswift

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From brian at freeswitch.org  Sun Aug 30 13:51:41 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 15:51:41 -0500
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
References: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
Message-ID: <7B458C2D-8DC2-40B9-8EF1-B7FC934A5F0F@freeswitch.org>

http://downloads.cepstral.com/cepstral/x86-64-linux/Cepstral_Allison_x86-64-linux_5.1.0.tar.gz

Make sure you download the 64bit version.

/b



On Aug 30, 2009, at 2:33 PM, Pete Mueller wrote:

>  I don't think Cepstral supports 64-bit, as the SDK is 32-bit.



From mike at jerris.com  Sun Aug 30 14:45:21 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 30 Aug 2009 17:45:21 -0400
Subject: [Freeswitch-users] Transferring to another extension is VERY
	slow.
In-Reply-To: <3d381e170908291121k3bd64ecnacf4f7941962e2ed@mail.gmail.com>
References: <3d381e170908291121k3bd64ecnacf4f7941962e2ed@mail.gmail.com>
Message-ID: <93769c20908301445l37bf8d65j6d00523f75eb3002@mail.gmail.com>

Try reviewing the debug log to see why it is delayed.  You may want to look
at the inline sip trace at all to see if anything is amiss there as well.
Mike


On Sat, Aug 29, 2009 at 2:21 PM, Erik Wickstrom <erik at erikwickstrom.com>wrote:

> Hi,
>
> I'm having some trouble debugging why the transfer command is so slow
> in my case.  I have a lua IVR setup which responds to user selections
> by transferring the call to another dialplan ext.  Pretty basic stuff.
>
> The problem is, it takes 30-40 seconds for the the transfer to
> actually start after the key press.  After about 30 seconds of
> silence, "hold music" finally starts and the other extension finally
> answers shortly thereafter.
>
> I've tried transferring to endpoint extensions, the 5000 sample IVR,
> 9999 hold music -- in all cases there is this long pause.
>
> Does anyone know what could be causing this?
>
> Thanks!
> Erik
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Sun Aug 30 14:49:13 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 30 Aug 2009 17:49:13 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <7B458C2D-8DC2-40B9-8EF1-B7FC934A5F0F@freeswitch.org>
References: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
	<7B458C2D-8DC2-40B9-8EF1-B7FC934A5F0F@freeswitch.org>
Message-ID: <c55d46950908301449u6bb3d43cifd92a353cca6cae5@mail.gmail.com>

Nop, the 64bit  cepstral doesn't work either:

Creating mod_cepstral.so...
/usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when searching
for -lswift
/usr/bin/ld: cannot find -lswift
collect2: ld returned 1 exit status
gcc -w -I/home/installs/freeswitch/src/include
-I/home/installs/freeswitch/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -I/opt/swift/include
-I/opt/swift/include -shared -o .libs/mod_cepstral.so -shared -Wl,-x
.libs/mod_cepstral.o  -L/opt/swift/lib -lswift -lm
/home/installs/freeswitch/.libs/libfreeswitch.so
-L/home/installs/freeswitch/libs/apr-util/xml/expat/lib
/home/installs/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/home/installs/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt
-lpthread -L/home/installs/freeswitch/libs/srtp -L/usr/kerberos/lib64 -lssl
-lcrypto -ldl -lz -lncurses   -Wl,--rpath -Wl,/usr/local/freeswitch/lib
-Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_cepstral.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_cepstral-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
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From brian at freeswitch.org  Sun Aug 30 14:53:11 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 16:53:11 -0500
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <c55d46950908301449u6bb3d43cifd92a353cca6cae5@mail.gmail.com>
References: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
	<7B458C2D-8DC2-40B9-8EF1-B7FC934A5F0F@freeswitch.org>
	<c55d46950908301449u6bb3d43cifd92a353cca6cae5@mail.gmail.com>
Message-ID: <399999D5-EF3E-444F-8E06-05A0006DD053@freeswitch.org>

Make sure you rm -rf /opt/swift then reinstall

/b

On Aug 30, 2009, at 4:49 PM, Max Bridgewater wrote:

> Nop, the 64bit  cepstral doesn't work either:
>
> Creating mod_cepstral.so...
> /usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when  
> searching for -lswift
> /usr/bin/ld: cannot find -lswift
> collect2: ld returned 1 exit status



From Prometheus001 at gmx.net  Sun Aug 30 15:25:01 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 31 Aug 2009 00:25:01 +0200
Subject: [Freeswitch-users] sofia_reg_external in odbc?
Message-ID: <4A9AFC3D.9060208@gmx.net>

Hello,

is there a chance to have sofia_reg_external in odbc/mysql instead of
sqlite?
In a B2BUA environment we have thousand of external registrations during
a migration phase, and it would be good to have easy external control
over the registered gateways (like in fs_internal. sip_registrations).

Best regards
Peter



From brian at freeswitch.org  Sun Aug 30 16:43:50 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 18:43:50 -0500
Subject: [Freeswitch-users] sofia_reg_external in odbc?
In-Reply-To: <4A9AFC3D.9060208@gmx.net>
References: <4A9AFC3D.9060208@gmx.net>
Message-ID: <F939B13C-83EB-44D2-928A-24728F2A6DDF@freeswitch.org>

<param name="odbc-dsn" value="dsn:user:pass"/>

On the profile.

/b


On Aug 30, 2009, at 5:25 PM, Peter P GMX wrote:

> Hello,
>
> is there a chance to have sofia_reg_external in odbc/mysql instead of
> sqlite?
> In a B2BUA environment we have thousand of external registrations  
> during
> a migration phase, and it would be good to have easy external control
> over the registered gateways (like in fs_internal. sip_registrations).
>
> Best regards
> Peter



From jim at evolutiontel.net  Sun Aug 30 19:54:50 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 31 Aug 2009 12:54:50 +1000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <FC48B8FE-9919-40EC-A878-5F06410186F2@runsolutions.com>
References: <C6BEE13A.5E372%krice@freeswitch.org>
	<A6E1A4F9-F0D5-4645-A72D-26365CF6FC7B@enterux.com>
	<B55A1F03-DC5E-4F32-99C8-3AB9D11B2C8A@freeswitch.org>
	<FC48B8FE-9919-40EC-A878-5F06410186F2@runsolutions.com>
Message-ID: <ddad80610908301954i37ecdf96i78b6cb3dc9a2c633@mail.gmail.com>

High Availability is always an interesting problem.  PSTN equipment
generally overcomes it through duplication of messaging and hardware.
Obviously at an E1 (T1 for the US folks) or Subscribers Line Level
this is not possible.  This duplication comes at a cost and lowers the
calls per minute a softswitch can handle for the amount of hardware
purchased.  Like everything it is a trade-off between cost and
reliability.

Per Dave's comment, duplication on a network level can be achieved
easily through duplicating network switches and using OSPF, VRRP or
network bonding to ensure continuity if a path is compromised.  From
experience this is exactly what the vendors do and it does work.
During failover the calls experience a minut breakup of voice quality
but they still hold up ok.

Conceptually in a pure SIP installation of Freeswitch, hot standby or
active standby could be achieved through duplication of messages
(INVITE, 200OK and BYE for calls) between 2 boxes and then using VRRP
to change the IP address when the box falls over.  Going this way
allows call state to be as up to date as possible.  Obviously this
would require logic added to sofia to transfer the RTP port
information and UUID information between the FS instances (Easily
achieved using additional SIP headers).  It would also require that
sofia does not forward duplicated messages to gateways or user agents.
 CDR information would also need to be generated correctly (My
experience of Radius with Opensips is that it generates start and stop
messages on INVITE/200OK then BYE respectively so this could be an
easy solution to the CDR issue).

Regards,



On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
> So, the first way would be to have a look on the sip stack, which is, in
> fact, sofia ..., well, sound's like a nice, fun, long-going hobby project to
> me :-)
> I will definitly at least look at the sip code to check if it is something i
> would myself willingly give over to ...
> But I *really* do want ?a setup where it does not matter if one specific FS
> instance is UP or NOT ...
>
> --
> Raimund Sacherer
> -
> RunSolutions
> ????Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 - ?Palma de Mallorca
> Baleares
> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>
> The sip stack needs to be modified to spin that data up into the state
> machine so that it can take over calls once the fail over takes place... its
> not an easy task.
> /b
> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>
> Is itnpossible to have a db cluster know the state of each and every call
> and then use Heartbeat on this db +
> fs cluster so that clients see only one ip where as internally all fs boxes
> refer db for call states, db again is under replication.
> This in the thioery can be written, but I am sure if we think bit more on
> this direction the problem seem to be getting addressed.
> Other guys also chip in their 2 cents, we just need 50 of em to make a full
> dollar.
>
> Thanks & Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions Pvt. Ltd.,
> The Enterprise Linux Company (r),
> http://www.enterux.com
> http://www.entVoice.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From krice at freeswitch.org  Sun Aug 30 20:23:40 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 30 Aug 2009 22:23:40 -0500
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <ddad80610908301954i37ecdf96i78b6cb3dc9a2c633@mail.gmail.com>
Message-ID: <C6C0AC6C.5E412%krice@freeswitch.org>

Your forgetting that with RTP theres the duplication of RTP that may be
required along with a mechanism that keeps media timers working properly...

Where many things don't support RTP timers they sure come in handy on a
regular basis especially dealing with several of the other software based
platforms out there that just love to have calls end but never want to send
a BYE


> From: Jim Burke <jim at evolutiontel.net>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 31 Aug 2009 12:54:50 +1000
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
> 
> Conceptually in a pure SIP installation of Freeswitch, hot standby or
> active standby could be achieved through duplication of messages
> (INVITE, 200OK and BYE for calls) between 2 boxes and then using VRRP
> to change the IP address when the box falls over.  Going this way
> allows call state to be as up to date as possible.  Obviously this
> would require logic added to sofia to transfer the RTP port
> information and UUID information between the FS instances (Easily
> achieved using additional SIP headers).  It would also require that
> sofia does not forward duplicated messages to gateways or user agents.
>  CDR information would also need to be generated correctly (My
> experience of Radius with Opensips is that it generates start and stop
> messages on INVITE/200OK then BYE respectively so this could be an
> easy solution to the CDR issue).
> 
> Regards,
> 
> 
> 
> On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
>> So, the first way would be to have a look on the sip stack, which is, in
>> fact, sofia ..., well, sound's like a nice, fun, long-going hobby project to
>> me :-)
>> I will definitly at least look at the sip code to check if it is something i
>> would myself willingly give over to ...
>> But I *really* do want ?a setup where it does not matter if one specific FS
>> instance is UP or NOT ...
>> 
>> --
>> Raimund Sacherer
>> -
>> RunSolutions
>> ????Open Source It Consulting
>> -
>> 
>> Parc Bit - Centro Empresarial Son Espanyol
>> Edificio Estel - Local 3D
>> 07121 - ?Palma de Mallorca
>> Baleares
>> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>> 
>> The sip stack needs to be modified to spin that data up into the state
>> machine so that it can take over calls once the fail over takes place... its
>> not an easy task.
>> /b
>> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>> 
>> Is itnpossible to have a db cluster know the state of each and every call
>> and then use Heartbeat on this db +
>> fs cluster so that clients see only one ip where as internally all fs boxes
>> refer db for call states, db again is under replication.
>> This in the thioery can be written, but I am sure if we think bit more on
>> this direction the problem seem to be getting addressed.
>> Other guys also chip in their 2 cents, we just need 50 of em to make a full
>> dollar.
>> 
>> Thanks & Regards,
>> Mitul Limbani,
>> Founder & CEO,
>> Enterux Solutions Pvt. Ltd.,
>> The Enterprise Linux Company (r),
>> http://www.enterux.com
>> http://www.entVoice.com
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> 
> -- 
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jim at evolutiontel.net  Sun Aug 30 21:32:22 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 31 Aug 2009 14:32:22 +1000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <C6C0AC6C.5E412%krice@freeswitch.org>
References: <ddad80610908301954i37ecdf96i78b6cb3dc9a2c633@mail.gmail.com>
	<C6C0AC6C.5E412%krice@freeswitch.org>
Message-ID: <ddad80610908302132i4804be0eg3ea7a7ae2c8a1bfa@mail.gmail.com>

Hi Ken,

I don't disagree, the scenario you mentioned is just the tip of the
iceberg when it comes to getting something like this working smoothly.
 However to try and cover them all in a single post (assuming that I
could) would have resulted in something akin to War and Peace and
probably bored the pants off of most readers :)  For any HA solution
to move from a concept into reality, it would require some serious
brainstorming to identify all scenarios and how to handle them.

Given that many posts in this thread mentioned transfer of call state
at time of failure or some form of virtualisation, I wanted to provide
another solution that would provide real time call state updates
between FS boxes utilising mechanisms that are largely already
in-place and thus (hopefully) reducing software development time if
someone was willing to cough up the $$$$$.

Regards,

On Mon, Aug 31, 2009 at 1:23 PM, Ken Rice<krice at freeswitch.org> wrote:
> Your forgetting that with RTP theres the duplication of RTP that may be
> required along with a mechanism that keeps media timers working properly...
>
> Where many things don't support RTP timers they sure come in handy on a
> regular basis especially dealing with several of the other software based
> platforms out there that just love to have calls end but never want to send
> a BYE
>
>
>> From: Jim Burke <jim at evolutiontel.net>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Mon, 31 Aug 2009 12:54:50 +1000
>> To: <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
>>
>> Conceptually in a pure SIP installation of Freeswitch, hot standby or
>> active standby could be achieved through duplication of messages
>> (INVITE, 200OK and BYE for calls) between 2 boxes and then using VRRP
>> to change the IP address when the box falls over. ?Going this way
>> allows call state to be as up to date as possible. ?Obviously this
>> would require logic added to sofia to transfer the RTP port
>> information and UUID information between the FS instances (Easily
>> achieved using additional SIP headers). ?It would also require that
>> sofia does not forward duplicated messages to gateways or user agents.
>> ?CDR information would also need to be generated correctly (My
>> experience of Radius with Opensips is that it generates start and stop
>> messages on INVITE/200OK then BYE respectively so this could be an
>> easy solution to the CDR issue).
>>
>> Regards,
>>
>>
>>
>> On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
>>> So, the first way would be to have a look on the sip stack, which is, in
>>> fact, sofia ..., well, sound's like a nice, fun, long-going hobby project to
>>> me :-)
>>> I will definitly at least look at the sip code to check if it is something i
>>> would myself willingly give over to ...
>>> But I *really* do want ?a setup where it does not matter if one specific FS
>>> instance is UP or NOT ...
>>>
>>> --
>>> Raimund Sacherer
>>> -
>>> RunSolutions
>>> ????Open Source It Consulting
>>> -
>>>
>>> Parc Bit - Centro Empresarial Son Espanyol
>>> Edificio Estel - Local 3D
>>> 07121 - ?Palma de Mallorca
>>> Baleares
>>> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>>>
>>> The sip stack needs to be modified to spin that data up into the state
>>> machine so that it can take over calls once the fail over takes place... its
>>> not an easy task.
>>> /b
>>> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>>>
>>> Is itnpossible to have a db cluster know the state of each and every call
>>> and then use Heartbeat on this db +
>>> fs cluster so that clients see only one ip where as internally all fs boxes
>>> refer db for call states, db again is under replication.
>>> This in the thioery can be written, but I am sure if we think bit more on
>>> this direction the problem seem to be getting addressed.
>>> Other guys also chip in their 2 cents, we just need 50 of em to make a full
>>> dollar.
>>>
>>> Thanks & Regards,
>>> Mitul Limbani,
>>> Founder & CEO,
>>> Enterux Solutions Pvt. Ltd.,
>>> The Enterprise Linux Company (r),
>>> http://www.enterux.com
>>> http://www.entVoice.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Jim Burke
>> Director Evolutiontel.
>> http://www.evolutiontel.net
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From krice at freeswitch.org  Sun Aug 30 22:37:00 2009
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 31 Aug 2009 00:37:00 -0500
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <ddad80610908302132i4804be0eg3ea7a7ae2c8a1bfa@mail.gmail.com>
Message-ID: <C6C0CBAC.5E420%krice@freeswitch.org>

I was thinking more like something the size of war and peace but written
with the style of an RFC (and about as interesting and an RFC)

I know there has already been some discussion on several fronts of atleast
getting the core and several other pieces to where they need to be for
stateful failover and I'm not sure if its been mentioned here, but sofia is
going to be a bit of work and estimates run in the 100K USD range. Now if we
could get a get a couple of corporate sponsors to help here it would be
great.

K


> From: Jim Burke <jim at evolutiontel.net>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 31 Aug 2009 14:32:22 +1000
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
> 
> Hi Ken,
> 
> I don't disagree, the scenario you mentioned is just the tip of the
> iceberg when it comes to getting something like this working smoothly.
>  However to try and cover them all in a single post (assuming that I
> could) would have resulted in something akin to War and Peace and
> probably bored the pants off of most readers :)  For any HA solution
> to move from a concept into reality, it would require some serious
> brainstorming to identify all scenarios and how to handle them.
> 
> Given that many posts in this thread mentioned transfer of call state
> at time of failure or some form of virtualisation, I wanted to provide
> another solution that would provide real time call state updates
> between FS boxes utilising mechanisms that are largely already
> in-place and thus (hopefully) reducing software development time if
> someone was willing to cough up the $$$$$.
> 
> Regards,
> 
> On Mon, Aug 31, 2009 at 1:23 PM, Ken Rice<krice at freeswitch.org> wrote:
>> Your forgetting that with RTP theres the duplication of RTP that may be
>> required along with a mechanism that keeps media timers working properly...
>> 
>> Where many things don't support RTP timers they sure come in handy on a
>> regular basis especially dealing with several of the other software based
>> platforms out there that just love to have calls end but never want to send
>> a BYE
>> 
>> 
>>> From: Jim Burke <jim at evolutiontel.net>
>>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>>> Date: Mon, 31 Aug 2009 12:54:50 +1000
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
>>> 
>>> Conceptually in a pure SIP installation of Freeswitch, hot standby or
>>> active standby could be achieved through duplication of messages
>>> (INVITE, 200OK and BYE for calls) between 2 boxes and then using VRRP
>>> to change the IP address when the box falls over. ?Going this way
>>> allows call state to be as up to date as possible. ?Obviously this
>>> would require logic added to sofia to transfer the RTP port
>>> information and UUID information between the FS instances (Easily
>>> achieved using additional SIP headers). ?It would also require that
>>> sofia does not forward duplicated messages to gateways or user agents.
>>> ?CDR information would also need to be generated correctly (My
>>> experience of Radius with Opensips is that it generates start and stop
>>> messages on INVITE/200OK then BYE respectively so this could be an
>>> easy solution to the CDR issue).
>>> 
>>> Regards,
>>> 
>>> 
>>> 
>>> On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com>
>>> wrote:
>>>> So, the first way would be to have a look on the sip stack, which is, in
>>>> fact, sofia ..., well, sound's like a nice, fun, long-going hobby project
>>>> to
>>>> me :-)
>>>> I will definitly at least look at the sip code to check if it is something
>>>> i
>>>> would myself willingly give over to ...
>>>> But I *really* do want ?a setup where it does not matter if one specific FS
>>>> instance is UP or NOT ...
>>>> 
>>>> --
>>>> Raimund Sacherer
>>>> -
>>>> RunSolutions
>>>> ????Open Source It Consulting
>>>> -
>>>> 
>>>> Parc Bit - Centro Empresarial Son Espanyol
>>>> Edificio Estel - Local 3D
>>>> 07121 - ?Palma de Mallorca
>>>> Baleares
>>>> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>>>> 
>>>> The sip stack needs to be modified to spin that data up into the state
>>>> machine so that it can take over calls once the fail over takes place...
>>>> its
>>>> not an easy task.
>>>> /b
>>>> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>>>> 
>>>> Is itnpossible to have a db cluster know the state of each and every call
>>>> and then use Heartbeat on this db +
>>>> fs cluster so that clients see only one ip where as internally all fs boxes
>>>> refer db for call states, db again is under replication.
>>>> This in the thioery can be written, but I am sure if we think bit more on
>>>> this direction the problem seem to be getting addressed.
>>>> Other guys also chip in their 2 cents, we just need 50 of em to make a full
>>>> dollar.
>>>> 
>>>> Thanks & Regards,
>>>> Mitul Limbani,
>>>> Founder & CEO,
>>>> Enterux Solutions Pvt. Ltd.,
>>>> The Enterprise Linux Company (r),
>>>> http://www.enterux.com
>>>> http://www.entVoice.com
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>> 
>>> 
>>> 
>>> --
>>> Jim Burke
>>> Director Evolutiontel.
>>> http://www.evolutiontel.net
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jason at jasonjgw.net  Sun Aug 30 22:57:44 2009
From: jason at jasonjgw.net (Jason White)
Date: Mon, 31 Aug 2009 15:57:44 +1000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <C6C0CBAC.5E420%krice@freeswitch.org>
References: <ddad80610908302132i4804be0eg3ea7a7ae2c8a1bfa@mail.gmail.com>
	<C6C0CBAC.5E420%krice@freeswitch.org>
Message-ID: <20090831055744.GA10477@jdc.jasonjgw.net>

Ken Rice <krice at freeswitch.org> wrote:
 
> I know there has already been some discussion on several fronts of atleast
> getting the core and several other pieces to where they need to be for
> stateful failover and I'm not sure if its been mentioned here, but sofia is
> going to be a bit of work and estimates run in the 100K USD range. Now if we
> could get a get a couple of corporate sponsors to help here it would be
> great.

I agree. Perhaps some of those corporations which are saving money over
proprietary solutions by using FreeSWITCH, and who also want the
high-availability functionality, would be ideal sponsors for the work.



From rs at runsolutions.com  Mon Aug 31 00:04:12 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Mon, 31 Aug 2009 09:04:12 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <20090831055744.GA10477@jdc.jasonjgw.net>
References: <ddad80610908302132i4804be0eg3ea7a7ae2c8a1bfa@mail.gmail.com>
	<C6C0CBAC.5E420%krice@freeswitch.org>
	<20090831055744.GA10477@jdc.jasonjgw.net>
Message-ID: <0F51835A-3E93-47D4-BBFF-623C25AC67D1@runsolutions.com>

Shouldn't be so hard, right? But there's human nature, even if the  
corporations get to save, like, 100k worth of licenses and hardware  
and other costs it is very hard to get them to spend a little back to  
the community which helped to achive that goal.

I think my first, very ambitous goal to make the sofia rewrite a  
personal long-term hobby is not going to be very realistic, because of  
my daytime job and my private life.

But I will try to persuade my employer to make at least a little  
contribution, as I really want this in FS, maybe we should

* make it a bounty project
* try to bring in some big sponsors (who are the biggest companys  
using FS anyway?)
* make some awareness in the voip scene??

seriously what else can be done to get the estimated 100k to become  
reality?

I really want this feature in FS! :-)

best,

-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 31, 2009, at 7:57 AM, Jason White wrote:

> Ken Rice <krice at freeswitch.org> wrote:
>
>> I know there has already been some discussion on several fronts of  
>> atleast
>> getting the core and several other pieces to where they need to be  
>> for
>> stateful failover and I'm not sure if its been mentioned here, but  
>> sofia is
>> going to be a bit of work and estimates run in the 100K USD range.  
>> Now if we
>> could get a get a couple of corporate sponsors to help here it  
>> would be
>> great.
>
> I agree. Perhaps some of those corporations which are saving money  
> over
> proprietary solutions by using FreeSWITCH, and who also want the
> high-availability functionality, would be ideal sponsors for the work.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From anatoliy at kounitskiy.com  Mon Aug 31 02:15:10 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Mon, 31 Aug 2009 12:15:10 +0300
Subject: [Freeswitch-users] Question about mod_limit
Message-ID: <1cd828b60908310215n60bdb242r3210a140b0811538@mail.gmail.com>

I have question about mod_limit: is there a reason to have difference
between the api applications and the dialplan? I mean when you use db
function from the cli you can use insert/select/delete but when you
use it from the dialplan you can only use insert/delete.

#define DB_USAGE "[insert|delete]/<realm>/<key>/<val>"
#define DB_DESC "save data"

SWITCH_STANDARD_APP(db_function)

but in the
SWITCH_STANDARD_API(db_api_function)
i have three options insert,delete and select

What i try to accomplish is to be able to insert values from the
dialplan and after that to be able to use them ( to select and use the
result) from another extension.

Thank you in advanced,
-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540


From pete at privateconnect.com  Mon Aug 31 03:33:43 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Mon, 31 Aug 2009 03:33:43 -0700
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Message-ID: <20090831033343.2ad02225396a31c9de30536f2e338977.a975af96c2.wbe@email04.secureserver.net>

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From erwin.davis at gmail.com  Mon Aug 31 04:31:52 2009
From: erwin.davis at gmail.com (Erwin Huang)
Date: Mon, 31 Aug 2009 07:31:52 -0400
Subject: [Freeswitch-users] some errors from FS startup
Message-ID: <ec9a78d50908310431g120729f8jc5a4302aa74b9840@mail.gmail.com>

Hi, I was able to build the FS successfully and make a couple of test calls.
But when I looked at the startup logs, I saw some errors. Please point out
how to fix them. Thanks,

Regards,

e

2009-08-31 06:59:55.54063 [ERR] mod_xml_curl.c:404 Binding has no url!
2009-08-31 06:59:55.54094 [CRIT] switch_loadable_module.c:871 Error Loading
modu
le /usr/local/freeswitch/mod/mod_xml_curl.so
**Module load routine returned an error**


2009-08-31 06:59:57.393133 [CONSOLE] mod_local_stream.c:142 Can't open
directory
: /usr/local/freeswitch/sounds/music/8000
2009-08-31 06:59:57.404228 [CONSOLE] mod_local_stream.c:142 Can't open
directory
: /usr/local/freeswitch/sounds/music/8000
2009-08-31 06:59:57.414708 [CONSOLE] mod_local_stream.c:142 Can't open
directory
: /usr/local/freeswitch/sounds/music/16000
2009-08-31 06:59:57.424568 [CONSOLE] mod_local_stream.c:142 Can't open
directory
: /usr/local/freeswitch/sounds/music/32000


// The errors below indicate the files donot exist. How to include these two
files into FS make build?

2009-08-31 06:59:57.477374 [CRIT] switch_loadable_module.c:871 Error Loading
mod
ule /usr/local/freeswitch/mod/mod_java.so
**/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No
such
 file or directory**


2009-08-31 06:59:57.485242 [CRIT] switch_loadable_module.c:871 Error Loading
mod
ule /usr/local/freeswitch/mod/mod_flite.so
**/usr/local/freeswitch/mod/mod_flite.so: cannot open shared object file: No
suc
h file or directory**
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From rs at runsolutions.com  Mon Aug 31 04:50:16 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Mon, 31 Aug 2009 13:50:16 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <20090831033343.2ad02225396a31c9de30536f2e338977.a975af96c2.wbe@email04.secureserver.net>
References: <20090831033343.2ad02225396a31c9de30536f2e338977.a975af96c2.wbe@email04.secureserver.net>
Message-ID: <237DA68E-B77C-489C-8CCB-35C241B7C8A8@runsolutions.com>

Hello Pete,

wow, i like it when you say getting the money is the easiest part!

I think if you are able to help procure the money we should have, as  
Jim Burke indicated, a rather good brainstorming session behind us.
I will create today a Wiki Page where we can define the Project, break  
up the development parts, generelly layout what has to be done and  
what will be the best and cleanest way to do it, as well define what  
and when to failover, etc.

If we have a pretty clear Idea on what will be the best way, you can  
get and fetch the money :-)

Seriously, i think we should discuss the possibility with Anthony how,  
if you/the community can come up with the money, this could be  
integrated into the development cycle!

Let's keep this rolling,
best

-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 31, 2009, at 12:33 PM, Pete Mueller wrote:

> Actually, getting the 100K is probably the easiest part of the  
> battle.  The problem will come in with the strings attached to the  
> 100K.  I could persuade my company to give FS devs the money, but  
> there would be conditions, such as:
> - There would need to be a defined timeline, milestones, and release  
> date
> - That release cannot be too far in the future (no one will fund an  
> 18 month project any more)
> - If those dates are missed, penalties will be applied
> - There will be some of branding/marketing/PR agreement (turning the  
> payment of development into a marketing tool)
> - There will be all sorts of legal waivers the devs will have to  
> sign (indemnifications, perpetual licenses, etc)
>
> From previous OSS experience, the "baggage" that comes with a large  
> payment is usually undesirable to dev groups.  Which leave  
> collecting a lot of little payments from many places.  Anyone that  
> has worked for some time fund raising for an organization will know  
> how much of a logistical mess that is. Not to mention people that  
> commit to pay, then back out.
>
> One good way to get funding for a project like this is to target the  
> government or non-industry related corporate entities.  Both cases  
> you target groups that don't compete in your market space (or rely  
> on your competitors), but would benefit from the outcome of the  
> project.
>
> An example would be Dell.  Dell would not see FS as competition in  
> any way, but if the case was made that adding feature X would  
> increase sales of Dell computers in the telcom space, or that  
> Company X would buy Dells to run FS if this features existed, then  
> Dell could look at investing in the project as a marketing expense.
>
> There are down sides with this method. Devs need to walk a fine line  
> with the corporation, ensuring that their product does not conflict  
> with some corporate initiative or large client.  Many times you will  
> be partnering with VARS or SIs to present the proposal, as there has  
> to be a project to justify the work.  Also, expect a long sales  
> cycle (5+ months). And, some purists would argue that the app is not  
> FOSS once your get some large corp to fund all/part of it.  Other  
> are just content with getting the job done.
>
> I offer my help in this area if the devs are interested in exploring  
> this style of fund raising.
> -pete
>
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
> From: Raimund Sacherer <rs at runsolutions.com>
> Date: Mon, August 31, 2009 12:04 am
> To: freeswitch-users at lists.freeswitch.org
>
> Shouldn't be so hard, right? But there's human nature, even if the
> corporations get to save, like, 100k worth of licenses and hardware
> and other costs it is very hard to get them to spend a little back to
> the community which helped to achive that goal.
>
> I think my first, very ambitous goal to make the sofia rewrite a
> personal long-term hobby is not going to be very realistic, because of
> my daytime job and my private life.
>
> But I will try to persuade my employer to make at least a little
> contribution, as I really want this in FS, maybe we should
>
> * make it a bounty project
> * try to bring in some big sponsors (who are the biggest companys
> using FS anyway?)
> * make some awareness in the voip scene??
>
> seriously what else can be done to get the estimated 100k to become
> reality?
>
> I really want this feature in FS! :-)
>
> best,
>
> -- 
> Raimund Sacherer
> -
> RunSolutions
> Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 - Palma de Mallorca
> Baleares
>
> On Aug 31, 2009, at 7:57 AM, Jason White wrote:
>
> > Ken Rice <krice at freeswitch.org> wrote:
> >
> >> I know there has already been some discussion on several fronts of
> >> atleast
> >> getting the core and several other pieces to where they need to be
> >> for
> >> stateful failover and I'm not sure if its been mentioned here, but
> >> sofia is
> >> going to be a bit of work and estimates run in the 100K USD range.
> >> Now if we
> >> could get a get a couple of corporate sponsors to help here it
> >> would be
> >> great.
> >
> > I agree. Perhaps some of those corporations which are saving money
> > over
> > proprietary solutions by using FreeSWITCH, and who also want the
> > high-availability functionality, would be ideal sponsors for the  
> work.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From dave at 3c.co.uk  Mon Aug 31 05:22:23 2009
From: dave at 3c.co.uk (David Knell)
Date: Mon, 31 Aug 2009 15:22:23 +0300
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <C6C0AC6C.5E412%krice@freeswitch.org>
References: <C6C0AC6C.5E412%krice@freeswitch.org>
Message-ID: <1251721344.6737.32.camel@dk-d820>

A random thought.  It strikes me that adding support for live call
failover to FreeSWITCH might well be the wrong thing to do - it's
something which is probably not too difficult to achieve if starting
from the ground up, but which is going to be a right PITA to do
retrospectively.  And, once you've done switching, there's conferencing,
IVR stuff...

Here's an alternative approach.  Have two FS boxes (which I'll call FS1
and FS2 - you can't fault *my* imagination) and a front-end B2BUA/RTP
proxy with a bit of a difference.  The B2BUA sends incoming stuff to
both FS1 and FS2 as two separate call legs, and the RTP proxy forwards
inbound RTP to both FS1 and FS2.  Assuming that FS1 is the current
"master", it forwards RTP and call state transitions from FS1 back to
the caller.  If it detects that FS1 has gone away, then - as FS2 should
be in roughly the same state as FS1 - it flips to using FS2 as its
source.

The internal state of the B2BUA/RTP proxy should be relatively simple,
and therefore should be able to be duplicated to a hot standby.

I can see a collection of gotchas, such as registration (the B2BUA needs
access to the user database), and I haven't even begun to try to think
of how to deal with calls *from* the FS boxes.  Actually, that's not
true: I have, and they clearly need to go through the same sort of
mechanism in reverse, which is fine until I start trying to work out how
to pair up calls from FS1 and FS2 reliably, and then my head hurts.

But I think that this idea has merit.  Firstly, it abstracts the
failover mechanism away into one place; secondly, it is general - FS1
and FS2 could just as easily be Asterisk1 and Asterisk2.

And I'll do it for just $98,650..

--Dave

> Your forgetting that with RTP theres the duplication of RTP that may be
> required along with a mechanism that keeps media timers working properly...
> 
> Where many things don't support RTP timers they sure come in handy on a
> regular basis especially dealing with several of the other software based
> platforms out there that just love to have calls end but never want to send
> a BYE
> 
> 
> > From: Jim Burke <jim at evolutiontel.net>
> > Reply-To: <freeswitch-users at lists.freeswitch.org>
> > Date: Mon, 31 Aug 2009 12:54:50 +1000
> > To: <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
> > 
> > Conceptually in a pure SIP installation of Freeswitch, hot standby or
> > active standby could be achieved through duplication of messages
> > (INVITE, 200OK and BYE for calls) between 2 boxes and then using VRRP
> > to change the IP address when the box falls over.  Going this way
> > allows call state to be as up to date as possible.  Obviously this
> > would require logic added to sofia to transfer the RTP port
> > information and UUID information between the FS instances (Easily
> > achieved using additional SIP headers).  It would also require that
> > sofia does not forward duplicated messages to gateways or user agents.
> >  CDR information would also need to be generated correctly (My
> > experience of Radius with Opensips is that it generates start and stop
> > messages on INVITE/200OK then BYE respectively so this could be an
> > easy solution to the CDR issue).
> > 
> > Regards,
> > 
> > 
> > 
> > On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
> >> So, the first way would be to have a look on the sip stack, which is, in
> >> fact, sofia ..., well, sound's like a nice, fun, long-going hobby project to
> >> me :-)
> >> I will definitly at least look at the sip code to check if it is something i
> >> would myself willingly give over to ...
> >> But I *really* do want  a setup where it does not matter if one specific FS
> >> instance is UP or NOT ...
> >> 
> >> --
> >> Raimund Sacherer
> >> -
> >> RunSolutions
> >>     Open Source It Consulting
> >> -
> >> 
> >> Parc Bit - Centro Empresarial Son Espanyol
> >> Edificio Estel - Local 3D
> >> 07121 -  Palma de Mallorca
> >> Baleares
> >> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
> >> 
> >> The sip stack needs to be modified to spin that data up into the state
> >> machine so that it can take over calls once the fail over takes place... its
> >> not an easy task.
> >> /b
> >> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
> >> 
> >> Is itnpossible to have a db cluster know the state of each and every call
> >> and then use Heartbeat on this db +
> >> fs cluster so that clients see only one ip where as internally all fs boxes
> >> refer db for call states, db again is under replication.
> >> This in the thioery can be written, but I am sure if we think bit more on
> >> this direction the problem seem to be getting addressed.
> >> Other guys also chip in their 2 cents, we just need 50 of em to make a full
> >> dollar.
> >> 
> >> Thanks & Regards,
> >> Mitul Limbani,
> >> Founder & CEO,
> >> Enterux Solutions Pvt. Ltd.,
> >> The Enterprise Linux Company (r),
> >> http://www.enterux.com
> >> http://www.entVoice.com
> >> 
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> 
> >> 
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> 
> >> 
> > 
> > 
> > 
> > -- 
> > Jim Burke
> > Director Evolutiontel.
> > http://www.evolutiontel.net
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



From fraunhofer.lists.freeswitch-001 at traced.net  Mon Aug 31 05:22:19 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Mon, 31 Aug 2009 14:22:19 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
	<3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
	<20090828001050.GA8108@jdc.jasonjgw.net>
	<86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com>
Message-ID: <a37510170908310522n73753a45sa346e6a3f255fcb9@mail.gmail.com>

Hello *,

still no luck. Removing the default dialplan with all those
hash-inserts made the memory-usage-curve flatter but it oom-coredumped
stating:

  freeswitch: src/switch_core_memory.c:443: switch_core_perform_alloc:
Assertion `ptr != ((void *)0)' failed.

after eating up 3.2 gig ram+swap.

Any chance or idea to track that down or any hint where to look at?

The other two freeswitch-machines in this scenario play well, they're
dumb load-sinks. what makes this machine different is on one hand the
lua-scripts that are run on it and on the other hand it's the only
machine who gets it's commands via xml_rpc.

Cheers
  Beni.


From rs at runsolutions.com  Mon Aug 31 05:36:48 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Mon, 31 Aug 2009 14:36:48 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <1251721344.6737.32.camel@dk-d820>
References: <C6C0AC6C.5E412%krice@freeswitch.org>
	<1251721344.6737.32.camel@dk-d820>
Message-ID: <C99339FA-1E87-4A23-9C25-FB7D564B83CB@runsolutions.com>

Hi David,

On Aug 31, 2009, at 2:22 PM, David Knell wrote:

> A random thought.  It strikes me that adding support for live call
> failover to FreeSWITCH might well be the wrong thing to do - it's
> something which is probably not too difficult to achieve if starting
> from the ground up, but which is going to be a right PITA to do
> retrospectively.  And, once you've done switching, there's  
> conferencing,
> IVR stuff...
>
> Here's an alternative approach.  Have two FS boxes (which I'll call  
> FS1
> and FS2 - you can't fault *my* imagination) and a front-end B2BUA/RTP
> proxy with a bit of a difference.  The B2BUA sends incoming stuff to
> both FS1 and FS2 as two separate call legs, and the RTP proxy forwards
> inbound RTP to both FS1 and FS2.  Assuming that FS1 is the current
> "master", it forwards RTP and call state transitions from FS1 back to
> the caller.  If it detects that FS1 has gone away, then - as FS2  
> should
> be in roughly the same state as FS1 - it flips to using FS2 as its
> source.
>
> The internal state of the B2BUA/RTP proxy should be relatively simple,
> and therefore should be able to be duplicated to a hot standby.
>
> I can see a collection of gotchas, such as registration (the B2BUA  
> needs
> access to the user database), and I haven't even begun to try to think
> of how to deal with calls *from* the FS boxes.  Actually, that's not
> true: I have, and they clearly need to go through the same sort of
> mechanism in reverse, which is fine until I start trying to work out  
> how
> to pair up calls from FS1 and FS2 reliably, and then my head hurts.
Yeah, sounds like a real challenge

>
> But I think that this idea has merit.  Firstly, it abstracts the
> failover mechanism away into one place; secondly, it is general - FS1
> and FS2 could just as easily be Asterisk1 and Asterisk2.
this, in theory, sound quite interesting ...
would be nice to have a proof of concept setup thought.

-> Sound's not quite THAT nice than a full-built-in solution to FS ...  
but if it works ...


>
> And I'll do it for just $98,650..
Haha, that's really funny :-)


>
> --Dave
>
>> Your forgetting that with RTP theres the duplication of RTP that  
>> may be
>> required along with a mechanism that keeps media timers working  
>> properly...
>>
>> Where many things don't support RTP timers they sure come in handy  
>> on a
>> regular basis especially dealing with several of the other software  
>> based
>> platforms out there that just love to have calls end but never want  
>> to send
>> a BYE
>>
>>
>>> From: Jim Burke <jim at evolutiontel.net>
>>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>>> Date: Mon, 31 Aug 2009 12:54:50 +1000
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
>>>
>>> Conceptually in a pure SIP installation of Freeswitch, hot standby  
>>> or
>>> active standby could be achieved through duplication of messages
>>> (INVITE, 200OK and BYE for calls) between 2 boxes and then using  
>>> VRRP
>>> to change the IP address when the box falls over.  Going this way
>>> allows call state to be as up to date as possible.  Obviously this
>>> would require logic added to sofia to transfer the RTP port
>>> information and UUID information between the FS instances (Easily
>>> achieved using additional SIP headers).  It would also require that
>>> sofia does not forward duplicated messages to gateways or user  
>>> agents.
>>> CDR information would also need to be generated correctly (My
>>> experience of Radius with Opensips is that it generates start and  
>>> stop
>>> messages on INVITE/200OK then BYE respectively so this could be an
>>> easy solution to the CDR issue).
>>>
>>> Regards,
>>>
>>>
>>>
>>> On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com 
>>> > wrote:
>>>> So, the first way would be to have a look on the sip stack, which  
>>>> is, in
>>>> fact, sofia ..., well, sound's like a nice, fun, long-going hobby  
>>>> project to
>>>> me :-)
>>>> I will definitly at least look at the sip code to check if it is  
>>>> something i
>>>> would myself willingly give over to ...
>>>> But I *really* do want  a setup where it does not matter if one  
>>>> specific FS
>>>> instance is UP or NOT ...
>>>>
>>>> --
>>>> Raimund Sacherer
>>>> -
>>>> RunSolutions
>>>>    Open Source It Consulting
>>>> -
>>>>
>>>> Parc Bit - Centro Empresarial Son Espanyol
>>>> Edificio Estel - Local 3D
>>>> 07121 -  Palma de Mallorca
>>>> Baleares
>>>> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>>>>
>>>> The sip stack needs to be modified to spin that data up into the  
>>>> state
>>>> machine so that it can take over calls once the fail over takes  
>>>> place... its
>>>> not an easy task.
>>>> /b
>>>> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>>>>
>>>> Is itnpossible to have a db cluster know the state of each and  
>>>> every call
>>>> and then use Heartbeat on this db +
>>>> fs cluster so that clients see only one ip where as internally  
>>>> all fs boxes
>>>> refer db for call states, db again is under replication.
>>>> This in the thioery can be written, but I am sure if we think bit  
>>>> more on
>>>> this direction the problem seem to be getting addressed.
>>>> Other guys also chip in their 2 cents, we just need 50 of em to  
>>>> make a full
>>>> dollar.
>>>>
>>>> Thanks & Regards,
>>>> Mitul Limbani,
>>>> Founder & CEO,
>>>> Enterux Solutions Pvt. Ltd.,
>>>> The Enterprise Linux Company (r),
>>>> http://www.enterux.com
>>>> http://www.entVoice.com
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> -- 
>>> Jim Burke
>>> Director Evolutiontel.
>>> http://www.evolutiontel.net
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
> -- 
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From enno.egbert at googlemail.com  Mon Aug 31 06:11:26 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Mon, 31 Aug 2009 06:11:26 -0700 (PDT)
Subject: [Freeswitch-users]  external Gateways
Message-ID: <25223286.post@talk.nabble.com>


Hello,

i use a sip gateway witch is only be able to allocate one sip channel.  How
can I bridge a call to a second gateway if the first gateway is in use. 

Now the dialplan look like this:

  <condition field="${toll_allow}" expression="domestic"/>
  <condition field="destination_number" expression="^04(\d+)$">
   <action application="set" data="effective_caller_id_number=300"/>
   <action application="set" data="sip_secure_media=false"/>
   <action application="bridge"
data="sofia/gateway/mygateway/04$1 at XXX.XXX.XXX.XXX"/>
  </condition> 

Thanks for help

 
-- 
View this message in context: http://www.nabble.com/external-Gateways-tp25223286p25223286.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.



From brian at freeswitch.org  Mon Aug 31 06:32:53 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 31 Aug 2009 08:32:53 -0500
Subject: [Freeswitch-users] memory leak
In-Reply-To: <a37510170908310522n73753a45sa346e6a3f255fcb9@mail.gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
	<3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
	<20090828001050.GA8108@jdc.jasonjgw.net>
	<86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com>
	<a37510170908310522n73753a45sa346e6a3f255fcb9@mail.gmail.com>
Message-ID: <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org>

Use valgrind.

/b

On Aug 31, 2009, at 7:22 AM, Benedikt Fraunhofer wrote:

>
> The other two freeswitch-machines in this scenario play well, they're
> dumb load-sinks. what makes this machine different is on one hand the
> lua-scripts that are run on it and on the other hand it's the only
> machine who gets it's commands via xml_rpc.



From rupa at rupa.com  Mon Aug 31 07:07:41 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 31 Aug 2009 09:07:41 -0500
Subject: [Freeswitch-users] memory leak
In-Reply-To: <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
	<3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
	<20090828001050.GA8108@jdc.jasonjgw.net>
	<86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com>
	<a37510170908310522n73753a45sa346e6a3f255fcb9@mail.gmail.com>
	<8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org>
Message-ID: <b39273530908310707k5b061aaaoc8fac345225ea3e6@mail.gmail.com>

Isn't there a known issue with lua+sql leaking memory on some platforms?

On Mon, Aug 31, 2009 at 8:32 AM, Brian West<brian at freeswitch.org> wrote:
> Use valgrind.
>
> /b
>
> On Aug 31, 2009, at 7:22 AM, Benedikt Fraunhofer wrote:
>
>>
>> The other two freeswitch-machines in this scenario play well, they're
>> dumb load-sinks. what makes this machine different is on one hand the
>> lua-scripts that are run on it and on the other hand it's the only
>> machine who gets it's commands via xml_rpc.

> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa


From anatoliy at kounitskiy.com  Mon Aug 31 07:32:05 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Mon, 31 Aug 2009 17:32:05 +0300
Subject: [Freeswitch-users] Question about mod_limit
In-Reply-To: <1cd828b60908310215n60bdb242r3210a140b0811538@mail.gmail.com>
References: <1cd828b60908310215n60bdb242r3210a140b0811538@mail.gmail.com>
Message-ID: <1cd828b60908310732p1c6db054kf3db1ea842e5a004@mail.gmail.com>

OK, I was able to answer one of the questions:
if I want to use the function db - i can insert and delete:
   <action application="db"
data="insert/test/${username}/${dialed_extension:3}"/>
   <action application="db"
data="delete/test/${username}/${dialed_extension:3}"/>
but if i want to select i have to use the API function db:
   <action application="set" data="test=${db(select/test/${username})}"/>

Probably there is a reason, that the dialplan application db can't do
select - someone input here?
I was looking something like this:
  <action application="db"
data="test=select/test/${username}/${dialed_extension:3}"/>
but at the end i found a solution. I'm just searching for a reason :)

Bear in mind that I'm newbie in the FreeSwitch :)


On Mon, Aug 31, 2009 at 12:15 PM, Anatoliy
Kounitskiy<anatoliy at kounitskiy.com> wrote:
> I have question about mod_limit: is there a reason to have difference
> between the api applications and the dialplan? I mean when you use db
> function from the cli you can use insert/select/delete but when you
> use it from the dialplan you can only use insert/delete.
>
> #define DB_USAGE "[insert|delete]/<realm>/<key>/<val>"
> #define DB_DESC "save data"
>
> SWITCH_STANDARD_APP(db_function)
>
> but in the
> SWITCH_STANDARD_API(db_api_function)
> i have three options insert,delete and select
>
> What i try to accomplish is to be able to insert values from the
> dialplan and after that to be able to use them ( to select and use the
> result) from another extension.
>
> Thank you in advanced,
> --
> Anatoliy Kounitskiy
> -------------------------
> E-mail: anatoliy at kounitskiy.com
> Mobile: +359898913540
>



-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540


From email.list.subscriber at gmail.com  Mon Aug 31 07:46:43 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Mon, 31 Aug 2009 10:46:43 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4A92566C.3030200@blueface.ie>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>	<4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>
	<4A92566C.3030200@blueface.ie>
Message-ID: <4a9be20d.22ba720a.5469.ffffb6ad@mx.google.com>

Sorry for the late reply, I was out of the office for the past two
weeks.  I'm not sure if there were updates over the mailing list
(still catching up), but while I was out, one of our sys admins was
able to get this compiled and has been verified to be functional on
our Solaris 10/x86 server.

Vladimir

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Bruce McAlister
Sent: Monday, August 24, 2009 4:59 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

Hi Vladimir,

Did you get an update on this at all?

Thanks


vmorales wrote:
> Hi Michal,
> 
> Just checking in to see if you've been able to take a stab at this.
> 
> Thanks,
> Vladimir
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michal Bielicki
> Sent: Tuesday, August 11, 2009 5:33 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
> 
> I'll retst it later today and give you a link with instructions
> 
> Am 10.08.2009 um 20:14 schrieb vmorales:
> 
>> By "./compile" I was referring to "./configure"
>>
>> Vladimir
>>
>> -----Original Message-----
>> From: vmorales [mailto:email.list.subscriber at gmail.com]
>> Sent: Monday, August 10, 2009 11:49 AM
>> To: 'freeswitch-users at lists.freeswitch.org'
>> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> Thanks for the response(s):
>>
>> I ran the "./compile" script with a set PREFIX.  This took a few
>> attempts with errors before it was able to complete error-free, as
I
>> had to install libtool.
>>
>> Since then, I have tried running 'make', 'gmake', and
>> '/opt/gnu/bin/make', but each results with an error.  This is the
>> error when running 'make' or 'gmake':
>>
>> <snip>
>> make: Fatal error: Command failed for target `all-recursive'
>> Current working directory /home/vmorales/freeswitch-1.0.4
>> *** Error code 1
>> make: Fatal error: Command failed for target `all'
>> </snip>
>>
>>
>> This is the error when running '/opt/gnu/bin/make':
>>
>> <snip>
>> make[5]: *** [mod_amr.so] Error 1
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_amr-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> Making all in build
>> +-------- FreeSWITCH Build Complete -----------+
>> + FreeSWITCH has been successfully built.      +
>> + Install by running:                          +
>> +                                              +
>> +               /opt/gnu/bin/make install                   +
>> +----------------------------------------------+
>> make[1]: *** [all-recursive] Error 1
>> make: *** [all] Error 2
>> </snip>
>>
>>
>> I re-untar'd before each compile/make attempt.  Let me know if this
> is
>> something that I can resolve.
>>
>> Vladimir
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Michael Jerris
>> Sent: Saturday, August 08, 2009 12:37 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> This is not currently a supported platform, it only builds on 64
bit
>> right now I think on solaris.
>>
>> Mike
>>
>> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>>
>>> Hello,
>>>
>>> Does anyone have, or know where to get, a pre-compiled copy of
>>> FreeSwitch for Solaris 10/x86?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>> http://www.freeswitch.org
> 
> Michal Bielicki
> Leiter der Niederlassung
> HaloKwadrat Sp. z o.o.
> Niederlassung Kleinmachnow
> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
> Ust.Id.: DE261885536
> Geschaeftsfuehrer: Aleksander Wiercinski
> Meiereifeld 2b, 14532 Kleinmachnow
> t. +49 33203 263220
> f. +49 33203 263229 sip. info at halokwadrat.de
> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
> Hauptgesch?ftsstelle:
> Halo Kwadrat Sp. z o.o.
> ul. Polna 46/14
> 00-644 Warszawa, Polen
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From kawarod at laposte.net  Mon Aug 31 08:00:59 2009
From: kawarod at laposte.net (rod)
Date: Mon, 31 Aug 2009 19:00:59 +0400
Subject: [Freeswitch-users] Set disable-transcoding in dialplan
Message-ID: <4A9BE5AB.4030304@laposte.net>

Hi all,

I'm wondering if I can do something like this:
    - in my internal profile, I have this because of some PEER using G729:
          - <param name="disable-transcoding" value="true"/>

But for a specific PEER, I'd like to activate transcoding:
          - for this PEER, only G711 is used
          - I'd like to transcode DTMF SIP INFO or RFC2833 to INBAND

So in my dialplan, I tried before bridging:

    - <action application="set" data="disable-transcoding=false"/>
    - <action application="start_dtmf_generate" data="true"/>

But I still see RFC2833 events between my FS and PEER and the DTMF are 
not working.

So 2 questions:
    - does application "start_dtmf_generate" requires transcoding
    - if yes, can I set the variable disable-transcoding in my dialplan

regards,
rod


From steveu at coppice.org  Mon Aug 31 08:09:01 2009
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 31 Aug 2009 23:09:01 +0800
Subject: [Freeswitch-users] T.38 with V.34 support
Message-ID: <4A9BE78D.5040000@coppice.org>

Hi,

Does anyone have access to a system that supports T.38 FAXing at V.34 
rates. These things seem to be as rare as bugless software.

Steve



From gregt at cgicommunications.com  Mon Aug 31 08:22:08 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Mon, 31 Aug 2009 11:22:08 -0400
Subject: [Freeswitch-users] Incorrect method of PHP call control?
Message-ID: <4E10B925-CEFD-4AF1-8527-282DCDEC0603@cgicommunications.com>

Hi. Before I go to far down this path, I wonder if what I intend to do  
is not a good practice.

I started using mod_xml_curl to use PHP on localhost to generate a  
dialplan dynamically, based on the Caller-Destination-Number variable  
that is posted. It prints out the XML that calls the javascript that  
then controls the call. For example,

$response = <<< XML
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
   <section name="dialplan" description="example_curl_dialplan">
     <context name="public">
       <extension name="curl_test">
         <condition field="destination_number" expression="^(\+1|1?) 
(5844111)$">
           <action application="javascript" data="demos/stest-examp- 
st.js" />
         </condition>
       </extension>
     </context>
   </section>
</document>
XML;

Then I thought, that's silly to go back out to javascript to handle  
the actions, playing files, using pocketsphinx, etc. I should just  
stay in PHP, using esl.php to answer and handle the call.

Then I rethought, is that a good practice to take over the call  
control from freeswitch at that point, while it is in the xml-curl  
dialplan hunt?

Then I also thought, is it even possible to do some of the things I  
need to do from the php esl, like the equivalent of this javascript:
session.collectInput(onInputsml, "emptyobject", 7000);
--
Greg Thoen


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From foxb at abv.bg  Mon Aug 31 08:49:55 2009
From: foxb at abv.bg (Hristo Benev)
Date: Mon, 31 Aug 2009 18:49:55 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <1590255305.13053.1251733804539.JavaMail.apache@mail21.abv.bg>

 Now it is correct.

Thank you for your time.



 >-------- ?????????? ????? --------
 >??:  rod 
 >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >??: freeswitch-users at lists.freeswitch.org
 >????????? ??: ??????, 2009, ?????? 29 08:43:24 EEST

 >Ok,
 >
 >I found what's happening. I probably did some change on the wiki to 
 >start reflect the new configuration, without having time enough to check 
 >the configuration. There is a mistake in your dialplan configuration. 
 >You should put this instead:
 >
 >wrong line:
 >
 >
 >
 >please correct with:
 >
 >
 >
 >This should match PEER_01 in dialplan instead of trying matching 
 >France at PEER_01.
 >
 >Let me know if this is right now.
 >
 >rod
 >
 >Hristo Benev a ?crit :
 >>  Hello Rod,
 >>
 >> I did the change.
 >>
 >> Here is extract of console:
 >> -----------------------------------------------------------------------------------------
 >> variable_continue_on_fail: [true]
 >> variable_sip_h_X-ROUTE: [LOOKUP]
 >> variable_export_vars: [sip_h_X-ROUTE]
 >> variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20]
 >> variable_sip_redirect_contact_0: [sip:France at PEER_01]
 >> variable_sip_redirected_to: [sip:France at PEER_01]
 >> variable_sip_redirect_contact_user_0: [France]  variable_sip_redirect_contact_host_0: [PEER_01]
 >> variable_sip_redirect_dialstring_0: [sofia/internal/sip:France at PEER_01]
 >> variable_sip_redirect_dialstring: [sofia/internal/sip:France at PEER_01]
 >> variable_proto_specific_hangup_cause: [sip:503]
 >> variable_sip_hangup_phrase: [DNS Error]
 >> variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE]
 >> variable_ROUTE_GW: [France]
 >> variable_AREA: [France]
 >> variable_current_application: [info]
 >> ------------------------------------------------------------------------------------------
 >> I have different value it is actually the description field as shown here:
 >>
 >> --------------------------------
 >>  /opt/kamailio/sbin/kamctl cr show
 >> cr carrier names
 >> +----+---------+
 >> | id | carrier |
 >> +----+---------+
 >> |  1 | default |
 >> +----+---------+
 >> cr domain names
 >> +----+---------+
 >> | id | domain  |
 >> +----+---------+
 >> |  1 | default |
 >> +----+---------+
 >> cr routes
 >> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
 >> | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description |
 >> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
 >> |  1 |       1 |      1 | 1000        |     0 |    0 |    1 |     0 | PEER_01      |                |                | France      |
 >> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
 >> -----------------------------------------
 >>
 >> And here is what I have in kamailio:
 >> -------------------------------------------------------------------------------------
 >> ####### Routing Logic ########
 >>
 >>
 >> # main request routing logic
 >>
 >> route{
 >>
 >>         if (!mf_process_maxfwd_header("10")) {
 >>                 sl_send_reply("483","Too Many Hops");
 >>                 exit;
 >>         }
 >>
 >>         t_check_trans();
 >>
 >>         if ($rU==NULL) {
 >>                 # request with no Username in RURI
 >>                 sl_send_reply("484","Address Incomplete");
 >>                 exit;
 >>         }
 >>
 >>         # LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP
 >>         if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){
 >>                 if(!cr_route("default", "default", "$rU", "$rU", "call_id","$avp(s:route_desc)")){
 >>                      #xlog("ROUTING FAILED: no route found for $rU");
 >>                      sl_send_reply("604", "Unable to route this call");
 >>                      exit;
 >>                 } else {
 >>                      xlog("LOOKUP FOUND: $rd $avp(s:route_desc)");
 >>                      avp_pushto("$ru/username", "$avp(s:route_desc)");
 >>                      sl_send_reply("302", "$rd");
 >>                      exit;
 >>                 }
 >>         }
 >> }
 >> ----------------------------------------------------------------------------------
 >>
 >> Another question...
 >> In that part of FreeSwitch dialplan.xml
 >>
 >> -----------------------------------
 >>   
 >>
 >>     
 >>       
 >>         
 >>
 >> -----------------------------------
 >>
 >> X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the regex is true.
 >>
 >>
 >> As for thanks - for sure by default they are also for the developers of both apps.
 >> I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning so it will be nice to have something working to start with.
 >>
 >>
 >>
 >>  >-------- ?????????? ????? --------
 >>  >??:  rod 
 >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >??: freeswitch-users at lists.freeswitch.org
 >>  >????????? ??: ?????, 2009, ?????? 28 09:54:00 EEST
 >>
 >>  >Hello,
 >>  >
 >>  >the trace seems good.
 >>  >If you check the answer from Kamailio, you'll see that Kamailio answers 
 >>  >with "302 PEER_01".
 >>  >
 >>  >As Michael Collins stated before, you can get the variable containing 
 >>  >"PEER_01", then this variable is stored in a custom variable.
 >>  >
 >>  >In your dialplan, may you please add:
 >>  >
 >>  >, just before the transfer line, eg:
 >>  >
 >>  > 
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >	 
 >>  >         
 >>  >      
 >>  >
 >>  >
 >>  >Using application Info, you'll see on the console (or CDR) the list of 
 >>  >variables used for this call.  You should see the content of 
 >>  >"${sip_redirect_contact_user_0}" that should contain  the value 
 >>  >"PEER_01". Please check this and let me know.
 >>  >
 >>  >For the new configuration file, no problem for sharing. But as I wrote 
 >>  >on the wiki page, I worked on this setup cause LCR module was not 
 >>  >available when I start working on FS, nor I'm a good programmer to write 
 >>  >a server side HTTP script (used by xml_curl) that could scale to my needs.
 >>  >
 >>  >I enhanced a bit this configuration with support for fallback routing, 
 >>  >almost realtime graph (every minutes using www.cacti.net and some 
 >>  >functions in FS like limit_hash) of number of concurrent calls per AREA, 
 >>  >PEER...
 >>  >
 >>  >Thanks for "the good tutorial", but don't forget the dev team who did 
 >>  >this great product ;-)
 >>  >
 >>  >rod.
 >>  >
 >>  >
 >>  >Hristo Benev a ?crit :
 >>  >>  I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).
 >>  >>
 >>  >> If you need additional info I'll provide it.
 >>  >>
 >>  >> Here is trace:
 >>  >>
 >>  >> ngrep -d any -nn -i '1000' port 5062 -W byline
 >>  >> interface: any
 >>  >> filter: (ip or ip6) and ( port 5062 )
 >>  >> match: 1000
 >>  >> #
 >>  >> U 10.10.10.10:5090 -> 127.0.0.1:5062
 >>  >> INVITE sip:1000 at 127.0.0.1:5062 SIP/2.0.
 >>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
 >>  >> Max-Forwards: 69.
 >>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
 >>  >> To: .
 >>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
 >>  >> CSeq: 119574771 INVITE.
 >>  >> Contact: .
 >>  >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
 >>  >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
 >>  >> Supported: timer, precondition, path, replaces.
 >>  >> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
 >>  >> Content-Type: application/sdp.
 >>  >> Content-Disposition: session.
 >>  >> Content-Length: 429.
 >>  >> X-ROUTE: LOOKUP.
 >>  >> Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off.
 >>  >> .
 >>  >> v=0.
 >>  >> o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
 >>  >> s=FreeSWITCH.
 >>  >> c=IN IP4 10.10.10.10.
 >>  >> t=0 0.
 >>  >> m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
 >>  >> a=rtpmap:0 PCMU/8000.
 >>  >> a=rtpmap:115 G7221/32000.
 >>  >> a=fmtp:115 bitrate=48000.
 >>  >> a=rtpmap:107 G7221/16000.
 >>  >> a=fmtp:107 bitrate=32000.
 >>  >> a=rtpmap:9 G722/8000.
 >>  >> a=rtpmap:8 PCMA/8000.
 >>  >> a=rtpmap:3 GSM/8000.
 >>  >> a=rtpmap:101 telephone-event/8000.
 >>  >> a=fmtp:101 0-16.
 >>  >> a=rtpmap:13 CN/8000.
 >>  >> a=ptime:20.
 >>  >>
 >>  >> #
 >>  >> U 127.0.0.1:5062 -> 10.10.10.10:5090
 >>  >> SIP/2.0 302 PEER_01.
 >>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
 >>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
 >>  >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
 >>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
 >>  >> CSeq: 119574771 INVITE.
 >>  >> Contact: sip:France at PEER_01.
 >>  >> Server: Kamailio (1.5.2-notls (i386/linux)).
 >>  >> Content-Length: 0.
 >>  >> .
 >>  >>
 >>  >> #
 >>  >> U 10.10.10.10:5090 -> 127.0.0.1:5062
 >>  >> ACK sip:1000 at 127.0.0.1:5062 SIP/2.0.
 >>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
 >>  >> Max-Forwards: 69.
 >>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
 >>  >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
 >>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
 >>  >> CSeq: 119574771 ACK.
 >>  >> Content-Length: 0.
 >>  >> .
 >>  >>
 >>  >>
 >>  >>
 >>  >>
 >>  >>
 >>  >>  >-------- ?????????? ????? --------
 >>  >>  >??:  Hristo Benev 
 >>  >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >>  >??: freeswitch-users at lists.freeswitch.org
 >>  >>  >????????? ??: ?????????, 2009, ?????? 27 14:58:53 EEST
 >>  >>
 >>  >>  > 
 >>  >>  >Bojnour,
 >>  >>  >
 >>  >>  >I'll send a trace ASAP.
 >>  >>  >
 >>  >>  >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
 >>  >>  >
 >>  >>  >I'm really interested to see the updates of the project.
 >>  >>  >
 >>  >>  >Thank you for the good tutorial.
 >>  >>  >
 >>  >>  >Hristo
 >>  >>  >
 >>  >>  >
 >>  >>  > >-------- ?????????? ????? --------
 >>  >>  > >??:  rod 
 >>  >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >>  > >??: freeswitch-users at lists.freeswitch.org
 >>  >>  > >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST
 >>  >>  >
 >>  >>  > >Hi Hristo,
 >>  >>  > >
 >>  >>  > >I'm the author of this setup and wiki page. I did a lot of modifications 
 >>  >>  > >on this setup (alternative routing if failure essentially) but don't 
 >>  >>  > >have too much time to update the wiki.
 >>  >>  > >
 >>  >>  > >May you please send me an ngrep trace when you call 1000:
 >>  >>  > >
 >>  >>  > >ngrep -d any -nn -i '1000' port 5060 -W byline 
 >>  >>  > >
 >>  >>  > >I will check what's happening.
 >>  >>  > >Do you have an entry for 1000 in your mysql database ?
 >>  >>  > >
 >>  >>  > >regards,
 >>  >>  > >rod
 >>  >>  > >
 >>  >>  > >Hristo Benev a ?crit :
 >>  >>  > >>  
 >>  >>  > >> It seems that the problem is on kamailio configuration.
 >>  >>  > >> Will ask on their list.
 >>  >>  > >>
 >>  >>  > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
 >>  >>  > >>
 >>  >>  > >> I have it setup as friend in asterisk, but still ???
 >>  >>  > >>
 >>  >>  > >> Any ideas?
 >>  >>  > >>
 >>  >>  > >> Thanks,
 >>  >>  > >>
 >>  >>  > >>
 >>  >>  > >>  >-------- ?????????? ????? --------
 >>  >>  > >>  >??:  Hristo Benev 
 >>  >>  > >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >>  > >>  >??: freeswitch-users at lists.freeswitch.org
 >>  >>  > >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
 >>  >>  > >>
 >>  >>  > >>  > I think that the problem is here:
 >>  >>  > >>  >-------------------------
 >>  >>  > >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
 >>  >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
 >>  >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
 >>  >>  > >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
 >>  >>  > >>  >--------------------------
 >>  >>  > >>  >
 >>  >>  > >>  >Actually Regex FAIL
 >>  >>  > >>  >
 >>  >>  > >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
 >>  >>  > >>  >Here is my  default.xml:
 >>  >>  > >>  >----------------
 >>  >>  > >>  >
 >>  >>  > >>  >
 >>  >>  > >>  >
 >>  >>  > >>  >  
 >>  >>  > >>  >
 >>  >>  > >>  >     
 >>  >>  > >>  >      
 >>  >>  > >>  >         
 >>  >>  > >>  >         
 >>  >>  > >>  >         
 >>  >>  > >>  >         
 >>  >>  > >>  >         
 >>  >>  > >>  >         
 >>  >>  > >>  >         
 >>  >>  > >>  >      
 >>  >>  > >>  >    
 >>  >>  > >>  >
 >>  >>  > >>  >   
 >>  >>  > >>  >
 >>  >>  > >>  >  
 >>  >>  > >>  >
 >>  >>  > >>  >    
 >>  >>  > >>  >      
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >      
 >>  >>  > >>  >    
 >>  >>  > >>  >
 >>  >>  > >>  >  
 >>  >>  > >>  >
 >>  >>  > >>  >
 >>  >>  > >>  >
 >>  >>  > >>  >--------------------------
 >>  >>  > >>  >
 >>  >>  > >>  >
 >>  >>  > >>  >
 >>  >>  > >>  > >-------- ?????????? ????? --------
 >>  >>  > >>  > >??:  Brian West 
 >>  >>  > >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >>  > >>  > >??: freeswitch-users at lists.freeswitch.org
 >>  >>  > >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
 >>  >>  > >>  >
 >>  >>  > >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
 >>  >>  > >>  >You have to process all 302's in the dialplan. 
 >>  >>  > >>  >Set this on your sofia profile
 >>  >>  > >>  >You can set these variables sip_redirect_profile,
 >>  >>  > >>  >sip_redirect_context,
 >>  >>  > >>  >sip_redirect_dialplan,
 >>  >>  > >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
 >>  >>  > >>  >sip_redirected_to,
 >>  >>  > >>  >sip_redirect_contact_user_%d,
 >>  >>  > >>  >sip_redirect_contact_host_%d,
 >>  >>  > >>  >sip_redirect_contact_params_%d,
 >>  >>  > >>  >sip_redirect_dialstring_%d,
 >>  >>  > >>  >sip_redirect_dialstring,
 >>  >>  > >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
 >>  >>  > >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
 >>  >>  > >>  > >
 >>  >>  > >>  >
 >>  >>  > >>  >_______________________________________________
 >>  >>  > >>  >FreeSWITCH-users mailing list
 >>  >>  > >>  >FreeSWITCH-users at lists.freeswitch.org
 >>  >>  > >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >>  > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >>  > >>  >http://www.freeswitch.org
 >>  >>  > >>  >
 >>  >>  > >>
 >>  >>  > >> _______________________________________________
 >>  >>  > >> FreeSWITCH-users mailing list
 >>  >>  > >> FreeSWITCH-users at lists.freeswitch.org
 >>  >>  > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >>  > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >>  > >> http://www.freeswitch.org
 >>  >>  > >>   
 >>  >>  > >
 >>  >>  > >_______________________________________________
 >>  >>  > >FreeSWITCH-users mailing list
 >>  >>  > >FreeSWITCH-users at lists.freeswitch.org
 >>  >>  > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >>  > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >>  > >http://www.freeswitch.org
 >>  >>  > >
 >>  >>  >
 >>  >>  >_______________________________________________
 >>  >>  >FreeSWITCH-users mailing list
 >>  >>  >FreeSWITCH-users at lists.freeswitch.org
 >>  >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >>  >http://www.freeswitch.org
 >>  >>  >
 >>  >>
 >>  >> _______________________________________________
 >>  >> FreeSWITCH-users mailing list
 >>  >> FreeSWITCH-users at lists.freeswitch.org
 >>  >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >> http://www.freeswitch.org
 >>  >>   
 >>  >
 >>  >_______________________________________________
 >>  >FreeSWITCH-users mailing list
 >>  >FreeSWITCH-users at lists.freeswitch.org
 >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >http://www.freeswitch.org
 >>  >
 >>
 >> _______________________________________________
 >> FreeSWITCH-users mailing list
 >> FreeSWITCH-users at lists.freeswitch.org
 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >> http://www.freeswitch.org
 >>   
 >
 >_______________________________________________
 >FreeSWITCH-users mailing list
 >FreeSWITCH-users at lists.freeswitch.org
 >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >http://www.freeswitch.org
 >


From msc at freeswitch.org  Mon Aug 31 09:15:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:15:33 -0700
Subject: [Freeswitch-users] external Gateways
In-Reply-To: <25223286.post@talk.nabble.com>
References: <25223286.post@talk.nabble.com>
Message-ID: <87f2f3b90908310915o45bd1b8bv441d82cfce887f4d@mail.gmail.com>

On Mon, Aug 31, 2009 at 6:11 AM, NOx-WHV <enno.egbert at googlemail.com> wrote:

>
> Hello,
>
> i use a sip gateway witch is only be able to allocate one sip channel.  How
> can I bridge a call to a second gateway if the first gateway is in use.
>
> Now the dialplan look like this:
>
>  <condition field="${toll_allow}" expression="domestic"/>
>  <condition field="destination_number" expression="^04(\d+)$">
>   <action application="set" data="effective_caller_id_number=300"/>
>   <action application="set" data="sip_secure_media=false"/>
>   <action application="bridge"
> data="sofia/gateway/mygateway/04$1 at XXX.XXX.XXX.XXX"/>
>  </condition>
>
> Thanks for help
>
>
Try pipe-delimited list of dialstrings in your bridge:

<action application="bridge"
data="sofia/gateway/mygateway/04$1 at XXX.XXX.XXX.XXX
|sofia/gateway/mygw2/foo at bar|sofia/gateway/mygw3/foo at bar"/>

-MC

>
> --
> View this message in context:
> http://www.nabble.com/external-Gateways-tp25223286p25223286.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Aug 31 09:17:22 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:17:22 -0700
Subject: [Freeswitch-users] liverpie as gui-proxy?
In-Reply-To: <6f7c60c40908291937h57c632d5l84b6ec5ac1286c48@mail.gmail.com>
References: <6f7c60c40908291937h57c632d5l84b6ec5ac1286c48@mail.gmail.com>
Message-ID: <87f2f3b90908310917g62b4ba2er1588314ecf25bd06@mail.gmail.com>

On Sat, Aug 29, 2009 at 7:37 PM, tom <tomabroad at gmail.com> wrote:

> hi,
>
> i was actually looking for something completely else, but reading wiki  led
> me to the tought:
> if mod_event can basically do all kinds of FS-things, why not having a
> standalone ruby-on-rails as as gui which interacts with the evnet-socket via
> liverpie?
>
>
Yes, this is totally possible, at least in theory. The actual execute, of
course, requires real work.
-MC

> im a total newbie to FS, just writing down my thoughts...
>
>
> thx for ur replies
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Aug 31 09:21:46 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:21:46 -0700
Subject: [Freeswitch-users] How can it be done with freeswitch?
In-Reply-To: <25199956.post@talk.nabble.com>
References: <25184845.post@talk.nabble.com>
	<87f2f3b90908280907w4812a8dbr7a97385f9afdd376@mail.gmail.com>
	<25199956.post@talk.nabble.com>
Message-ID: <87f2f3b90908310921h57685f14o1da57ec4ec6390cb@mail.gmail.com>

On Fri, Aug 28, 2009 at 9:26 PM, Nagalenoj <nagalenoj at gmail.com> wrote:

>
> Thanks for your help. Is it the only possible way.?
> Else, will it be done with the functions listed in the below url,
> http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes
>
>
Yes, these classes and functions are the building blocks that you'll use to
construct your application. We have a few Perl examples you can use to get
you going. At this point you will probably just need to roll up your sleeves
and start hacking away at it. Definitely email or drop by #freeswitch on
irc.freenode.net if you have more questions.

-MC
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From msc at freeswitch.org  Mon Aug 31 09:25:01 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:25:01 -0700
Subject: [Freeswitch-users] What is the difference between preAnswer and
	ring_ready?
In-Reply-To: <6B5963B0-C53A-4005-A14F-DD3140F29560@avgs.ca>
References: <c55d46950908291830w6b83a6a7h17a6faf67732d58a@mail.gmail.com>
	<20090830015030.GA3163@jdc.jasonjgw.net>
	<6B5963B0-C53A-4005-A14F-DD3140F29560@avgs.ca>
Message-ID: <87f2f3b90908310925p2fa672a4n11d5bb3bef599f5b@mail.gmail.com>

Well said, Math. BTW, this info is available on the wiki:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pre_answer

-MC

On Sat, Aug 29, 2009 at 8:54 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> In SIP, ring_ready will send a 180 ringing, pre answer will send a 183
> with a session description, so that some audio can be exchanged while
> the call isnt answered yet (used a lot for inband error messages etc).
>
> Regards,
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 29-Aug-09, at 9:50 PM, Jason White wrote:
>
> > Max Bridgewater <max.bridgewater at gmail.com> wrote:
> >> Hi,
> >> Assuming an inbound call, I have trouble understanding what the
> >> supposed
> >> difference between the following two set of instructions is:
> >>
> >> session.execute("ring_ready");
> >> session.set("set", "ringback=/home/ring.wav");
> >
> > Ring_ready sends a SIP "ringing" message and will use your device's
> > default
> > ringback sound - at least, that has been my experience. I should
> > remember the
> > message number in the SIP protocol, but I don't (the SIP experts
> > here will
> > undoubtedly have the details at the ready).
> >>
> >> and
> >>
> >> session.preAnswer();
> >> session.streamFile("/home/ring.wav")
> >
> > I think that will play the file as early media.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Mon Aug 31 09:31:59 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 31 Aug 2009 12:31:59 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <399999D5-EF3E-444F-8E06-05A0006DD053@freeswitch.org>
References: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
	<7B458C2D-8DC2-40B9-8EF1-B7FC934A5F0F@freeswitch.org>
	<c55d46950908301449u6bb3d43cifd92a353cca6cae5@mail.gmail.com>
	<399999D5-EF3E-444F-8E06-05A0006DD053@freeswitch.org>
Message-ID: <c55d46950908310931i56d4a272t937514c4ae253d7f@mail.gmail.com>

I thought i did that before trying. But i tried again and everything works.

Thanks for you unyielding patience.
Max.

On Sun, Aug 30, 2009 at 5:53 PM, Brian West <brian at freeswitch.org> wrote:

> Make sure you rm -rf /opt/swift then reinstall
>
> /b
>
> On Aug 30, 2009, at 4:49 PM, Max Bridgewater wrote:
>
> > Nop, the 64bit  cepstral doesn't work either:
> >
> > Creating mod_cepstral.so...
> > /usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when
> > searching for -lswift
> > /usr/bin/ld: cannot find -lswift
> > collect2: ld returned 1 exit status
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Aug 31 09:38:35 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:38:35 -0700
Subject: [Freeswitch-users] Set disable-transcoding in dialplan
In-Reply-To: <4A9BE5AB.4030304@laposte.net>
References: <4A9BE5AB.4030304@laposte.net>
Message-ID: <87f2f3b90908310938oa7e838dkd6e07579f224de37@mail.gmail.com>

Check out this page:
http://wiki.freeswitch.org/wiki/Codec_negotiation

Late negotiation will probably let you handle all the cases you need.
-MC

On Mon, Aug 31, 2009 at 8:00 AM, rod <kawarod at laposte.net> wrote:

> Hi all,
>
> I'm wondering if I can do something like this:
>    - in my internal profile, I have this because of some PEER using G729:
>          - <param name="disable-transcoding" value="true"/>
>
> But for a specific PEER, I'd like to activate transcoding:
>          - for this PEER, only G711 is used
>          - I'd like to transcode DTMF SIP INFO or RFC2833 to INBAND
>
> So in my dialplan, I tried before bridging:
>
>    - <action application="set" data="disable-transcoding=false"/>
>    - <action application="start_dtmf_generate" data="true"/>
>
> But I still see RFC2833 events between my FS and PEER and the DTMF are
> not working.
>
> So 2 questions:
>    - does application "start_dtmf_generate" requires transcoding
>    - if yes, can I set the variable disable-transcoding in my dialplan
>
> regards,
> rod
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Aug 31 09:45:00 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:45:00 -0700
Subject: [Freeswitch-users] some errors from FS startup
In-Reply-To: <ec9a78d50908310431g120729f8jc5a4302aa74b9840@mail.gmail.com>
References: <ec9a78d50908310431g120729f8jc5a4302aa74b9840@mail.gmail.com>
Message-ID: <87f2f3b90908310945i2f2118eam3fa4330a121f4090@mail.gmail.com>

Most likely you've enabled a few modules in
conf/autoload_configs/modules.conf.xml but you haven't built them in the
src/freeswitch.trunk/modules.conf file. Be sure to open modules.conf,
uncomment the modules that are throwing errors, like mod_xml_curl. Then run
"make install" again. For the music errors most likely you just need to
install the MOH files. Be sure to get the regular sound files too:

make cd-moh-install && make cd-sounds-install

-MC

On Mon, Aug 31, 2009 at 4:31 AM, Erwin Huang <erwin.davis at gmail.com> wrote:

> Hi, I was able to build the FS successfully and make a couple of test
> calls. But when I looked at the startup logs, I saw some errors. Please
> point out how to fix them. Thanks,
>
> Regards,
>
> e
>
> 2009-08-31 06:59:55.54063 [ERR] mod_xml_curl.c:404 Binding has no url!
> 2009-08-31 06:59:55.54094 [CRIT] switch_loadable_module.c:871 Error Loading
> modu
> le /usr/local/freeswitch/mod/mod_xml_curl.so
> **Module load routine returned an error**
>
>
> 2009-08-31 06:59:57.393133 [CONSOLE] mod_local_stream.c:142 Can't open
> directory
> : /usr/local/freeswitch/sounds/music/8000
> 2009-08-31 06:59:57.404228 [CONSOLE] mod_local_stream.c:142 Can't open
> directory
> : /usr/local/freeswitch/sounds/music/8000
> 2009-08-31 06:59:57.414708 [CONSOLE] mod_local_stream.c:142 Can't open
> directory
> : /usr/local/freeswitch/sounds/music/16000
> 2009-08-31 06:59:57.424568 [CONSOLE] mod_local_stream.c:142 Can't open
> directory
> : /usr/local/freeswitch/sounds/music/32000
>
>
> // The errors below indicate the files donot exist. How to include these
> two files into FS make build?
>
> 2009-08-31 06:59:57.477374 [CRIT] switch_loadable_module.c:871 Error
> Loading mod
> ule /usr/local/freeswitch/mod/mod_java.so
> **/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No
> such
>  file or directory**
>
>
> 2009-08-31 06:59:57.485242 [CRIT] switch_loadable_module.c:871 Error
> Loading mod
> ule /usr/local/freeswitch/mod/mod_flite.so
> **/usr/local/freeswitch/mod/mod_flite.so: cannot open shared object file:
> No suc
> h file or directory**
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bruce.mcalister at blueface.ie  Mon Aug 31 09:49:00 2009
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Mon, 31 Aug 2009 17:49:00 +0100
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4a9be20d.22ba720a.5469.ffffb6ad@mx.google.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>	<4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>	<4A92566C.3030200@blueface.ie>
	<4a9be20d.22ba720a.5469.ffffb6ad@mx.google.com>
Message-ID: <4A9BFEFC.2090800@blueface.ie>

Hi Vladimir,

Would you mind letting us know what your sysadmin did to achieve a 
successful build?

I am still having issues building freeswitch.

Thanks
Bruce

vmorales wrote:
> Sorry for the late reply, I was out of the office for the past two
> weeks.  I'm not sure if there were updates over the mailing list
> (still catching up), but while I was out, one of our sys admins was
> able to get this compiled and has been verified to be functional on
> our Solaris 10/x86 server.
> 
> Vladimir
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Bruce McAlister
> Sent: Monday, August 24, 2009 4:59 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
> 
> Hi Vladimir,
> 
> Did you get an update on this at all?
> 
> Thanks
> 
> 
> vmorales wrote:
>> Hi Michal,
>>
>> Just checking in to see if you've been able to take a stab at this.
>>
>> Thanks,
>> Vladimir
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Michal Bielicki
>> Sent: Tuesday, August 11, 2009 5:33 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> I'll retst it later today and give you a link with instructions
>>
>> Am 10.08.2009 um 20:14 schrieb vmorales:
>>
>>> By "./compile" I was referring to "./configure"
>>>
>>> Vladimir
>>>
>>> -----Original Message-----
>>> From: vmorales [mailto:email.list.subscriber at gmail.com]
>>> Sent: Monday, August 10, 2009 11:49 AM
>>> To: 'freeswitch-users at lists.freeswitch.org'
>>> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>>> 10/x86
>>>
>>> Thanks for the response(s):
>>>
>>> I ran the "./compile" script with a set PREFIX.  This took a few
>>> attempts with errors before it was able to complete error-free, as
> I
>>> had to install libtool.
>>>
>>> Since then, I have tried running 'make', 'gmake', and
>>> '/opt/gnu/bin/make', but each results with an error.  This is the
>>> error when running 'make' or 'gmake':
>>>
>>> <snip>
>>> make: Fatal error: Command failed for target `all-recursive'
>>> Current working directory /home/vmorales/freeswitch-1.0.4
>>> *** Error code 1
>>> make: Fatal error: Command failed for target `all'
>>> </snip>
>>>
>>>
>>> This is the error when running '/opt/gnu/bin/make':
>>>
>>> <snip>
>>> make[5]: *** [mod_amr.so] Error 1
>>> make[4]: *** [all] Error 1
>>> make[3]: *** [mod_amr-all] Error 1
>>> make[2]: *** [all-recursive] Error 1
>>> Making all in build
>>> +-------- FreeSWITCH Build Complete -----------+
>>> + FreeSWITCH has been successfully built.      +
>>> + Install by running:                          +
>>> +                                              +
>>> +               /opt/gnu/bin/make install                   +
>>> +----------------------------------------------+
>>> make[1]: *** [all-recursive] Error 1
>>> make: *** [all] Error 2
>>> </snip>
>>>
>>>
>>> I re-untar'd before each compile/make attempt.  Let me know if this
>> is
>>> something that I can resolve.
>>>
>>> Vladimir
>>>
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Michael Jerris
>>> Sent: Saturday, August 08, 2009 12:37 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>>> 10/x86
>>>
>>> This is not currently a supported platform, it only builds on 64
> bit
>>> right now I think on solaris.
>>>
>>> Mike
>>>
>>> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>>>
>>>> Hello,
>>>>
>>>> Does anyone have, or know where to get, a pre-compiled copy of
>>>> FreeSwitch for Solaris 10/x86?
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> rs
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>>> http://www.freeswitch.org
>> Michal Bielicki
>> Leiter der Niederlassung
>> HaloKwadrat Sp. z o.o.
>> Niederlassung Kleinmachnow
>> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
>> Ust.Id.: DE261885536
>> Geschaeftsfuehrer: Aleksander Wiercinski
>> Meiereifeld 2b, 14532 Kleinmachnow
>> t. +49 33203 263220
>> f. +49 33203 263229 sip. info at halokwadrat.de
>> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
>> Hauptgesch?ftsstelle:
>> Halo Kwadrat Sp. z o.o.
>> ul. Polna 46/14
>> 00-644 Warszawa, Polen
>> EIngetragen im HRB Warszawa, KRS 0000153539
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 


From msc at freeswitch.org  Mon Aug 31 09:55:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:55:54 -0700
Subject: [Freeswitch-users] Incorrect method of PHP call control?
In-Reply-To: <4E10B925-CEFD-4AF1-8527-282DCDEC0603@cgicommunications.com>
References: <4E10B925-CEFD-4AF1-8527-282DCDEC0603@cgicommunications.com>
Message-ID: <87f2f3b90908310955u399f5c5v1214d58e6004e392@mail.gmail.com>

On Mon, Aug 31, 2009 at 8:22 AM, Greg Thoen <gregt at cgicommunications.com>wrote:

> Hi. Before I go to far down this path, I wonder if what I intend to do is
> not a good practice.
> I started using mod_xml_curl to use PHP on localhost to generate a dialplan
> dynamically, based on the Caller-Destination-Number variable that is posted.
> It prints out the XML that calls the javascript that then controls the call.
> For example,
>
> $response = <<< XML
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="dialplan" description="example_curl_dialplan">
>     <context name="public">
>       <extension name="curl_test">
>         <condition field="destination_number"
> expression="^(\+1|1?)(5844111)$">
>           <action application="javascript" data="demos/stest-examp-st.js"
> />
>         </condition>
>       </extension>
>     </context>
>   </section>
> </document>
> XML;
>
> Then I thought, that's silly to go back out to javascript to handle the
> actions, playing files, using pocketsphinx, etc. I should just stay in PHP,
> using esl.php to answer and handle the call.
>
> Then I rethought, is that a good practice to take over the call control
> from freeswitch at that point, while it is in the xml-curl dialplan hunt?
>
> Then I also thought, is it even possible to do some of the things I need to
> do from the php esl, like the equivalent of this javascript:
> session.collectInput(onInputsml, "emptyobject", 7000);
>
> --
>
> Greg Thoen
>
>
Just remember that you're dealing with two somewhat related but still
distinctly separate entities: generating a dialplan and executing some sort
of call control from the dialplan. You need some sort of dialplan no matter
what, so the issue there is whether you need a dynamic one or not. If you're
just going to drop calls to an extension that opens an outbound socket to
your call control program then you may not need the dynamic dp generation
that mod_xml_curl gives you. You'll have to decide on static vs. dynamic
based on your needs. In either case, once the call is connected to your
socket you've got all sorts of control options. PHP has an ESL abstraction
just like the other languages so there shouldn't be any issue about PHP
lacking the ability to control calls.

I say start hacking away at it and see what happens. :) Definitely join us
in #freeswitch on irc.freenode.net if you want to discuss this more in
realtime.
-MC
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From email.list.subscriber at gmail.com  Mon Aug 31 10:16:13 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Mon, 31 Aug 2009 13:16:13 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4A9BFEFC.2090800@blueface.ie>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>	<4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>	<4A92566C.3030200@blueface.ie>	<4a9be20d.22ba720a.5469.ffffb6ad@mx.google.com>
	<4A9BFEFC.2090800@blueface.ie>
Message-ID: <4a9c0516.19bd720a.23b1.27df@mx.google.com>

Using Sun Studio and exporting the following flag:

	export CFLAGS="-lresolv"

Is how our sysadmin was able to get FS to build.

Vladimir

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Bruce McAlister
Sent: Monday, August 31, 2009 12:49 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

Hi Vladimir,

Would you mind letting us know what your sysadmin did to achieve a 
successful build?

I am still having issues building freeswitch.

Thanks
Bruce

vmorales wrote:
> Sorry for the late reply, I was out of the office for the past two
> weeks.  I'm not sure if there were updates over the mailing list
> (still catching up), but while I was out, one of our sys admins was
> able to get this compiled and has been verified to be functional on
> our Solaris 10/x86 server.
> 
> Vladimir
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Bruce McAlister
> Sent: Monday, August 24, 2009 4:59 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
> 
> Hi Vladimir,
> 
> Did you get an update on this at all?
> 
> Thanks
> 
> 
> vmorales wrote:
>> Hi Michal,
>>
>> Just checking in to see if you've been able to take a stab at this.
>>
>> Thanks,
>> Vladimir
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Michal Bielicki
>> Sent: Tuesday, August 11, 2009 5:33 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> I'll retst it later today and give you a link with instructions
>>
>> Am 10.08.2009 um 20:14 schrieb vmorales:
>>
>>> By "./compile" I was referring to "./configure"
>>>
>>> Vladimir
>>>
>>> -----Original Message-----
>>> From: vmorales [mailto:email.list.subscriber at gmail.com]
>>> Sent: Monday, August 10, 2009 11:49 AM
>>> To: 'freeswitch-users at lists.freeswitch.org'
>>> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for
Solaris
>>> 10/x86
>>>
>>> Thanks for the response(s):
>>>
>>> I ran the "./compile" script with a set PREFIX.  This took a few
>>> attempts with errors before it was able to complete error-free, as
> I
>>> had to install libtool.
>>>
>>> Since then, I have tried running 'make', 'gmake', and
>>> '/opt/gnu/bin/make', but each results with an error.  This is the
>>> error when running 'make' or 'gmake':
>>>
>>> <snip>
>>> make: Fatal error: Command failed for target `all-recursive'
>>> Current working directory /home/vmorales/freeswitch-1.0.4
>>> *** Error code 1
>>> make: Fatal error: Command failed for target `all'
>>> </snip>
>>>
>>>
>>> This is the error when running '/opt/gnu/bin/make':
>>>
>>> <snip>
>>> make[5]: *** [mod_amr.so] Error 1
>>> make[4]: *** [all] Error 1
>>> make[3]: *** [mod_amr-all] Error 1
>>> make[2]: *** [all-recursive] Error 1
>>> Making all in build
>>> +-------- FreeSWITCH Build Complete -----------+
>>> + FreeSWITCH has been successfully built.      +
>>> + Install by running:                          +
>>> +                                              +
>>> +               /opt/gnu/bin/make install                   +
>>> +----------------------------------------------+
>>> make[1]: *** [all-recursive] Error 1
>>> make: *** [all] Error 2
>>> </snip>
>>>
>>>
>>> I re-untar'd before each compile/make attempt.  Let me know if
this
>> is
>>> something that I can resolve.
>>>
>>> Vladimir
>>>
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
Of
>>> Michael Jerris
>>> Sent: Saturday, August 08, 2009 12:37 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for
Solaris
>>> 10/x86
>>>
>>> This is not currently a supported platform, it only builds on 64
> bit
>>> right now I think on solaris.
>>>
>>> Mike
>>>
>>> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>>>
>>>> Hello,
>>>>
>>>> Does anyone have, or know where to get, a pre-compiled copy of
>>>> FreeSwitch for Solaris 10/x86?
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> rs
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>>> http://www.freeswitch.org
>> Michal Bielicki
>> Leiter der Niederlassung
>> HaloKwadrat Sp. z o.o.
>> Niederlassung Kleinmachnow
>> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
>> Ust.Id.: DE261885536
>> Geschaeftsfuehrer: Aleksander Wiercinski
>> Meiereifeld 2b, 14532 Kleinmachnow
>> t. +49 33203 263220
>> f. +49 33203 263229 sip. info at halokwadrat.de
>> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
>> Hauptgesch?ftsstelle:
>> Halo Kwadrat Sp. z o.o.
>> ul. Polna 46/14
>> 00-644 Warszawa, Polen
>> EIngetragen im HRB Warszawa, KRS 0000153539
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
> http://www.freeswitch.org
> 
> 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
http://www.freeswitch.org



From e.schmidbauer at gmail.com  Mon Aug 31 10:20:44 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Mon, 31 Aug 2009 13:20:44 -0400
Subject: [Freeswitch-users] remote endpoints
Message-ID: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>

I am unable to call a user outside of our local area network. the user
is registered on the external profile but there is no way to call the
phone. does anyone have any suggestions how to do this?


From msc at freeswitch.org  Mon Aug 31 10:42:21 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 10:42:21 -0700
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
Message-ID: <87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>

Could you give a few more details? For example, could you pastebin the
output of "sofia status profile external"?
-MC

On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer <e.schmidbauer at gmail.com>wrote:

> I am unable to call a user outside of our local area network. the user
> is registered on the external profile but there is no way to call the
> phone. does anyone have any suggestions how to do this?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Aug 31 11:01:47 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 11:01:47 -0700
Subject: [Freeswitch-users] prompt playback volume using mod_shout
In-Reply-To: <158584.63541.qm@web50801.mail.re2.yahoo.com>
References: <158584.63541.qm@web50801.mail.re2.yahoo.com>
Message-ID: <87f2f3b90908311101m5a4343d9w963535d11e4e55ec@mail.gmail.com>

If I may ask, what "insane" values did you try? Just curious.
-MC

On Sat, Aug 29, 2009 at 8:24 PM, Marc Orenberg <marc at kasteris.com> wrote:

> Hi,
>
> I'm finding that when I record a caller's voice using mod_shout in MP3
> format, the recording is too soft when I play it back.
> I see there's a "volume" setting in shout.conf.xml, but it says, "Dont
> change these unless you are insane".
> When I insanely try to increase this value, I don't here any difference in
> volume.
> Is this the right knob, or is there some other way to do this?
>
> Thanks,
> Marc
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From marc at kasteris.com  Mon Aug 31 11:17:05 2009
From: marc at kasteris.com (Marc Orenberg)
Date: Mon, 31 Aug 2009 11:17:05 -0700 (PDT)
Subject: [Freeswitch-users] prompt playback volume using mod_shout
In-Reply-To: <87f2f3b90908311101m5a4343d9w963535d11e4e55ec@mail.gmail.com>
References: <158584.63541.qm@web50801.mail.re2.yahoo.com>
	<87f2f3b90908311101m5a4343d9w963535d11e4e55ec@mail.gmail.com>
Message-ID: <879878.54860.qm@web50812.mail.re2.yahoo.com>

Michael,

Thanks for the reply.
It was initially set to .1.
So I first tried 1, and then 5.
Then, after a few beers to further impair my judgement, I raised it to 10.
None of these had any effect.

Thanks,
Marc








________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Monday, August 31, 2009 2:01:47 PM
Subject: Re: [Freeswitch-users] prompt playback volume using mod_shout

If I may ask, what "insane" values did you try? Just curious. 
-MC


On Sat, Aug 29, 2009 at 8:24 PM, Marc Orenberg <marc at kasteris.com> wrote:

Hi, 
>
>
>I'm finding that when I record a caller's voice using mod_shout in MP3 format, the recording is too soft when I play it back.
>I see there's a "volume" setting in shout.conf.xml, but it says, "Dont change these unless you are insane".
>When I insanely try to increase this value, I don't here any difference in volume.
>Is this the right knob, or is there some other way to do this?
>
>
>Thanks,
>Marc
>
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Aug 31 11:39:47 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 11:39:47 -0700
Subject: [Freeswitch-users] prompt playback volume using mod_shout
In-Reply-To: <879878.54860.qm@web50812.mail.re2.yahoo.com>
References: <158584.63541.qm@web50801.mail.re2.yahoo.com>
	<87f2f3b90908311101m5a4343d9w963535d11e4e55ec@mail.gmail.com>
	<879878.54860.qm@web50812.mail.re2.yahoo.com>
Message-ID: <87f2f3b90908311139o6936fb5fpca9b23919553d180@mail.gmail.com>

Hmmm... in the libmpg123 docs it says that vol is a double. Try doing
smaller values like .01 and see if you can affect the volume that way.
-MC

On Mon, Aug 31, 2009 at 11:17 AM, Marc Orenberg <marc at kasteris.com> wrote:

> Michael,
>
> Thanks for the reply.
> It was initially set to .1.
> So I first tried 1, and then 5.
> Then, after a few beers to further impair my judgement, I raised it to 10.
> None of these had any effect.
>
> Thanks,
> Marc
>
>
>
>
>
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, August 31, 2009 2:01:47 PM
> *Subject:* Re: [Freeswitch-users] prompt playback volume using mod_shout
>
> If I may ask, what "insane" values did you try? Just curious.
> -MC
>
> On Sat, Aug 29, 2009 at 8:24 PM, Marc Orenberg <marc at kasteris.com> wrote:
>
>> Hi,
>>
>> I'm finding that when I record a caller's voice using mod_shout in MP3
>> format, the recording is too soft when I play it back.
>> I see there's a "volume" setting in shout.conf.xml, but it says, "Dont
>> change these unless you are insane".
>> When I insanely try to increase this value, I don't here any difference in
>> volume.
>> Is this the right knob, or is there some other way to do this?
>>
>> Thanks,
>> Marc
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From e.schmidbauer at gmail.com  Mon Aug 31 11:54:52 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Mon, 31 Aug 2009 14:54:52 -0400
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
Message-ID: <2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>

=================================================================================================
Name                    external
Domain Name             N/A
DBName                  sofia_reg_external
Pres Hosts
Dialplan                XML
Context                 public
Challenge Realm         auto_to
RTP-IP                  192.168.0.125
Ext-RTP-IP              98.118.151.30
SIP-IP                  192.168.0.125
Ext-SIP-IP              98.118.151.30
URL                     sip:mod_sofia at 192.168.0.125:5080
BIND-URL                sip:mod_sofia at 192.168.0.125:5080
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS
CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h@20i,PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                17
FAILED-CALLS-IN         11
CALLS-OUT               9
FAILED-CALLS-OUT        9

Registrations:
=================================================================================================
Call-ID:        MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
User:           1000 at server1.altpressonline.com
Contact:        "1000" <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
Agent:          X-Lite release 1103k stamp 53621
Status:         Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
Host:           server1.altpressonline.com
IP:             69.204.30.67
Port:           16006
Auth-User:      1000
Auth-Realm:     server1.altpressonline.com

=================================================================================================

sorry, i don't know how to login to pastebin

On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org> wrote:
> Could you give a few more details? For example, could you pastebin the
> output of "sofia status profile external"?
> -MC
>
> On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer <e.schmidbauer at gmail.com>
> wrote:
>>
>> I am unable to call a user outside of our local area network. the user
>> is registered on the external profile but there is no way to call the
>> phone. does anyone have any suggestions how to do this?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jmesquita at freeswitch.org  Mon Aug 31 12:01:01 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 31 Aug 2009 16:01:01 -0300
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
Message-ID: <ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>

Check the password dialog. It will tell you what the username/password is.

post the logs for a call as well, please.

Regards,

jmesquita

On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer <e.schmidbauer at gmail.com>wrote:

>
> =================================================================================================
> Name                    external
> Domain Name             N/A
> DBName                  sofia_reg_external
> Pres Hosts
> Dialplan                XML
> Context                 public
> Challenge Realm         auto_to
> RTP-IP                  192.168.0.125
> Ext-RTP-IP              98.118.151.30
> SIP-IP                  192.168.0.125
> Ext-SIP-IP              98.118.151.30
> URL                     sip:mod_sofia at 192.168.0.125:5080
> BIND-URL                sip:mod_sofia at 192.168.0.125:5080
> HOLD-MUSIC              local_stream://moh
> OUTBOUND-PROXY          N/A
> CODECS
> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h@20i,PCMU,PCMA,GSM
> TEL-EVENT               101
> DTMF-MODE               rfc2833
> CNG                     13
> SESSION-TO              0
> MAX-DIALOG              0
> NOMEDIA                 false
> LATE-NEG                false
> PROXY-MEDIA             false
> AGGRESSIVENAT           false
> STUN-ENABLED            true
> STUN-AUTO-DISABLE       false
> CALLS-IN                17
> FAILED-CALLS-IN         11
> CALLS-OUT               9
> FAILED-CALLS-OUT        9
>
> Registrations:
>
> =================================================================================================
> Call-ID:        MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
> User:           1000 at server1.altpressonline.com
> Contact:        "1000" <sip:1000 at 69.204.30.67:16006
> ;rinstance=50cbd6140e1ab991>
> Agent:          X-Lite release 1103k stamp 53621
> Status:         Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
> Host:           server1.altpressonline.com
> IP:             69.204.30.67
> Port:           16006
> Auth-User:      1000
> Auth-Realm:     server1.altpressonline.com
>
>
> =================================================================================================
>
> sorry, i don't know how to login to pastebin
>
> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org>
> wrote:
> > Could you give a few more details? For example, could you pastebin the
> > output of "sofia status profile external"?
> > -MC
> >
> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer <e.schmidbauer at gmail.com
> >
> > wrote:
> >>
> >> I am unable to call a user outside of our local area network. the user
> >> is registered on the external profile but there is no way to call the
> >> phone. does anyone have any suggestions how to do this?
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From e.schmidbauer at gmail.com  Mon Aug 31 12:07:33 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Mon, 31 Aug 2009 15:07:33 -0400
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
Message-ID: <2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>

thanks...heres the pastebin:
http://pastebin.freeswitch.org/10171

2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Check the password dialog. It will tell you what the username/password is.
>
> post the logs for a call as well, please.
>
> Regards,
>
> jmesquita
>
> On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer <e.schmidbauer at gmail.com>
> wrote:
>>
>>
>> =================================================================================================
>> Name ? ? ? ? ? ? ? ? ? ?external
>> Domain Name ? ? ? ? ? ? N/A
>> DBName ? ? ? ? ? ? ? ? ?sofia_reg_external
>> Pres Hosts
>> Dialplan ? ? ? ? ? ? ? ?XML
>> Context ? ? ? ? ? ? ? ? public
>> Challenge Realm ? ? ? ? auto_to
>> RTP-IP ? ? ? ? ? ? ? ? ?192.168.0.125
>> Ext-RTP-IP ? ? ? ? ? ? ?98.118.151.30
>> SIP-IP ? ? ? ? ? ? ? ? ?192.168.0.125
>> Ext-SIP-IP ? ? ? ? ? ? ?98.118.151.30
>> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.0.125:5080
>> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.0.125:5080
>> HOLD-MUSIC ? ? ? ? ? ? ?local_stream://moh
>> OUTBOUND-PROXY ? ? ? ? ?N/A
>> CODECS
>> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h@20i,PCMU,PCMA,GSM
>> TEL-EVENT ? ? ? ? ? ? ? 101
>> DTMF-MODE ? ? ? ? ? ? ? rfc2833
>> CNG ? ? ? ? ? ? ? ? ? ? 13
>> SESSION-TO ? ? ? ? ? ? ?0
>> MAX-DIALOG ? ? ? ? ? ? ?0
>> NOMEDIA ? ? ? ? ? ? ? ? false
>> LATE-NEG ? ? ? ? ? ? ? ?false
>> PROXY-MEDIA ? ? ? ? ? ? false
>> AGGRESSIVENAT ? ? ? ? ? false
>> STUN-ENABLED ? ? ? ? ? ?true
>> STUN-AUTO-DISABLE ? ? ? false
>> CALLS-IN ? ? ? ? ? ? ? ?17
>> FAILED-CALLS-IN ? ? ? ? 11
>> CALLS-OUT ? ? ? ? ? ? ? 9
>> FAILED-CALLS-OUT ? ? ? ?9
>>
>> Registrations:
>>
>> =================================================================================================
>> Call-ID: ? ? ? ?MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
>> User: ? ? ? ? ? 1000 at server1.altpressonline.com
>> Contact: ? ? ? ?"1000"
>> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
>> Agent: ? ? ? ? ?X-Lite release 1103k stamp 53621
>> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
>> Host: ? ? ? ? ? server1.altpressonline.com
>> IP: ? ? ? ? ? ? 69.204.30.67
>> Port: ? ? ? ? ? 16006
>> Auth-User: ? ? ?1000
>> Auth-Realm: ? ? server1.altpressonline.com
>>
>>
>> =================================================================================================
>>
>> sorry, i don't know how to login to pastebin
>>
>> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org>
>> wrote:
>> > Could you give a few more details? For example, could you pastebin the
>> > output of "sofia status profile external"?
>> > -MC
>> >
>> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
>> > <e.schmidbauer at gmail.com>
>> > wrote:
>> >>
>> >> I am unable to call a user outside of our local area network. the user
>> >> is registered on the external profile but there is no way to call the
>> >> phone. does anyone have any suggestions how to do this?
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From jmesquita at freeswitch.org  Mon Aug 31 12:30:11 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 31 Aug 2009 16:30:11 -0300
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
	<2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
Message-ID: <ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>

   1. 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel
   sofia/external/Anonymous at anonymous.invalid entering state [terminated][
   487]

The far end seems to be replying with 487 - Request Terminated...

Nothing wrong on FS, seems to be a problem with your endpoints. Can you
enable a sip trace?

jmesquita


On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <e.schmidbauer at gmail.com>wrote:

> thanks...heres the pastebin:
> http://pastebin.freeswitch.org/10171
>
> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > Check the password dialog. It will tell you what the username/password
> is.
> >
> > post the logs for a call as well, please.
> >
> > Regards,
> >
> > jmesquita
> >
> > On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer <e.schmidbauer at gmail.com>
> > wrote:
> >>
> >>
> >>
> =================================================================================================
> >> Name                    external
> >> Domain Name             N/A
> >> DBName                  sofia_reg_external
> >> Pres Hosts
> >> Dialplan                XML
> >> Context                 public
> >> Challenge Realm         auto_to
> >> RTP-IP                  192.168.0.125
> >> Ext-RTP-IP              98.118.151.30
> >> SIP-IP                  192.168.0.125
> >> Ext-SIP-IP              98.118.151.30
> >> URL                     sip:mod_sofia at 192.168.0.125:5080
> >> BIND-URL                sip:mod_sofia at 192.168.0.125:5080
> >> HOLD-MUSIC              local_stream://moh
> >> OUTBOUND-PROXY          N/A
> >> CODECS
> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h@20i,PCMU,PCMA,GSM
> >> TEL-EVENT               101
> >> DTMF-MODE               rfc2833
> >> CNG                     13
> >> SESSION-TO              0
> >> MAX-DIALOG              0
> >> NOMEDIA                 false
> >> LATE-NEG                false
> >> PROXY-MEDIA             false
> >> AGGRESSIVENAT           false
> >> STUN-ENABLED            true
> >> STUN-AUTO-DISABLE       false
> >> CALLS-IN                17
> >> FAILED-CALLS-IN         11
> >> CALLS-OUT               9
> >> FAILED-CALLS-OUT        9
> >>
> >> Registrations:
> >>
> >>
> =================================================================================================
> >> Call-ID:        MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
> >> User:           1000 at server1.altpressonline.com
> >> Contact:        "1000"
> >> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
> >> Agent:          X-Lite release 1103k stamp 53621
> >> Status:         Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
> >> Host:           server1.altpressonline.com
> >> IP:             69.204.30.67
> >> Port:           16006
> >> Auth-User:      1000
> >> Auth-Realm:     server1.altpressonline.com
> >>
> >>
> >>
> =================================================================================================
> >>
> >> sorry, i don't know how to login to pastebin
> >>
> >> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org>
> >> wrote:
> >> > Could you give a few more details? For example, could you pastebin the
> >> > output of "sofia status profile external"?
> >> > -MC
> >> >
> >> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
> >> > <e.schmidbauer at gmail.com>
> >> > wrote:
> >> >>
> >> >> I am unable to call a user outside of our local area network. the
> user
> >> >> is registered on the external profile but there is no way to call the
> >> >> phone. does anyone have any suggestions how to do this?
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Mon Aug 31 12:45:58 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 31 Aug 2009 21:45:58 +0200
Subject: [Freeswitch-users] T.38 with V.34 support
In-Reply-To: <4A9BE78D.5040000@coppice.org>
References: <4A9BE78D.5040000@coppice.org>
Message-ID: <65d96fc80908311245j51026fd5g3c93213c67ba7e73@mail.gmail.com>

Hi Steve,

what exactly do you need?

i was experimenting with V.17/V.29 <-> V.34 fax interworking....

well... I have V.34 machines ready over here... I just need to reconfigure
some CPE to point towards FS and thats it.... what setup do you need ?

T.


On Mon, Aug 31, 2009 at 5:09 PM, Steve Underwood <steveu at coppice.org> wrote:

> Hi,
>
> Does anyone have access to a system that supports T.38 FAXing at V.34
> rates. These things seem to be as rare as bugless software.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From juanma.v82 at gmail.com  Mon Aug 31 12:56:48 2009
From: juanma.v82 at gmail.com (Juan Manuel Vicente)
Date: Mon, 31 Aug 2009 16:56:48 -0300
Subject: [Freeswitch-users] Load Balacing
Message-ID: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>

Hello:

I am using two boxes, with same domain, to register about 1000 users working
together a DNS box doing round robin to resolve two differentes Ips one to
each box. This two boxes are using the same database (through unixodbc).

When I use the funcion sofia_contact or sofia profile status both boxes show
the status of the user. Does not matter where the resgistration was did.

But when I try to call the user this call fails, the call only works if the
call is sent through the same box where the user is registered.

Any ideas where i need to looking for ?

Thx in Advance.
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From e.schmidbauer at gmail.com  Mon Aug 31 13:18:56 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Mon, 31 Aug 2009 16:18:56 -0400
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
	<2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
	<ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>
Message-ID: <2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com>

here's the sip trace:
http://pastebin.freeswitch.org/10172



2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel
> sofia/external/Anonymous at anonymous.invalid entering state [terminated][487]
>
> The far end seems to be replying with 487 - Request Terminated...
>
> Nothing wrong on FS, seems to be a problem with your endpoints. Can you
> enable a sip trace?
>
> jmesquita
>
>
> On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <e.schmidbauer at gmail.com>
> wrote:
>>
>> thanks...heres the pastebin:
>> http://pastebin.freeswitch.org/10171
>>
>> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> > Check the password dialog. It will tell you what the username/password
>> > is.
>> >
>> > post the logs for a call as well, please.
>> >
>> > Regards,
>> >
>> > jmesquita
>> >
>> > On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer <e.schmidbauer at gmail.com>
>> > wrote:
>> >>
>> >>
>> >>
>> >> =================================================================================================
>> >> Name ? ? ? ? ? ? ? ? ? ?external
>> >> Domain Name ? ? ? ? ? ? N/A
>> >> DBName ? ? ? ? ? ? ? ? ?sofia_reg_external
>> >> Pres Hosts
>> >> Dialplan ? ? ? ? ? ? ? ?XML
>> >> Context ? ? ? ? ? ? ? ? public
>> >> Challenge Realm ? ? ? ? auto_to
>> >> RTP-IP ? ? ? ? ? ? ? ? ?192.168.0.125
>> >> Ext-RTP-IP ? ? ? ? ? ? ?98.118.151.30
>> >> SIP-IP ? ? ? ? ? ? ? ? ?192.168.0.125
>> >> Ext-SIP-IP ? ? ? ? ? ? ?98.118.151.30
>> >> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.0.125:5080
>> >> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.0.125:5080
>> >> HOLD-MUSIC ? ? ? ? ? ? ?local_stream://moh
>> >> OUTBOUND-PROXY ? ? ? ? ?N/A
>> >> CODECS
>> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h@20i,PCMU,PCMA,GSM
>> >> TEL-EVENT ? ? ? ? ? ? ? 101
>> >> DTMF-MODE ? ? ? ? ? ? ? rfc2833
>> >> CNG ? ? ? ? ? ? ? ? ? ? 13
>> >> SESSION-TO ? ? ? ? ? ? ?0
>> >> MAX-DIALOG ? ? ? ? ? ? ?0
>> >> NOMEDIA ? ? ? ? ? ? ? ? false
>> >> LATE-NEG ? ? ? ? ? ? ? ?false
>> >> PROXY-MEDIA ? ? ? ? ? ? false
>> >> AGGRESSIVENAT ? ? ? ? ? false
>> >> STUN-ENABLED ? ? ? ? ? ?true
>> >> STUN-AUTO-DISABLE ? ? ? false
>> >> CALLS-IN ? ? ? ? ? ? ? ?17
>> >> FAILED-CALLS-IN ? ? ? ? 11
>> >> CALLS-OUT ? ? ? ? ? ? ? 9
>> >> FAILED-CALLS-OUT ? ? ? ?9
>> >>
>> >> Registrations:
>> >>
>> >>
>> >> =================================================================================================
>> >> Call-ID: ? ? ? ?MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
>> >> User: ? ? ? ? ? 1000 at server1.altpressonline.com
>> >> Contact: ? ? ? ?"1000"
>> >> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
>> >> Agent: ? ? ? ? ?X-Lite release 1103k stamp 53621
>> >> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
>> >> Host: ? ? ? ? ? server1.altpressonline.com
>> >> IP: ? ? ? ? ? ? 69.204.30.67
>> >> Port: ? ? ? ? ? 16006
>> >> Auth-User: ? ? ?1000
>> >> Auth-Realm: ? ? server1.altpressonline.com
>> >>
>> >>
>> >>
>> >> =================================================================================================
>> >>
>> >> sorry, i don't know how to login to pastebin
>> >>
>> >> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org>
>> >> wrote:
>> >> > Could you give a few more details? For example, could you pastebin
>> >> > the
>> >> > output of "sofia status profile external"?
>> >> > -MC
>> >> >
>> >> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
>> >> > <e.schmidbauer at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> I am unable to call a user outside of our local area network. the
>> >> >> user
>> >> >> is registered on the external profile but there is no way to call
>> >> >> the
>> >> >> phone. does anyone have any suggestions how to do this?
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From pjintheusa at gmail.com  Mon Aug 31 13:23:36 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 31 Aug 2009 16:23:36 -0400
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <191c3a030908281412h51b75088s8c40bd0a53688928@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
	<191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
	<367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>
	<191c3a030908281333w39f001bcx7b9914f7bdda4f8f@mail.gmail.com>
	<191c3a030908281340m83d3c55q96f1257341250b86@mail.gmail.com>
	<367751820908281353l69b26639va13b3d63321b89e5@mail.gmail.com>
	<191c3a030908281412h51b75088s8c40bd0a53688928@mail.gmail.com>
Message-ID: <367751820908311323v3ce593fej6e5fbf13e75c4e61@mail.gmail.com>

That makes perfect sense - I now send the call to a managed app instead of
park and every thing works great. thanks.



On Fri, Aug 28, 2009 at 5:12 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Taking over the session while it's parked like that from your code and
> asking it to play a file is making it do 2 things at once.
> The session's thread is already busy in the park loop so you would have to
> change it's state to something passive like soft_execute so
> it's thread was not doing anything or send it a message so the park loop
> would pick it up by using the uuid_broadcast app::arg
>
> there are variables you can set that begin group_confirm_* that you can set
> to do this all without even using a script.
>
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_variables
>
>
>
>
>
> On Fri, Aug 28, 2009 at 3:53 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> The dialplan is is kicking off the C# script:
>>
>> <condition field="destination_number" expression="^(12125553666)$">
>>     * <action application="managed" data="MyFSProjects.ProcessCall"/>*
>> </condition>
>>
>> I launch a thread and specify the uuid to originate:
>>
>> fsApi.Execute("originate",
>> string.Format("{{ignore_early_media=true,absolute_codec_string='PCMU'}}[*
>> origination_uuid={0}*,origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>> &park", *call.Uuid*, call.OutgoingCallerID, call.NumberToDial));
>>
>> in the callback i try and play the wav file:
>>
>> FreeSWITCH.Native.ManagedSession blegSession = new
>> FreeSWITCH.Native.ManagedSession(callID);
>>
>> if (blegSession.Ready() && blegSession.mediaReady())
>> {
>>       blegSession.Answer();
>>       blegSession.Execute("playback",
>> "prompts/press-1-to-accept-call-from.wav");
>> }
>>
>> I go on to bridge the leg a and b - that bits works fine.
>>
>>
>>
>> On Fri, Aug 28, 2009 at 4:40 PM, Anthony Minessale<
>> anthony.minessale at gmail.com> wrote:
>> > nevermind,
>> >
>> > I didn't look closely at your code.
>> >
>> > how did you end up with blegSession ?
>> > Are you running this script from the CLI or via the application
>> interface.
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > On Fri, Aug 28, 2009 at 3:33 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> the L16 codec is used because the wav files is raw PCM audio
>> >> which is being encoded to PCMU
>> >>
>> >> Have you tried doing this to a locally registered phone?
>> >>
>> >>
>> >>
>> >> On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones <pjintheusa at gmail.com>
>> >> wrote:
>> >>>
>> >>> thanks for the reply.
>> >>>
>> >>> Tried this and exactly the same result. The prompt is slowed - as if
>> >>> the person is drunk.
>> >>>
>> >>> I do notice that the originating call SDP shows:
>> >>>
>> >>> a=rtpmap:0 PCMU/8000
>> >>>
>> >>> but the play tries to use:
>> >>>
>> >>> EXECUTE sofia/external/6093693828
>> >>> playback(prompts/press-1-to-accept-call-from.wav)
>> >>> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
>> >>> Activated L16 at 8000hz 1 channels 20ms
>> >>>
>> >>>
>> >>> Should the codec be PCMU at 8000hz as opposed to L16 at 8000hz? If so of
>> >>> there a way of forcing this codec for play.
>> >>>
>> >>> Thanks
>> >>>
>> >>>
>> >>> Phillip
>> >>>
>> >>>
>> >>> Full code & logs
>> >>> http://pastebin.freeswitch.org/10147
>> >>> http://pastebin.freeswitch.org/10150
>> >>>
>> >>>
>> >>>
>> >>> On Thu, Aug 27, 2009 at 4:30 PM, Anthony
>> >>> Minessale<anthony.minessale at gmail.com> wrote:
>> >>> > try wedging {ignore_early_media=true} before the first [ in your
>> dial
>> >>> > string
>> >>> > and eliminate the code waiting for answer.
>> >>> >
>> >>> >
>> >>> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <
>> pjintheusa at gmail.com>
>> >>> > wrote:
>> >>> >>
>> >>> >> Hi there,
>> >>> >>
>> >>> >> I know there are other ways of doing this. I am just trying get to
>> >>> >> know have fun with the FreeSWITCH API.
>> >>> >>
>> >>> >> I am using originate and park a call:
>> >>> >>
>> >>> >> fsApi.Execute("originate",
>> >>> >>
>> >>> >>
>> >>> >>
>> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>> >>> >> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
>> >>> >>
>> >>> >> That works great, the phone rings. I want to play something to this
>> >>> >> called party when they pick up.
>> >>> >>
>> >>> >>  while (!blegSession.answered())
>> >>> >>  {
>> >>> >>        Log.WriteLine(LogLevel.Alert,
>> >>> >> "Inside::CallReturns:!Session.answered::Loop");
>> >>> >>        blegSession.sleep(500, 1);
>> >>> >>  }
>> >>> >>
>> >>> >> string promptFile = "prompts/whisper.wav";
>> >>> >> blegSession.StreamFile(promptFile, 0);
>> >>> >>
>> >>> >>
>> >>> >> This works - but the audio is choppy and slow.
>> >>> >>
>> >>> >> Is there something I need to do to that parked call before
>> streaming
>> >>> >> that
>> >>> >> file?
>> >>> >>
>> >>> >>
>> >>> >> Thanks for any input.
>> >>> >>
>> >>> >> Phil
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> FreeSWITCH-users mailing list
>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>
>> >>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>> >
>> >>> > --
>> >>> > Anthony Minessale II
>> >>> >
>> >>> > FreeSWITCH http://www.freeswitch.org/
>> >>> > ClueCon http://www.cluecon.com/
>> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >
>> >>> > AIM: anthm
>> >>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >>> > IRC: irc.freenode.net #freeswitch
>> >>> >
>> >>> > FreeSWITCH Developer Conference
>> >>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >>> > iax:guest at conference.freeswitch.org/888
>> >>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >>> > pstn:213-799-1400
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> pstn:213-799-1400
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Mon Aug 31 14:01:05 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 31 Aug 2009 17:01:05 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4a9c0516.19bd720a.23b1.27df@mx.google.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>	<4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>	<4A92566C.3030200@blueface.ie>	<4a9be20d.22ba720a.5469.ffffb6ad@mx.google.com>
	<4A9BFEFC.2090800@blueface.ie>
	<4a9c0516.19bd720a.23b1.27df@mx.google.com>
Message-ID: <CAF0BC3A-32AC-4B97-9035-B16534FA4749@jerris.com>

What was the error building (which lib) ?  We should probably push  
this in so configure supplies this to the build system


Mike

On Aug 31, 2009, at 1:16 PM, vmorales wrote:

> Using Sun Studio and exporting the following flag:
>
> 	export CFLAGS="-lresolv"
>
> Is how our sysadmin was able to get FS to build.
>
> Vladimir
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Bruce McAlister
> Sent: Monday, August 31, 2009 12:49 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> Hi Vladimir,
>
> Would you mind letting us know what your sysadmin did to achieve a
> successful build?
>
> I am still having issues building freeswitch.
>
> Thanks
> Bruce
>
> vmorales wrote:
>> Sorry for the late reply, I was out of the office for the past two
>> weeks.  I'm not sure if there were updates over the mailing list
>> (still catching up), but while I was out, one of our sys admins was
>> able to get this compiled and has been verified to be functional on
>> our Solaris 10/x86 server.
>>
>> Vladimir
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Bruce McAlister
>> Sent: Monday, August 24, 2009 4:59 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> Hi Vladimir,
>>
>> Did you get an update on this at all?
>>
>> Thanks
>>
>>
>> vmorales wrote:
>>> Hi Michal,
>>>
>>> Just checking in to see if you've been able to take a stab at this.
>>>
>>> Thanks,
>>> Vladimir
>>>
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Michal Bielicki
>>> Sent: Tuesday, August 11, 2009 5:33 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>>> 10/x86
>>>
>>> I'll retst it later today and give you a link with instructions
>>>
>>> Am 10.08.2009 um 20:14 schrieb vmorales:
>>>
>>>> By "./compile" I was referring to "./configure"
>>>>
>>>> Vladimir
>>>>
>>>> -----Original Message-----
>>>> From: vmorales [mailto:email.list.subscriber at gmail.com]
>>>> Sent: Monday, August 10, 2009 11:49 AM
>>>> To: 'freeswitch-users at lists.freeswitch.org'
>>>> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for
> Solaris
>>>> 10/x86
>>>>
>>>> Thanks for the response(s):
>>>>
>>>> I ran the "./compile" script with a set PREFIX.  This took a few
>>>> attempts with errors before it was able to complete error-free, as
>> I
>>>> had to install libtool.
>>>>
>>>> Since then, I have tried running 'make', 'gmake', and
>>>> '/opt/gnu/bin/make', but each results with an error.  This is the
>>>> error when running 'make' or 'gmake':
>>>>
>>>> <snip>
>>>> make: Fatal error: Command failed for target `all-recursive'
>>>> Current working directory /home/vmorales/freeswitch-1.0.4
>>>> *** Error code 1
>>>> make: Fatal error: Command failed for target `all'
>>>> </snip>
>>>>
>>>>
>>>> This is the error when running '/opt/gnu/bin/make':
>>>>
>>>> <snip>
>>>> make[5]: *** [mod_amr.so] Error 1
>>>> make[4]: *** [all] Error 1
>>>> make[3]: *** [mod_amr-all] Error 1
>>>> make[2]: *** [all-recursive] Error 1
>>>> Making all in build
>>>> +-------- FreeSWITCH Build Complete -----------+
>>>> + FreeSWITCH has been successfully built.      +
>>>> + Install by running:                          +
>>>> +                                              +
>>>> +               /opt/gnu/bin/make install                   +
>>>> +----------------------------------------------+
>>>> make[1]: *** [all-recursive] Error 1
>>>> make: *** [all] Error 2
>>>> </snip>
>>>>
>>>>
>>>> I re-untar'd before each compile/make attempt.  Let me know if
> this
>>> is
>>>> something that I can resolve.
>>>>
>>>> Vladimir
>>>>
>>>>
>>>> -----Original Message-----
>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
> Of
>>>> Michael Jerris
>>>> Sent: Saturday, August 08, 2009 12:37 AM
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for
> Solaris
>>>> 10/x86
>>>>
>>>> This is not currently a supported platform, it only builds on 64
>> bit
>>>> right now I think on solaris.
>>>>
>>>> Mike
>>>>
>>>> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> Does anyone have, or know where to get, a pre-compiled copy of
>>>>> FreeSwitch for Solaris 10/x86?
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>>> rs
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> rs
>>>> http://www.freeswitch.org
>>> Michal Bielicki
>>> Leiter der Niederlassung
>>> HaloKwadrat Sp. z o.o.
>>> Niederlassung Kleinmachnow
>>> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
>>> Ust.Id.: DE261885536
>>> Geschaeftsfuehrer: Aleksander Wiercinski
>>> Meiereifeld 2b, 14532 Kleinmachnow
>>> t. +49 33203 263220
>>> f. +49 33203 263229 sip. info at halokwadrat.de
>>> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
>>> Hauptgesch?ftsstelle:
>>> Halo Kwadrat Sp. z o.o.
>>> ul. Polna 46/14
>>> 00-644 Warszawa, Polen
>>> EIngetragen im HRB Warszawa, KRS 0000153539
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From mike at jerris.com  Mon Aug 31 14:02:18 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 31 Aug 2009 17:02:18 -0400
Subject: [Freeswitch-users] Load Balacing
In-Reply-To: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>
References: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>
Message-ID: <B84A71A8-9C66-44F1-8D11-FDC7BB99AB60@jerris.com>

nat mapping on the client box not allowing the invite from another ip?

Mike

On Aug 31, 2009, at 3:56 PM, Juan Manuel Vicente wrote:

> Hello:
>
> I am using two boxes, with same domain, to register about 1000 users  
> working together a DNS box doing round robin to resolve two  
> differentes Ips one to each box. This two boxes are using the same  
> database (through unixodbc).
>
> When I use the funcion sofia_contact or sofia profile status both  
> boxes show the status of the user. Does not matter where the  
> resgistration was did.
>
> But when I try to call the user this call fails, the call only works  
> if the call is sent through the same box where the user is registered.
>
> Any ideas where i need to looking for ?
>
> Thx in Advance.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From krice at freeswitch.org  Mon Aug 31 14:35:22 2009
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 31 Aug 2009 16:35:22 -0500
Subject: [Freeswitch-users] Load Balacing
In-Reply-To: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>
Message-ID: <C6C1AC4A.5E4D4%krice@freeswitch.org>

You need to write you a small application to use the sip_registration table
from your odbc datasource and route the call to the appropriate registration
server and then have that server send the calls out... I am doing this exact
same thing now and this is what I had to do to get it to work properly with
a stack of clients behind nat



From: Juan Manuel Vicente <juanma.v82 at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Mon, 31 Aug 2009 16:56:48 -0300
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Load Balacing

Hello:

I am using two boxes, with same domain, to register about 1000 users working
together a DNS box doing round robin to resolve two differentes Ips one to
each box. This two boxes are using the same database (through unixodbc).

When I use the funcion sofia_contact or sofia profile status both boxes show
the status of the user. Does not matter where the resgistration was did.

But when I try to call the user this call fails, the call only works if the
call is sent through the same box where the user is registered.

Any ideas where i need to looking for ?

Thx in Advance.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From andrew at hijacked.us  Mon Aug 31 14:57:29 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Mon, 31 Aug 2009 17:57:29 -0400
Subject: [Freeswitch-users] Load Balacing
In-Reply-To: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>
References: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>
Message-ID: <20090831215728.GD20919@hijacked.us>

On Mon, Aug 31, 2009 at 04:56:48PM -0300, Juan Manuel Vicente wrote:
> Hello:
> 
> I am using two boxes, with same domain, to register about 1000 users working
> together a DNS box doing round robin to resolve two differentes Ips one to
> each box. This two boxes are using the same database (through unixodbc).
> 
> When I use the funcion sofia_contact or sofia profile status both boxes show
> the status of the user. Does not matter where the resgistration was did.
> 
> But when I try to call the user this call fails, the call only works if the
> call is sent through the same box where the user is registered.
>

I had the same problem except that I was using the multicast module to
replicate registrations, I added a sofia config parameter to optionally
rewrite the fs_path part of the sip URI on replicated packets so I could
hairpin the sip messages through the original registrar. This doesn't
help you directly but maybe it'll give you an idea?

Andrew


From tculjaga at gmail.com  Mon Aug 31 15:48:50 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 1 Sep 2009 00:48:50 +0200
Subject: [Freeswitch-users] mod_opal
Message-ID: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>

hello,

i'm trying to build mod_opal and getting this error:



making all mod_logfile

making all mod_loopback

making all mod_native_file

making all mod_opal
Compiling mod_opal.cpp...
quiet_libtool: compile:  g++ -g -ggdb -I.
-I/home/tculjaga/freeswitch-trunk/src/include
-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
-D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal
-DHAVE_CONFIG_H -c mod_opal.cpp  -fPIC -DPIC -o .libs/mod_opal.o
In file included from mod_opal.cpp:25:
mod_opal.h:151: error: conflicting return type specified for ?virtual
OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)?
/usr/include/opal/opal/localep.h:267: error:   overriding ?virtual
ptlib_virtual_function_changed_or_removed******
OpalLocalEndPoint::CreateConnection(OpalCall&, void*)?
mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&,
FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
switch_channel_t*)?:
mod_opal.cpp:564: error: no matching function for call to
?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)?
/usr/include/opal/opal/localep.h:290: note: candidates are:
OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&,
void*, unsigned int, OpalConnection::StringOptions*, char)
/usr/include/opal/opal/localep.h:276: note:
OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
/usr/include/opal/opal/patch.h: In member function ?switch_status_t
FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)?:
/usr/include/opal/opal/patch.h:272: error: ?virtual
ptlib_virtual_function_changed_or_removed******
OpalMediaPatch::OnPatchStart()? is private
mod_opal.cpp:1277: error: within this context
mod_opal.cpp:1277: warning: ?virtual
ptlib_virtual_function_changed_or_removed******
OpalMediaPatch::OnPatchStart()? is deprecated (declared at
/usr/include/opal/opal/patch.h:272)
mod_opal.cpp:1277: warning: ignoring return value of function declared with
attribute warn_unused_result
/usr/include/opal/opal/patch.h: In member function ?switch_status_t
FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)?:
/usr/include/opal/opal/patch.h:272: error: ?virtual
ptlib_virtual_function_changed_or_removed******
OpalMediaPatch::OnPatchStart()? is private
mod_opal.cpp:1399: error: within this context
mod_opal.cpp:1399: warning: ?virtual
ptlib_virtual_function_changed_or_removed******
OpalMediaPatch::OnPatchStart()? is deprecated (declared at
/usr/include/opal/opal/patch.h:272)
mod_opal.cpp:1399: warning: ignoring return value of function declared with
attribute warn_unused_result
make[5]: *** [mod_opal.lo] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_opal-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
tculjaga at nemesis:~/freeswitch-trunk$
tculjaga at nemesis:~/freeswitch-trunk$
tculjaga at nemesis:~/freeswitch-trunk$



what ptlib/opal/fs version did you use to build it?


I tried with trunk (ptlib, opal, fs)... and as you can see :)



T.
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From msc at freeswitch.org  Mon Aug 31 16:00:39 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 16:00:39 -0700
Subject: [Freeswitch-users] mod_opal
In-Reply-To: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>
References: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>
Message-ID: <87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com>

On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hello,
>
> i'm trying to build mod_opal and getting this error:
>
>
>
> making all mod_logfile
>
> making all mod_loopback
>
> making all mod_native_file
>
> making all mod_opal
> Compiling mod_opal.cpp...
> quiet_libtool: compile:  g++ -g -ggdb -I.
> -I/home/tculjaga/freeswitch-trunk/src/include
> -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
> -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal
> -DHAVE_CONFIG_H -c mod_opal.cpp  -fPIC -DPIC -o .libs/mod_opal.o
> In file included from mod_opal.cpp:25:
> mod_opal.h:151: error: conflicting return type specified for ?virtual
> OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)?
> /usr/include/opal/opal/localep.h:267: error:   overriding ?virtual
> ptlib_virtual_function_changed_or_removed******
> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)?
> mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&,
> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
> switch_channel_t*)?:
> mod_opal.cpp:564: error: no matching function for call to
> ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)?
> /usr/include/opal/opal/localep.h:290: note: candidates are:
> OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&,
> void*, unsigned int, OpalConnection::StringOptions*, char)
> /usr/include/opal/opal/localep.h:276: note:
> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
> /usr/include/opal/opal/patch.h: In member function ?switch_status_t
> FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)?:
> /usr/include/opal/opal/patch.h:272: error: ?virtual
> ptlib_virtual_function_changed_or_removed******
> OpalMediaPatch::OnPatchStart()? is private
> mod_opal.cpp:1277: error: within this context
> mod_opal.cpp:1277: warning: ?virtual
> ptlib_virtual_function_changed_or_removed******
> OpalMediaPatch::OnPatchStart()? is deprecated (declared at
> /usr/include/opal/opal/patch.h:272)
> mod_opal.cpp:1277: warning: ignoring return value of function declared with
> attribute warn_unused_result
> /usr/include/opal/opal/patch.h: In member function ?switch_status_t
> FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)?:
> /usr/include/opal/opal/patch.h:272: error: ?virtual
> ptlib_virtual_function_changed_or_removed******
> OpalMediaPatch::OnPatchStart()? is private
> mod_opal.cpp:1399: error: within this context
> mod_opal.cpp:1399: warning: ?virtual
> ptlib_virtual_function_changed_or_removed******
> OpalMediaPatch::OnPatchStart()? is deprecated (declared at
> /usr/include/opal/opal/patch.h:272)
> mod_opal.cpp:1399: warning: ignoring return value of function declared with
> attribute warn_unused_result
> make[5]: *** [mod_opal.lo] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_opal-all] Error 1
> make[2]: *** [all-recursive] Error 1
> Making all in build
>  +-------- FreeSWITCH Build Complete -----------+
>  + FreeSWITCH has been successfully built.      +
>  + Install by running:                          +
>  +                                              +
>  +               make install                   +
>  +----------------------------------------------+
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> tculjaga at nemesis:~/freeswitch-trunk$
> tculjaga at nemesis:~/freeswitch-trunk$
> tculjaga at nemesis:~/freeswitch-trunk$
>
>
>
> what ptlib/opal/fs version did you use to build it?
>
>
> I tried with trunk (ptlib, opal, fs)... and as you can see :)
>

Did you run the buildopal.sh script in src/build ?
-MC
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From jmesquita at freeswitch.org  Mon Aug 31 16:37:46 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 31 Aug 2009 20:37:46 -0300
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
	<2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
	<ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>
	<2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com>
Message-ID: <ebbc2d550908311637q22ef76aaj6621a5d2345885e8@mail.gmail.com>

Problem is definetly on far end.

If you look at the siptrace, you have the following sequence:

1. Asterisk calls in
2. FreeSWITCH replies with a Trying(100) to complete call right away and
proceeds to dialplan
3. FreeSWITCH invites (calls) 7 times the final destination that never
responds.
4. Asterisk sends a CANCEL message

In all that, your final endpoint never responds to any message. Are you sure
you can reach it?

jmesquita

On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer <e.schmidbauer at gmail.com>wrote:

> here's the sip trace:
> http://pastebin.freeswitch.org/10172
>
>
>
> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel
> > sofia/external/Anonymous at anonymous.invalid entering state
> [terminated][487]
> >
> > The far end seems to be replying with 487 - Request Terminated...
> >
> > Nothing wrong on FS, seems to be a problem with your endpoints. Can you
> > enable a sip trace?
> >
> > jmesquita
> >
> >
> > On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <e.schmidbauer at gmail.com>
> > wrote:
> >>
> >> thanks...heres the pastebin:
> >> http://pastebin.freeswitch.org/10171
> >>
> >> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> >> > Check the password dialog. It will tell you what the username/password
> >> > is.
> >> >
> >> > post the logs for a call as well, please.
> >> >
> >> > Regards,
> >> >
> >> > jmesquita
> >> >
> >> > On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer <
> e.schmidbauer at gmail.com>
> >> > wrote:
> >> >>
> >> >>
> >> >>
> >> >>
> =================================================================================================
> >> >> Name                    external
> >> >> Domain Name             N/A
> >> >> DBName                  sofia_reg_external
> >> >> Pres Hosts
> >> >> Dialplan                XML
> >> >> Context                 public
> >> >> Challenge Realm         auto_to
> >> >> RTP-IP                  192.168.0.125
> >> >> Ext-RTP-IP              98.118.151.30
> >> >> SIP-IP                  192.168.0.125
> >> >> Ext-SIP-IP              98.118.151.30
> >> >> URL                     sip:mod_sofia at 192.168.0.125:5080
> >> >> BIND-URL                sip:mod_sofia at 192.168.0.125:5080
> >> >> HOLD-MUSIC              local_stream://moh
> >> >> OUTBOUND-PROXY          N/A
> >> >> CODECS
> >> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h
> @20i,PCMU,PCMA,GSM
> >> >> TEL-EVENT               101
> >> >> DTMF-MODE               rfc2833
> >> >> CNG                     13
> >> >> SESSION-TO              0
> >> >> MAX-DIALOG              0
> >> >> NOMEDIA                 false
> >> >> LATE-NEG                false
> >> >> PROXY-MEDIA             false
> >> >> AGGRESSIVENAT           false
> >> >> STUN-ENABLED            true
> >> >> STUN-AUTO-DISABLE       false
> >> >> CALLS-IN                17
> >> >> FAILED-CALLS-IN         11
> >> >> CALLS-OUT               9
> >> >> FAILED-CALLS-OUT        9
> >> >>
> >> >> Registrations:
> >> >>
> >> >>
> >> >>
> =================================================================================================
> >> >> Call-ID:        MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
> >> >> User:           1000 at server1.altpressonline.com
> >> >> Contact:        "1000"
> >> >> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
> >> >> Agent:          X-Lite release 1103k stamp 53621
> >> >> Status:         Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
> >> >> Host:           server1.altpressonline.com
> >> >> IP:             69.204.30.67
> >> >> Port:           16006
> >> >> Auth-User:      1000
> >> >> Auth-Realm:     server1.altpressonline.com
> >> >>
> >> >>
> >> >>
> >> >>
> =================================================================================================
> >> >>
> >> >> sorry, i don't know how to login to pastebin
> >> >>
> >> >> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org>
> >> >> wrote:
> >> >> > Could you give a few more details? For example, could you pastebin
> >> >> > the
> >> >> > output of "sofia status profile external"?
> >> >> > -MC
> >> >> >
> >> >> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
> >> >> > <e.schmidbauer at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> I am unable to call a user outside of our local area network. the
> >> >> >> user
> >> >> >> is registered on the external profile but there is no way to call
> >> >> >> the
> >> >> >> phone. does anyone have any suggestions how to do this?
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From e.schmidbauer at gmail.com  Mon Aug 31 17:06:52 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Mon, 31 Aug 2009 20:06:52 -0400
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <ebbc2d550908311637q22ef76aaj6621a5d2345885e8@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
	<2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
	<ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>
	<2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com>
	<ebbc2d550908311637q22ef76aaj6621a5d2345885e8@mail.gmail.com>
Message-ID: <2cef777b0908311706o4173312bw9af8b217aca47392@mail.gmail.com>

i cannot reach the remote endpoint. the remote endpoint can reach a
locally registered endpoint. any idea why this is happening? the
remote endpoint is behind a linksys with upnp enabled.

2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Problem is definetly on far end.
>
> If you look at the siptrace, you have the following sequence:
>
> 1. Asterisk calls in
> 2. FreeSWITCH replies with a Trying(100) to complete call right away and
> proceeds to dialplan
> 3. FreeSWITCH invites (calls) 7 times the final destination that never
> responds.
> 4. Asterisk sends a CANCEL message
>
> In all that, your final endpoint never responds to any message. Are you sure
> you can reach it?
>
> jmesquita
>
> On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer <e.schmidbauer at gmail.com>
> wrote:
>>
>> here's the sip trace:
>> http://pastebin.freeswitch.org/10172
>>
>>
>>
>> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> > 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel
>> > sofia/external/Anonymous at anonymous.invalid entering state
>> > [terminated][487]
>> >
>> > The far end seems to be replying with 487 - Request Terminated...
>> >
>> > Nothing wrong on FS, seems to be a problem with your endpoints. Can you
>> > enable a sip trace?
>> >
>> > jmesquita
>> >
>> >
>> > On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <e.schmidbauer at gmail.com>
>> > wrote:
>> >>
>> >> thanks...heres the pastebin:
>> >> http://pastebin.freeswitch.org/10171
>> >>
>> >> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> >> > Check the password dialog. It will tell you what the
>> >> > username/password
>> >> > is.
>> >> >
>> >> > post the logs for a call as well, please.
>> >> >
>> >> > Regards,
>> >> >
>> >> > jmesquita
>> >> >
>> >> > On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer
>> >> > <e.schmidbauer at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >> Name ? ? ? ? ? ? ? ? ? ?external
>> >> >> Domain Name ? ? ? ? ? ? N/A
>> >> >> DBName ? ? ? ? ? ? ? ? ?sofia_reg_external
>> >> >> Pres Hosts
>> >> >> Dialplan ? ? ? ? ? ? ? ?XML
>> >> >> Context ? ? ? ? ? ? ? ? public
>> >> >> Challenge Realm ? ? ? ? auto_to
>> >> >> RTP-IP ? ? ? ? ? ? ? ? ?192.168.0.125
>> >> >> Ext-RTP-IP ? ? ? ? ? ? ?98.118.151.30
>> >> >> SIP-IP ? ? ? ? ? ? ? ? ?192.168.0.125
>> >> >> Ext-SIP-IP ? ? ? ? ? ? ?98.118.151.30
>> >> >> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.0.125:5080
>> >> >> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.0.125:5080
>> >> >> HOLD-MUSIC ? ? ? ? ? ? ?local_stream://moh
>> >> >> OUTBOUND-PROXY ? ? ? ? ?N/A
>> >> >> CODECS
>> >> >>
>> >> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h@20i,PCMU,PCMA,GSM
>> >> >> TEL-EVENT ? ? ? ? ? ? ? 101
>> >> >> DTMF-MODE ? ? ? ? ? ? ? rfc2833
>> >> >> CNG ? ? ? ? ? ? ? ? ? ? 13
>> >> >> SESSION-TO ? ? ? ? ? ? ?0
>> >> >> MAX-DIALOG ? ? ? ? ? ? ?0
>> >> >> NOMEDIA ? ? ? ? ? ? ? ? false
>> >> >> LATE-NEG ? ? ? ? ? ? ? ?false
>> >> >> PROXY-MEDIA ? ? ? ? ? ? false
>> >> >> AGGRESSIVENAT ? ? ? ? ? false
>> >> >> STUN-ENABLED ? ? ? ? ? ?true
>> >> >> STUN-AUTO-DISABLE ? ? ? false
>> >> >> CALLS-IN ? ? ? ? ? ? ? ?17
>> >> >> FAILED-CALLS-IN ? ? ? ? 11
>> >> >> CALLS-OUT ? ? ? ? ? ? ? 9
>> >> >> FAILED-CALLS-OUT ? ? ? ?9
>> >> >>
>> >> >> Registrations:
>> >> >>
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >> Call-ID: ? ? ? ?MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
>> >> >> User: ? ? ? ? ? 1000 at server1.altpressonline.com
>> >> >> Contact: ? ? ? ?"1000"
>> >> >> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
>> >> >> Agent: ? ? ? ? ?X-Lite release 1103k stamp 53621
>> >> >> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
>> >> >> Host: ? ? ? ? ? server1.altpressonline.com
>> >> >> IP: ? ? ? ? ? ? 69.204.30.67
>> >> >> Port: ? ? ? ? ? 16006
>> >> >> Auth-User: ? ? ?1000
>> >> >> Auth-Realm: ? ? server1.altpressonline.com
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >>
>> >> >> sorry, i don't know how to login to pastebin
>> >> >>
>> >> >> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org>
>> >> >> wrote:
>> >> >> > Could you give a few more details? For example, could you pastebin
>> >> >> > the
>> >> >> > output of "sofia status profile external"?
>> >> >> > -MC
>> >> >> >
>> >> >> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
>> >> >> > <e.schmidbauer at gmail.com>
>> >> >> > wrote:
>> >> >> >>
>> >> >> >> I am unable to call a user outside of our local area network. the
>> >> >> >> user
>> >> >> >> is registered on the external profile but there is no way to call
>> >> >> >> the
>> >> >> >> phone. does anyone have any suggestions how to do this?
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


From msc at freeswitch.org  Mon Aug 31 17:23:34 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 17:23:34 -0700
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <2cef777b0908311706o4173312bw9af8b217aca47392@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
	<2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
	<ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>
	<2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com>
	<ebbc2d550908311637q22ef76aaj6621a5d2345885e8@mail.gmail.com>
	<2cef777b0908311706o4173312bw9af8b217aca47392@mail.gmail.com>
Message-ID: <87f2f3b90908311723v5c41801l5a8c8edcf9bc9904@mail.gmail.com>

What kind of linksys? I had a 54WRT-G that didn't work until I used Tomato
firmware. After that it all worked very well.
-MC

On Mon, Aug 31, 2009 at 5:06 PM, e schmidbauer <e.schmidbauer at gmail.com>wrote:

> i cannot reach the remote endpoint. the remote endpoint can reach a
> locally registered endpoint. any idea why this is happening? the
> remote endpoint is behind a linksys with upnp enabled.
>
> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > Problem is definetly on far end.
> >
> > If you look at the siptrace, you have the following sequence:
> >
> > 1. Asterisk calls in
> > 2. FreeSWITCH replies with a Trying(100) to complete call right away and
> > proceeds to dialplan
> > 3. FreeSWITCH invites (calls) 7 times the final destination that never
> > responds.
> > 4. Asterisk sends a CANCEL message
> >
> > In all that, your final endpoint never responds to any message. Are you
> sure
> > you can reach it?
> >
> > jmesquita
> >
> > On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer <e.schmidbauer at gmail.com>
> > wrote:
> >>
> >> here's the sip trace:
> >> http://pastebin.freeswitch.org/10172
> >>
> >>
> >>
> >> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> >> > 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel
> >> > sofia/external/Anonymous at anonymous.invalid entering state
> >> > [terminated][487]
> >> >
> >> > The far end seems to be replying with 487 - Request Terminated...
> >> >
> >> > Nothing wrong on FS, seems to be a problem with your endpoints. Can
> you
> >> > enable a sip trace?
> >> >
> >> > jmesquita
> >> >
> >> >
> >> > On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <
> e.schmidbauer at gmail.com>
> >> > wrote:
> >> >>
> >> >> thanks...heres the pastebin:
> >> >> http://pastebin.freeswitch.org/10171
> >> >>
> >> >> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> >> >> > Check the password dialog. It will tell you what the
> >> >> > username/password
> >> >> > is.
> >> >> >
> >> >> > post the logs for a call as well, please.
> >> >> >
> >> >> > Regards,
> >> >> >
> >> >> > jmesquita
> >> >> >
> >> >> > On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer
> >> >> > <e.schmidbauer at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >> Name                    external
> >> >> >> Domain Name             N/A
> >> >> >> DBName                  sofia_reg_external
> >> >> >> Pres Hosts
> >> >> >> Dialplan                XML
> >> >> >> Context                 public
> >> >> >> Challenge Realm         auto_to
> >> >> >> RTP-IP                  192.168.0.125
> >> >> >> Ext-RTP-IP              98.118.151.30
> >> >> >> SIP-IP                  192.168.0.125
> >> >> >> Ext-SIP-IP              98.118.151.30
> >> >> >> URL                     sip:mod_sofia at 192.168.0.125:5080
> >> >> >> BIND-URL                sip:mod_sofia at 192.168.0.125:5080
> >> >> >> HOLD-MUSIC              local_stream://moh
> >> >> >> OUTBOUND-PROXY          N/A
> >> >> >> CODECS
> >> >> >>
> >> >> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h
> @20i,PCMU,PCMA,GSM
> >> >> >> TEL-EVENT               101
> >> >> >> DTMF-MODE               rfc2833
> >> >> >> CNG                     13
> >> >> >> SESSION-TO              0
> >> >> >> MAX-DIALOG              0
> >> >> >> NOMEDIA                 false
> >> >> >> LATE-NEG                false
> >> >> >> PROXY-MEDIA             false
> >> >> >> AGGRESSIVENAT           false
> >> >> >> STUN-ENABLED            true
> >> >> >> STUN-AUTO-DISABLE       false
> >> >> >> CALLS-IN                17
> >> >> >> FAILED-CALLS-IN         11
> >> >> >> CALLS-OUT               9
> >> >> >> FAILED-CALLS-OUT        9
> >> >> >>
> >> >> >> Registrations:
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >> Call-ID:        MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
> >> >> >> User:           1000 at server1.altpressonline.com
> >> >> >> Contact:        "1000"
> >> >> >> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
> >> >> >> Agent:          X-Lite release 1103k stamp 53621
> >> >> >> Status:         Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
> >> >> >> Host:           server1.altpressonline.com
> >> >> >> IP:             69.204.30.67
> >> >> >> Port:           16006
> >> >> >> Auth-User:      1000
> >> >> >> Auth-Realm:     server1.altpressonline.com
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >>
> >> >> >> sorry, i don't know how to login to pastebin
> >> >> >>
> >> >> >> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<
> msc at freeswitch.org>
> >> >> >> wrote:
> >> >> >> > Could you give a few more details? For example, could you
> pastebin
> >> >> >> > the
> >> >> >> > output of "sofia status profile external"?
> >> >> >> > -MC
> >> >> >> >
> >> >> >> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
> >> >> >> > <e.schmidbauer at gmail.com>
> >> >> >> > wrote:
> >> >> >> >>
> >> >> >> >> I am unable to call a user outside of our local area network.
> the
> >> >> >> >> user
> >> >> >> >> is registered on the external profile but there is no way to
> call
> >> >> >> >> the
> >> >> >> >> phone. does anyone have any suggestions how to do this?
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jim at evolutiontel.net  Mon Aug 31 18:26:19 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Tue, 1 Sep 2009 11:26:19 +1000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <1251721344.6737.32.camel@dk-d820>
References: <C6C0AC6C.5E412%krice@freeswitch.org>
	<1251721344.6737.32.camel@dk-d820>
Message-ID: <ddad80610908311826v3f0ec02ai195cd7772095c41@mail.gmail.com>

Actually Dave, our idea's are very similar.

The difference being is that you fork the calls into 2 separate legs
at a B2BUA and sending each to an FS box, while I would duplicate
messages to an FS box acting in Standby mode using extra SIP headers
to let the standby machine know the RTP port number and the UUID.
These 2 FS boxes would essentially become the core switching units in
Active/Standby configuration and be used to handle switching logic and
billing.  Thus if FS2 became the Active box, it would have exactly the
same call information as FS1 had.  Theoretically there would nothing
stopping FS1 and FS2 being master and standby for each other at the
same time and both handling traffic, although this has the potential
to cause over load if something did fail.

i.e FS1 receives INVITE, handles call and forwards INVITE to Gateway
and then sends a second INVITE (Duplication of Original message,
utilising extra SIP headers to pass on required information) to FS2.
FS1 fails FS2 takes over and handles call.

Then you would look at way's to distribute functionality across other
boxes to allow scalability.  An example would be to use dedicated FS
boxes for TDM cards and Opensips to handle registrations, SIP proxy or
SBC controllers.

Of course this assumes you are looking to build a softswitch that can
handle large call numbers and have the $$$ to do so.  Sadly, I have
niether the money or the requirement.




On Mon, Aug 31, 2009 at 10:22 PM, David Knell<dave at 3c.co.uk> wrote:
> A random thought. ?It strikes me that adding support for live call
> failover to FreeSWITCH might well be the wrong thing to do - it's
> something which is probably not too difficult to achieve if starting
> from the ground up, but which is going to be a right PITA to do
> retrospectively. ?And, once you've done switching, there's conferencing,
> IVR stuff...
>
> Here's an alternative approach. ?Have two FS boxes (which I'll call FS1
> and FS2 - you can't fault *my* imagination) and a front-end B2BUA/RTP
> proxy with a bit of a difference. ?The B2BUA sends incoming stuff to
> both FS1 and FS2 as two separate call legs, and the RTP proxy forwards
> inbound RTP to both FS1 and FS2. ?Assuming that FS1 is the current
> "master", it forwards RTP and call state transitions from FS1 back to
> the caller. ?If it detects that FS1 has gone away, then - as FS2 should
> be in roughly the same state as FS1 - it flips to using FS2 as its
> source.
>
> The internal state of the B2BUA/RTP proxy should be relatively simple,
> and therefore should be able to be duplicated to a hot standby.
>
> I can see a collection of gotchas, such as registration (the B2BUA needs
> access to the user database), and I haven't even begun to try to think
> of how to deal with calls *from* the FS boxes. ?Actually, that's not
> true: I have, and they clearly need to go through the same sort of
> mechanism in reverse, which is fine until I start trying to work out how
> to pair up calls from FS1 and FS2 reliably, and then my head hurts.
>
> But I think that this idea has merit. ?Firstly, it abstracts the
> failover mechanism away into one place; secondly, it is general - FS1
> and FS2 could just as easily be Asterisk1 and Asterisk2.
>
> And I'll do it for just $98,650..
>
> --Dave
>
>> Your forgetting that with RTP theres the duplication of RTP that may be
>> required along with a mechanism that keeps media timers working properly...
>>
>> Where many things don't support RTP timers they sure come in handy on a
>> regular basis especially dealing with several of the other software based
>> platforms out there that just love to have calls end but never want to send
>> a BYE
>>
>>
>> > From: Jim Burke <jim at evolutiontel.net>
>> > Reply-To: <freeswitch-users at lists.freeswitch.org>
>> > Date: Mon, 31 Aug 2009 12:54:50 +1000
>> > To: <freeswitch-users at lists.freeswitch.org>
>> > Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
>> >
>> > Conceptually in a pure SIP installation of Freeswitch, hot standby or
>> > active standby could be achieved through duplication of messages
>> > (INVITE, 200OK and BYE for calls) between 2 boxes and then using VRRP
>> > to change the IP address when the box falls over. ?Going this way
>> > allows call state to be as up to date as possible. ?Obviously this
>> > would require logic added to sofia to transfer the RTP port
>> > information and UUID information between the FS instances (Easily
>> > achieved using additional SIP headers). ?It would also require that
>> > sofia does not forward duplicated messages to gateways or user agents.
>> > ?CDR information would also need to be generated correctly (My
>> > experience of Radius with Opensips is that it generates start and stop
>> > messages on INVITE/200OK then BYE respectively so this could be an
>> > easy solution to the CDR issue).
>> >
>> > Regards,
>> >
>> >
>> >
>> > On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
>> >> So, the first way would be to have a look on the sip stack, which is, in
>> >> fact, sofia ..., well, sound's like a nice, fun, long-going hobby project to
>> >> me :-)
>> >> I will definitly at least look at the sip code to check if it is something i
>> >> would myself willingly give over to ...
>> >> But I *really* do want ?a setup where it does not matter if one specific FS
>> >> instance is UP or NOT ...
>> >>
>> >> --
>> >> Raimund Sacherer
>> >> -
>> >> RunSolutions
>> >> ? ? Open Source It Consulting
>> >> -
>> >>
>> >> Parc Bit - Centro Empresarial Son Espanyol
>> >> Edificio Estel - Local 3D
>> >> 07121 - ?Palma de Mallorca
>> >> Baleares
>> >> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>> >>
>> >> The sip stack needs to be modified to spin that data up into the state
>> >> machine so that it can take over calls once the fail over takes place... its
>> >> not an easy task.
>> >> /b
>> >> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>> >>
>> >> Is itnpossible to have a db cluster know the state of each and every call
>> >> and then use Heartbeat on this db +
>> >> fs cluster so that clients see only one ip where as internally all fs boxes
>> >> refer db for call states, db again is under replication.
>> >> This in the thioery can be written, but I am sure if we think bit more on
>> >> this direction the problem seem to be getting addressed.
>> >> Other guys also chip in their 2 cents, we just need 50 of em to make a full
>> >> dollar.
>> >>
>> >> Thanks & Regards,
>> >> Mitul Limbani,
>> >> Founder & CEO,
>> >> Enterux Solutions Pvt. Ltd.,
>> >> The Enterprise Linux Company (r),
>> >> http://www.enterux.com
>> >> http://www.entVoice.com
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > Jim Burke
>> > Director Evolutiontel.
>> > http://www.evolutiontel.net
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From jim at evolutiontel.net  Mon Aug 31 18:27:14 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Tue, 1 Sep 2009 11:27:14 +1000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <C99339FA-1E87-4A23-9C25-FB7D564B83CB@runsolutions.com>
References: <C6C0AC6C.5E412%krice@freeswitch.org>
	<1251721344.6737.32.camel@dk-d820>
	<C99339FA-1E87-4A23-9C25-FB7D564B83CB@runsolutions.com>
Message-ID: <ddad80610908311827u538f64f2ra0a1e6b8ce593af6@mail.gmail.com>

> And I'll do it for just $98,650..

Haha, that's really funny :-)

Dave is talking New Zealand Dollars ;)

On Mon, Aug 31, 2009 at 10:36 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
> Hi David,
>
> On Aug 31, 2009, at 2:22 PM, David Knell wrote:
>
>> A random thought. ?It strikes me that adding support for live call
>> failover to FreeSWITCH might well be the wrong thing to do - it's
>> something which is probably not too difficult to achieve if starting
>> from the ground up, but which is going to be a right PITA to do
>> retrospectively. ?And, once you've done switching, there's
>> conferencing,
>> IVR stuff...
>>
>> Here's an alternative approach. ?Have two FS boxes (which I'll call
>> FS1
>> and FS2 - you can't fault *my* imagination) and a front-end B2BUA/RTP
>> proxy with a bit of a difference. ?The B2BUA sends incoming stuff to
>> both FS1 and FS2 as two separate call legs, and the RTP proxy forwards
>> inbound RTP to both FS1 and FS2. ?Assuming that FS1 is the current
>> "master", it forwards RTP and call state transitions from FS1 back to
>> the caller. ?If it detects that FS1 has gone away, then - as FS2
>> should
>> be in roughly the same state as FS1 - it flips to using FS2 as its
>> source.
>>
>> The internal state of the B2BUA/RTP proxy should be relatively simple,
>> and therefore should be able to be duplicated to a hot standby.
>>
>> I can see a collection of gotchas, such as registration (the B2BUA
>> needs
>> access to the user database), and I haven't even begun to try to think
>> of how to deal with calls *from* the FS boxes. ?Actually, that's not
>> true: I have, and they clearly need to go through the same sort of
>> mechanism in reverse, which is fine until I start trying to work out
>> how
>> to pair up calls from FS1 and FS2 reliably, and then my head hurts.
> Yeah, sounds like a real challenge
>
>>
>> But I think that this idea has merit. ?Firstly, it abstracts the
>> failover mechanism away into one place; secondly, it is general - FS1
>> and FS2 could just as easily be Asterisk1 and Asterisk2.
> this, in theory, sound quite interesting ...
> would be nice to have a proof of concept setup thought.
>
> -> Sound's not quite THAT nice than a full-built-in solution to FS ...
> but if it works ...
>
>
>>
>> And I'll do it for just $98,650..
> Haha, that's really funny :-)
>
>
>>
>> --Dave
>>
>>> Your forgetting that with RTP theres the duplication of RTP that
>>> may be
>>> required along with a mechanism that keeps media timers working
>>> properly...
>>>
>>> Where many things don't support RTP timers they sure come in handy
>>> on a
>>> regular basis especially dealing with several of the other software
>>> based
>>> platforms out there that just love to have calls end but never want
>>> to send
>>> a BYE
>>>
>>>
>>>> From: Jim Burke <jim at evolutiontel.net>
>>>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>>>> Date: Mon, 31 Aug 2009 12:54:50 +1000
>>>> To: <freeswitch-users at lists.freeswitch.org>
>>>> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
>>>>
>>>> Conceptually in a pure SIP installation of Freeswitch, hot standby
>>>> or
>>>> active standby could be achieved through duplication of messages
>>>> (INVITE, 200OK and BYE for calls) between 2 boxes and then using
>>>> VRRP
>>>> to change the IP address when the box falls over. ?Going this way
>>>> allows call state to be as up to date as possible. ?Obviously this
>>>> would require logic added to sofia to transfer the RTP port
>>>> information and UUID information between the FS instances (Easily
>>>> achieved using additional SIP headers). ?It would also require that
>>>> sofia does not forward duplicated messages to gateways or user
>>>> agents.
>>>> CDR information would also need to be generated correctly (My
>>>> experience of Radius with Opensips is that it generates start and
>>>> stop
>>>> messages on INVITE/200OK then BYE respectively so this could be an
>>>> easy solution to the CDR issue).
>>>>
>>>> Regards,
>>>>
>>>>
>>>>
>>>> On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com
>>>> > wrote:
>>>>> So, the first way would be to have a look on the sip stack, which
>>>>> is, in
>>>>> fact, sofia ..., well, sound's like a nice, fun, long-going hobby
>>>>> project to
>>>>> me :-)
>>>>> I will definitly at least look at the sip code to check if it is
>>>>> something i
>>>>> would myself willingly give over to ...
>>>>> But I *really* do want ?a setup where it does not matter if one
>>>>> specific FS
>>>>> instance is UP or NOT ...
>>>>>
>>>>> --
>>>>> Raimund Sacherer
>>>>> -
>>>>> RunSolutions
>>>>> ? ?Open Source It Consulting
>>>>> -
>>>>>
>>>>> Parc Bit - Centro Empresarial Son Espanyol
>>>>> Edificio Estel - Local 3D
>>>>> 07121 - ?Palma de Mallorca
>>>>> Baleares
>>>>> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>>>>>
>>>>> The sip stack needs to be modified to spin that data up into the
>>>>> state
>>>>> machine so that it can take over calls once the fail over takes
>>>>> place... its
>>>>> not an easy task.
>>>>> /b
>>>>> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>>>>>
>>>>> Is itnpossible to have a db cluster know the state of each and
>>>>> every call
>>>>> and then use Heartbeat on this db +
>>>>> fs cluster so that clients see only one ip where as internally
>>>>> all fs boxes
>>>>> refer db for call states, db again is under replication.
>>>>> This in the thioery can be written, but I am sure if we think bit
>>>>> more on
>>>>> this direction the problem seem to be getting addressed.
>>>>> Other guys also chip in their 2 cents, we just need 50 of em to
>>>>> make a full
>>>>> dollar.
>>>>>
>>>>> Thanks & Regards,
>>>>> Mitul Limbani,
>>>>> Founder & CEO,
>>>>> Enterux Solutions Pvt. Ltd.,
>>>>> The Enterprise Linux Company (r),
>>>>> http://www.enterux.com
>>>>> http://www.entVoice.com
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>>> freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>>> freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Jim Burke
>>>> Director Evolutiontel.
>>>> http://www.evolutiontel.net
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>> --
>> David Knell, Director, 3C Limited
>> T: +44 20 3298 2000
>> E: dave at 3c.co.uk
>> W: http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From jim at evolutiontel.net  Mon Aug 31 18:40:13 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Tue, 1 Sep 2009 11:40:13 +1000
Subject: [Freeswitch-users] Load Balacing
In-Reply-To: <20090831215728.GD20919@hijacked.us>
References: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>
	<20090831215728.GD20919@hijacked.us>
Message-ID: <ddad80610908311840u5bae139q38496c49507b897a@mail.gmail.com>

Could you use the server host saved in the database at registration
and then use it with path to get to the other FS box?

http://wiki.freeswitch.org/wiki/Sofia#Specifying_SIP_Proxy_With_fs_path

On Tue, Sep 1, 2009 at 7:57 AM, Andrew Thompson<andrew at hijacked.us> wrote:
> On Mon, Aug 31, 2009 at 04:56:48PM -0300, Juan Manuel Vicente wrote:
>> Hello:
>>
>> I am using two boxes, with same domain, to register about 1000 users working
>> together a DNS box doing round robin to resolve two differentes Ips one to
>> each box. This two boxes are using the same database (through unixodbc).
>>
>> When I use the funcion sofia_contact or sofia profile status both boxes show
>> the status of the user. Does not matter where the resgistration was did.
>>
>> But when I try to call the user this call fails, the call only works if the
>> call is sent through the same box where the user is registered.
>>
>
> I had the same problem except that I was using the multicast module to
> replicate registrations, I added a sofia config parameter to optionally
> rewrite the fs_path part of the sip URI on replicated packets so I could
> hairpin the sip messages through the original registrar. This doesn't
> help you directly but maybe it'll give you an idea?
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net


From se02005-mec at yahoo.com  Mon Aug 31 19:53:12 2009
From: se02005-mec at yahoo.com (Merle J. Ebbert)
Date: Mon, 31 Aug 2009 19:53:12 -0700
Subject: [Freeswitch-users] Can an Instant Message (IM) be sent to a cell
 phone with FS under Windows?
Message-ID: <4A9C8C98.3020205@yahoo.com>

An HTML attachment was scrubbed...
URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090831/b994ec5c/attachment.html 

From help at pdscc.com  Mon Aug 31 22:43:44 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Mon, 31 Aug 2009 22:43:44 -0700
Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3
In-Reply-To: <20090823185152.D17845FE@sinclaire.sibble.net>
References: <20090823185152.D17845FE@sinclaire.sibble.net>
Message-ID: <20090901054337.7317F65B@sinclaire.sibble.net>

Haven't had any responses, anyone have any ideas on the problem with 
compiling the ESL modules as below?

On 23 Aug 2009 at 11:51, Harondel J. Sibble wrote:

> Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server, 
> then went to install FreePBX v3, I've gotten all the prerequisities in the
> wizard fixed except for ESL
> 
> As per
> 
> http://wiki.freeswitch.org/wiki/Event_Socket_Library
> http://wiki.freeswitch.org/wiki/Event_Socket
> 
> I go into my FS source dir
> 
> /home/sibbleh/freeswitch-1.0.4/libs/esl
> 
> Run make and then "sudo make phpmod-install" 
> 
> and I get 
> 
> 
> $ sudo make phpmod-install
> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-
> I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> -ggdb 
> -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-
> variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
> CXXFLAGS="-
> I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> -ggdb 
> -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" 
> CXX_CFLAGS="" -C php
> make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> g++  -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE
> -g 
> -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -
> I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -
> I/usr/include/php5/Zend -I/usr/include/php5/ext -
> I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64
> -
> Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> cc1plus: warnings being treated as errors
> esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1047: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1073: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:1111: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:1141: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1172: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1198: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1234: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1269: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1294: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**,
> zval*, 
> int)':
> esl_wrap.cpp:1346: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1403: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1441: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1478: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1508: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1538: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1571: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_addHeader(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1611: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_delHeader(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1644: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_firstHeader(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1674: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_nextHeader(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1704: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_0(int, zval*,
> zval**, zval*, int)':
> esl_wrap.cpp:1744: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_1(int, zval*,
> zval**, zval*, int)':
> esl_wrap.cpp:1770: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLconnection(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1803: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_socketDescriptor(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:1846: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_connected(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1872: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_getInfo(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1898: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_send(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1931: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendRecv(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1964: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_api(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:2007: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_bgapi(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:2050: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendEvent(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2082: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEvent(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2108: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEventTimed(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:2141: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_filter(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2181: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_events(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2221: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_execute(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2272: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_executeAsync(int, zval*,
> zval**, zval*, int)':
> esl_wrap.cpp:2323: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_setAsyncExecute(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:2356: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_setEventLock(int, zval*,
> zval**, zval*, int)':
> esl_wrap.cpp:2389: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_disconnect(int, zval*,
> zval**, zval*, int)':
> esl_wrap.cpp:2415: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_eslSetLogLevel(int, zval*, zval**, 
> zval*, int)':
> esl_wrap.cpp:2438: error: format not a string literal and no format
> arguments
> make[1]: *** [esl_wrap.o] Error 1
> make[1]: Leaving directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> make: *** [phpmod] Error 2
> 
> Same thing happens if I try sudo make everymod
> 
> Checking the list archives I found this thread
> 
> http://www.nabble.com/ESL-Wrapper-td22209991.html#a22222338
> 
> I've made sure that the php-dev packages are installed. Any suggestions on
> what to do next?
> -- 

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)



From tculjaga at gmail.com  Mon Aug 31 23:09:00 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 1 Sep 2009 08:09:00 +0200
Subject: [Freeswitch-users] mod_opal
In-Reply-To: <87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com>
References: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>
	<87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com>
Message-ID: <65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>

hhmmm :))

is there any doc following up mod_opal ?
I really don't want to waste your time :)

T.


On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> hello,
>>
>> i'm trying to build mod_opal and getting this error:
>>
>>
>>
>> making all mod_logfile
>>
>> making all mod_loopback
>>
>> making all mod_native_file
>>
>> making all mod_opal
>> Compiling mod_opal.cpp...
>> quiet_libtool: compile:  g++ -g -ggdb -I.
>> -I/home/tculjaga/freeswitch-trunk/src/include
>> -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
>> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
>> -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal
>> -DHAVE_CONFIG_H -c mod_opal.cpp  -fPIC -DPIC -o .libs/mod_opal.o
>> In file included from mod_opal.cpp:25:
>> mod_opal.h:151: error: conflicting return type specified for ?virtual
>> OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)?
>> /usr/include/opal/opal/localep.h:267: error:   overriding ?virtual
>> ptlib_virtual_function_changed_or_removed******
>> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)?
>> mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&,
>> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
>> switch_channel_t*)?:
>> mod_opal.cpp:564: error: no matching function for call to
>> ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)?
>> /usr/include/opal/opal/localep.h:290: note: candidates are:
>> OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&,
>> void*, unsigned int, OpalConnection::StringOptions*, char)
>> /usr/include/opal/opal/localep.h:276: note:
>> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
>> /usr/include/opal/opal/patch.h: In member function ?switch_status_t
>> FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)?:
>> /usr/include/opal/opal/patch.h:272: error: ?virtual
>> ptlib_virtual_function_changed_or_removed******
>> OpalMediaPatch::OnPatchStart()? is private
>> mod_opal.cpp:1277: error: within this context
>> mod_opal.cpp:1277: warning: ?virtual
>> ptlib_virtual_function_changed_or_removed******
>> OpalMediaPatch::OnPatchStart()? is deprecated (declared at
>> /usr/include/opal/opal/patch.h:272)
>> mod_opal.cpp:1277: warning: ignoring return value of function declared
>> with attribute warn_unused_result
>> /usr/include/opal/opal/patch.h: In member function ?switch_status_t
>> FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)?:
>> /usr/include/opal/opal/patch.h:272: error: ?virtual
>> ptlib_virtual_function_changed_or_removed******
>> OpalMediaPatch::OnPatchStart()? is private
>> mod_opal.cpp:1399: error: within this context
>> mod_opal.cpp:1399: warning: ?virtual
>> ptlib_virtual_function_changed_or_removed******
>> OpalMediaPatch::OnPatchStart()? is deprecated (declared at
>> /usr/include/opal/opal/patch.h:272)
>> mod_opal.cpp:1399: warning: ignoring return value of function declared
>> with attribute warn_unused_result
>> make[5]: *** [mod_opal.lo] Error 1
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_opal-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> Making all in build
>>  +-------- FreeSWITCH Build Complete -----------+
>>  + FreeSWITCH has been successfully built.      +
>>  + Install by running:                          +
>>  +                                              +
>>  +               make install                   +
>>  +----------------------------------------------+
>> make[1]: *** [all-recursive] Error 1
>> make: *** [all] Error 2
>> tculjaga at nemesis:~/freeswitch-trunk$
>> tculjaga at nemesis:~/freeswitch-trunk$
>> tculjaga at nemesis:~/freeswitch-trunk$
>>
>>
>>
>> what ptlib/opal/fs version did you use to build it?
>>
>>
>> I tried with trunk (ptlib, opal, fs)... and as you can see :)
>>
>
> Did you run the buildopal.sh script in src/build ?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon Aug 31 23:29:24 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 1 Sep 2009 08:29:24 +0200
Subject: [Freeswitch-users] mod_opal
In-Reply-To: <65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>
References: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>
	<87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com>
	<65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>
Message-ID: <65d96fc80908312329l7fb63744q97adaa5ea7d18997@mail.gmail.com>

hi, It went well

obviously FS needs v3_6 opal :)

thx.



On Tue, Sep 1, 2009 at 8:09 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hhmmm :))
>
> is there any doc following up mod_opal ?
> I really don't want to waste your time :)
>
> T.
>
>
> On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> hello,
>>>
>>> i'm trying to build mod_opal and getting this error:
>>>
>>>
>>>
>>> making all mod_logfile
>>>
>>> making all mod_loopback
>>>
>>> making all mod_native_file
>>>
>>> making all mod_opal
>>> Compiling mod_opal.cpp...
>>> quiet_libtool: compile:  g++ -g -ggdb -I.
>>> -I/home/tculjaga/freeswitch-trunk/src/include
>>> -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
>>> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
>>> -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal
>>> -DHAVE_CONFIG_H -c mod_opal.cpp  -fPIC -DPIC -o .libs/mod_opal.o
>>> In file included from mod_opal.cpp:25:
>>> mod_opal.h:151: error: conflicting return type specified for ?virtual
>>> OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)?
>>> /usr/include/opal/opal/localep.h:267: error:   overriding ?virtual
>>> ptlib_virtual_function_changed_or_removed******
>>> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)?
>>> mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&,
>>> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
>>> switch_channel_t*)?:
>>> mod_opal.cpp:564: error: no matching function for call to
>>> ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)?
>>> /usr/include/opal/opal/localep.h:290: note: candidates are:
>>> OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&,
>>> void*, unsigned int, OpalConnection::StringOptions*, char)
>>> /usr/include/opal/opal/localep.h:276: note:
>>> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
>>> /usr/include/opal/opal/patch.h: In member function ?switch_status_t
>>> FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)?:
>>> /usr/include/opal/opal/patch.h:272: error: ?virtual
>>> ptlib_virtual_function_changed_or_removed******
>>> OpalMediaPatch::OnPatchStart()? is private
>>> mod_opal.cpp:1277: error: within this context
>>> mod_opal.cpp:1277: warning: ?virtual
>>> ptlib_virtual_function_changed_or_removed******
>>> OpalMediaPatch::OnPatchStart()? is deprecated (declared at
>>> /usr/include/opal/opal/patch.h:272)
>>> mod_opal.cpp:1277: warning: ignoring return value of function declared
>>> with attribute warn_unused_result
>>> /usr/include/opal/opal/patch.h: In member function ?switch_status_t
>>> FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)?:
>>> /usr/include/opal/opal/patch.h:272: error: ?virtual
>>> ptlib_virtual_function_changed_or_removed******
>>> OpalMediaPatch::OnPatchStart()? is private
>>> mod_opal.cpp:1399: error: within this context
>>> mod_opal.cpp:1399: warning: ?virtual
>>> ptlib_virtual_function_changed_or_removed******
>>> OpalMediaPatch::OnPatchStart()? is deprecated (declared at
>>> /usr/include/opal/opal/patch.h:272)
>>> mod_opal.cpp:1399: warning: ignoring return value of function declared
>>> with attribute warn_unused_result
>>> make[5]: *** [mod_opal.lo] Error 1
>>> make[4]: *** [all] Error 1
>>> make[3]: *** [mod_opal-all] Error 1
>>> make[2]: *** [all-recursive] Error 1
>>> Making all in build
>>>  +-------- FreeSWITCH Build Complete -----------+
>>>  + FreeSWITCH has been successfully built.      +
>>>  + Install by running:                          +
>>>  +                                              +
>>>  +               make install                   +
>>>  +----------------------------------------------+
>>> make[1]: *** [all-recursive] Error 1
>>> make: *** [all] Error 2
>>> tculjaga at nemesis:~/freeswitch-trunk$
>>> tculjaga at nemesis:~/freeswitch-trunk$
>>> tculjaga at nemesis:~/freeswitch-trunk$
>>>
>>>
>>>
>>> what ptlib/opal/fs version did you use to build it?
>>>
>>>
>>> I tried with trunk (ptlib, opal, fs)... and as you can see :)
>>>
>>
>> Did you run the buildopal.sh script in src/build ?
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Mon Aug 31 23:29:28 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 23:29:28 -0700
Subject: [Freeswitch-users] mod_opal
In-Reply-To: <65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>
References: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>
	<87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com>
	<65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>
Message-ID: <87f2f3b90908312329g17b642adj9c0073a863bf28d7@mail.gmail.com>

On Mon, Aug 31, 2009 at 11:09 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> hhmmm :))
>
> is there any doc following up mod_opal ?
> I really don't want to waste your time :)
>
> T.
>
Last I heard this was it:
http://jira.freeswitch.org/browse/MODOPAL-10
-MC
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From tculjaga at gmail.com  Mon Aug 31 23:32:00 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 1 Sep 2009 08:32:00 +0200
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <2cef777b0908311706o4173312bw9af8b217aca47392@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
	<2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
	<ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>
	<2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com>
	<ebbc2d550908311637q22ef76aaj6621a5d2345885e8@mail.gmail.com>
	<2cef777b0908311706o4173312bw9af8b217aca47392@mail.gmail.com>
Message-ID: <65d96fc80908312332h48c92eden4cdd28ae2f66e4f7@mail.gmail.com>

oh good, on remote router/dsl modem (whatever doing NAT) never use upnp,
never use ALG, just do a simple NAT and it is alway gonna work!

T.

On Tue, Sep 1, 2009 at 2:06 AM, e schmidbauer <e.schmidbauer at gmail.com>wrote:

> i cannot reach the remote endpoint. the remote endpoint can reach a
> locally registered endpoint. any idea why this is happening? the
> remote endpoint is behind a linksys with upnp enabled.
>
> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > Problem is definetly on far end.
> >
> > If you look at the siptrace, you have the following sequence:
> >
> > 1. Asterisk calls in
> > 2. FreeSWITCH replies with a Trying(100) to complete call right away and
> > proceeds to dialplan
> > 3. FreeSWITCH invites (calls) 7 times the final destination that never
> > responds.
> > 4. Asterisk sends a CANCEL message
> >
> > In all that, your final endpoint never responds to any message. Are you
> sure
> > you can reach it?
> >
> > jmesquita
> >
> > On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer <e.schmidbauer at gmail.com>
> > wrote:
> >>
> >> here's the sip trace:
> >> http://pastebin.freeswitch.org/10172
> >>
> >>
> >>
> >> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> >> > 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel
> >> > sofia/external/Anonymous at anonymous.invalid entering state
> >> > [terminated][487]
> >> >
> >> > The far end seems to be replying with 487 - Request Terminated...
> >> >
> >> > Nothing wrong on FS, seems to be a problem with your endpoints. Can
> you
> >> > enable a sip trace?
> >> >
> >> > jmesquita
> >> >
> >> >
> >> > On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <
> e.schmidbauer at gmail.com>
> >> > wrote:
> >> >>
> >> >> thanks...heres the pastebin:
> >> >> http://pastebin.freeswitch.org/10171
> >> >>
> >> >> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> >> >> > Check the password dialog. It will tell you what the
> >> >> > username/password
> >> >> > is.
> >> >> >
> >> >> > post the logs for a call as well, please.
> >> >> >
> >> >> > Regards,
> >> >> >
> >> >> > jmesquita
> >> >> >
> >> >> > On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer
> >> >> > <e.schmidbauer at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >> Name                    external
> >> >> >> Domain Name             N/A
> >> >> >> DBName                  sofia_reg_external
> >> >> >> Pres Hosts
> >> >> >> Dialplan                XML
> >> >> >> Context                 public
> >> >> >> Challenge Realm         auto_to
> >> >> >> RTP-IP                  192.168.0.125
> >> >> >> Ext-RTP-IP              98.118.151.30
> >> >> >> SIP-IP                  192.168.0.125
> >> >> >> Ext-SIP-IP              98.118.151.30
> >> >> >> URL                     sip:mod_sofia at 192.168.0.125:5080
> >> >> >> BIND-URL                sip:mod_sofia at 192.168.0.125:5080
> >> >> >> HOLD-MUSIC              local_stream://moh
> >> >> >> OUTBOUND-PROXY          N/A
> >> >> >> CODECS
> >> >> >>
> >> >> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h
> @20i,PCMU,PCMA,GSM
> >> >> >> TEL-EVENT               101
> >> >> >> DTMF-MODE               rfc2833
> >> >> >> CNG                     13
> >> >> >> SESSION-TO              0
> >> >> >> MAX-DIALOG              0
> >> >> >> NOMEDIA                 false
> >> >> >> LATE-NEG                false
> >> >> >> PROXY-MEDIA             false
> >> >> >> AGGRESSIVENAT           false
> >> >> >> STUN-ENABLED            true
> >> >> >> STUN-AUTO-DISABLE       false
> >> >> >> CALLS-IN                17
> >> >> >> FAILED-CALLS-IN         11
> >> >> >> CALLS-OUT               9
> >> >> >> FAILED-CALLS-OUT        9
> >> >> >>
> >> >> >> Registrations:
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >> Call-ID:        MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
> >> >> >> User:           1000 at server1.altpressonline.com
> >> >> >> Contact:        "1000"
> >> >> >> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
> >> >> >> Agent:          X-Lite release 1103k stamp 53621
> >> >> >> Status:         Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
> >> >> >> Host:           server1.altpressonline.com
> >> >> >> IP:             69.204.30.67
> >> >> >> Port:           16006
> >> >> >> Auth-User:      1000
> >> >> >> Auth-Realm:     server1.altpressonline.com
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >>
> >> >> >> sorry, i don't know how to login to pastebin
> >> >> >>
> >> >> >> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<
> msc at freeswitch.org>
> >> >> >> wrote:
> >> >> >> > Could you give a few more details? For example, could you
> pastebin
> >> >> >> > the
> >> >> >> > output of "sofia status profile external"?
> >> >> >> > -MC
> >> >> >> >
> >> >> >> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
> >> >> >> > <e.schmidbauer at gmail.com>
> >> >> >> > wrote:
> >> >> >> >>
> >> >> >> >> I am unable to call a user outside of our local area network.
> the
> >> >> >> >> user
> >> >> >> >> is registered on the external profile but there is no way to
> call
> >> >> >> >> the
> >> >> >> >> phone. does anyone have any suggestions how to do this?
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From msc at freeswitch.org  Mon Aug 31 23:33:50 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 23:33:50 -0700
Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3
In-Reply-To: <20090901054337.7317F65B@sinclaire.sibble.net>
References: <20090823185152.D17845FE@sinclaire.sibble.net>
	<20090901054337.7317F65B@sinclaire.sibble.net>
Message-ID: <87f2f3b90908312333s5bf47d6et94ec24fbc51d6d24@mail.gmail.com>

Did the simple "make" in the libs/esl directory run properly? Just curious.
I'll have to defer to the Ubuntu gurus out there for thoughts on what else
could be wrong.
-MC

On Mon, Aug 31, 2009 at 10:43 PM, Harondel J. Sibble <help at pdscc.com> wrote:

> Haven't had any responses, anyone have any ideas on the problem with
> compiling the ESL modules as below?
>
> On 23 Aug 2009 at 11:51, Harondel J. Sibble wrote:
>
> > Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server,
> > then went to install FreePBX v3, I've gotten all the prerequisities in
> the
> > wizard fixed except for ESL
> >
> > As per
> >
> > http://wiki.freeswitch.org/wiki/Event_Socket_Library
> > http://wiki.freeswitch.org/wiki/Event_Socket
> >
> > I go into my FS source dir
> >
> > /home/sibbleh/freeswitch-1.0.4/libs/esl
> >
> > Run make and then "sudo make phpmod-install"
> >
> > and I get
> >
> >
> > $ sudo make phpmod-install
> > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-
> > I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> > -ggdb
> > -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-
> > variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
> > CXXFLAGS="-
> > I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> > -ggdb
> > -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable"
> > CXX_CFLAGS="" -C php
> > make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> > g++  -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include
> -DHAVE_EDITLINE
> > -g
> > -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable
> -
> > I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -
> > I/usr/include/php5/Zend -I/usr/include/php5/ext -
> > I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE
> -D_FILE_OFFSET_BITS=64
> > -
> > Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> > cc1plus: warnings being treated as errors
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1047: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1073: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:1111: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:1141: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1172: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1198: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1234: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1269: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1294: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**,
> > zval*,
> > int)':
> > esl_wrap.cpp:1346: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1403: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1441: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1478: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1508: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1538: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1571: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_addHeader(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1611: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_delHeader(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1644: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_firstHeader(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1674: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_nextHeader(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1704: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_0(int,
> zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1744: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_1(int,
> zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1770: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1803: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_socketDescriptor(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:1846: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_connected(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1872: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_getInfo(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1898: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_send(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1931: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendRecv(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1964: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_api(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:2007: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_bgapi(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:2050: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendEvent(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2082: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEvent(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2108: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEventTimed(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:2141: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_filter(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2181: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_events(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2221: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_execute(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2272: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_executeAsync(int,
> zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2323: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_setAsyncExecute(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:2356: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_setEventLock(int,
> zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2389: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_disconnect(int,
> zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2415: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_eslSetLogLevel(int, zval*, zval**,
> > zval*, int)':
> > esl_wrap.cpp:2438: error: format not a string literal and no format
> > arguments
> > make[1]: *** [esl_wrap.o] Error 1
> > make[1]: Leaving directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> > make: *** [phpmod] Error 2
> >
> > Same thing happens if I try sudo make everymod
> >
> > Checking the list archives I found this thread
> >
> > http://www.nabble.com/ESL-Wrapper-td22209991.html#a22222338
> >
> > I've made sure that the php-dev packages are installed. Any suggestions
> on
> > what to do next?
> > --
>
> --
> Harondel J. Sibble
> Sibble Computer Consulting
> Creating Solutions for the small and medium business computer user.
> help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
> (604) 739-3709 (voice)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Mon Aug 31 23:37:41 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 1 Sep 2009 08:37:41 +0200
Subject: [Freeswitch-users] mod_opal
In-Reply-To: <65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>
References: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>
	<87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com>
	<65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C52ED0F4743@cooper>

Please look into MODOPAL-10 in jira. You need to apply a patch if you're using latest opal trunk, ro else you need to use the latest stable version of opal. However, I'm not sure how automated this is in the build process in Linux. I've only done this on Windows builds lately.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Tihomir Culjaga
Skickat: den 1 september 2009 08:09
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] mod_opal

hhmmm :))

is there any doc following up mod_opal ?
I really don't want to waste your time :)

T.

On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:

On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga <tculjaga at gmail.com<mailto:tculjaga at gmail.com>> wrote:
hello,

i'm trying to build mod_opal and getting this error:



making all mod_logfile

making all mod_loopback

making all mod_native_file

making all mod_opal
Compiling mod_opal.cpp...
quiet_libtool: compile:  g++ -g -ggdb -I. -I/home/tculjaga/freeswitch-trunk/src/include -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal -DHAVE_CONFIG_H -c mod_opal.cpp  -fPIC -DPIC -o .libs/mod_opal.o
In file included from mod_opal.cpp:25:
mod_opal.h:151: error: conflicting return type specified for 'virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)'
/usr/include/opal/opal/localep.h:267: error:   overriding 'virtual ptlib_virtual_function_changed_or_removed****** OpalLocalEndPoint::CreateConnection(OpalCall&, void*)'
mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall&, FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)':
mod_opal.cpp:564: error: no matching function for call to 'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)'
/usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&, void*, unsigned int, OpalConnection::StringOptions*, char)
/usr/include/opal/opal/localep.h:276: note:                 OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
/usr/include/opal/opal/patch.h: In member function 'switch_status_t FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)':
/usr/include/opal/opal/patch.h:272: error: 'virtual ptlib_virtual_function_changed_or_removed****** OpalMediaPatch::OnPatchStart()' is private
mod_opal.cpp:1277: error: within this context
mod_opal.cpp:1277: warning: 'virtual ptlib_virtual_function_changed_or_removed****** OpalMediaPatch::OnPatchStart()' is deprecated (declared at /usr/include/opal/opal/patch.h:272)
mod_opal.cpp:1277: warning: ignoring return value of function declared with attribute warn_unused_result
/usr/include/opal/opal/patch.h: In member function 'switch_status_t FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)':
/usr/include/opal/opal/patch.h:272: error: 'virtual ptlib_virtual_function_changed_or_removed****** OpalMediaPatch::OnPatchStart()' is private
mod_opal.cpp:1399: error: within this context
mod_opal.cpp:1399: warning: 'virtual ptlib_virtual_function_changed_or_removed****** OpalMediaPatch::OnPatchStart()' is deprecated (declared at /usr/include/opal/opal/patch.h:272)
mod_opal.cpp:1399: warning: ignoring return value of function declared with attribute warn_unused_result
make[5]: *** [mod_opal.lo] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_opal-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
tculjaga at nemesis:~/freeswitch-trunk$
tculjaga at nemesis:~/freeswitch-trunk$
tculjaga at nemesis:~/freeswitch-trunk$



what ptlib/opal/fs version did you use to build it?


I tried with trunk (ptlib, opal, fs)... and as you can see :)

Did you run the buildopal.sh script in src/build ?
-MC


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From thangappan143 at gmail.com  Sat Aug  1 00:18:12 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Sat, 1 Aug 2009 12:48:12 +0530
Subject: [Freeswitch-users] ODBC problem
Message-ID: <7aa29e790908010018k71502d49ob69665a2d48b2047@mail.gmail.com>

While installing mod_lcr I got the following problem.
freeswitch at debian> load mod_lcr
API CALL [load(mod_lcr)] output:
-ERR [module load file routine returned an error]

freeswitch at debian> 2009-08-01 18:12:15 [INFO] mod_lcr.c:522
lcr_load_config() odbc_dsn is 192.168.1.222:freeswitch:freeswitch
2009-08-01 18:12:15 [INFO] mod_lcr.c:536 lcr_load_config() dsn is
"192.168.1.222", user is "freeswitch", and password is "freeswitch"
2009-08-01 18:12:15 [ERR] switch_odbc.c:164 switch_odbc_handle_connect()
STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not
found, and no default driver specified

2009-08-01 18:12:15 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot Open ODBC
Database!
2009-08-01 18:12:15 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to load lcr
config file
2009-08-01 18:12:15 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_lcr.so
**Module load routine returned an error**

What could the error?
How can I resolve it?

-- 
Regards,
Thangappan.M
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From dome at tel.co.th  Sat Aug  1 01:17:42 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 1 Aug 2009 15:17:42 +0700
Subject: [Freeswitch-users] ODBC problem
In-Reply-To: <7aa29e790908010018k71502d49ob69665a2d48b2047@mail.gmail.com>
References: <7aa29e790908010018k71502d49ob69665a2d48b2047@mail.gmail.com>
Message-ID: <8ccbff060908010117j6e2fe01ale4b72a12e70203a8@mail.gmail.com>

please check
/etc/odbc.ini
/opdbinst.ini

Dome C.

2009/8/1 Thangappan.M <thangappan143 at gmail.com>:
> While installing mod_lcr I got the following problem.
> freeswitch at debian> load mod_lcr
> API CALL [load(mod_lcr)] output:
> -ERR [module load file routine returned an error]
>
> freeswitch at debian> 2009-08-01 18:12:15 [INFO] mod_lcr.c:522
> lcr_load_config() odbc_dsn is 192.168.1.222:freeswitch:freeswitch
> 2009-08-01 18:12:15 [INFO] mod_lcr.c:536 lcr_load_config() dsn is
> "192.168.1.222", user is "freeswitch", and password is "freeswitch"
> 2009-08-01 18:12:15 [ERR] switch_odbc.c:164 switch_odbc_handle_connect()
> STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not
> found, and no default driver specified
>
> 2009-08-01 18:12:15 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot Open ODBC
> Database!
> 2009-08-01 18:12:15 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to load lcr
> config file
> 2009-08-01 18:12:15 [CRIT] switch_loadable_module.c:839
> switch_loadable_module_load_file() Error Loading module
> /usr/local/freeswitch/mod/mod_lcr.so
> **Module load routine returned an error**
>
> What could the error?
> How can I resolve it?
>
> --
> Regards,
> Thangappan.M
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From freeswitch at peely.com  Sat Aug  1 03:36:29 2009
From: freeswitch at peely.com (peely)
Date: Sat, 1 Aug 2009 03:36:29 -0700 (PDT)
Subject: [Freeswitch-users] LUA: Independent control of each call leg.
In-Reply-To: <87f2f3b90907311153n23a6a9e6h838206f281bc7d07@mail.gmail.com>
References: <24744087.post@talk.nabble.com>
	<87f2f3b90907311153n23a6a9e6h838206f281bc7d07@mail.gmail.com>
Message-ID: <24767998.post@talk.nabble.com>


Hi,

Thanks for your response. It's a real shame I can't get the async behavior I
want from Freeswitch/LUA as this is exactly the kind of abstraction layer I
hoped for in a SIP Application Server.

Most of the apps I want to develop would be served using the LUA environment
as-is but if a few scenarios I want to be able to perform a small amount of
activity whilst both legs are in a connected state, a prime example is a
b-leg "whisper" where you are still playing "ringing" to the a-party while
the b-party answers and hears a message just before connection to the
a-party.

Regards,


Neil.



mercutioviz wrote:
> 
> The level of control you need really isn't served by doing scripting from
> the dialplan. I highly recommend using ESL and the event socket. It will
> mean a bit of a paradigm shift in your coding, but with that shift comes a
> lot of power and control over what you can do with the calls - really
> limited only by your imagination.
> 
> -MC
> 
> On Fri, Jul 31, 2009 at 8:01 AM, peely <freeswitch at peely.com> wrote:
> 
>>
>> Hi,
>>
>> I'm trying to develop an application using lua and need to control the
>> inbound and outbound legs independently, even when they are switched
>> together.
>>
>> I can initiate the outbound session but I can't seem to bridge without
>> losing control of the script.
>>
>> Does anyone know a way I can allow ingress to egress calling whilst still
>> maintaining script control mid-call? I also need to ingress to hear
>> provisional speech during outbound connect. I've looked at conferencing
>> but
>> there seems to be quite a lot of automated messaging.
>>
>>
>>
>> Thanks,
>>
>>
>> Neil.
>>
> 
> 
> 

-- 
View this message in context: http://www.nabble.com/LUA%3A-Independent-control-of-each-call-leg.-tp24744087p24767998.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From krice at freeswitch.org  Sat Aug  1 04:52:16 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 01 Aug 2009 06:52:16 -0500
Subject: [Freeswitch-users] Connect to PostgreSQL database
In-Reply-To: <7aa29e790907312331v56770633m188a88aef5d4a9cc@mail.gmail.com>
Message-ID: <C69996A0.5A439%krice@freeswitch.org>

You need to look at using ODBC... That is what pretty much everything
uses... Connect FS to ODBC and ODBC to pgsql...



From: "Thangappan.M" <thangappan143 at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Sat, 1 Aug 2009 12:01:36 +0530
To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Connect to PostgreSQL database

Dear all,

?I installed postgresql database in my machine. So now I need to connect the
database from freeswitch.When I searched about the site.They told , to load
the mod_lcr module.

I followed the following steps.
? * Edit the modules.conf file and uncommented the applications/mod_lcr line
?* make mod_lcr-install
?* edit /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml and
uncomment the mod_lcr line.
?* Reload the freeswitch


Where I made a mistake?
Tell the steps to connect the postreSQL database.

-- 
Regards,
Thangappan.M


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From brian at freeswitch.org  Sat Aug  1 09:04:54 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 1 Aug 2009 11:04:54 -0500
Subject: [Freeswitch-users] LUA: Independent control of each call leg.
In-Reply-To: <24767998.post@talk.nabble.com>
References: <24744087.post@talk.nabble.com>
	<87f2f3b90907311153n23a6a9e6h838206f281bc7d07@mail.gmail.com>
	<24767998.post@talk.nabble.com>
Message-ID: <BCC8B127-B1A6-4AC6-B729-D9E389DBC0C9@freeswitch.org>

You can do all this via ESL-Lua cd libs/esl; make luamod

/b

On Aug 1, 2009, at 5:36 AM, peely wrote:

>
> Hi,
>
> Thanks for your response. It's a real shame I can't get the async  
> behavior I
> want from Freeswitch/LUA as this is exactly the kind of abstraction  
> layer I
> hoped for in a SIP Application Server.
>
> Most of the apps I want to develop would be served using the LUA  
> environment
> as-is but if a few scenarios I want to be able to perform a small  
> amount of
> activity whilst both legs are in a connected state, a prime example  
> is a
> b-leg "whisper" where you are still playing "ringing" to the a-party  
> while
> the b-party answers and hears a message just before connection to the
> a-party.
>
> Regards,
>
>
> Neil.




From gshfreesw at gmail.com  Sat Aug  1 09:35:09 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Sat, 1 Aug 2009 12:35:09 -0400
Subject: [Freeswitch-users] Connect to PostgreSQL database
In-Reply-To: <C69996A0.5A439%krice@freeswitch.org>
References: <7aa29e790907312331v56770633m188a88aef5d4a9cc@mail.gmail.com>
	<C69996A0.5A439%krice@freeswitch.org>
Message-ID: <5070fcbd0908010935n2d481397h210ce38057f03933@mail.gmail.com>

Thangappa,

Are you using Perl Event Socket? In that use, you can use Perl DBI to
connect to any DB instead of using the FS's ODBC module.

On Sat, Aug 1, 2009 at 7:52 AM, Ken Rice <krice at freeswitch.org> wrote:

>  You need to look at using ODBC... That is what pretty much everything
> uses... Connect FS to ODBC and ODBC to pgsql...
>
>
> ------------------------------
> *From: *"Thangappan.M" <thangappan143 at gmail.com>
> *Reply-To: *<freeswitch-users at lists.freeswitch.org>
> *Date: *Sat, 1 Aug 2009 12:01:36 +0530
> *To: *freeswitch-users <freeswitch-users at lists.freeswitch.org>
> *Subject: *[Freeswitch-users] Connect to PostgreSQL database
>
>
> Dear all,
>
>  I installed postgresql database in my machine. So now I need to connect
> the database from freeswitch.When I searched about the site.They told , to
> load the mod_lcr module.
>
> I followed the following steps.
>   * Edit the modules.conf file and uncommented the applications/mod_lcr
> line
>  * make mod_lcr-install
>  * edit /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml and
> uncomment the mod_lcr line.
>  * Reload the freeswitch
>
>
> Where I made a mistake?
> Tell the steps to connect the postreSQL database.
>
> --
> Regards,
> Thangappan.M
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From krice at freeswitch.org  Sat Aug  1 09:43:28 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 01 Aug 2009 11:43:28 -0500
Subject: [Freeswitch-users] Connect to PostgreSQL database
In-Reply-To: <5070fcbd0908010935n2d481397h210ce38057f03933@mail.gmail.com>
Message-ID: <C699DAE0.5A45A%krice@freeswitch.org>

He specifically said from FreeSWITCH... The ONLY way to connect FreeSWITCH
directory to any database (this does not includes external scripts running
as a separate process using ESL) is to either a) write your own custom
module to do so or b) use ODBC on the modules that support that now



From: Shameem Shiek <gshfreesw at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Sat, 1 Aug 2009 12:35:09 -0400
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Connect to PostgreSQL database

Thangappa,

Are you using Perl Event Socket? In that use, you can use Perl DBI to
connect to any DB instead of using the FS's ODBC module.

On Sat, Aug 1, 2009 at 7:52 AM, Ken Rice <krice at freeswitch.org> wrote:
> You need to look at using ODBC... That is what pretty much everything uses...
> Connect FS to ODBC and ODBC to pgsql...
> 
> 
> 
> From: "Thangappan.M" <thangappan143 at gmail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 1 Aug 2009 12:01:36 +0530
> To: freeswitch-users <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Connect to PostgreSQL database
> 
> 
> Dear all,
> 
> ?I installed postgresql database in my machine. So now I need to connect the
> database from freeswitch.When I searched about the site.They told , to load
> the mod_lcr module.
> 
> I followed the following steps.
> ? * Edit the modules.conf file and uncommented the applications/mod_lcr line
> ?* make mod_lcr-install
> ?* edit /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml and
> uncomment the mod_lcr line.
> ?* Reload the freeswitch
> 
> 
> Where I made a mistake?
> Tell the steps to connect the postreSQL database.
> 
> -- 
> Regards,
> Thangappan.M
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From darren at dmmhosting.co.uk  Sat Aug  1 03:53:06 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Sat, 1 Aug 2009 11:53:06 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <54141a3f-017f-4469-965a-1717526c5b2f@dmmhosting.co.uk>

I am considering using freeswitch and would like to know if this is possible.
 
The provider I use has a host that sits behind an OpenSER proxy. The hostname cannot get resolved by DNS on the internet.
 
Using freeswitch, at the moment, I am getting a DNS failure message for the host.
 
Is there a way of registering to this host and making calls through it by making all traffic go through the outbound proxy?
 
TIA
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4292 (20090730) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From gmaruzz at celliax.org  Sat Aug  1 11:34:34 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 1 Aug 2009 20:34:34 +0200
Subject: [Freeswitch-users] if using centos you should read this
In-Reply-To: <7b197bef0907302321p6aea0f2fhc542551b70392d6e@mail.gmail.com>
References: <6C1D282197D14BB9935770FD34B8F42A@noblesys.com> 
	<92D4AB13-73E0-4989-AC2F-A6703354D14B@freeswitch.org>
	<b16156850907302157p7898e599m8ae98c2b8f133d49@mail.gmail.com> 
	<87f2f3b90907302221u67ce4e32mf2e8faa8a4a13771@mail.gmail.com> 
	<b16156850907302236i696d3941i6f2a24ca59d9ff85@mail.gmail.com> 
	<7b197bef0907302321p6aea0f2fhc542551b70392d6e@mail.gmail.com>
Message-ID: <7b197bef0908011134u4ceeb820k13fdf6a222374235@mail.gmail.com>

http://linux.slashdot.org/story/09/08/01/1443221/CentOS-Administrator-Reappears

str8edge sends word that Lance Davis, the CentOS project administrator
who had mysteriously gone absent, has now returned and is working with
the development team to get things back on track. From their
announcement: "The CentOS Development team had a routine meeting today
with Lance Davis in attendance. During the meeting a majority of
issues were resolved immediately and a working agreement was reached
with deadlines for remaining unresolved issues. There should be no
impact to any CentOS users going forward. The CentOS project is now in
control of the CentOS.org and CentOS.info domains and owns all
trademarks, materials, and artwork in the CentOS distributions. We
look forward to working with Lance to quickly complete all the agreed
upon issues. More information will follow soon."

On Fri, Jul 31, 2009 at 8:21 AM, Giovanni Maruzzelli<gmaruzz at celliax.org> wrote:
> :-)!
>
>
> On Fri, Jul 31, 2009 at 7:36 AM, Muhammad
> Shahzad<shaheryarkh at googlemail.com> wrote:
>> Please read my email as,
>>
>>> CentOS has been a trusted platfrom for me from last 3+ years. I have
>>> developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS
>>> boxes, and 27 out of 49 Asterisk box are still running on CentOS in
>>> production environment. I really wish and hope this great project continues.
>>>
>>> I don't know any of its developers personally but i am quite sure they
>>> will resolve their differences professionally and put this project back on
>>> track.
>>
>> This damn Google Spell made meaning of my entire post the possite. ;-(
>>
>> Thank you.
>>
>>
>> On Fri, Jul 31, 2009 at 11:21 AM, Michael Collins <msc at freeswitch.org>
>> wrote:
>>>
>>>
>>> On Thu, Jul 30, 2009 at 9:57 PM, Muhammad Shahzad
>>> <shaheryarkh at googlemail.com> wrote:
>>>>
>>>> CentOS has been a trusted platfrom for me from last 3+ years. I have
>>>> developed and deployed many FS and Asterisk solutions on it, 9 out of 13 FS
>>>> boxes, and 27 out of 49 Asterisk box are still ruining on CentOS in
>>>> production environment. I really wish and hope this great project continues.
>>>>
>>>> I don't know any of its developers personally but i am quite sure they
>>>> will resolve their differences professionally and put this project back on
>>>> track.
>>>
>>> The guys doing the work have vowed to continue the project. The only real
>>> issues are who controls the centos.org domain name and how to handle
>>> donations to the project. CentOS isn't going anywhere but forward.
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Muhammad Shahzad
>> -----------------------------------
>> CISCO Rich Media Communication Specialist (CRMCS)
>> CISCO Certified Network Associate (CCNA)
>> Cell: +92 334 422 40 88
>> MSN: shari_786pk at hotmail.com
>> Email: shaheryarkh at googlemail.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>



From d at unwire.it  Sat Aug  1 11:57:35 2009
From: d at unwire.it (Darin Weeks)
Date: Sat, 1 Aug 2009 11:57:35 -0700
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <54141a3f-017f-4469-965a-1717526c5b2f@dmmhosting.co.uk>
References: <54141a3f-017f-4469-965a-1717526c5b2f@dmmhosting.co.uk>
Message-ID: <989132e70908011157m67680efbm427282aaab87255a@mail.gmail.com>

Have you looked over various example configurations on the wiki?  see: SIP
Provider Examples

I'm a bit confused by your questions... to make voip calls work, you
basically need to register/authenticate with proxies/gateways to send and
receive calls through them.  Sounds like you need to find out from your
provider's documentation as to how they expect you to connect with them.
Why are you trying to connect to the host behind the proxy rather than the
proxy?


On Sat, Aug 1, 2009 at 3:53 AM, Darren Williams <darren at dmmhosting.co.uk>wrote:

>  I am considering using freeswitch and would like to know if this is
> possible.
>
>
>
> The provider I use has a host that sits behind an OpenSER proxy. The
> hostname cannot get resolved by DNS on the internet.
>
>
>
> Using freeswitch, at the moment, I am getting a DNS failure message for the
> host.
>
>
>
> Is there a way of registering to this host and making calls through it by
> making all traffic go through the outbound proxy?
>
>
>
> TIA
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4292 (20090730) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 

) ) ) ) ) ) ) ) ) UNWIRE.IT ( ( ( ( ( ( ( ( (
broadband internet for west hollywood
   d at unwire.it        http://unwire.it
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From pjintheusa at gmail.com  Sat Aug  1 12:36:09 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 1 Aug 2009 15:36:09 -0400
Subject: [Freeswitch-users] LUA: Independent control of each call leg.
In-Reply-To: <24744087.post@talk.nabble.com>
References: <24744087.post@talk.nabble.com>
Message-ID: <367751820908011236p5db2fcfbu859bdbef60d6de7e@mail.gmail.com>

 >>a prime example is a b-leg "whisper" where you are still playing
"ringing" to the a-party while
>>the b-party answers and hears a message just before connection to the
>>a-party.

You should be able to do this particular function using group_confirm. See
this page in the wiki.

http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#exec_in_answer_confirm

In your case the test.js would contain the whisper. Note that you can send
args to this script.


On Fri, Jul 31, 2009 at 11:01 AM, peely <freeswitch at peely.com> wrote:

>
> Hi,
>
> I'm trying to develop an application using lua and need to control the
> inbound and outbound legs independently, even when they are switched
> together.
>
> I can initiate the outbound session but I can't seem to bridge without
> losing control of the script.
>
> For example, if I use:
>
>
>        local api = freeswitch.API();
>        inSession = session;
>        inSession:answer();
>        inSession:setAutoHangup(false);
>
>
>        egSession = freeswitch.Session("sofia/default/mynum at mydomain.com");
>        egSession:setAutoHangup(false);
>
>        if egSession:ready() then
>                api:execute("uuid_bridge",inSession.uuid .. " " ..
> egSession.uuid);
>        end
>
>        while egSession:ready() do
>                inSession:sleep(1000);
>        end
>
> Then I lose the script entirely, and if I use:
>
>        inSession:execute("bridge", "sofia/default/mynum at mydomain.com")
>
> Then I lose the ability to control the call whilst the outbound is in
> progress.
>
> Does anyone know a way I can allow ingress to egress calling whilst still
> maintaining script control mid-call? I also need to ingress to hear
> provisional speech during outbound connect. I've looked at conferencing but
> there seems to be quite a lot of automated messaging.
>
>
>
> Thanks,
>
>
> Neil.
>
> --
> View this message in context:
> http://www.nabble.com/LUA%3A-Independent-control-of-each-call-leg.-tp24744087p24744087.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sun Aug  2 01:33:32 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 Aug 2009 03:33:32 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <989132e70908011157m67680efbm427282aaab87255a@mail.gmail.com>
References: <54141a3f-017f-4469-965a-1717526c5b2f@dmmhosting.co.uk>
	<989132e70908011157m67680efbm427282aaab87255a@mail.gmail.com>
Message-ID: <65978701-BC97-4C7F-9BC5-EC66F769F3ED@freeswitch.org>

Fill out proxy... with the fake hostname... then fill out register- 
proxy and/or outbound-proxy.

/b

On Aug 1, 2009, at 1:57 PM, Darin Weeks wrote:

> Have you looked over various example configurations on the wiki?   
> see: SIP Provider Examples
>
> I'm a bit confused by your questions... to make voip calls work, you  
> basically need to register/authenticate with proxies/gateways to  
> send and receive calls through them.  Sounds like you need to find  
> out from your provider's documentation as to how they expect you to  
> connect with them.  Why are you trying to connect to the host behind  
> the proxy rather than the proxy?
>
>
> On Sat, Aug 1, 2009 at 3:53 AM, Darren Williams <darren at dmmhosting.co.uk 
> > wrote:
> I am considering using freeswitch and would like to know if this is  
> possible.
>
>
> The provider I use has a host that sits behind an OpenSER proxy. The  
> hostname cannot get resolved by DNS on the internet.
>
>
> Using freeswitch, at the moment, I am getting a DNS failure message  
> for the host.
>
>
> Is there a way of registering to this host and making calls through  
> it by making all traffic go through the outbound proxy?
>
>
> TIA
>

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From markmorreny at gmail.com  Sun Aug  2 02:04:18 2009
From: markmorreny at gmail.com (mark morreny)
Date: Sun, 2 Aug 2009 17:04:18 +0800
Subject: [Freeswitch-users] H248 support
Message-ID: <20ad6b920908020204p6a6bec6dt32f17638d778a4e0@mail.gmail.com>

Hi,

Someone told me to check here.  I am looking for a H248 supported gateway.
Does freeswitch support H248 or is there anyway to make it supportable in
freeSWITCH?

Thanks,
Mark
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From brian at freeswitch.org  Sun Aug  2 02:09:06 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 Aug 2009 04:09:06 -0500
Subject: [Freeswitch-users] H248 support
In-Reply-To: <20ad6b920908020204p6a6bec6dt32f17638d778a4e0@mail.gmail.com>
References: <20ad6b920908020204p6a6bec6dt32f17638d778a4e0@mail.gmail.com>
Message-ID: <D489C23D-0858-4221-A37B-AB5AFF821B4C@freeswitch.org>

Not at this time.. you could try contacting consulting at freeswitch.org  
to see what it might take to gain this support by funding it.

/b

On Aug 2, 2009, at 4:04 AM, mark morreny wrote:

> Hi,
>
> Someone told me to check here.  I am looking for a H248 supported  
> gateway.  Does freeswitch support H248 or is there anyway to make it  
> supportable in freeSWITCH?
>
> Thanks,
> Mark




From darren at dmmhosting.co.uk  Sun Aug  2 03:31:17 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Sun, 2 Aug 2009 11:31:17 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <52d777d0-f93e-4e41-9293-d6c403f60080@dmmhosting.co.uk>

?and/or outbound proxy?, I do not seem to be able to find this parameter.
 
It is probably my terminology which is confusing too and what exactly I want to do. All I know is that on my Thomson ST2030. If I enter:
 
Registrar Server Address: bmnha-01.bt.com
Proxy Server Address: bmnha-01.bt.com
Outbound Proxy Server:www.bbvservice-560129.bt.com
 
Everything works fine, I would just like to be able to transfer this config to freeswitch.
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 02 August 2009 09:34
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
Fill out proxy... with the fake hostname... then fill out register-proxy and/or outbound-proxy.
 
/b
 
On Aug 1, 2009, at 1:57 PM, Darin Weeks wrote:



Have you looked over various example configurations on the wiki?  see: SIP Provider Examples

I'm a bit confused by your questions... to make voip calls work, you basically need to register/authenticate with proxies/gateways to send and receive calls through them.  Sounds like you need to find out from your provider's documentation as to how they expect you to connect with them.  Why are you trying to connect to the host behind the proxy rather than the proxy?


On Sat, Aug 1, 2009 at 3:53 AM, Darren Williams <darren at dmmhosting.co.uk> wrote:
I am considering using freeswitch and would like to know if this is possible.
 
The provider I use has a host that sits behind an OpenSER proxy. The hostname cannot get resolved by DNS on the internet.
 
Using freeswitch, at the moment, I am getting a DNS failure message for the host.
 
Is there a way of registering to this host and making calls through it by making all traffic go through the outbound proxy?
 
TIA
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4297 (20090801) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4297 (20090801) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From rdenert at tng.de  Sun Aug  2 05:18:50 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Sun, 2 Aug 2009 14:18:50 +0200 (CEST)
Subject: [Freeswitch-users] Language of the speech
In-Reply-To: <6CE276B1-75B1-4470-BDD0-0E51721D3658@jerris.com>
Message-ID: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>

Hello,

sorry but I don't understand your answer.

BR

----- Urspr?ngliche Mail -----
Von: "Michael Jerris" <mike at jerris.com>
An: freeswitch-users at lists.freeswitch.org
Gesendet: Freitag, 31. Juli 2009 21:34:16 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: Re: [Freeswitch-users] Language of the speech

we don;t have any german sound files?

On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:

> Hi again,
>
> does anybody know why my freeswitch doesn't play German speech-files?
>
> Here is my construct:
>
> I'm generating a random number in my lua script:
> rand = math.random(11, 1000);
>
> The I want that the freeswitch says the "random number":
> session:execute("say", "de name_spelled iterated " ..test_nummer);
>
> It works fine in English but I don't bring the freeswitch to say the  
> number in german. I installed the freeswitch-lang- 
> de_1.0.3-1_i386.deb, change the line in freeswitch.xml:
> <section name="phrases" description="Speech Phrase Management">
>    <macros>
>      <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>    </macros>
> </section>
>
> Here is an extraction from the fs_cli:
> 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152  
> switch_core_standard_on_execute() sofia/external/anonymous at anonymous.invalid 
>  Execute set(default_language=de)
> 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous at anonymous.invalid 
>  SET [default_language]=[de]
>
> But I don't here any German speech only the English on
>
> Does anybody of you have an idea? Thanks a lot again.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
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unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
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Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.



From mike at jerris.com  Sun Aug  2 11:06:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 2 Aug 2009 14:06:58 -0400
Subject: [Freeswitch-users] Language of the speech
In-Reply-To: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>
References: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>
Message-ID: <A8ED0B8A-A1AD-4240-82AB-F5F125E2ACAA@jerris.com>

how is going to play files that we don't have?

On Aug 2, 2009, at 8:18 AM, Rudolf Denert wrote:

> Hello,
>
> sorry but I don't understand your answer.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Michael Jerris" <mike at jerris.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Freitag, 31. Juli 2009 21:34:16 GMT +01:00 Amsterdam/ 
> Berlin/Bern/Rom/Stockholm/Wien
> Betreff: Re: [Freeswitch-users] Language of the speech
>
> we don;t have any german sound files?
>
> On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:
>
>> Hi again,
>>
>> does anybody know why my freeswitch doesn't play German speech-files?
>>
>> Here is my construct:
>>
>> I'm generating a random number in my lua script:
>> rand = math.random(11, 1000);
>>
>> The I want that the freeswitch says the "random number":
>> session:execute("say", "de name_spelled iterated " ..test_nummer);
>>
>> It works fine in English but I don't bring the freeswitch to say the
>> number in german. I installed the freeswitch-lang-
>> de_1.0.3-1_i386.deb, change the line in freeswitch.xml:
>> <section name="phrases" description="Speech Phrase Management">
>>   <macros>
>>     <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>>   </macros>
>> </section>
>>
>> Here is an extraction from the fs_cli:
>> 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152
>> switch_core_standard_on_execute() sofia/external/anonymous at anonymous.invalid
>> Execute set(default_language=de)
>> 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous at anonymous.invalid
>> SET [default_language]=[de]
>>
>> But I don't here any German speech only the English on
>>
>> Does anybody of you have an idea? Thanks a lot again.
>




From wiltingtree at gmail.com  Sun Aug  2 12:36:07 2009
From: wiltingtree at gmail.com (Adam Wilt)
Date: Sun, 2 Aug 2009 15:36:07 -0400
Subject: [Freeswitch-users] Bridging a call to an extension on another PBX.
Message-ID: <eb7797d70908021236x18eabbb4t6882553da99642a0@mail.gmail.com>

Hello,
I'm trying to conference-in a call from FreeSWITCH to an extension on
another PBX using sip.

According to the documentation, I think it should look like this:

       conference abc at default dial
{sip_auth_username=myuser,sip_auth_password=mypassword}sofia/external/
101 at 1.2.3.4

where 1.2.3.4 is the ip address of the remote pbx, and 101 is the extension.

I've tried adding a gateway for it in the sip profiles, and then doing this:

      conference abc at default dial sofia/mygateway/701

Both of these methods give me a result of:

      Call Requested: result: [DESTINATION_OUT_OF_ORDER]

I set-up my soft phone to register to the same ip address with the same
credentials, and it allows me to call the extension properly.

Can somebody please tell me what I'm doing wrong?

Thanks,
Adam
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From nik.middleton at noblesolutions.co.uk  Sun Aug  2 12:38:55 2009
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Sun, 2 Aug 2009 20:38:55 +0100
Subject: [Freeswitch-users] Outbound socket question
Message-ID: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>

Hi Guys,

 

I'm using an outbound socket to control calls, and it works a charm.
However, what I'd like to do is send a custom event regarding the call
on hang-up.  The way I see things happening at the moment, and I could
be wrong, is that the socket is closed when a hang-up occurs, so am I
taking a chance trying to send the event then? (try to sneak out the
event before socket closure happens)  The other option is of course to
open an inbound socket and send the event, but I'd rather not do that if
possible.

 

Regards

  

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From jmesquita at gmail.com  Sun Aug  2 13:21:54 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sun, 2 Aug 2009 17:21:54 -0300
Subject: [Freeswitch-users] Bridging a call to an extension on another
	PBX.
In-Reply-To: <eb7797d70908021236x18eabbb4t6882553da99642a0@mail.gmail.com>
References: <eb7797d70908021236x18eabbb4t6882553da99642a0@mail.gmail.com>
Message-ID: <5a8712120908021321n750e549cl3e2f994813cc3b2d@mail.gmail.com>

Adam,

Pastebin the logs. Also, a sip dump of both situations can really help.

To enable sip traces on FreeSWITCH all you have to do is type on the CLI:

sofia profile <profile> siptrace on/off


jmesquita

On Sun, Aug 2, 2009 at 4:36 PM, Adam Wilt <wiltingtree at gmail.com> wrote:

> Hello,
> I'm trying to conference-in a call from FreeSWITCH to an extension on
> another PBX using sip.
>
> According to the documentation, I think it should look like this:
>
>        conference abc at default dial
> {sip_auth_username=myuser,sip_auth_password=mypassword}sofia/external/
> 101 at 1.2.3.4
>
> where 1.2.3.4 is the ip address of the remote pbx, and 101 is the
> extension.
>
> I've tried adding a gateway for it in the sip profiles, and then doing
> this:
>
>       conference abc at default dial sofia/mygateway/701
>
> Both of these methods give me a result of:
>
>       Call Requested: result: [DESTINATION_OUT_OF_ORDER]
>
> I set-up my soft phone to register to the same ip address with the same
> credentials, and it allows me to call the extension properly.
>
> Can somebody please tell me what I'm doing wrong?
>
> Thanks,
> Adam
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From merul at mac.com  Sun Aug  2 14:34:08 2009
From: merul at mac.com (Merul Patel)
Date: Sun, 02 Aug 2009 22:34:08 +0100
Subject: [Freeswitch-users] Configuring Sangoma U100
Message-ID: <27E91460-3D50-4DF4-AF1D-95D97634112C@mac.com>

I'm new to FS, and experimenting with it on a constrained environment  
(PCEngines ALIX board running Voyage Linux 0.62).

So far, FS has compiled fine, and I can register multiple softphones  
and make calls between them, but I'm lost at how to configure a  
Sangoma U100 so I can make and receive calls over an analogue line.

I've installed the wanpipe drivers (3.5.4) from Sangoma, and the  
wanrouter utility detects the USB device, and I've compiled it to  
support the TDM API.

FS was compiled with the Openzap module - as best as I can tell.

I thought that I would be able to use the wancfg_tdmapi utility to  
configure the /etc/wanpipe/wanpipe1.conf and then use the generated  
configuration file as the basis for configuring autoload_configs/ 
openzap.conf.xml.

However, the wancfg utility doesn't generate the wanpipe1.conf, and  
I'm stumped.

Any pointers would be much appreciated.

Best regards,

Merul
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From msc at freeswitch.org  Sun Aug  2 15:35:01 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sun, 2 Aug 2009 15:35:01 -0700
Subject: [Freeswitch-users] Outbound socket question
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
Message-ID: <87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>

On Sun, Aug 2, 2009 at 12:38 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

>  Hi Guys,
>
>
>
> I?m using an outbound socket to control calls, and it works a charm.
> However, what I?d like to do is send a custom event regarding the call on
> hang-up.  The way I see things happening at the moment, and I could be
> wrong, is that the socket is closed when a hang-up occurs, so am I taking a
> chance trying to send the event then? (try to sneak out the event before
> socket closure happens)  The other option is of course to open an inbound
> socket and send the event, but I?d rather not do that if possible.
>
Nik,

Perhaps the "linger" event socket command will do what you need? Check out
this commit:
http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009391.html

Let me know if it works for you and I'll be sure to get it documented
properly. If you get it working I'd love to see a code snippet so we can
wikify this knowledge. :)

Thanks,
MC
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From mike at jerris.com  Sun Aug  2 15:42:57 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 2 Aug 2009 18:42:57 -0400
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <27E91460-3D50-4DF4-AF1D-95D97634112C@mac.com>
References: <27E91460-3D50-4DF4-AF1D-95D97634112C@mac.com>
Message-ID: <4241B9E6-0F1F-4F9E-A1DF-CC0A1E7BE8F9@jerris.com>


On Aug 2, 2009, at 5:34 PM, Merul Patel wrote:

> I'm new to FS, and experimenting with it on a constrained  
> environment (PCEngines ALIX board running Voyage Linux 0.62).
>
> So far, FS has compiled fine, and I can register multiple softphones  
> and make calls between them, but I'm lost at how to configure a  
> Sangoma U100 so I can make and receive calls over an analogue line.
>
> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the  
> wanrouter utility detects the USB device, and I've compiled it to  
> support the TDM API.
>
> FS was compiled with the Openzap module - as best as I can tell.
>
> I thought that I would be able to use the wancfg_tdmapi utility to  
> configure the /etc/wanpipe/wanpipe1.conf and then use the generated  
> configuration file as the basis for configuring autoload_configs/ 
> openzap.conf.xml.

try wancfg_fs

> However, the wancfg utility doesn't generate the wanpipe1.conf, and  
> I'm stumped.
>
> Any pointers would be much appreciated.
>
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From brian at freeswitch.org  Sun Aug  2 16:10:48 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 2 Aug 2009 18:10:48 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <52d777d0-f93e-4e41-9293-d6c403f60080@dmmhosting.co.uk>
References: <52d777d0-f93e-4e41-9293-d6c403f60080@dmmhosting.co.uk>
Message-ID: <1305B47F-256B-494A-B3D4-E1850E134A2C@freeswitch.org>

set the proxy, register-proxy to bmnha-01.bt.com and the outbound- 
proxy to www.bbvservice-560129.bt.com, I regard this type of config B  
R O K EN and the provider shouldn't be doing this with DNS names that  
do not exist in my opinion... if they are doing this for security sake  
they should just setup VPN or direct access... this type of setup must  
makes my skin crawl.

/b

On Aug 2, 2009, at 5:31 AM, Darren Williams wrote:

> ?and/or outbound proxy?, I do not seem to be able to find this  
> parameter.
>
> It is probably my terminology which is confusing too and what  
> exactly I want to do. All I know is that on my Thomson ST2030. If I  
> enter:
>
> Registrar Server Address: bmnha-01.bt.com
> Proxy Server Address: bmnha-01.bt.com
> Outbound Proxy Server:www.bbvservice-560129.bt.com
>
> Everything works fine, I would just like to be able to transfer this  
> config to freeswitch.

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From mrene_lists at avgs.ca  Sun Aug  2 18:11:11 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 2 Aug 2009 21:11:11 -0400
Subject: [Freeswitch-users] Language of the speech
In-Reply-To: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>
References: <14294094.208231249215530037.JavaMail.root@zimbra.tng.de>
Message-ID: <B4C913FC-6332-489C-B0A1-887CBEECF96D@avgs.ca>

Da gibst keine eigentliches DE audio-Datei, nur text fuer text-to- 
speech, deshalb funktionniert es nicht.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 2-Aug-09 um 8:18 AM schrieb Rudolf Denert:

> Hello,
>
> sorry but I don't understand your answer.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Michael Jerris" <mike at jerris.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Freitag, 31. Juli 2009 21:34:16 GMT +01:00 Amsterdam/ 
> Berlin/Bern/Rom/Stockholm/Wien
> Betreff: Re: [Freeswitch-users] Language of the speech
>
> we don;t have any german sound files?
>
> On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:
>
>> Hi again,
>>
>> does anybody know why my freeswitch doesn't play German speech-files?
>>
>> Here is my construct:
>>
>> I'm generating a random number in my lua script:
>> rand = math.random(11, 1000);
>>
>> The I want that the freeswitch says the "random number":
>> session:execute("say", "de name_spelled iterated " ..test_nummer);
>>
>> It works fine in English but I don't bring the freeswitch to say the
>> number in german. I installed the freeswitch-lang-
>> de_1.0.3-1_i386.deb, change the line in freeswitch.xml:
>> <section name="phrases" description="Speech Phrase Management">
>>   <macros>
>>     <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>>   </macros>
>> </section>
>>
>> Here is an extraction from the fs_cli:
>> 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152
>> switch_core_standard_on_execute() sofia/external/anonymous at anonymous.invalid
>> Execute set(default_language=de)
>> 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous at anonymous.invalid
>> SET [default_language]=[de]
>>
>> But I don't here any German speech only the English on
>>
>> Does anybody of you have an idea? Thanks a lot again.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -- 
> -- 
> Rudolf Denert, Technical Support
> TNG AG, NGN
> Projensdorfer Str. 324, D-24106 Kiel, Germany
> phone: +49 431 7097-10, fax: +49 431 7097-555
> mailto: rdenert at tng.de http://www.tng.de
> -
> Register: Amtsgericht Kiel, HRB 6596 KI
> Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
> Executive board (Vorstand): Dr. Sven Willert (Vorsitz),  Dr. Thomas
> Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg
> Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
> -
> This e-mail may contain confidential and/or privileged information. If
> you are not the intended recipient (or have received this e-mail in
> error) please notify the sender immediately and destroy this e-mail.  
> Any
> unauthorized copying, disclosure or distribution of the material in  
> this
> e-mail is strictly forbidden.
>
> Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
> Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
> E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
> Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
> die unbefugte Weitergabe dieser Mail ist nicht gestattet.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From thangappan143 at gmail.com  Sun Aug  2 22:09:23 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 3 Aug 2009 10:39:23 +0530
Subject: [Freeswitch-users] Problem in spidermonkey_odbc
Message-ID: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>

Dear all,

 I am not yet installed odbc in my machine for accessing the psql database.
I have got the following error while tried to load the mod_lcr command.

2009-08-03 15:45:47 [ERR] switch_odbc.c:164 switch_odbc_handle_connect()
STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not
found, and no default driver specified

2009-08-03 15:45:47 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot Open ODBC
Database!
2009-08-03 15:45:47 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to load lcr
config file
2009-08-03 15:45:47 [CRIT] switch_loadable_module.c:839
switch_loadable_module_load_file() Error Loading module
/usr/local/freeswitch/mod/mod_lcr.so
**Module load routine returned an error**

I have set the correct driver informations in the odbc.ini(/etc and /home)
and odbcinst.ini(/etc).
I have create the symbolic links in the /usr/loca/freeswitch/etc/

Where I made a problem?

After some time I found that there is no spidermonket_odbc.so file.This has
been found while executing the freeswitch command. The error is,

2009-08-03 15:26:06 [ERR] mod_spidermonkey.c:931 sm_load_file() Error
Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
**/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: cannot open shared
object file: No such file or directory**

So please help me?
I am doing this for more than 8 hours.





-- 
Regards,
Thangappan.M
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From mattdfong at gmail.com  Sun Aug  2 22:22:47 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Sun, 2 Aug 2009 22:22:47 -0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
Message-ID: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>

A few users of mine have been getting hung-up on after leg b of the bridge
hangsups. I looked in the logs and they are being hungup with
an EXCHANGE_ROUTING_ERROR hangup cause. The problem is
only occurring intermittently and only when both leg a and leg b are both
passed thru external gateways (the problem does *not* exist if leg a is a
softphone connected directly to FS).
I'm wondering if there are any known incompatibilities with equipment out
there that would cause this error. Otherwise, I assume the best way to
diagnosis this further is to ngrep, which I'll do, but thought I'd ask here
first. Thanks.

--matt
hello hunter - hosted predictive dialer & voice broadcasting
http://www.hellohunter.com
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From mike at jerris.com  Sun Aug  2 23:12:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Aug 2009 01:12:51 -0500
Subject: [Freeswitch-users] Problem in spidermonkey_odbc
In-Reply-To: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
References: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
Message-ID: <D1107F74-5E72-485C-89EF-ED66A3A1AFEC@jerris.com>

You need to first install unixodbc and it's related devel packages and  
then run configure and re build freeswitch.

On Aug 3, 2009, at 12:09 AM, "Thangappan.M" <thangappan143 at gmail.com>  
wrote:

> Dear all,
>
>  I am not yet installed odbc in my machine for accessing the psql  
> database. I have got the following error while tried to load the  
> mod_lcr command.
>
> 2009-08-03 15:45:47 [ERR] switch_odbc.c:164  
> switch_odbc_handle_connect() STATE: IM002 CODE 0 ERROR: [unixODBC] 
> [Driver Manager]Data source name not found, and no default driver  
> specified
>
> 2009-08-03 15:45:47 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot  
> Open ODBC Database!
> 2009-08-03 15:45:47 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to  
> load lcr config file
> 2009-08-03 15:45:47 [CRIT] switch_loadable_module.c:839  
> switch_loadable_module_load_file() Error Loading module /usr/local/ 
> freeswitch/mod/mod_lcr.so
> **Module load routine returned an error**
>
> I have set the correct driver informations in the odbc.ini(/etc and / 
> home) and odbcinst.ini(/etc).
> I have create the symbolic links in the /usr/loca/freeswitch/etc/
>
> Where I made a problem?
>
> After some time I found that there is no spidermonket_odbc.so  
> file.This has been found while executing the freeswitch command. The  
> error is,
>
> 2009-08-03 15:26:06 [ERR] mod_spidermonkey.c:931 sm_load_file()  
> Error Loading module /usr/local/freeswitch/mod/ 
> mod_spidermonkey_odbc.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: cannot open  
> shared object file: No such file or directory**
>
> So please help me?
> I am doing this for more than 8 hours.
>
>
>
>
>
> -- 
> Regards,
> Thangappan.M
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mrene_lists at avgs.ca  Sun Aug  2 23:20:35 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 3 Aug 2009 02:20:35 -0400
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
Message-ID: <177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>

Hi,

Digging a bit in mod_sofia releaved that it can be caused by a SIP  
code 482 (loop detected), 483 (too many hops) or 484 (address  
incomplete).

Do a SIP trace to sched more light on what's happening.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 3-Aug-09 um 1:22 AM schrieb Matthew Fong:

> EXCHANGE_ROUTING_ERROR




From thangappan143 at gmail.com  Mon Aug  3 00:07:29 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 3 Aug 2009 12:37:29 +0530
Subject: [Freeswitch-users] Problem in spidermonkey_odbc
In-Reply-To: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
References: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
Message-ID: <7aa29e790908030007l1478b8b1mfa460a556349342f@mail.gmail.com>

I have installed unixodbc at first.But I am getting the same error.

On Mon, Aug 3, 2009 at 10:39 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Dear all,
>
>  I am not yet installed odbc in my machine for accessing the psql database.
> I have got the following error while tried to load the mod_lcr command.
>
> 2009-08-03 15:45:47 [ERR] switch_odbc.c:164 switch_odbc_handle_connect()
> STATE: IM002 CODE 0 ERROR: [unixODBC][Driver Manager]Data source name not
> found, and no default driver specified
>
> 2009-08-03 15:45:47 [CRIT] mod_lcr.c:546 lcr_load_config() Cannot Open ODBC
> Database!
> 2009-08-03 15:45:47 [ERR] mod_lcr.c:985 mod_lcr_load() Unable to load lcr
> config file
> 2009-08-03 15:45:47 [CRIT] switch_loadable_module.c:839
> switch_loadable_module_load_file() Error Loading module
> /usr/local/freeswitch/mod/mod_lcr.so
> **Module load routine returned an error**
>
> I have set the correct driver informations in the odbc.ini(/etc and /home)
> and odbcinst.ini(/etc).
> I have create the symbolic links in the /usr/loca/freeswitch/etc/
>
> Where I made a problem?
>
> After some time I found that there is no spidermonket_odbc.so file.This has
> been found while executing the freeswitch command. The error is,
>
> 2009-08-03 15:26:06 [ERR] mod_spidermonkey.c:931 sm_load_file() Error
> Loading module /usr/local/freeswitch/mod/mod_spidermonkey_odbc.so
> **/usr/local/freeswitch/mod/mod_spidermonkey_odbc.so: cannot open shared
> object file: No such file or directory**
>
> So please help me?
> I am doing this for more than 8 hours.
>
>
>
>
>
> --
> Regards,
> Thangappan.M
>



-- 
Regards,
Thangappan.M
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From thangappan143 at gmail.com  Mon Aug  3 00:12:14 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Mon, 3 Aug 2009 12:42:14 +0530
Subject: [Freeswitch-users] Need Help In IVR
Message-ID: <7aa29e790908030012rcce847fu8320d833d2c85530@mail.gmail.com>

Dear all,

I am in the process of implementing IVR in Perl using outbound socket.

In the case of the XML macro I can easily specify the timeout,inter digit
timeout value as
 <menu name="demo_ivr"
          greet-long="phrase:demo_ivr_main_menu"
          greet-short="phrase:demo_ivr_main_menu_short"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout ="10000"
          inter-digit-timeout="2000"
          max-failures="3"
          max-timeouts="3"
          digit-len="4">
      <entry action="menu-exec-app" digits="1" param="bridge
sofia/$${domain}/888 at conference.freeswitch.org"/>
    </menu>

Is there any way for Perl to configure this values. Where are the variables
resides?
I am struggling to implement this?




-- 
Regards,
Thangappan.M
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From rdenert at tng.de  Mon Aug  3 00:57:37 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Mon, 3 Aug 2009 09:57:37 +0200 (CEST)
Subject: [Freeswitch-users] Language of the speech
In-Reply-To: <8128254.210221249286232840.JavaMail.root@zimbra.tng.de>
Message-ID: <14701697.210241249286257372.JavaMail.root@zimbra.tng.de>

Hallo,

ich habe also keine Chance mir eine Zahl vorlesen zu lassen, welche per Zufall generiere? :-/ Ich frage nur deshalb, da ich in einem Subordner diverse Soundfiles von Zahlen habe, welche ich von einem Debianpaket entpackt habe.

Gru? aus Kiel

Translation:

Hello,

there is no chance to read an random generated number from the freeswitch? I?m asking again because I have several german soundfiles of numbers which I extracted form a debianpacket. 

BR from Kiel

----- Urspr?ngliche Mail -----
Von: "Mathieu Rene" <mrene_lists at avgs.ca>
An: freeswitch-users at lists.freeswitch.org
Gesendet: Montag, 3. August 2009 03:11:11 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: Re: [Freeswitch-users] Language of the speech

Da gibst keine eigentliches DE audio-Datei, nur text fuer text-to- 
speech, deshalb funktionniert es nicht.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 2-Aug-09 um 8:18 AM schrieb Rudolf Denert:

> Hello,
>
> sorry but I don't understand your answer.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Michael Jerris" <mike at jerris.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Freitag, 31. Juli 2009 21:34:16 GMT +01:00 Amsterdam/ 
> Berlin/Bern/Rom/Stockholm/Wien
> Betreff: Re: [Freeswitch-users] Language of the speech
>
> we don;t have any german sound files?
>
> On Jul 31, 2009, at 10:31 AM, Rudolf Denert wrote:
>
>> Hi again,
>>
>> does anybody know why my freeswitch doesn't play German speech-files?
>>
>> Here is my construct:
>>
>> I'm generating a random number in my lua script:
>> rand = math.random(11, 1000);
>>
>> The I want that the freeswitch says the "random number":
>> session:execute("say", "de name_spelled iterated " ..test_nummer);
>>
>> It works fine in English but I don't bring the freeswitch to say the
>> number in german. I installed the freeswitch-lang-
>> de_1.0.3-1_i386.deb, change the line in freeswitch.xml:
>> <section name="phrases" description="Speech Phrase Management">
>>   <macros>
>>     <X-PRE-PROCESS cmd="include" data="lang/de/*.xml"/>
>>   </macros>
>> </section>
>>
>> Here is an extraction from the fs_cli:
>> 2009-07-31 16:23:47 [DEBUG] switch_core_state_machine.c:152
>> switch_core_standard_on_execute() sofia/external/anonymous at anonymous.invalid
>> Execute set(default_language=de)
>> 2009-07-31 16:23:47 [DEBUG] mod_dptools.c:711 set_function() sofia/external/anonymous at anonymous.invalid
>> SET [default_language]=[de]
>>
>> But I don't here any German speech only the English on
>>
>> Does anybody of you have an idea? Thanks a lot again.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> -- 
> -- 
> Rudolf Denert, Technical Support
> TNG AG, NGN
> Projensdorfer Str. 324, D-24106 Kiel, Germany
> phone: +49 431 7097-10, fax: +49 431 7097-555
> mailto: rdenert at tng.de http://www.tng.de
> -
> Register: Amtsgericht Kiel, HRB 6596 KI
> Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
> Executive board (Vorstand): Dr. Sven Willert (Vorsitz),  Dr. Thomas
> Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg
> Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
> -
> This e-mail may contain confidential and/or privileged information. If
> you are not the intended recipient (or have received this e-mail in
> error) please notify the sender immediately and destroy this e-mail.  
> Any
> unauthorized copying, disclosure or distribution of the material in  
> this
> e-mail is strictly forbidden.
>
> Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
> Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
> E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
> Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
> die unbefugte Weitergabe dieser Mail ist nicht gestattet.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
-
Register: Amtsgericht Kiel, HRB 6596 KI
Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
Executive board (Vorstand): Dr. Sven Willert (Vorsitz),  Dr. Thomas
Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg
Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
-
This e-mail may contain confidential and/or privileged information. If
you are not the intended recipient (or have received this e-mail in
error) please notify the sender immediately and destroy this e-mail. Any
unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.



From merul at mac.com  Mon Aug  3 01:12:54 2009
From: merul at mac.com (Merul Patel)
Date: Mon, 03 Aug 2009 09:12:54 +0100
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <mailman.801.1249261906.2787.freeswitch-users@lists.freeswitch.org>
References: <mailman.801.1249261906.2787.freeswitch-users@lists.freeswitch.org>
Message-ID: <97532242-8290-47A9-900C-49E7567F4A4E@mac.com>

>> I'm new to FS, and experimenting with it on a constrained  
>> environment (PCEngines ALIX board running Voyage Linux 0.62).
>>
>> So far, FS has compiled fine, and I can register multiple  
>> softphones and make calls between them, but I'm lost at how to  
>> configure a Sangoma U100 so I can make and receive calls over an  
>> analogue line.
>>
>> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the  
>> wanrouter utility detects the USB device, and I've compiled it to  
>> support the TDM API.
>>
>> FS was compiled with the Openzap module - as best as I can tell.
>>
>> I thought that I would be able to use the wancfg_tdmapi utility to  
>> configure the /etc/wanpipe/wanpipe1.conf and then use the generated  
>> configuration file as the basis for configuring autoload_configs/ 
>> openzap.conf.xml.
>
> try wancfg_fs

Thanks for the suggestion Michael, but the same result occurs as when  
I try wancfg_tdmapi, ie:

"No Sangoma voice compatible cards found/configured"

>
>> However, the wancfg utility doesn't generate the wanpipe1.conf, and  
>> I'm stumped.
>>
>> Any pointers would be much appreciated.
>>




From darren at dmmhosting.co.uk  Mon Aug  3 02:12:24 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Mon, 3 Aug 2009 10:12:24 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <2d4a177f-88f4-44e9-94d9-dae6235f718f@dmmhosting.co.uk>

Brian, this ?broken? business explains a lot, I just assumed this was a normal practise. This ?outbound-proxy? parameter, I don?t see any reference to this anywhere.
 
This just causes Registration Failed with status DNS Error [503]
 
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 03 August 2009 00:11
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
set the proxy, register-proxy to bmnha-01.bt.com and the outbound-proxy to www.bbvservice-560129.bt.com, I regard this type of config B R O K EN and the provider shouldn't be doing this with DNS names that do not exist in my opinion... if they are doing this for security sake they should just setup VPN or direct access... this type of setup must makes my skin crawl.
 
/b
 
On Aug 2, 2009, at 5:31 AM, Darren Williams wrote:



?and/or outbound proxy?, I do not seem to be able to find this parameter.
 
It is probably my terminology which is confusing too and what exactly I want to do. All I know is that on my Thomson ST2030. If I enter:
 
Registrar Server Address: bmnha-01.bt.com
Proxy Server Address: bmnha-01.bt.com
Outbound Proxy Server:www.bbvservice-560129.bt.com
 
Everything works fine, I would just like to be able to transfer this config to freeswitch.
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


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From raffaele.p.guidi at gmail.com  Mon Aug  3 04:43:01 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Mon, 3 Aug 2009 13:43:01 +0200
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
Message-ID: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>

I found the answer by myself while I had finished writing the e-mail. The
correct call url is loopback/<extension> (in this case the command
is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
future reference (can't find any example of that anywhere). Is the project
wiki accesible for anyone to contribute or do I have to ask for an
authorization?

Regards,
   Raffaele

*********** ORIGINAL QUESTION *************

Hi, I'm trying to call an extension wich is not associated to a user from
the ESL (or the CLI as well) using the "ORIGINATE" command.
Now, while

originate user/1001 1000


works perfectly

with:

originate user/fakecall 1000


I have an error:

2009-08-03 13:21:19.406250 [ERR] switch_ivr_originate.c:1494 Cannot create
outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]


...this is not surprising ("fakecall" of course is not an user), but I
cannot figure out what is the correct CALL URL for this extension. Same
error is reported using sofia/internal/fakecall. It seems that the
EXECUTE_EXTENSION method could do for the magic (it works when issued from
an other extension in the dialplan) but it is not available from the CLI nor
the event socket (I'm using a binary version for windows - freeswitch.msi -
dated jul, 11th 2009).

PS: Calling "fakecall" from a registered phone (or from portaudio) works as
expected.
PPS: the configuration of fakecall in dialplan/default.xml
...
<context name="default">
   <extension name="fakecall">
<condition field="destination_number" expression="^fakecall$">
 <action application="lua" data="lua/helloworld.lua test $1"/>
</condition>
 </extension>
....
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From chad at apartmentlines.com  Mon Aug  3 03:46:10 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Mon, 3 Aug 2009 06:46:10 -0400
Subject: [Freeswitch-users] arriving today for ClueCon
Message-ID: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>

i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this  
afternoon?  anybody need help with setup?



From juanbackson at gmail.com  Mon Aug  3 04:45:21 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Mon, 3 Aug 2009 19:45:21 +0800
Subject: [Freeswitch-users] Question about dynamic registration
Message-ID: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>

Hi,

Other than curl, is there anyway to do dynamic registration?
It there anyway to embed a script in freeswitch to do the authorization?

Thanks,
Anne
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From brian at freeswitch.org  Mon Aug  3 05:40:48 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 07:40:48 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <2d4a177f-88f4-44e9-94d9-dae6235f718f@dmmhosting.co.uk>
References: <2d4a177f-88f4-44e9-94d9-dae6235f718f@dmmhosting.co.uk>
Message-ID: <184C69C8-EBEA-487D-A921-B35C478F7607@freeswitch.org>

You must be on SVN trunk.

/b

On Aug 3, 2009, at 4:12 AM, Darren Williams wrote:

> Brian, this ?broken? business explains a lot, I just assumed this  
> was a normal practise. This ?outbound-proxy? parameter, I don?t see  
> any reference to this anywhere.
>
> This just causes Registration Failed with status DNS Error [503]
>

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From brian at freeswitch.org  Mon Aug  3 05:41:41 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 07:41:41 -0500
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
Message-ID: <6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>

Just look for large groups of people with laptops.  I'm sure you can't  
miss us.

/b

On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:

> i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
> afternoon?  anybody need help with setup?




From brian at freeswitch.org  Mon Aug  3 05:42:33 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 07:42:33 -0500
Subject: [Freeswitch-users] Question about dynamic registration
In-Reply-To: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>
References: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>
Message-ID: <9FE0F6B8-C4BF-4820-8CC2-6825C5EE8422@freeswitch.org>

You could build your own module to do it how ever you please.  But  
forking a script every time to auth is not very scalable.

/b

On Aug 3, 2009, at 6:45 AM, Juan Backson wrote:

> Hi,
>
> Other than curl, is there anyway to do dynamic registration?
> It there anyway to embed a script in freeswitch to do the  
> authorization?
>
> Thanks,
> Anne
> ______




From asannucci at gmail.com  Mon Aug  3 05:54:49 2009
From: asannucci at gmail.com (bakko)
Date: Mon, 3 Aug 2009 14:54:49 +0200
Subject: [Freeswitch-users] Problem in spidermonkey_odbc
In-Reply-To: <7aa29e790908030007l1478b8b1mfa460a556349342f@mail.gmail.com>
References: <7aa29e790908022209n2ddb9b43xa5ebc1b5028686d3@mail.gmail.com>
	<7aa29e790908030007l1478b8b1mfa460a556349342f@mail.gmail.com>
Message-ID: <CC687DA31ED44F2B8541E3878F54CF7F@voztovoice>

If your linux distribution is Centos you have to install  unixODBC-devel postgresql-odbc unixODBC then compile freeswitch.

Look at this wiki page:

http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc

BR
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From msc at freeswitch.org  Mon Aug  3 05:59:34 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Aug 2009 05:59:34 -0700
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
In-Reply-To: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
References: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
Message-ID: <87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>

On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> I found the answer by myself while I had finished writing the e-mail. The
> correct call url is loopback/<extension> (in this case the command
> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
> future reference (can't find any example of that anywhere). Is the project
> wiki accesible for anyone to contribute or do I have to ask for an
> authorization?
>

All you need to do is sign up for a free account on the wiki and you can
start editing. It's a community resource and all FS users are invited to add
their respective knowledge.

As for not finding what you were looking for, does this page not have it?
http://wiki.freeswitch.org/wiki/Loopback

If not then please feel free to add to this page whatever your specific
scenario entails and give some examples.

-MC
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From a.afzali2003 at gmail.com  Mon Aug  3 06:38:03 2009
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 3 Aug 2009 17:08:03 +0330
Subject: [Freeswitch-users] Missing mod_curl
Message-ID: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>

Hi,

I'll appreciate if somebody tell me where has gone the mod_curl ? I just
need to use it for http method calls.

Regards,
-- afshin
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From jgonzalez at sqli.com  Mon Aug  3 06:41:09 2009
From: jgonzalez at sqli.com (julien)
Date: Mon, 03 Aug 2009 15:41:09 +0200
Subject: [Freeswitch-users] Authentication problem when calling softphones
	from ipphones
Message-ID: <4A76E8F5.9070804@sqli.com>

Hello everyone,
I'm using a SIP trunk to link my PBX and FS.
My problem is when I try to call a softphone on FS from my ipphone, I've 
the following error on FS during Authentication :

2009-08-03 14:58:27.123817 [DEBUG] sofia.c:4554 IP [PBX IP@] Rejected by 
acl "domains". Falling back to Digest auth.
2009-08-03 14:58:27.235492 [DEBUG] sofia.c:4554 IP [PBX IP@] Rejected by 
acl "domains". Falling back to Digest auth.
2009-08-03 14:58:27.235492 [WARNING] sofia_reg.c:1755 Can't find user 
[@[FS IP@]]
You must define a domain called '[FS IP@]' in your directory and add a 
user with the id="" attribute
and you must configure your device to use the proper domain in it's 
authentication credentials.

I defined my gateway to the PBX this way :

<include>
  <gateway name="pbxlyon">
  <param name="username" value="pbxlyon"/>
  <param name="realm" value="[PBX IP@]"/>
  <param name="password" value="pbxlyon"/>
  <param name="register" value="false"/>
  <param name="register-transport" value="udp"/>
  <param name="retry_seconds" value="300"/>
  </gateway>
</include>

I don't want the PBX to try to authenticate because I can't define a 
username nor a password for the authentication.
Thank for you time.
Best regards,
Julien GONZALEZ.








From mike at jerris.com  Mon Aug  3 06:59:35 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Aug 2009 08:59:35 -0500
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <97532242-8290-47A9-900C-49E7567F4A4E@mac.com>
References: <mailman.801.1249261906.2787.freeswitch-users@lists.freeswitch.org>
	<97532242-8290-47A9-900C-49E7567F4A4E@mac.com>
Message-ID: <FCA3464E-2A85-40F3-93AA-F8154470B399@jerris.com>

What is the output of wantouter hwprobe?

On Aug 3, 2009, at 3:12 AM, Merul Patel <merul at mac.com> wrote:

>>> I'm new to FS, and experimenting with it on a constrained
>>> environment (PCEngines ALIX board running Voyage Linux 0.62).
>>>
>>> So far, FS has compiled fine, and I can register multiple
>>> softphones and make calls between them, but I'm lost at how to
>>> configure a Sangoma U100 so I can make and receive calls over an
>>> analogue line.
>>>
>>> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the
>>> wanrouter utility detects the USB device, and I've compiled it to
>>> support the TDM API.
>>>
>>> FS was compiled with the Openzap module - as best as I can tell.
>>>
>>> I thought that I would be able to use the wancfg_tdmapi utility to
>>> configure the /etc/wanpipe/wanpipe1.conf and then use the generated
>>> configuration file as the basis for configuring autoload_configs/
>>> openzap.conf.xml.
>>
>> try wancfg_fs
>
> Thanks for the suggestion Michael, but the same result occurs as when
> I try wancfg_tdmapi, ie:
>
> "No Sangoma voice compatible cards found/configured"
>
>>
>>> However, the wancfg utility doesn't generate the wanpipe1.conf, and
>>> I'm stumped.
>>>
>>> Any pointers would be much appreciated.
>>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From mike at jerris.com  Mon Aug  3 07:03:08 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Aug 2009 09:03:08 -0500
Subject: [Freeswitch-users] Missing mod_curl
In-Reply-To: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
References: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
Message-ID: <396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>

It is still there.

On Aug 3, 2009, at 8:38 AM, afshin afzali <a.afzali2003 at gmail.com>  
wrote:

> Hi,
>
> I'll appreciate if somebody tell me where has gone the mod_curl ? I  
> just need to use it for http method calls.
>
> Regards,
> -- afshin
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From hoaianh at gmx.de  Mon Aug  3 07:30:44 2009
From: hoaianh at gmx.de (Ngo-Vi Hoai-Anh)
Date: Mon, 03 Aug 2009 16:30:44 +0200
Subject: [Freeswitch-users] telnet to event socket
Message-ID: <4A76F494.5020101@gmx.de>

Hi,

I'm taking a close look at event socket on FS 1.0.3. Configuration is 
the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and 
fsconsole.pl work but I was not able to telnet to port 8021. As I've 
done that I received somewhat like:
#>auth/request

I typed in: auth ClueCon

After some seconds I've got the message 'connection close by foreign host'

Any ideas?

Thank you
Hoai-Anh



From vkozak at abisoft.spb.ru  Mon Aug  3 07:37:43 2009
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Mon, 3 Aug 2009 18:37:43 +0400
Subject: [Freeswitch-users] Fw: FreeSwitch doesn't play music on hold
	forbriged channel
Message-ID: <C6AE09778C1441F6842140EFFC55DF20@abisoft.biz>

No. I didn't. Moreover, I tried to set it explicitly using api uuid_setvar.
  ----- Original Message ----- 
  From: Kozak Vladimir 
  To: ?????? ?????? 
  Sent: Monday, August 03, 2009 5:28 PM
  Subject: Fw: [Freeswitch-users] FreeSwitch doesn't play music on hold forbriged channel



  ----- Original Message ----- 
  From: Brian West 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Friday, July 31, 2009 5:22 PM
  Subject: Re: [Freeswitch-users] FreeSwitch doesn't play music on hold forbriged channel


  I don't see the variable hold_music ... did you remove it? 


  /b


  On Jul 31, 2009, at 5:24 AM, Kozak Vladimir wrote:



    The scenario is the following:
    FS User A dial an extension
    Extention opens outbound socket channel to my application
    My application bridges the call to FS User B
    The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id
    The application sends hold to the bridged channel using SendMsg with Other-leg-unique-id
    User B is placed on hold but no music on hold is played to the caller (User A)


    I have outbound socket channel and the following sequence of commands/event:
    listening on [any] 8084 ...
    connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250] 34000
    connect





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From rdenert at tng.de  Mon Aug  3 07:49:19 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Mon, 3 Aug 2009 16:49:19 +0200 (CEST)
Subject: [Freeswitch-users] telnet to event socket
In-Reply-To: <4A76F494.5020101@gmx.de>
Message-ID: <2076715.215161249310959264.JavaMail.root@zimbra.tng.de>

Hi,

you only have to write "auth <password>" and hit enter twice. The default password is something like ClueCon.

BR

----- Urspr?ngliche Mail -----
Von: "Ngo-Vi Hoai-Anh" <hoaianh at gmx.de>
An: freeswitch-users at lists.freeswitch.org
Gesendet: Montag, 3. August 2009 16:30:44 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: [Freeswitch-users] telnet to event socket

Hi,

I'm taking a close look at event socket on FS 1.0.3. Configuration is 
the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and 
fsconsole.pl work but I was not able to telnet to port 8021. As I've 
done that I received somewhat like:
#>auth/request

I typed in: auth ClueCon

After some seconds I've got the message 'connection close by foreign host'

Any ideas?

Thank you
Hoai-Anh

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
-
Register: Amtsgericht Kiel, HRB 6596 KI
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Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
-
This e-mail may contain confidential and/or privileged information. If
you are not the intended recipient (or have received this e-mail in
error) please notify the sender immediately and destroy this e-mail. Any
unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.



From dujinfang at gmail.com  Mon Aug  3 07:51:06 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 3 Aug 2009 22:51:06 +0800
Subject: [Freeswitch-users] telnet to event socket
In-Reply-To: <4A76F494.5020101@gmx.de>
References: <4A76F494.5020101@gmx.de>
Message-ID: <645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>


On Aug 3, 2009, at 10:30 PM, Ngo-Vi Hoai-Anh wrote:
> Hi,
>
> I'm taking a close look at event socket on FS 1.0.3. Configuration is
> the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and
> fsconsole.pl work but I was not able to telnet to port 8021. As I've
> done that I received somewhat like:
> #>auth/request
>
> I typed in: auth ClueCon
>
followed by two Enters (\n\n).

> After some seconds I've got the message 'connection close by foreign  
> host'
>
> Any ideas?
>
> Thank you
> Hoai-Anh
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From patj at linklocal.net  Mon Aug  3 08:31:11 2009
From: patj at linklocal.net (Pat Jensen)
Date: Mon, 3 Aug 2009 08:31:11 -0700
Subject: [Freeswitch-users] Authentication problem when calling
 softphones	from ipphones
In-Reply-To: <4A76E8F5.9070804@sqli.com>
References: <4A76E8F5.9070804@sqli.com>
Message-ID: <0F432CFDE6E44442BB35719DF1034F6D5F011ACFB7@ws2008.linklocal.net>

Julien,

Place an ACL entry with the IP address for your PBX in the following file:

/usr/local/freeswitch/conf/autoload_configs/acl.conf.xml

 <list name="domains" default="deny">
<node type="allow" cidr="123.12.12.1/32"/>   <!-- your pbx ip -->
</list>

This should allow unauthenticated invites from your PBX to hit FS.  Hope this helps.

Pat
________________________________________
From: julien [jgonzalez at sqli.com]
Sent: Monday, August 03, 2009 6:41 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Authentication problem when calling softphones      from ipphones

Hello everyone,
I'm using a SIP trunk to link my PBX and FS.
My problem is when I try to call a softphone on FS from my ipphone, I've
the following error on FS during Authentication :

2009-08-03 14:58:27.123817 [DEBUG] sofia.c:4554 IP [PBX IP@] Rejected by
acl "domains". Falling back to Digest auth.
2009-08-03 14:58:27.235492 [DEBUG] sofia.c:4554 IP [PBX IP@] Rejected by
acl "domains". Falling back to Digest auth.
2009-08-03 14:58:27.235492 [WARNING] sofia_reg.c:1755 Can't find user
[@[FS IP@]]
You must define a domain called '[FS IP@]' in your directory and add a
user with the id="" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.

I defined my gateway to the PBX this way :

<include>
  <gateway name="pbxlyon">
  <param name="username" value="pbxlyon"/>
  <param name="realm" value="[PBX IP@]"/>
  <param name="password" value="pbxlyon"/>
  <param name="register" value="false"/>
  <param name="register-transport" value="udp"/>
  <param name="retry_seconds" value="300"/>
  </gateway>
</include>

I don't want the PBX to try to authenticate because I can't define a
username nor a password for the authentication.
Thank for you time.
Best regards,
Julien GONZALEZ.










From a.afzali2003 at gmail.com  Mon Aug  3 09:01:34 2009
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 3 Aug 2009 20:31:34 +0430
Subject: [Freeswitch-users] Missing mod_curl
In-Reply-To: <396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>
References: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
	<396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>
Message-ID: <debff5a80908030901w4f88767aw4f5e39d34adcaf9a@mail.gmail.com>

I've gotten 1.0.4pre9 , but i can not see it :(
-- afshin


On Mon, Aug 3, 2009 at 6:33 PM, Michael Jerris <mike at jerris.com> wrote:

> It is still there.
>
> On Aug 3, 2009, at 8:38 AM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
>
> > Hi,
> >
> > I'll appreciate if somebody tell me where has gone the mod_curl ? I
> > just need to use it for http method calls.
> >
> > Regards,
> > -- afshin
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Mon Aug  3 09:12:17 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 4 Aug 2009 00:12:17 +0800
Subject: [Freeswitch-users] Missing mod_curl
In-Reply-To: <debff5a80908030901w4f88767aw4f5e39d34adcaf9a@mail.gmail.com>
References: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
	<396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>
	<debff5a80908030901w4f88767aw4f5e39d34adcaf9a@mail.gmail.com>
Message-ID: <A948E259-A60C-484C-9113-E79A5F0E4C64@gmail.com>


On Aug 4, 2009, at 12:01 AM, afshin afzali wrote:
> I've gotten 1.0.4pre9 , but i can not see it :(
> -- afshin
>

In trunk.

>
> On Mon, Aug 3, 2009 at 6:33 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> It is still there.
>
> On Aug 3, 2009, at 8:38 AM, afshin afzali <a.afzali2003 at gmail.com>
> wrote:
>
> > Hi,
> >
> > I'll appreciate if somebody tell me where has gone the mod_curl ? I
> > just need to use it for http method calls.
> >
> > Regards,
> > -- afshin
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From a.afzali2003 at gmail.com  Mon Aug  3 09:19:17 2009
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Mon, 3 Aug 2009 20:49:17 +0430
Subject: [Freeswitch-users] Missing mod_curl
In-Reply-To: <A948E259-A60C-484C-9113-E79A5F0E4C64@gmail.com>
References: <debff5a80908030638y72c7afa1w42db3c7c397ca640@mail.gmail.com>
	<396D78F0-F5C0-46CC-BCDD-4CAE73B35F52@jerris.com>
	<debff5a80908030901w4f88767aw4f5e39d34adcaf9a@mail.gmail.com>
	<A948E259-A60C-484C-9113-E79A5F0E4C64@gmail.com>
Message-ID: <debff5a80908030919k7f182f3aifef607846dc9fb88@mail.gmail.com>

Thanks

On Mon, Aug 3, 2009 at 8:42 PM, Seven Du <dujinfang at gmail.com> wrote:

>
> On Aug 4, 2009, at 12:01 AM, afshin afzali wrote:
> > I've gotten 1.0.4pre9 , but i can not see it :(
> > -- afshin
> >
>
> In trunk.
>
> >
> > On Mon, Aug 3, 2009 at 6:33 PM, Michael Jerris <mike at jerris.com>
> > wrote:
> > It is still there.
> >
> > On Aug 3, 2009, at 8:38 AM, afshin afzali <a.afzali2003 at gmail.com>
> > wrote:
> >
> > > Hi,
> > >
> > > I'll appreciate if somebody tell me where has gone the mod_curl ? I
> > > just need to use it for http method calls.
> > >
> > > Regards,
> > > -- afshin
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From darren at dmmhosting.co.uk  Mon Aug  3 10:25:37 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Mon, 3 Aug 2009 18:25:37 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <718419f1-d02d-4623-bd5a-7852608df505@dmmhosting.co.uk>

Definitely the same reply:
 
FreeSWITCH Version 1.0.trunk (14457)
 
[ERR] sofia_reg.c:1460 05061292117 Registration Failed with status DNS Error [503]. failure #1 and so on.
 
The complete config is:
 
<include>
  <gateway name="05061292117">
  <param name="username" value="05061292117"/>
  <param name="realm" value="bmnha-01.bt.com"/>
  <param name="from-user" value="05061292117"/>
  <param name="from-domain" value="bmnha-01.bt.com"/>
  <param name="password" value="XXXXXXXXXXXXXXX"/>
  <param name="proxy" value="bmnha-01.bt.com"/>
  <param name="register-proxy" value="bmnha-01.bt.com"/>
  <param name="outbound-proxy" value="www.bbvservice-560129.bt.com"/>
  <param name="register" value="true"/>
  <param name="register-transport" value="udp"/>
  <param name="retry-seconds" value="10"/>
  </gateway>
</include>
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 03 August 2009 13:41
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
You must be on SVN trunk.
 
/b
 
On Aug 3, 2009, at 4:12 AM, Darren Williams wrote:



Brian, this ?broken? business explains a lot, I just assumed this was a normal practise. This ?outbound-proxy? parameter, I don?t see any reference to this anywhere.
 
This just causes Registration Failed with status DNS Error [503]
 
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From merul at mac.com  Mon Aug  3 10:56:43 2009
From: merul at mac.com (Merul Patel)
Date: Mon, 03 Aug 2009 18:56:43 +0100
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <mailman.816.1249310297.2787.freeswitch-users@lists.freeswitch.org>
References: <mailman.816.1249310297.2787.freeswitch-users@lists.freeswitch.org>
Message-ID: <EE4A46F0-42E4-4305-A2DC-EFC4F7401DE1@mac.com>

> What is the output of wantouter hwprobe?

voyage:~# wanrouter hwprobe

-------------------------------
| Wanpipe Hardware Probe Info |
-------------------------------
1 . U100       : BUSID=1-1 : V=00

Card Cnt: U100=1

voyage:~# dahdi_hardware
usb:001/002          wanpipe-     10c4:8461 Sangoma WANPIPE USB-FXO  
Device


>
> On Aug 3, 2009, at 3:12 AM, Merul Patel <merul at mac.com> wrote:
>
>>>> I'm new to FS, and experimenting with it on a constrained
>>>> environment (PCEngines ALIX board running Voyage Linux 0.62).
>>>>
>>>> So far, FS has compiled fine, and I can register multiple
>>>> softphones and make calls between them, but I'm lost at how to
>>>> configure a Sangoma U100 so I can make and receive calls over an
>>>> analogue line.
>>>>
>>>> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the
>>>> wanrouter utility detects the USB device, and I've compiled it to
>>>> support the TDM API.
>>>>
>>>> FS was compiled with the Openzap module - as best as I can tell.
>>>>
>>>> I thought that I would be able to use the wancfg_tdmapi utility to
>>>> configure the /etc/wanpipe/wanpipe1.conf and then use the generated
>>>> configuration file as the basis for configuring autoload_configs/
>>>> openzap.conf.xml.
>>>
>>> try wancfg_fs
>>
>> Thanks for the suggestion Michael, but the same result occurs as when
>> I try wancfg_tdmapi, ie:
>>
>> "No Sangoma voice compatible cards found/configured"
>>
>>>
>>>> However, the wancfg utility doesn't generate the wanpipe1.conf, and
>>>> I'm stumped.
>>>>
>>>> Any pointers would be much appreciated.
>>>>




From dave at 3c.co.uk  Mon Aug  3 10:57:14 2009
From: dave at 3c.co.uk (David Knell)
Date: Mon, 3 Aug 2009 18:57:14 +0100
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
Message-ID: <01A26C17-A286-4A71-8B9C-8E558E963501@3c.co.uk>



-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk



On 3 Aug 2009, at 13:41, Brian West <brian at freeswitch.org> wrote:

> Just look for large groups of people with laptops.  I'm sure you can't
> miss us.
>
> /b
>
> On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:
>
>> i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
>> afternoon?  anybody need help with setup?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mike at jerris.com  Mon Aug  3 11:14:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 3 Aug 2009 13:14:27 -0500
Subject: [Freeswitch-users] Configuring Sangoma U100
In-Reply-To: <EE4A46F0-42E4-4305-A2DC-EFC4F7401DE1@mac.com>
References: <mailman.816.1249310297.2787.freeswitch-users@lists.freeswitch.org>
	<EE4A46F0-42E4-4305-A2DC-EFC4F7401DE1@mac.com>
Message-ID: <019E4A58-9EC6-4198-A041-DFE2C1219E14@jerris.com>

I have yet to configure one of these cards for freeswitch so it's  
possible it's not in the config util yet.  I suggest contacting  
sangoma support for assistance

On Aug 3, 2009, at 12:56 PM, Merul Patel <merul at mac.com> wrote:

>> What is the output of wantouter hwprobe?
>
> voyage:~# wanrouter hwprobe
>
> -------------------------------
> | Wanpipe Hardware Probe Info |
> -------------------------------
> 1 . U100       : BUSID=1-1 : V=00
>
> Card Cnt: U100=1
>
> voyage:~# dahdi_hardware
> usb:001/002          wanpipe-     10c4:8461 Sangoma WANPIPE USB-FXO
> Device
>
>
>>
>> On Aug 3, 2009, at 3:12 AM, Merul Patel <merul at mac.com> wrote:
>>
>>>>> I'm new to FS, and experimenting with it on a constrained
>>>>> environment (PCEngines ALIX board running Voyage Linux 0.62).
>>>>>
>>>>> So far, FS has compiled fine, and I can register multiple
>>>>> softphones and make calls between them, but I'm lost at how to
>>>>> configure a Sangoma U100 so I can make and receive calls over an
>>>>> analogue line.
>>>>>
>>>>> I've installed the wanpipe drivers (3.5.4) from Sangoma, and the
>>>>> wanrouter utility detects the USB device, and I've compiled it to
>>>>> support the TDM API.
>>>>>
>>>>> FS was compiled with the Openzap module - as best as I can tell.
>>>>>
>>>>> I thought that I would be able to use the wancfg_tdmapi utility to
>>>>> configure the /etc/wanpipe/wanpipe1.conf and then use the  
>>>>> generated
>>>>> configuration file as the basis for configuring autoload_configs/
>>>>> openzap.conf.xml.
>>>>
>>>> try wancfg_fs
>>>
>>> Thanks for the suggestion Michael, but the same result occurs as  
>>> when
>>> I try wancfg_tdmapi, ie:
>>>
>>> "No Sangoma voice compatible cards found/configured"
>>>
>>>>
>>>>> However, the wancfg utility doesn't generate the wanpipe1.conf,  
>>>>> and
>>>>> I'm stumped.
>>>>>
>>>>> Any pointers would be much appreciated.
>>>>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From msc at freeswitch.org  Mon Aug  3 11:13:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Aug 2009 11:13:51 -0700
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <01A26C17-A286-4A71-8B9C-8E558E963501@3c.co.uk>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
	<01A26C17-A286-4A71-8B9C-8E558E963501@3c.co.uk>
Message-ID: <87f2f3b90908031113u441f1091h63289b79f12ddc84@mail.gmail.com>

Excellent! Many of us are at the hotel already, getting everything set up.
-MC

On Mon, Aug 3, 2009 at 10:57 AM, David Knell <dave at 3c.co.uk> wrote:

>
>
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
>
> On 3 Aug 2009, at 13:41, Brian West <brian at freeswitch.org> wrote:
>
> > Just look for large groups of people with laptops.  I'm sure you can't
> > miss us.
> >
> > /b
> >
> > On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:
> >
> >> i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
> >> afternoon?  anybody need help with setup?
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Aug  3 11:16:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 3 Aug 2009 11:16:33 -0700
Subject: [Freeswitch-users] telnet to event socket
In-Reply-To: <645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>
References: <4A76F494.5020101@gmx.de>
	<645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>
Message-ID: <87f2f3b90908031116i2ab7d7b5o38fda7ce8d9b36b5@mail.gmail.com>

On Mon, Aug 3, 2009 at 7:51 AM, Seven Du <dujinfang at gmail.com> wrote:

>
> On Aug 3, 2009, at 10:30 PM, Ngo-Vi Hoai-Anh wrote:
> > Hi,
> >
> > I'm taking a close look at event socket on FS 1.0.3. Configuration is
> > the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and
> > fsconsole.pl work but I was not able to telnet to port 8021. As I've
> > done that I received somewhat like:
> > #>auth/request
> >
> > I typed in: auth ClueCon
> >
> followed by two Enters (\n\n).
>
> > After some seconds I've got the message 'connection close by foreign
> > host'
> >
>

Don't forget that you can change the password by modifying the value in
freeswitch/conf/autoload_configs/event_socket.conf.xml"

<param name="password" value="ClueCon"/>

-MC
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From testa at voicetechnology.com.br  Mon Aug  3 11:17:27 2009
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Mon, 3 Aug 2009 15:17:27 -0300
Subject: [Freeswitch-users] FS beats Aculab Prosody S on subjective test
	on lay users for conference quality
In-Reply-To: <4A71AA86.90707@coppice.org>
References: <9cb0e15e0907290753q6ec3d45x680f5020a07699ed@mail.gmail.com>
	<1248886530.3818.2.camel@dk-d820> <4A70F593.5080204@coppice.org>
	<1248958990.4428.15.camel@dk-d820> <4A71AA86.90707@coppice.org>
Message-ID: <9cb0e15e0908031117p79355d51qf49821277d009f7d@mail.gmail.com>

Hi, a bit late on answering some of the questions, but, here we go.On
Aculab, all codecs were G711u. The same codec we have on FS:

freeswitch at conference> show channels
API CALL [show(channels)] output:
uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,write_codec,write_rate,secure
c9cafeb8-803a-11de-8ceb-cb8648fd1ccf,inbound,2009-08-03
11:34:44,1249310084,sofia/internal/1000 at 192.168.0.40,CS_EXECUTE,Teste
Testa,1000,192.168.0.165,3200,conference,3200-192.168.0.40 at ultrawideband
,XML,default,L16,8000,PCMU,8000,
cc5edb90-803a-11de-8ceb-cb8648fd1ccf,inbound,2009-08-03
11:34:49,1249310089,sofia/internal/1000 at 192.168.0.40
,CS_EXECUTE,F.G.Testa,1000,192.168.0.249,3200,conference,3200-192.168.0.40 at ultrawideband
,XML,default,L16,8000,PCMU,8000,

2 total.

freeswitch at conference> conference list
API CALL [conference(list)] output:
Conference 3200-192.168.0.40 (2 members)
2;sofia/internal/1000 at 192.168.0.40
;cc5edb90-803a-11de-8ceb-cb8648fd1ccf;F.G.Testa;1000;hear|speak;0;0;300
1;sofia/internal/1000 at 192.168.0.40;c9cafeb8-803a-11de-8ceb-cb8648fd1ccf;Teste
Testa;1000;hear|speak|talking|floor;0;0;300

I think this answers some questions from Michael.
A packet dump I don't have right now.

Fernando G. Testa


On Thu, Jul 30, 2009 at 11:13 AM, Steve Underwood <steveu at coppice.org>wrote:

> David Knell wrote:
> > On Thu, 2009-07-30 at 09:21 +0800, Steve Underwood wrote:
> >
> >>
> >> High quality conferencing is a difficult task, and still a research
> >> topic. No two conferencing systems perform alike. The interesting thing
> >> about this and other reports is that the conferencing in Freeswitch is
> >> not very clever right now, yet people are already saying it beats
> >> various other offerings, including long time commercial offerings.
> >>
> >
> > It may well be that a simplistic implementation (noise gate, add them
> > all up) is all that's required for dealing with small groups or, more
> > generally, groups of any size which have a small number of active
> > speakers at any one time: it's predictable and unlikely to introduce
> > unpleasant side effects.
> >
> This is one of those situations where when you've experienced something
> better you make that your baseline for acceptability. I would consider a
> noise gate horribly crude, and VAD as the minimum for acceptable
> performance. If you've only used a noise gate you get used to it. If
> you're not sufficiently versed in the art you may well think nothing
> better is even possible.
>
> The fact that even the simple scheme, with noise gating, in Freeswitch
> is getting high praise, is pretty damning of mature commercial products.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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From pgrondin at ip5.com  Mon Aug  3 11:53:11 2009
From: pgrondin at ip5.com (Patrick Grondin)
Date: Mon, 3 Aug 2009 14:53:11 -0400
Subject: [Freeswitch-users] Using tone_detect application
In-Reply-To: <6C525B211BA8F44CAC96A4F68333A9D119326E0813@VMBX107.ihostexchange.net>
References: <6C525B211BA8F44CAC96A4F68333A9D119326E0813@VMBX107.ihostexchange.net>
Message-ID: <6C525B211BA8F44CAC96A4F68333A9D119326E10A6@VMBX107.ihostexchange.net>

Hi,

What's the correct way to detect a number changed SIT tone in Freeswitch ?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Patrick Grondin
Sent: July-28-09 3:44 PM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Using tone_detect application

Hi,

I'm doing some tests between 2 FS to understand how the tone_detect application works.

I'm trying to detect a SIT tone, but I can't seem to detect the 3 tones. I only get the first activated tone.

If I have all tones activated - - - > I detect only the first tone of my wav file.
If I have 2nd segment tones and up activated - - - > I detect only the second tone of my wav file.
If I have the 3rd segment tones activated - - - > I detect only the third tone of my wav file.

I see that tone_detect can detect all 3 tones, but never at the same time. Does anyone have an idea of what I could be doing wrong ? Thanks !

I'm using FreeSWITCH Version 1.0.trunk (14397).

My dialplan looks like this :

    <extension name="incoming">
      <condition field="destination_number" expression="^(514[0-9][0-9][0-9][0-9][0-9][0-9][0-9])$">
        <action application="answer"/>
        <action application="tone_detect" data="SIT1LO 913.8 w +25000 set SIT1LO=true"/>   <!--1st segment low -->
        <action application="tone_detect" data="SIT1HI 985.2 w +25000 set SIT1HI=true"/>   <!--1st segment high -->
        <action application="tone_detect" data="SIT2LO 1370.6 w +25000 set SIT2LO=true"/>   <!--2nd segment low -->
        <action application="tone_detect" data="SIT2HI 1428.5 w +25000 set SIT2HI=true"/>   <!--2nd segment high -->
        <action application="tone_detect" data="SIT3LO 1776.7 w +25000 set SIT3LO=true"/>  <!--3rd segment low -->
        <action application="tone_detect" data="SITITU1 950 w +25000 set SITITU1=true"/>   <!--2nd segment low -->
        <action application="tone_detect" data="SITITU2 1400 w +25000 set SITITU2=true"/>   <!--2nd segment high -->
        <action application="tone_detect" data="SITITU3 1800 w +25000 set SITITU3=true"/>  <!--3rd segment low -->
        <action application="bridge" data="sofia/internal/$1 at 192.168.100.9"/>
      </condition>
    </extension>
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From raffaele.p.guidi at gmail.com  Mon Aug  3 12:13:14 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Mon, 3 Aug 2009 21:13:14 +0200
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
In-Reply-To: <87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>
References: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
	<87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>
Message-ID: <b0a8792f0908031213r37c10f80m9d076f195b1ba83f@mail.gmail.com>

Yeah, it's there but, as you can see it from the google queries below it's
not easy to find that page - unless you know what exactly you are looking
for and search for "loopback". A simple example (i.e. in Sofia Syntax) would
fill the gap (I'll be happy to do it ASAP)

http://www.google.com/search?hl=en&safe=off&q=freeswitch+call+url&aq=f&oq=&aqi=

http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=<http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=>
http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+esl&aq=f&oq=&aqi=

Also, this email now shows up first in googling for "non user extension
freeswitch" (which were the keywords I was looking for). I think this has
been useful! ;)

Many thanks and regards,
   Raffaele

On Mon, Aug 3, 2009 at 14:59, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> I found the answer by myself while I had finished writing the e-mail. The
>> correct call url is loopback/<extension> (in this case the command
>> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
>> future reference (can't find any example of that anywhere). Is the project
>> wiki accesible for anyone to contribute or do I have to ask for an
>> authorization?
>>
>
> All you need to do is sign up for a free account on the wiki and you can
> start editing. It's a community resource and all FS users are invited to add
> their respective knowledge.
>
> As for not finding what you were looking for, does this page not have it?
> http://wiki.freeswitch.org/wiki/Loopback
>
> If not then please feel free to add to this page whatever your specific
> scenario entails and give some examples.
>
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From Prometheus001 at gmx.net  Mon Aug  3 12:40:36 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 03 Aug 2009 21:40:36 +0200
Subject: [Freeswitch-users] TDM API: CMD: 18 : Operation not supported
Message-ID: <4A773D34.300@gmx.net>

Hello,

I setup libpri and a sangoma card A108DE, but I cannot dial out.
At startup I receive on the D channel
    TDM API: CMD: 18 : Operation not supported

When dialling Libpri debug shows that the numbering plan is fine and
that it accepts the screened number, but then it finally hangs up with:
       Ext: 1  Cause: Info. element nonexist or not implemented (99),
class = Protocol Error (e.g. unknown message) (6) ]

Does lipri sent any incorrect message here? Protocol is EuroISDN
(Q.931/Q.921).
Anybody has discovered this already?

I am on trunk 14419.

See debug and configs below.

Best regards
Peter

Starting FS:
2009-08-03 21:37:18.264829 [DEBUG] zap_io.c:2281 span 1 [d-channel]=[1:16]
TDM API: CMD: 18
: Operation not supported
2009-08-03 21:37:18.264915 [INFO] ozmod_wanpipe.c:287 configuring device
s1c16 as OpenZAP device 1:16 fd:55 DTMF: none
2009-08-03 21:37:18.264929 [DEBUG] zap_io.c:2281 span 1
[b-channel]=[1:17-31]

Dialling:
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 Handling message
for SAPI/TEI=0/0
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- ACKing all
packets from 3 to (but not including) 4
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Since there was
nothing left, stopping T200 counter
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Stopping T203
counter since we got an ACK
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Nothing left,
starting T203 counter
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Protocol
Discriminator: Q.931 (8)  len=14
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Call Ref: len= 2
(reference 5/0x5) (Terminator)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Message type:
STATUS (125)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < [08 04 82 e3 98 74]
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Cause (len= 6) [
Ext: 1  Coding: CCITT (ITU) standard (0)  Spare: 0  Location: Public
network serving the local user (2)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 <                 
Ext: 1  Cause: Info. element nonexist or not implemented (99), class =
Protocol Error (e.g. unknown message) (6) ]
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 <             
Cause data 1: 98 (152, Non-Locking Shift To Codeset 0 IE)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 <             
Cause data 2: 74 (116, Redirecting Number IE)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < [14 01 01]
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 < Call State (len=
3) [ Ext: 0  Coding: CCITT (ITU) standard (0)  Call state: Call
Initiated (1)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Processing IE 8
(cs0, Cause)
2009-08-03 21:16:19.441416 [DEBUG] ozmod_libpri.c:106 -- Processing IE
20 (cs0, Call State)
2009-08-03 21:16:19.441416 [ERR] ozmod_libpri.c:88 Received unsolicited
status: Info. element nonexist or not implemented

openzap.conf
[span wanpipe PRI_1]
number => 1
trunk_type => E1
b-channel => 1:1-15
d-channel => 1:16
b-channel => 1:17-31

openzap.conf.xml
<configuration name="openzap.conf" description="OpenZAP Configuration">
   <settings>
     <param name="debug" value="10"/>
   </settings>
   <libpri_spans>
     <span name="PRI_1">
        <param name="node" value="cpe"/>
        <param name="dp" value="national"/>
        <param name="l1" value="alaw"/>
        <param name="switch" value="euroisdn"/>
        <param name="dialplan" value="XML"/>
        <param name="context" value="public"/>
      </span>
   </libpri_spans>





From brian at freeswitch.org  Mon Aug  3 14:24:11 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 16:24:11 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <718419f1-d02d-4623-bd5a-7852608df505@dmmhosting.co.uk>
References: <718419f1-d02d-4623-bd5a-7852608df505@dmmhosting.co.uk>
Message-ID: <C925BB2D-18A5-4741-95A3-F82D9B1F546C@freeswitch.org>

which of the hostnames is fake?

/b

On Aug 3, 2009, at 12:25 PM, Darren Williams wrote:

> Definitely the same reply:
>

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From darren at dmmhosting.co.uk  Mon Aug  3 14:39:00 2009
From: darren at dmmhosting.co.uk (Darren Williams)
Date: Mon, 3 Aug 2009 22:39:00 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <c857d204-9c47-4185-ba83-56737ac47462@dmmhosting.co.uk>

The bmnha-01.bt.com is the one that doesn?t resolve
www.bbvservice-560129.bt.com resolves to 62.239.15.140
 
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 03 August 2009 22:24
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
which of the hostnames is fake?
 
/b
 
On Aug 3, 2009, at 12:25 PM, Darren Williams wrote:



Definitely the same reply:
 
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From dule.maillist at gmail.com  Mon Aug  3 15:06:46 2009
From: dule.maillist at gmail.com (Dan Le)
Date: Mon, 3 Aug 2009 18:06:46 -0400
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
In-Reply-To: <b0a8792f0908031213r37c10f80m9d076f195b1ba83f@mail.gmail.com>
References: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
	<87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>
	<b0a8792f0908031213r37c10f80m9d076f195b1ba83f@mail.gmail.com>
Message-ID: <914fc92a0908031506t285d2719need5f8937c0bb2a6@mail.gmail.com>

Technically, loopback is not meant to specifically allow you to call
non-user extensions, it simply allows you to hit the dialplan.
For example,

originate sofia/gw/gwname/fakecall 1000

(where gwname is the gateway routing your calls out, and 'fakecall' the
non-user extension)

will also allow you to dial to non-user extensions. This information would
be on any of the wiki pages detailing the originate command.

Dan


On Mon, Aug 3, 2009 at 3:13 PM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> Yeah, it's there but, as you can see it from the google queries below it's
> not easy to find that page - unless you know what exactly you are looking
> for and search for "loopback". A simple example (i.e. in Sofia Syntax) would
> fill the gap (I'll be happy to do it ASAP)
>
>
> http://www.google.com/search?hl=en&safe=off&q=freeswitch+call+url&aq=f&oq=&aqi=
>
>
> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=
> <http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=>
> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+esl&aq=f&oq=&aqi=
>
> Also, this email now shows up first in googling for "non user extension
> freeswitch" (which were the keywords I was looking for). I think this has
> been useful! ;)
>
> Many thanks and regards,
>    Raffaele
>
> On Mon, Aug 3, 2009 at 14:59, Michael Collins <msc at freeswitch.org> wrote:
>
>>
>>
>> On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
>> raffaele.p.guidi at gmail.com> wrote:
>>
>>> I found the answer by myself while I had finished writing the e-mail. The
>>> correct call url is loopback/<extension> (in this case the command
>>> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
>>> future reference (can't find any example of that anywhere). Is the project
>>> wiki accesible for anyone to contribute or do I have to ask for an
>>> authorization?
>>>
>>
>> All you need to do is sign up for a free account on the wiki and you can
>> start editing. It's a community resource and all FS users are invited to add
>> their respective knowledge.
>>
>> As for not finding what you were looking for, does this page not have it?
>> http://wiki.freeswitch.org/wiki/Loopback
>>
>> If not then please feel free to add to this page whatever your specific
>> scenario entails and give some examples.
>>
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From raffaele.p.guidi at gmail.com  Mon Aug  3 15:38:43 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Tue, 4 Aug 2009 00:38:43 +0200
Subject: [Freeswitch-users] How to call a non-user extension from
	"originate" [SOLVED]
In-Reply-To: <914fc92a0908031506t285d2719need5f8937c0bb2a6@mail.gmail.com>
References: <b0a8792f0908030443y4fe89c94pea5470c45182ae97@mail.gmail.com>
	<87f2f3b90908030559g63093224qea04be7cdb0d2b26@mail.gmail.com>
	<b0a8792f0908031213r37c10f80m9d076f195b1ba83f@mail.gmail.com>
	<914fc92a0908031506t285d2719need5f8937c0bb2a6@mail.gmail.com>
Message-ID: <b0a8792f0908031538g20abf2e0yba8dab8ac1283305@mail.gmail.com>

uhm... does it?

freeswitch at W2GZ8VNR01> originate sofia/gateway/callwithus.com/fakecall 1001
....
API CALL [originate(sofia/gateway/callwithus.com/fakecall 1001)] output:
-ERR CALL_REJECTED


Anyway, I clearly understand that loopback allows to "hit the dialplan",
maybe I couldn't find it before because of my poor english (but googling for
"freeswitch hit the dialplan" doesn't help either). Really, I simply had
problems finding it - the information was there (is it a matter of SEO -
Search Engine Optimization ;)?

Regards,
    Raffaele

On Tue, Aug 4, 2009 at 00:06, Dan Le <dule.maillist at gmail.com> wrote:

> Technically, loopback is not meant to specifically allow you to call
> non-user extensions, it simply allows you to hit the dialplan.
> For example,
>
> originate sofia/gw/gwname/fakecall 1000
>
> (where gwname is the gateway routing your calls out, and 'fakecall' the
> non-user extension)
>
> will also allow you to dial to non-user extensions. This information would
> be on any of the wiki pages detailing the originate command.
>
> Dan
>
>
> On Mon, Aug 3, 2009 at 3:13 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> Yeah, it's there but, as you can see it from the google queries below it's
>> not easy to find that page - unless you know what exactly you are looking
>> for and search for "loopback". A simple example (i.e. in Sofia Syntax) would
>> fill the gap (I'll be happy to do it ASAP)
>>
>>
>> http://www.google.com/search?hl=en&safe=off&q=freeswitch+call+url&aq=f&oq=&aqi=
>>
>>
>> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=
>> <http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+cli+&aq=f&oq=&aqi=>
>> http://www.google.com/search?hl=en&safe=off&q=freeswitch+execute+non+user+extension+from+the+esl&aq=f&oq=&aqi=
>>
>> Also, this email now shows up first in googling for "non user extension
>> freeswitch" (which were the keywords I was looking for). I think this has
>> been useful! ;)
>>
>> Many thanks and regards,
>>    Raffaele
>>
>> On Mon, Aug 3, 2009 at 14:59, Michael Collins <msc at freeswitch.org> wrote:
>>
>>>
>>>
>>> On Mon, Aug 3, 2009 at 4:43 AM, Raffaele P. Guidi <
>>> raffaele.p.guidi at gmail.com> wrote:
>>>
>>>> I found the answer by myself while I had finished writing the e-mail.
>>>> The correct call url is loopback/<extension> (in this case the command
>>>> is originate loopback/fakecall 1000 ).  I'm sending the e-mail anyway for
>>>> future reference (can't find any example of that anywhere). Is the project
>>>> wiki accesible for anyone to contribute or do I have to ask for an
>>>> authorization?
>>>>
>>>
>>> All you need to do is sign up for a free account on the wiki and you can
>>> start editing. It's a community resource and all FS users are invited to add
>>> their respective knowledge.
>>>
>>> As for not finding what you were looking for, does this page not have it?
>>> http://wiki.freeswitch.org/wiki/Loopback
>>>
>>> If not then please feel free to add to this page whatever your specific
>>> scenario entails and give some examples.
>>>
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Aug  3 20:36:19 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 3 Aug 2009 22:36:19 -0500
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <c857d204-9c47-4185-ba83-56737ac47462@dmmhosting.co.uk>
References: <c857d204-9c47-4185-ba83-56737ac47462@dmmhosting.co.uk>
Message-ID: <A48DE4E2-997A-4E49-AB47-A12D5E9ABFF7@freeswitch.org>

Put the real hostname in register-proxy and outbound-proxy and the  
proxy needs to hold the fake one.

/b

On Aug 3, 2009, at 4:39 PM, Darren Williams wrote:

> The bmnha-01.bt.com is the one that doesn?t resolve
> www.bbvservice-560129.bt.com resolves to 62.239.15.140
>

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From thangappan143 at gmail.com  Mon Aug  3 22:47:03 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Tue, 4 Aug 2009 11:17:03 +0530
Subject: [Freeswitch-users] Fwd: Need Help In IVR
In-Reply-To: <7aa29e790908030012rcce847fu8320d833d2c85530@mail.gmail.com>
References: <7aa29e790908030012rcce847fu8320d833d2c85530@mail.gmail.com>
Message-ID: <7aa29e790908032247h2a93e838m1ac5fd3f8a3946c7@mail.gmail.com>

Can you please help me?


---------- Forwarded message ----------
From: Thangappan.M <thangappan143 at gmail.com>
Date: Mon, Aug 3, 2009 at 12:42 PM
Subject: Need Help In IVR
To: freeswitch-users <freeswitch-users at lists.freeswitch.org>


Dear all,

I am in the process of implementing IVR in Perl using outbound socket.

In the case of the XML macro I can easily specify the timeout,inter digit
timeout value as
 <menu name="demo_ivr"
          greet-long="phrase:demo_ivr_main_menu"
          greet-short="phrase:demo_ivr_main_menu_short"
          invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav"
          exit-sound="voicemail/vm-goodbye.wav"
          timeout ="10000"
          inter-digit-timeout="2000"
          max-failures="3"
          max-timeouts="3"
          digit-len="4">
      <entry action="menu-exec-app" digits="1" param="bridge
sofia/$${domain}/888 at conference.freeswitch.org"/>
    </menu>

Is there any way for Perl to configure this values. Where are the variables
resides?
I am struggling to implement this?




-- 
Regards,
Thangappan.M



-- 
Regards,
Thangappan.M
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From dujinfang at gmail.com  Tue Aug  4 00:17:32 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 4 Aug 2009 15:17:32 +0800
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH Story
Message-ID: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>

Hello All -   In the spirit of ClueCon (which we are missing this year, but
hopefully not next), we wanted to document our "FreeSWITCH Story".  We've
posted it to the wiki(
http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com) and
it is copied below.

Thank you all and enjoy a good conference!

Seven Du (seven)
Jonathan Palley (jpalley_idapted)
Idapted Ltd.


*How FreeSWITCH has created hundreds of job opportunities and changed
lives. *

We want to share our experience working with FreeSWITCH.  FreeSWITCH has
been a key enabler of our business.  We hope this story can be a small way
to say a very big THANK YOU ALL.

"Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
really allowed us to do just that.

What We Do:
We are not a telephony business; we are an educational technology and
service business. In Asia (China, in our case) students must pass English
examinations to study or work abroad and gain new experiences.  However,
there is limited access to native English speakers and the access students
can gain is typically very expensive.  At the same time, in the U.S., there
are many professionals looking for work-at-home opportunities - people who
need jobs and would create great teachers.  Through our technology and
content we empower these people to be effective English teachers.  Does it
work?  Yes.  The majority of our students are getting test scores that many
failed for years to get.  Just hours ago one student called one of our sales
agents crying with joy.  And for our teachers, they are now working in an
industry that was previously unavailable to those living in the U.S.
http://www.idapted.com

Why FreeSWITCH Enables This:
FreeSWITCH has been a key enabler of our business.  Recording calls,
controlling routing, integrating with various web-based interfaces, enabling
multiple endpoints - these are all key features of what we must do.  Most
importantly, setting up various servers and routes to mitigate cross-Pacific
and country-specific network challenges is key.  Doing what we are doing
with commercial solutions would have made the business unworkable.

Our Experiences with FreeSWITCH:
We started using FreeSWITCH as our VoIP Platform in April 2008, after
receiving unsatisfactory results with other open source solutions.  It took
one day of reading through the FreeSWITCH source code to know, "this is it.
 This is the VoIP platform we build our business on".  It took a few days of
working with the extremely competent and focused community to re-affirm this
commitment.

Our Setup:
Our teachers use a custom software that integrates a VoIP client with our
web based platform. Students connect to our teachers "on-demand".  Simply
put, on a web-based comet interface the student enters a phone number (or a
skype name or a gtalk account) and our platform bridges the best available
trainer and the student.  At the same time a web-based interface is being
updated.

The challenge for us is the connection between teachers and students over a
cross-continent network. For example, we experienced problems earlier this
year when a Asis-Pacific communication fiber broken... So, we've learned to
setup multi servers in multiple datacenters for redundancy.

We run multi instances of FreeSWITCH so we can always use the cutting edge
and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
only loads mod_dingaling. Here is one beauty of FS: We just had to create
different conf dirs (/usr/local/freeswitch, /usr/local/skype,
/usr/local/gtalk etc). This allows us to run the same code base over
different configurations, and call skype and gtalk accounts just like a
normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
abnormally or crashes, we can easily change to another FS-skype server (we
run other servers located in various places in China and HK for
redundancy).

FS --|
     |---PSTN gateways
     |--- FS-skype
     |--- FS-gtalk
     |--- FS-skype2
     |--- more ...



COMMUNITY:

The community's commitment cannot be undervalued.  The insightful, modular
design of FreeSWITCH allows anyone to contribute, whereever their skills
lie.  It also allows us to easily make modifications to the underlying code
to suit our specific use-cases  We want to highlight a few key people and
modules in the FS ecosystem:

mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
clients.  PSTN is zero-conf for the user and mitigates troubles with the end
users network/microphone, etc (which is significant with our user base).
 However, cheap providers fail randomly and FreeSWITCH's ability to control
routing, use multiple endpoints all while clearly seeing what is going on is
key.
Most importantly, anthm and the core team have been super helpful in getting
SIP to work with us.  Back in the pre 1.0 days anthm made significant
changes to mod-sofia to enable clients behind nats without STUN.  Its
important to point out that he didn't just make the changes -he forced us to
really make a compelling case as to why the changes were important for
FreeSWITCH.  This is a good thing.

skype (mod_skypiax): Due to the facts that users prefer skype, we configured
skypiax. It was unstable at the beginning and that's one of the reason we
started running that separate FS instance.  To be fair, it has caused a lot
of trouble - but we know this, its new software that takes a big risk and
implements a complex hack.  What is important is that the author of
skypiax(Giovanni Maruzzelli) has been a huge help. He's been very active
fixing bugs and logging in to our box to help trouble shoot. We owe him a
*big* thanks.

To make Skypiax more useful, we also created some patches including the ANY
and RR interfaces for sequential and round robin line hunting, some bug
fixes and other features like continue-load-on-fail and auto-skype-user
which haven't been merged into trunk yet. Thanks a community that gives us a
platform where we can all benefit and contribute.

erlang (mod_erlang_events): Another key enabler of the next release of our
system is the erlang interface.  We have a complex realtime queue routing
system has it handles input not just from freeswitch, but numerous other web
interfaces and sockets.  Erlang was the perfect technology to implement this
in and luckily an Erlang module for FreeSWITCH was already written.
Beautiful.

THE MORAL OF THE STORY:
FreeSWITCH is a great piece of software that has enabled new technologies
and business models.  The design has allowed (and the core team has
nurtured) a vibrant and exciting community that has made the software even
better.  Every day we go to work excited to push the boundaries of what can
be done with telephony technology and are confident this is the platform of
the future.

Thank you all.


Sincerely,

Du Jinfang (Seven) - Technical Operations/VoIP Manager
Jonathan Palley - CTO
Idapted Ltd.
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From hoaianh at gmx.de  Tue Aug  4 00:39:24 2009
From: hoaianh at gmx.de (Ngo-Vi Hoai-Anh)
Date: Tue, 04 Aug 2009 09:39:24 +0200
Subject: [Freeswitch-users] telnet to event socket
In-Reply-To: <645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>
References: <4A76F494.5020101@gmx.de>
	<645C7F91-6E96-4C9A-AF9B-DB5D89A87A6B@gmail.com>
Message-ID: <4A77E5AC.9050501@gmx.de>

Hi,

Thank you. It works now. The clue is to hit 'Enter' twice.

Seven Du schrieb:
> On Aug 3, 2009, at 10:30 PM, Ngo-Vi Hoai-Anh wrote:
>   
>> Hi,
>>
>> I'm taking a close look at event socket on FS 1.0.3. Configuration is
>> the same as on http://wiki.freeswitch.org/wiki/Event_Socket. fs.pl and
>> fsconsole.pl work but I was not able to telnet to port 8021. As I've
>> done that I received somewhat like:
>> #>auth/request
>>
>> I typed in: auth ClueCon
>>
>>     
> followed by two Enters (\n\n).
>
>   
>> After some seconds I've got the message 'connection close by foreign  
>> host'
>>
>> Any ideas?
>>
>> Thank you
>> Hoai-Anh
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   




From digitaldaz100 at gmail.com  Tue Aug  4 02:10:10 2009
From: digitaldaz100 at gmail.com (Darren Williams)
Date: Tue, 4 Aug 2009 10:10:10 +0100
Subject: [Freeswitch-users] Outbound Proxy
Message-ID: <26f14763-6f40-4468-91f3-e60a46892183@dmmhosting.co.uk>

This produces:
 
send 639 bytes to udp/[62.239.15.140]:5060 at 09:06:29.640488:
   ------------------------------------------------------------------------
   REGISTER sip:bmnha-01.bt.com SIP/2.0
   Via: SIP/2.0/UDP 91.121.159.57:5080;rport;branch=z9hG4bKQFFy1Ug5DcU0S
   Max-Forwards: 70
   From: <sip:05061292117 at bmnha-01.bt.com;transport=udp>;tag=NKaDDXrtme3Sr
   To: <sip:05061292117 at bmnha-01.bt.com;transport=udp>
   Call-ID: 88de957f-0907-4488-9b19-043ff13391f4
   CSeq: 118570058 REGISTER
   Contact: <sip:gw+05061292117 at 91.121.159.57:5080;transport=udp>
   Expires: 3600
   User-Agent: THOMSON ST2030 hw5 fw1.56 00-1F-9F-16-4E-99
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
   Supported: timer, precondition, path, replaces
   Content-Length: 0
 
   ------------------------------------------------------------------------
recv 726 bytes from udp/[62.239.15.140]:5060 at 09:06:29.656368:
   ------------------------------------------------------------------------
   SIP/2.0 401 Unauthorized
   Via: SIP/2.0/UDP 91.121.159.57:5080;received=91.121.159.57;branch=z9hG4bKQFFy1Ug5DcU0S;rport=5080
   From: <sip:05061292117 at bmnha-01.bt.com;transport=udp>;tag=NKaDDXrtme3Sr
   To: <sip:05061292117 at bmnha-01.bt.com;transport=udp>;tag=SD567ec99-8a92eaf49d827b084b51e364ec68ae70.bc35
   Call-ID: 88de957f-0907-4488-9b19-043ff13391f4
   CSeq: 118570058 REGISTER
   WWW-Authenticate: Digest realm="bmnha-01.bt.com", nonce="4a77fb4b6fe33ffccf77e6bdd6f7a7a397cdfd17", qop="auth"
   Server: Sip EXpress router (0.9.6 (sparc/solaris))
   Content-Length: 0
   Warning: 392 sip:5060 "Noisy feedback tells:  pid=9150 req_src_ip=172.20.92.61 req_src_port=5060 in_uri=sip:bmnha-01.bt.com out_uri=sip:bmnha-01.bt.com via_cnt==1"
 
   ------------------------------------------------------------------------
2009-08-04 11:06:29.655585 [ERR] sofia_reg.c:1460 05061292117 Registration Failed with status Operation has no matching challenge  [904]. failure #1 and so on.
 
Should my side not have responded at this point with another REGISTER with auth information in?
 
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: 04 August 2009 04:36
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Outbound Proxy
 
Put the real hostname in register-proxy and outbound-proxy and the proxy needs to hold the fake one.
 
/b
 
On Aug 3, 2009, at 4:39 PM, Darren Williams wrote:



The bmnha-01.bt.com is the one that doesn?t resolve
www.bbvservice-560129.bt.com resolves to 62.239.15.140
 
 


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com


__________ Information from ESET NOD32 Antivirus, version of virus signature database 4299 (20090802) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com
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From markmorreny at gmail.com  Tue Aug  4 04:08:36 2009
From: markmorreny at gmail.com (mark morreny)
Date: Tue, 4 Aug 2009 19:08:36 +0800
Subject: [Freeswitch-users] event socket vs erlang
Message-ID: <20ad6b920908040408x48802b84mbf83e20a2cb5d2f@mail.gmail.com>

Hi,

I have seen people using both event socket and erlang to control freeSWITCH
externally.

What is the pros and cons of using event socket vs erlang?

Thanks,
Mark
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From rdenert at tng.de  Tue Aug  4 04:14:43 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Tue, 4 Aug 2009 13:14:43 +0200 (CEST)
Subject: [Freeswitch-users] Module in Lua not working
In-Reply-To: <19314518.221201249383746271.JavaMail.root@zimbra.tng.de>
Message-ID: <7203453.221221249384483704.JavaMail.root@zimbra.tng.de>

Hello again!

I need some help again, because I have little trouble with a few modules. The first one is luasocket, the seconde one is luasql. I always get the error:

error loading module 'socket' from file '/usr/local/lib/lua/5.1/socket/core.so': /usr/local/lib/lua/5.1/socket/core.so: undefined symbol: lua_getmetatable

and

error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': /usr/local/lib/lua/5.1/luasql/mysql.so: undefined symbol: lua_pushlstring

in the fs_cli

This looks like a conflict with an older version of these modules. Is this right? If yes, what should i update?

Thanks again.

BR

-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
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die unbefugte Weitergabe dieser Mail ist nicht gestattet.



From gmaruzz at celliax.org  Tue Aug  4 06:35:29 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 4 Aug 2009 15:35:29 +0200
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
Message-ID: <7b197bef0908040635k330c73cayc354748f40a3bf19@mail.gmail.com>

Cool!



On Tue, Aug 4, 2009 at 9:17 AM, Seven Du<dujinfang at gmail.com> wrote:
> Hello All -
> ?? In the spirit of ClueCon (which we are missing this year, but hopefully
> not next), we wanted to document our "FreeSWITCH Story". ?We've posted it to
> the
> wiki(http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
> and it is copied below.
> Thank you all and enjoy a good conference!
> Seven Du (seven)
> Jonathan Palley (jpalley_idapted)
> Idapted Ltd.
>
> How FreeSWITCH has created hundreds of job opportunities and changed lives.
> We want to share our experience working with FreeSWITCH. ?FreeSWITCH has
> been a key enabler of our business. ?We hope this story can be a small way
> to say a very big THANK YOU ALL.
> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
> really allowed us to do just that.
> What We Do:
> We are not a telephony business; we are an educational technology and
> service business. In Asia (China, in our case) students must pass English
> examinations to study or work abroad and gain new experiences. ?However,
> there is limited access to native English speakers and the access students
> can gain is typically very expensive. ?At the same time, in the U.S., there
> are many professionals looking for work-at-home opportunities - people who
> need jobs and would create great teachers. ?Through our technology and
> content we empower these people to be effective English teachers. ?Does it
> work? ?Yes. ?The majority of our students are getting test scores that many
> failed for years to get. ?Just hours ago one student called one of our sales
> agents crying with joy. ?And for our teachers, they are now working in an
> industry that was previously unavailable to those living in the U.S.
> ?http://www.idapted.com
> Why FreeSWITCH Enables This:
> FreeSWITCH has been a key enabler of our business. ?Recording calls,
> controlling routing, integrating with various web-based interfaces, enabling
> multiple endpoints - these are all key features of what we must do. ?Most
> importantly, setting up various servers and routes to mitigate cross-Pacific
> and country-specific network challenges is key. ?Doing what we are doing
> with commercial solutions would have made the business unworkable.
> Our Experiences with FreeSWITCH:
> We started using FreeSWITCH as our VoIP Platform in April 2008, after
> receiving unsatisfactory results with other open source solutions. ?It took
> one day of reading through the FreeSWITCH source code to know, "this is it.
> ?This is the VoIP platform we build our business on". ?It took a few days of
> working with the extremely competent and focused community to re-affirm this
> commitment.
> Our Setup:
> Our teachers use a custom software that integrates a VoIP client with our
> web based platform. Students connect to our teachers "on-demand". ?Simply
> put, on a web-based comet interface the student enters a phone number (or a
> skype name or a gtalk account) and our platform bridges the best available
> trainer and the student. ?At the same time a web-based interface is being
> updated.
> The challenge for us is the connection between teachers and students over a
> cross-continent network. For example, we experienced problems earlier this
> year when a Asis-Pacific communication fiber broken... So, we've learned to
> setup multi servers in multiple datacenters for redundancy.
>
> We run multi instances of FreeSWITCH so we can always use the cutting edge
> and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
> connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
> only loads mod_dingaling. Here is one beauty of FS: We just had to create
> different conf dirs (/usr/local/freeswitch, /usr/local/skype,
> /usr/local/gtalk etc). This allows us to run the same code base over
> different configurations, and call skype and gtalk accounts just like a
> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
> abnormally or crashes, we can easily change to another FS-skype server (we
> run other servers located in various places in China and HK for
> redundancy).
> FS --|
> ?? ? |---PSTN gateways
> ?? ? |--- FS-skype
> ?? ? |--- FS-gtalk
> ?? ? |--- FS-skype2
> ?? ? |--- more ...
>
>
> COMMUNITY:
> The community's commitment cannot be undervalued. ?The insightful, modular
> design of FreeSWITCH allows anyone to contribute, whereever their skills
> lie. ?It also allows us to easily make modifications to the underlying code
> to suit our specific use-cases ?We want to highlight a few key people and
> modules in the FS ecosystem:
> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
> clients. ?PSTN is zero-conf for the user and mitigates troubles with the end
> users network/microphone, etc (which is significant with our user base).
> ?However, cheap providers fail randomly and FreeSWITCH's ability to control
> routing, use multiple endpoints all while clearly seeing what is going on is
> key.
> Most importantly, anthm and the core team have been super helpful in getting
> SIP to work with us. ?Back in the pre 1.0 days anthm made significant
> changes to mod-sofia to enable clients behind nats without STUN. ?Its
> important to point out that he didn't just make the changes -he forced us to
> really make a compelling case as to why the changes were important for
> FreeSWITCH. ?This is a good thing.
> skype (mod_skypiax): Due to the facts that users prefer skype, we configured
> skypiax. It was unstable at the beginning and that's one of the reason we
> started running that separate FS instance. ?To be fair, it has caused a lot
> of trouble - but we know this, its new software that takes a big risk and
> implements a complex hack. ?What is important is that the author of
> skypiax(Giovanni Maruzzelli) has been a huge help. He's been very active
> fixing bugs and logging in to our box to help trouble shoot. We owe him a
> *big* thanks.
> To make Skypiax more useful, we also created some patches including the ANY
> and RR interfaces for sequential and round robin line hunting, some bug
> fixes and other features like continue-load-on-fail and auto-skype-user
> which haven't been merged into trunk yet. Thanks a community that gives us a
> platform where we can all benefit and contribute.
> erlang (mod_erlang_events): Another key enabler of the next release of our
> system is the erlang interface. ?We have a complex realtime queue routing
> system has it handles input not just from freeswitch, but numerous other web
> interfaces and sockets. ?Erlang was the perfect technology to implement this
> in and luckily an Erlang module for FreeSWITCH was already written.
> Beautiful.
> THE MORAL OF THE STORY:
> FreeSWITCH is a great piece of software that has enabled new technologies
> and business models. ?The design has allowed (and the core team has
> nurtured) a vibrant and exciting community that has made the software even
> better. ?Every day we go to work excited to push the boundaries of what can
> be done with telephony technology and are confident this is the platform of
> the future.
> Thank you all.
>
> Sincerely,
> Du Jinfang (Seven) - Technical Operations/VoIP Manager
> Jonathan Palley - CTO
> Idapted Ltd.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From andrew at hijacked.us  Tue Aug  4 06:49:27 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Tue, 4 Aug 2009 09:49:27 -0400
Subject: [Freeswitch-users] event socket vs erlang
In-Reply-To: <20ad6b920908040408x48802b84mbf83e20a2cb5d2f@mail.gmail.com>
References: <20ad6b920908040408x48802b84mbf83e20a2cb5d2f@mail.gmail.com>
Message-ID: <20090804134926.GA27629@hijacked.us>

On Tue, Aug 04, 2009 at 07:08:36PM +0800, mark morreny wrote:
> Hi,
> 
> I have seen people using both event socket and erlang to control freeSWITCH
> externally.
> 
> What is the pros and cons of using event socket vs erlang?
>

It depends on if you want to use erlang or not. The erlang module
provides most of the event socket functionality plus a couple extras
(dynamic XML bindings ala xml_curl, intelligent message delivery, etc).
If you're not already planning to use erlang, it's probably better to
dig out the relevant event socket library module for your langague
instead.

Andrew - author of the erlang module



From jmesquita at gmail.com  Tue Aug  4 07:04:03 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 4 Aug 2009 11:04:03 -0300
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
Message-ID: <5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>

If this is good for me to hear, I would imagine to the core team.

Despite of this not being a group support meeting, I have to say that: Thank
you for sharing, Seven.

jmesquita

On Tue, Aug 4, 2009 at 4:17 AM, Seven Du <dujinfang at gmail.com> wrote:

> Hello All -   In the spirit of ClueCon (which we are missing this year,
> but hopefully not next), we wanted to document our "FreeSWITCH Story".
>  We've posted it to the wiki(
> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com) and
> it is copied below.
>
> Thank you all and enjoy a good conference!
>
> Seven Du (seven)
> Jonathan Palley (jpalley_idapted)
> Idapted Ltd.
>
>
> *How FreeSWITCH has created hundreds of job opportunities and changed
> lives. *
>
> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
> been a key enabler of our business.  We hope this story can be a small way
> to say a very big THANK YOU ALL.
>
> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
> really allowed us to do just that.
>
> What We Do:
> We are not a telephony business; we are an educational technology and
> service business. In Asia (China, in our case) students must pass English
> examinations to study or work abroad and gain new experiences.  However,
> there is limited access to native English speakers and the access students
> can gain is typically very expensive.  At the same time, in the U.S., there
> are many professionals looking for work-at-home opportunities - people who
> need jobs and would create great teachers.  Through our technology and
> content we empower these people to be effective English teachers.  Does it
> work?  Yes.  The majority of our students are getting test scores that many
> failed for years to get.  Just hours ago one student called one of our sales
> agents crying with joy.  And for our teachers, they are now working in an
> industry that was previously unavailable to those living in the U.S.
> http://www.idapted.com
>
> Why FreeSWITCH Enables This:
> FreeSWITCH has been a key enabler of our business.  Recording calls,
> controlling routing, integrating with various web-based interfaces, enabling
> multiple endpoints - these are all key features of what we must do.  Most
> importantly, setting up various servers and routes to mitigate cross-Pacific
> and country-specific network challenges is key.  Doing what we are doing
> with commercial solutions would have made the business unworkable.
>
> Our Experiences with FreeSWITCH:
> We started using FreeSWITCH as our VoIP Platform in April 2008, after
> receiving unsatisfactory results with other open source solutions.  It took
> one day of reading through the FreeSWITCH source code to know, "this is it.
>  This is the VoIP platform we build our business on".  It took a few days of
> working with the extremely competent and focused community to re-affirm this
> commitment.
>
> Our Setup:
> Our teachers use a custom software that integrates a VoIP client with our
> web based platform. Students connect to our teachers "on-demand".  Simply
> put, on a web-based comet interface the student enters a phone number (or a
> skype name or a gtalk account) and our platform bridges the best available
> trainer and the student.  At the same time a web-based interface is being
> updated.
>
> The challenge for us is the connection between teachers and students over a
> cross-continent network. For example, we experienced problems earlier this
> year when a Asis-Pacific communication fiber broken... So, we've learned to
> setup multi servers in multiple datacenters for redundancy.
>
> We run multi instances of FreeSWITCH so we can always use the cutting edge
> and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
> connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
> only loads mod_dingaling. Here is one beauty of FS: We just had to create
> different conf dirs (/usr/local/freeswitch, /usr/local/skype,
> /usr/local/gtalk etc). This allows us to run the same code base over
> different configurations, and call skype and gtalk accounts just like a
> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
> abnormally or crashes, we can easily change to another FS-skype server (we
> run other servers located in various places in China and HK for
> redundancy).
>
> FS --|
>      |---PSTN gateways
>      |--- FS-skype
>      |--- FS-gtalk
>      |--- FS-skype2
>      |--- more ...
>
>
>
> COMMUNITY:
>
> The community's commitment cannot be undervalued.  The insightful, modular
> design of FreeSWITCH allows anyone to contribute, whereever their skills
> lie.  It also allows us to easily make modifications to the underlying code
> to suit our specific use-cases  We want to highlight a few key people and
> modules in the FS ecosystem:
>
> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
> clients.  PSTN is zero-conf for the user and mitigates troubles with the end
> users network/microphone, etc (which is significant with our user base).
>  However, cheap providers fail randomly and FreeSWITCH's ability to control
> routing, use multiple endpoints all while clearly seeing what is going on is
> key.
> Most importantly, anthm and the core team have been super helpful in
> getting SIP to work with us.  Back in the pre 1.0 days anthm made
> significant changes to mod-sofia to enable clients behind nats without STUN.
>  Its important to point out that he didn't just make the changes -he forced
> us to really make a compelling case as to why the changes were important for
> FreeSWITCH.  This is a good thing.
>
> skype (mod_skypiax): Due to the facts that users prefer skype, we
> configured skypiax. It was unstable at the beginning and that's one of the
> reason we started running that separate FS instance.  To be fair, it has
> caused a lot of trouble - but we know this, its new software that takes a
> big risk and implements a complex hack.  What is important is that the
> author of skypiax(Giovanni Maruzzelli) has been a huge help. He's been very
> active fixing bugs and logging in to our box to help trouble shoot. We owe
> him a *big* thanks.
>
> To make Skypiax more useful, we also created some patches including the ANY
> and RR interfaces for sequential and round robin line hunting, some bug
> fixes and other features like continue-load-on-fail and auto-skype-user
> which haven't been merged into trunk yet. Thanks a community that gives us a
> platform where we can all benefit and contribute.
>
> erlang (mod_erlang_events): Another key enabler of the next release of our
> system is the erlang interface.  We have a complex realtime queue routing
> system has it handles input not just from freeswitch, but numerous other web
> interfaces and sockets.  Erlang was the perfect technology to implement this
> in and luckily an Erlang module for FreeSWITCH was already written.
> Beautiful.
>
> THE MORAL OF THE STORY:
> FreeSWITCH is a great piece of software that has enabled new technologies
> and business models.  The design has allowed (and the core team has
> nurtured) a vibrant and exciting community that has made the software even
> better.  Every day we go to work excited to push the boundaries of what can
> be done with telephony technology and are confident this is the platform of
> the future.
>
> Thank you all.
>
>
> Sincerely,
>
> Du Jinfang (Seven) - Technical Operations/VoIP Manager
> Jonathan Palley - CTO
> Idapted Ltd.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rdenert at tng.de  Tue Aug  4 08:41:18 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Tue, 4 Aug 2009 17:41:18 +0200 (CEST)
Subject: [Freeswitch-users] Module in Lua not working
In-Reply-To: <7203453.221221249384483704.JavaMail.root@zimbra.tng.de>
Message-ID: <17585545.224751249400478914.JavaMail.root@zimbra.tng.de>

Hello,

there is no more problem. I installed 1.0.4 and everthing is fine. :-)

BR

----- Urspr?ngliche Mail -----
Von: "Rudolf Denert" <rdenert at tng.de>
An: "freeswitch-users" <freeswitch-users at lists.freeswitch.org>
Gesendet: Dienstag, 4. August 2009 13:14:43 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: [Freeswitch-users] Module in Lua not working

Hello again!

I need some help again, because I have little trouble with a few modules. The first one is luasocket, the seconde one is luasql. I always get the error:

error loading module 'socket' from file '/usr/local/lib/lua/5.1/socket/core.so': /usr/local/lib/lua/5.1/socket/core.so: undefined symbol: lua_getmetatable

and

error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': /usr/local/lib/lua/5.1/luasql/mysql.so: undefined symbol: lua_pushlstring

in the fs_cli

This looks like a conflict with an older version of these modules. Is this right? If yes, what should i update?

Thanks again.

BR

-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
-
Register: Amtsgericht Kiel, HRB 6596 KI
Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
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-
This e-mail may contain confidential and/or privileged information. If
you are not the intended recipient (or have received this e-mail in
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unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
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Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

-- 
-- 
Rudolf Denert, Technical Support
TNG AG, NGN
Projensdorfer Str. 324, D-24106 Kiel, Germany
phone: +49 431 7097-10, fax: +49 431 7097-555
mailto: rdenert at tng.de http://www.tng.de
-
Register: Amtsgericht Kiel, HRB 6596 KI
Supervisory board (Aufsichtsrat): Stefan Scheuermann (Vorsitz)
Executive board (Vorstand): Dr. Sven Willert (Vorsitz),  Dr. Thomas
Rohwer, Sven Schade, Carsten Tolkmit, Dr.-Ing. Volkmar Hausberg
Tax-Id (Steuernr.): 1929112637, VAT-Id (USt-Id): DE225201428
-
This e-mail may contain confidential and/or privileged information. If
you are not the intended recipient (or have received this e-mail in
error) please notify the sender immediately and destroy this e-mail. Any
unauthorized copying, disclosure or distribution of the material in this
e-mail is strictly forbidden.

Diese E-Mail enth?lt vertrauliche und/oder rechtlich geschuetzte
Informationen. Wenn Sie nicht der richtige Adressat sind oder diese
E-Mail irrt?mlich erhalten haben, informieren Sie bitte sofort den
Absender und vernichten Sie diese Mail. Das unerlaubte Kopieren sowie
die unbefugte Weitergabe dieser Mail ist nicht gestattet.



From matt at hellohunter.com  Tue Aug  4 11:50:36 2009
From: matt at hellohunter.com (Matt Hunter)
Date: Tue, 4 Aug 2009 11:50:36 -0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
Message-ID: <4256bf830908041150x5a450797j62389d4513997f9c@mail.gmail.com>

Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the
logs below, but I am still at a loss at being able to identify the error or
reproduce it consistently. The below log indicates to me that my FS server
is initiating sending 2 BYE message to my DID provider (didforsale.com). Is
there a way I can look further inside FreeSWITCH to see why it is sending
this BYE packet?

sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
Route: <sip:65.217.40.210:5060;lr>
Route: <sip:63.110.102.238;lr>
Max-Forwards: 70
From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
To: <sip:+1909635XXXX at 199.173.100.144:5060
;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
CSeq: 118584736 BYE
Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0


sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
Route: <sip:65.217.40.210:5060;lr>
Route: <sip:63.110.102.238;lr>
Max-Forwards: 70
From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
To: <sip:+1909635XXXX at 199.173.100.144:5060
;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
CSeq: 118584736 BYE
Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0



On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> Digging a bit in mod_sofia releaved that it can be caused by a SIP
> code 482 (loop detected), 483 (too many hops) or 484 (address
> incomplete).
>
> Do a SIP trace to sched more light on what's happening.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> Am 3-Aug-09 um 1:22 AM schrieb Matthew Fong:
>
> > EXCHANGE_ROUTING_ERROR
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mattdfong at gmail.com  Tue Aug  4 11:51:03 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Tue, 4 Aug 2009 11:51:03 -0700
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
Message-ID: <4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>

Hi Mathieu, thanks for the reply. I enabled sip trace logging and got the
logs below, but I am still at a loss at being able to identify the error or
reproduce it consistently. The below log indicates to me that my FS server
is initiating sending 2 BYE message to my DID provider (didforsale.com). Is
there a way I can look further inside FreeSWITCH to see why it is sending
this BYE packet?

sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
Route: <sip:65.217.40.210:5060;lr>
Route: <sip:63.110.102.238;lr>
Max-Forwards: 70
From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
To: <sip:+1909635XXXX at 199.173.100.144:5060
;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
CSeq: 118584736 BYE
Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0


sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
Route: <sip:65.217.40.210:5060;lr>
Route: <sip:63.110.102.238;lr>
Max-Forwards: 70
From: <sip:+1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
To: <sip:+1909635XXXX at 199.173.100.144:5060
;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
CSeq: 118584736 BYE
Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
Content-Length: 0

On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> Digging a bit in mod_sofia releaved that it can be caused by a SIP
> code 482 (loop detected), 483 (too many hops) or 484 (address
> incomplete).
>
> Do a SIP trace to sched more light on what's happening.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> Am 3-Aug-09 um 1:22 AM schrieb Matthew Fong:
>
> > EXCHANGE_ROUTING_ERROR
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gregt at cgicommunications.com  Tue Aug  4 12:12:25 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Tue, 4 Aug 2009 15:12:25 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>
Message-ID: <E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>

Hi, does anyone have an example of a simple PHP socket script that  
will listen and spawn of a process that handles the incoming call?

I understand the inbound socket and code such as this
http://wiki.freeswitch.org/wiki/PHP_Event_Socket
that will let me initiate operations. It's the constantly running php  
socket program that I can't get my head around, and how it will spawn  
another php script that will be able to do things like answer the  
session, get dtmf, etc.
--
Greg


On Aug 2, 2009, at 6:35 PM, Michael Collins wrote:

>
>
> On Sun, Aug 2, 2009 at 12:38 PM, Nik Middleton <nik.middleton at noblesolutions.co.uk 
> > wrote:
> Hi Guys,
>
>
> I?m using an outbound socket to control calls, and it works a  
> charm.  However, what I?d like to do is send a custom event  
> regarding the call on hang-up.  The way I see things happening at  
> the moment, and I could be wrong, is that the socket is closed when  
> a hang-up occurs, so am I taking a chance trying to send the event  
> then? (try to sneak out the event before socket closure happens)   
> The other option is of course to open an inbound socket and send the  
> event, but I?d rather not do that if possible.
>
> Nik,
>
> Perhaps the "linger" event socket command will do what you need?  
> Check out this commit:
> http://lists.freeswitch.org/pipermail/freeswitch-svn/2009-January/009391.html
>
> Let me know if it works for you and I'll be sure to get it  
> documented properly. If you get it working I'd love to see a code  
> snippet so we can wikify this knowledge. :)
>
> Thanks,
> MC
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From william.suffill at gmail.com  Tue Aug  4 16:04:08 2009
From: william.suffill at gmail.com (William Suffill)
Date: Tue, 4 Aug 2009 19:04:08 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local> 
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com> 
	<E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>
Message-ID: <6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>

I wrote some notes on this but have yet to wiki it.

example of an outbound socket connection where the call is answered, a
variable is set then perhaps play one of the pre-installed files and
hangup.
ivrd

fs_ivrd comes with freeswitch. It being a small daemon just invokes
the script defined in a variable and passes data from it via STDIN/OUT

Since this is an outbound socket connections it needs to be defined in
the dialplan.



Ex:

<extension name="outbound-socket">

    <condition field="destination_number" expression="^55(522)$">

        <action application="set"
data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>

        <action application="socket" data="127.0.0.1:8084 async full"/>

    </condition>

  </extension>



The above dialplan sample would invoke ivr-demo.php when 55522 is
called as long as fs_ivrd is running. To start fs_ivrd:

/usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 8004



It takes 2 arguments -h for hostname and -p for port.



PHP Code

#!/usr/bin/php -q

<?php



    // set a couple of things so we dont kill the system

    ob_implicit_flush(true);

    set_time_limit(30);





    // Open stdin so we can read the AGI data in

    $in = fopen("php://stdin", "r");

    // Connect

    echo "connect\n\n";

    // Answer

    echo "sendmsg\n";

    echo "call-command: execute\n";

    echo "execute-app-name: answer\n\n";



    // Play a prompt

    echo "sendmsg\n";

    echo "call-command: execute\n";

    echo "execute-app-name: playback\n";

    echo "execute-app-arg:
/usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr-welcome_to_freeswitch.wav\n\n";



    // Wait

    sleep(5);



   // Hangup

   echo "sendmsg\n";

   echo "call-command: hangup\n\n";



fclose($in);



?>



ivrd will call this script for each call. All itdoes is answer the
channel tell FreeSWITCH to play the ?welcome to freeswitch? prompt.
Since the script is now controlling all call flow I needed to add a
wait or it would send the hangup immediately before the prompt was
played.

Some improvements possible but that's 1 way to do it. It would be
possible to do the socket directly in PHP but fs_ivrd is a nice option
too.

-- W



From diego.viola at gmail.com  Tue Aug  4 17:54:34 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 20:54:34 -0400
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
Message-ID: <86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>

Very cool, and yes, FreeSWITCH does rock =D

Both the software and the community ;)

2009/8/4 Jo?o Mesquita <jmesquita at gmail.com>

> If this is good for me to hear, I would imagine to the core team.
>
> Despite of this not being a group support meeting, I have to say that:
> Thank you for sharing, Seven.
>
> jmesquita
>
> On Tue, Aug 4, 2009 at 4:17 AM, Seven Du <dujinfang at gmail.com> wrote:
>
>> Hello All -   In the spirit of ClueCon (which we are missing this year,
>> but hopefully not next), we wanted to document our "FreeSWITCH Story".
>>  We've posted it to the wiki(
>> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
>> and it is copied below.
>>
>> Thank you all and enjoy a good conference!
>>
>> Seven Du (seven)
>> Jonathan Palley (jpalley_idapted)
>> Idapted Ltd.
>>
>>
>> *How FreeSWITCH has created hundreds of job opportunities and changed
>> lives. *
>>
>> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
>> been a key enabler of our business.  We hope this story can be a small way
>> to say a very big THANK YOU ALL.
>>
>> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
>> really allowed us to do just that.
>>
>> What We Do:
>> We are not a telephony business; we are an educational technology and
>> service business. In Asia (China, in our case) students must pass English
>> examinations to study or work abroad and gain new experiences.  However,
>> there is limited access to native English speakers and the access students
>> can gain is typically very expensive.  At the same time, in the U.S., there
>> are many professionals looking for work-at-home opportunities - people who
>> need jobs and would create great teachers.  Through our technology and
>> content we empower these people to be effective English teachers.  Does it
>> work?  Yes.  The majority of our students are getting test scores that many
>> failed for years to get.  Just hours ago one student called one of our sales
>> agents crying with joy.  And for our teachers, they are now working in an
>> industry that was previously unavailable to those living in the U.S.
>> http://www.idapted.com
>>
>> Why FreeSWITCH Enables This:
>> FreeSWITCH has been a key enabler of our business.  Recording calls,
>> controlling routing, integrating with various web-based interfaces, enabling
>> multiple endpoints - these are all key features of what we must do.  Most
>> importantly, setting up various servers and routes to mitigate cross-Pacific
>> and country-specific network challenges is key.  Doing what we are doing
>> with commercial solutions would have made the business unworkable.
>>
>> Our Experiences with FreeSWITCH:
>> We started using FreeSWITCH as our VoIP Platform in April 2008, after
>> receiving unsatisfactory results with other open source solutions.  It took
>> one day of reading through the FreeSWITCH source code to know, "this is it.
>>  This is the VoIP platform we build our business on".  It took a few days of
>> working with the extremely competent and focused community to re-affirm this
>> commitment.
>>
>> Our Setup:
>> Our teachers use a custom software that integrates a VoIP client with our
>> web based platform. Students connect to our teachers "on-demand".  Simply
>> put, on a web-based comet interface the student enters a phone number (or a
>> skype name or a gtalk account) and our platform bridges the best available
>> trainer and the student.  At the same time a web-based interface is being
>> updated.
>>
>> The challenge for us is the connection between teachers and students over
>> a cross-continent network. For example, we experienced problems earlier this
>> year when a Asis-Pacific communication fiber broken... So, we've learned to
>> setup multi servers in multiple datacenters for redundancy.
>>
>> We run multi instances of FreeSWITCH so we can always use the cutting edge
>> and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS) instance
>> connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and FS-gtalk
>> only loads mod_dingaling. Here is one beauty of FS: We just had to create
>> different conf dirs (/usr/local/freeswitch, /usr/local/skype,
>> /usr/local/gtalk etc). This allows us to run the same code base over
>> different configurations, and call skype and gtalk accounts just like a
>> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
>> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
>> abnormally or crashes, we can easily change to another FS-skype server (we
>> run other servers located in various places in China and HK for
>> redundancy).
>>
>> FS --|
>>      |---PSTN gateways
>>      |--- FS-skype
>>      |--- FS-gtalk
>>      |--- FS-skype2
>>      |--- more ...
>>
>>
>>
>> COMMUNITY:
>>
>> The community's commitment cannot be undervalued.  The insightful, modular
>> design of FreeSWITCH allows anyone to contribute, whereever their skills
>> lie.  It also allows us to easily make modifications to the underlying code
>> to suit our specific use-cases  We want to highlight a few key people and
>> modules in the FS ecosystem:
>>
>> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
>> clients.  PSTN is zero-conf for the user and mitigates troubles with the end
>> users network/microphone, etc (which is significant with our user base).
>>  However, cheap providers fail randomly and FreeSWITCH's ability to control
>> routing, use multiple endpoints all while clearly seeing what is going on is
>> key.
>> Most importantly, anthm and the core team have been super helpful in
>> getting SIP to work with us.  Back in the pre 1.0 days anthm made
>> significant changes to mod-sofia to enable clients behind nats without STUN.
>>  Its important to point out that he didn't just make the changes -he forced
>> us to really make a compelling case as to why the changes were important for
>> FreeSWITCH.  This is a good thing.
>>
>> skype (mod_skypiax): Due to the facts that users prefer skype, we
>> configured skypiax. It was unstable at the beginning and that's one of the
>> reason we started running that separate FS instance.  To be fair, it has
>> caused a lot of trouble - but we know this, its new software that takes a
>> big risk and implements a complex hack.  What is important is that the
>> author of skypiax(Giovanni Maruzzelli) has been a huge help. He's been very
>> active fixing bugs and logging in to our box to help trouble shoot. We owe
>> him a *big* thanks.
>>
>> To make Skypiax more useful, we also created some patches including the
>> ANY and RR interfaces for sequential and round robin line hunting, some bug
>> fixes and other features like continue-load-on-fail and auto-skype-user
>> which haven't been merged into trunk yet. Thanks a community that gives us a
>> platform where we can all benefit and contribute.
>>
>> erlang (mod_erlang_events): Another key enabler of the next release of our
>> system is the erlang interface.  We have a complex realtime queue routing
>> system has it handles input not just from freeswitch, but numerous other web
>> interfaces and sockets.  Erlang was the perfect technology to implement this
>> in and luckily an Erlang module for FreeSWITCH was already written.
>> Beautiful.
>>
>> THE MORAL OF THE STORY:
>> FreeSWITCH is a great piece of software that has enabled new technologies
>> and business models.  The design has allowed (and the core team has
>> nurtured) a vibrant and exciting community that has made the software even
>> better.  Every day we go to work excited to push the boundaries of what can
>> be done with telephony technology and are confident this is the platform of
>> the future.
>>
>> Thank you all.
>>
>>
>> Sincerely,
>>
>> Du Jinfang (Seven) - Technical Operations/VoIP Manager
>> Jonathan Palley - CTO
>> Idapted Ltd.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Tue Aug  4 17:55:22 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 20:55:22 -0400
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
Message-ID: <86a32abc0908041755v7400796dwb21a13d039635539@mail.gmail.com>

Cool, I wish I could be there, next year =D

Any pics or videos of ClueCon? :)

On Mon, Aug 3, 2009 at 8:41 AM, Brian West <brian at freeswitch.org> wrote:

> Just look for large groups of people with laptops.  I'm sure you can't
> miss us.
>
> /b
>
> On Aug 3, 2009, at 5:46 AM, Chad Phillips -- Apartment Lines wrote:
>
> > i'm arriving around 12:30PM -- are there any pre-ClueCon meetups this
> > afternoon?  anybody need help with setup?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Tue Aug  4 17:56:27 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 20:56:27 -0400
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
	<86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>
Message-ID: <86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>

Maybe you can link your testimonial or put it here also? :D

http://wiki.freeswitch.org/wiki/Testimonials

On Tue, Aug 4, 2009 at 8:54 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Very cool, and yes, FreeSWITCH does rock =D
>
> Both the software and the community ;)
>
> 2009/8/4 Jo?o Mesquita <jmesquita at gmail.com>
>
> If this is good for me to hear, I would imagine to the core team.
>>
>> Despite of this not being a group support meeting, I have to say that:
>> Thank you for sharing, Seven.
>>
>> jmesquita
>>
>> On Tue, Aug 4, 2009 at 4:17 AM, Seven Du <dujinfang at gmail.com> wrote:
>>
>>>  Hello All -   In the spirit of ClueCon (which we are missing this year,
>>> but hopefully not next), we wanted to document our "FreeSWITCH Story".
>>>  We've posted it to the wiki(
>>> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
>>> and it is copied below.
>>>
>>> Thank you all and enjoy a good conference!
>>>
>>> Seven Du (seven)
>>> Jonathan Palley (jpalley_idapted)
>>> Idapted Ltd.
>>>
>>>
>>> *How FreeSWITCH has created hundreds of job opportunities and changed
>>> lives. *
>>>
>>> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
>>> been a key enabler of our business.  We hope this story can be a small way
>>> to say a very big THANK YOU ALL.
>>>
>>> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH has
>>> really allowed us to do just that.
>>>
>>> What We Do:
>>> We are not a telephony business; we are an educational technology and
>>> service business. In Asia (China, in our case) students must pass English
>>> examinations to study or work abroad and gain new experiences.  However,
>>> there is limited access to native English speakers and the access students
>>> can gain is typically very expensive.  At the same time, in the U.S., there
>>> are many professionals looking for work-at-home opportunities - people who
>>> need jobs and would create great teachers.  Through our technology and
>>> content we empower these people to be effective English teachers.  Does it
>>> work?  Yes.  The majority of our students are getting test scores that many
>>> failed for years to get.  Just hours ago one student called one of our sales
>>> agents crying with joy.  And for our teachers, they are now working in an
>>> industry that was previously unavailable to those living in the U.S.
>>> http://www.idapted.com
>>>
>>> Why FreeSWITCH Enables This:
>>> FreeSWITCH has been a key enabler of our business.  Recording calls,
>>> controlling routing, integrating with various web-based interfaces, enabling
>>> multiple endpoints - these are all key features of what we must do.  Most
>>> importantly, setting up various servers and routes to mitigate cross-Pacific
>>> and country-specific network challenges is key.  Doing what we are doing
>>> with commercial solutions would have made the business unworkable.
>>>
>>> Our Experiences with FreeSWITCH:
>>> We started using FreeSWITCH as our VoIP Platform in April 2008, after
>>> receiving unsatisfactory results with other open source solutions.  It took
>>> one day of reading through the FreeSWITCH source code to know, "this is it.
>>>  This is the VoIP platform we build our business on".  It took a few days of
>>> working with the extremely competent and focused community to re-affirm this
>>> commitment.
>>>
>>> Our Setup:
>>> Our teachers use a custom software that integrates a VoIP client with our
>>> web based platform. Students connect to our teachers "on-demand".  Simply
>>> put, on a web-based comet interface the student enters a phone number (or a
>>> skype name or a gtalk account) and our platform bridges the best available
>>> trainer and the student.  At the same time a web-based interface is being
>>> updated.
>>>
>>> The challenge for us is the connection between teachers and students over
>>> a cross-continent network. For example, we experienced problems earlier this
>>> year when a Asis-Pacific communication fiber broken... So, we've learned to
>>> setup multi servers in multiple datacenters for redundancy.
>>>
>>> We run multi instances of FreeSWITCH so we can always use the cutting
>>> edge and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS)
>>> instance connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and
>>> FS-gtalk only loads mod_dingaling. Here is one beauty of FS: We just had to
>>> create different conf dirs (/usr/local/freeswitch, /usr/local/skype,
>>> /usr/local/gtalk etc). This allows us to run the same code base over
>>> different configurations, and call skype and gtalk accounts just like a
>>> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
>>> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
>>> abnormally or crashes, we can easily change to another FS-skype server (we
>>> run other servers located in various places in China and HK for
>>> redundancy).
>>>
>>> FS --|
>>>      |---PSTN gateways
>>>      |--- FS-skype
>>>      |--- FS-gtalk
>>>      |--- FS-skype2
>>>      |--- more ...
>>>
>>>
>>>
>>> COMMUNITY:
>>>
>>> The community's commitment cannot be undervalued.  The insightful,
>>> modular design of FreeSWITCH allows anyone to contribute, whereever their
>>> skills lie.  It also allows us to easily make modifications to the
>>> underlying code to suit our specific use-cases  We want to highlight a few
>>> key people and modules in the FS ecosystem:
>>>
>>> mod_sofia: SIP is how we connect to our PSTN gateways and to our teachers
>>> clients.  PSTN is zero-conf for the user and mitigates troubles with the end
>>> users network/microphone, etc (which is significant with our user base).
>>>  However, cheap providers fail randomly and FreeSWITCH's ability to control
>>> routing, use multiple endpoints all while clearly seeing what is going on is
>>> key.
>>> Most importantly, anthm and the core team have been super helpful in
>>> getting SIP to work with us.  Back in the pre 1.0 days anthm made
>>> significant changes to mod-sofia to enable clients behind nats without STUN.
>>>  Its important to point out that he didn't just make the changes -he forced
>>> us to really make a compelling case as to why the changes were important for
>>> FreeSWITCH.  This is a good thing.
>>>
>>> skype (mod_skypiax): Due to the facts that users prefer skype, we
>>> configured skypiax. It was unstable at the beginning and that's one of the
>>> reason we started running that separate FS instance.  To be fair, it has
>>> caused a lot of trouble - but we know this, its new software that takes a
>>> big risk and implements a complex hack.  What is important is that the
>>> author of skypiax(Giovanni Maruzzelli) has been a huge help. He's been very
>>> active fixing bugs and logging in to our box to help trouble shoot. We owe
>>> him a *big* thanks.
>>>
>>> To make Skypiax more useful, we also created some patches including the
>>> ANY and RR interfaces for sequential and round robin line hunting, some bug
>>> fixes and other features like continue-load-on-fail and auto-skype-user
>>> which haven't been merged into trunk yet. Thanks a community that gives us a
>>> platform where we can all benefit and contribute.
>>>
>>> erlang (mod_erlang_events): Another key enabler of the next release of
>>> our system is the erlang interface.  We have a complex realtime queue
>>> routing system has it handles input not just from freeswitch, but numerous
>>> other web interfaces and sockets.  Erlang was the perfect technology to
>>> implement this in and luckily an Erlang module for FreeSWITCH was already
>>> written. Beautiful.
>>>
>>> THE MORAL OF THE STORY:
>>> FreeSWITCH is a great piece of software that has enabled new technologies
>>> and business models.  The design has allowed (and the core team has
>>> nurtured) a vibrant and exciting community that has made the software even
>>> better.  Every day we go to work excited to push the boundaries of what can
>>> be done with telephony technology and are confident this is the platform of
>>> the future.
>>>
>>> Thank you all.
>>>
>>>
>>> Sincerely,
>>>
>>> Du Jinfang (Seven) - Technical Operations/VoIP Manager
>>> Jonathan Palley - CTO
>>> Idapted Ltd.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From gabe at gundy.org  Tue Aug  4 18:53:23 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Tue, 4 Aug 2009 19:53:23 -0600
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <86a32abc0908041755v7400796dwb21a13d039635539@mail.gmail.com>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
	<86a32abc0908041755v7400796dwb21a13d039635539@mail.gmail.com>
Message-ID: <903da5680908041853y7f828878pc539470b328066dc@mail.gmail.com>

On Tue, Aug 4, 2009 at 6:55 PM, Diego Viola<diego.viola at gmail.com> wrote:
> Cool, I wish I could be there, next year =D

Tomorrow is my wedding anniversary 8/5.  If ClueCon keeps getting
scheduled on that date, I don't know if I'll ever get to go :(


> Any pics or videos of ClueCon? :)

Yes, bring on the vids.

Best,
Gabe



From diego.viola at gmail.com  Tue Aug  4 18:56:27 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 21:56:27 -0400
Subject: [Freeswitch-users] arriving today for ClueCon
In-Reply-To: <903da5680908041853y7f828878pc539470b328066dc@mail.gmail.com>
References: <2CAC3064-B765-4979-85F6-9D628F0A7090@apartmentlines.com>
	<6525CB65-04B6-477B-9B58-DD3BC5C680DB@freeswitch.org>
	<86a32abc0908041755v7400796dwb21a13d039635539@mail.gmail.com>
	<903da5680908041853y7f828878pc539470b328066dc@mail.gmail.com>
Message-ID: <86a32abc0908041856u4d4a5c4byff3c9f73be0e5031@mail.gmail.com>

Looking forward to this talk =D

"FreeSWITCH: Learning to Think Fourth Dimensionally"

;)

On Tue, Aug 4, 2009 at 9:53 PM, Gabriel Gunderson <gabe at gundy.org> wrote:

> On Tue, Aug 4, 2009 at 6:55 PM, Diego Viola<diego.viola at gmail.com> wrote:
> > Cool, I wish I could be there, next year =D
>
> Tomorrow is my wedding anniversary 8/5.  If ClueCon keeps getting
> scheduled on that date, I don't know if I'll ever get to go :(
>
>
> > Any pics or videos of ClueCon? :)
>
> Yes, bring on the vids.
>
> Best,
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Tue Aug  4 19:22:07 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 4 Aug 2009 22:22:07 -0400
Subject: [Freeswitch-users] Looking for some FreeSWITCH job
Message-ID: <86a32abc0908041922j5dac756g49b2d87db74479b1@mail.gmail.com>

Hi,

I'm currently looking for some FS jobs, I really need one, I'm currently
unemployed and looking for some serious FreeSWITCH jobs.

Anyone?

P.S: I also do FS and web development with any language, PHP, Ruby, etc.
anything really.

Thanks,

Diego
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From jpalley at idapted.com  Tue Aug  4 21:33:32 2009
From: jpalley at idapted.com (Jonathan Palley)
Date: Wed, 5 Aug 2009 12:33:32 +0800
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com> 
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com> 
	<86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com> 
	<86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>
Message-ID: <2d8777c00908042133o6e2ffd33g37da13392ac6ccb@mail.gmail.com>

Diego -   Already done.  See the bottom of the page (we linked to another
page because of its length)!

:)
Jonathan Palley
Idapted Ltd.

On Wed, Aug 5, 2009 at 8:56 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Maybe you can link your testimonial or put it here also? :D
>
> http://wiki.freeswitch.org/wiki/Testimonials
>
>
> On Tue, Aug 4, 2009 at 8:54 PM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Very cool, and yes, FreeSWITCH does rock =D
>>
>> Both the software and the community ;)
>>
>> 2009/8/4 Jo?o Mesquita <jmesquita at gmail.com>
>>
>> If this is good for me to hear, I would imagine to the core team.
>>>
>>> Despite of this not being a group support meeting, I have to say that:
>>> Thank you for sharing, Seven.
>>>
>>> jmesquita
>>>
>>> On Tue, Aug 4, 2009 at 4:17 AM, Seven Du <dujinfang at gmail.com> wrote:
>>>
>>>>  Hello All -   In the spirit of ClueCon (which we are missing this
>>>> year, but hopefully not next), we wanted to document our "FreeSWITCH Story".
>>>>  We've posted it to the wiki(
>>>> http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com)
>>>> and it is copied below.
>>>>
>>>> Thank you all and enjoy a good conference!
>>>>
>>>> Seven Du (seven)
>>>> Jonathan Palley (jpalley_idapted)
>>>> Idapted Ltd.
>>>>
>>>>
>>>> *How FreeSWITCH has created hundreds of job opportunities and changed
>>>> lives. *
>>>>
>>>> We want to share our experience working with FreeSWITCH.  FreeSWITCH has
>>>> been a key enabler of our business.  We hope this story can be a small way
>>>> to say a very big THANK YOU ALL.
>>>>
>>>> "Changing lives" is an over-used cliche, but in this case, FreeSWITCH
>>>> has really allowed us to do just that.
>>>>
>>>> What We Do:
>>>> We are not a telephony business; we are an educational technology and
>>>> service business. In Asia (China, in our case) students must pass English
>>>> examinations to study or work abroad and gain new experiences.  However,
>>>> there is limited access to native English speakers and the access students
>>>> can gain is typically very expensive.  At the same time, in the U.S., there
>>>> are many professionals looking for work-at-home opportunities - people who
>>>> need jobs and would create great teachers.  Through our technology and
>>>> content we empower these people to be effective English teachers.  Does it
>>>> work?  Yes.  The majority of our students are getting test scores that many
>>>> failed for years to get.  Just hours ago one student called one of our sales
>>>> agents crying with joy.  And for our teachers, they are now working in an
>>>> industry that was previously unavailable to those living in the U.S.
>>>> http://www.idapted.com
>>>>
>>>> Why FreeSWITCH Enables This:
>>>> FreeSWITCH has been a key enabler of our business.  Recording calls,
>>>> controlling routing, integrating with various web-based interfaces, enabling
>>>> multiple endpoints - these are all key features of what we must do.  Most
>>>> importantly, setting up various servers and routes to mitigate cross-Pacific
>>>> and country-specific network challenges is key.  Doing what we are doing
>>>> with commercial solutions would have made the business unworkable.
>>>>
>>>> Our Experiences with FreeSWITCH:
>>>> We started using FreeSWITCH as our VoIP Platform in April 2008, after
>>>> receiving unsatisfactory results with other open source solutions.  It took
>>>> one day of reading through the FreeSWITCH source code to know, "this is it.
>>>>  This is the VoIP platform we build our business on".  It took a few days of
>>>> working with the extremely competent and focused community to re-affirm this
>>>> commitment.
>>>>
>>>> Our Setup:
>>>> Our teachers use a custom software that integrates a VoIP client with
>>>> our web based platform. Students connect to our teachers "on-demand".
>>>>  Simply put, on a web-based comet interface the student enters a phone
>>>> number (or a skype name or a gtalk account) and our platform bridges the
>>>> best available trainer and the student.  At the same time a web-based
>>>> interface is being updated.
>>>>
>>>> The challenge for us is the connection between teachers and students
>>>> over a cross-continent network. For example, we experienced problems earlier
>>>> this year when a Asis-Pacific communication fiber broken... So, we've
>>>> learned to setup multi servers in multiple datacenters for redundancy.
>>>>
>>>> We run multi instances of FreeSWITCH so we can always use the cutting
>>>> edge and mitigate the effects of bugs. A main, "stable" FreeSWITCH(FS)
>>>> instance connect to other FreeSWITCHes - Fs-skype only loads mod_skypiax and
>>>> FS-gtalk only loads mod_dingaling. Here is one beauty of FS: We just had to
>>>> create different conf dirs (/usr/local/freeswitch, /usr/local/skype,
>>>> /usr/local/gtalk etc). This allows us to run the same code base over
>>>> different configurations, and call skype and gtalk accounts just like a
>>>> normal PSTN gateway (sofia/gateway/pstn/.... or sofia/gateway/skype/.... or
>>>> sofia/gateway/gtalk/.... ). More important, if one FS (say FS-skype) behaves
>>>> abnormally or crashes, we can easily change to another FS-skype server (we
>>>> run other servers located in various places in China and HK for
>>>> redundancy).
>>>>
>>>> FS --|
>>>>      |---PSTN gateways
>>>>      |--- FS-skype
>>>>      |--- FS-gtalk
>>>>      |--- FS-skype2
>>>>      |--- more ...
>>>>
>>>>
>>>>
>>>> COMMUNITY:
>>>>
>>>> The community's commitment cannot be undervalued.  The insightful,
>>>> modular design of FreeSWITCH allows anyone to contribute, whereever their
>>>> skills lie.  It also allows us to easily make modifications to the
>>>> underlying code to suit our specific use-cases  We want to highlight a few
>>>> key people and modules in the FS ecosystem:
>>>>
>>>> mod_sofia: SIP is how we connect to our PSTN gateways and to our
>>>> teachers clients.  PSTN is zero-conf for the user and mitigates troubles
>>>> with the end users network/microphone, etc (which is significant with our
>>>> user base).  However, cheap providers fail randomly and FreeSWITCH's ability
>>>> to control routing, use multiple endpoints all while clearly seeing what is
>>>> going on is key.
>>>> Most importantly, anthm and the core team have been super helpful in
>>>> getting SIP to work with us.  Back in the pre 1.0 days anthm made
>>>> significant changes to mod-sofia to enable clients behind nats without STUN.
>>>>  Its important to point out that he didn't just make the changes -he forced
>>>> us to really make a compelling case as to why the changes were important for
>>>> FreeSWITCH.  This is a good thing.
>>>>
>>>> skype (mod_skypiax): Due to the facts that users prefer skype, we
>>>> configured skypiax. It was unstable at the beginning and that's one of the
>>>> reason we started running that separate FS instance.  To be fair, it has
>>>> caused a lot of trouble - but we know this, its new software that takes a
>>>> big risk and implements a complex hack.  What is important is that the
>>>> author of skypiax(Giovanni Maruzzelli) has been a huge help. He's been very
>>>> active fixing bugs and logging in to our box to help trouble shoot. We owe
>>>> him a *big* thanks.
>>>>
>>>> To make Skypiax more useful, we also created some patches including the
>>>> ANY and RR interfaces for sequential and round robin line hunting, some bug
>>>> fixes and other features like continue-load-on-fail and auto-skype-user
>>>> which haven't been merged into trunk yet. Thanks a community that gives us a
>>>> platform where we can all benefit and contribute.
>>>>
>>>> erlang (mod_erlang_events): Another key enabler of the next release of
>>>> our system is the erlang interface.  We have a complex realtime queue
>>>> routing system has it handles input not just from freeswitch, but numerous
>>>> other web interfaces and sockets.  Erlang was the perfect technology to
>>>> implement this in and luckily an Erlang module for FreeSWITCH was already
>>>> written. Beautiful.
>>>>
>>>> THE MORAL OF THE STORY:
>>>> FreeSWITCH is a great piece of software that has enabled new
>>>> technologies and business models.  The design has allowed (and the core team
>>>> has nurtured) a vibrant and exciting community that has made the software
>>>> even better.  Every day we go to work excited to push the boundaries of what
>>>> can be done with telephony technology and are confident this is the platform
>>>> of the future.
>>>>
>>>> Thank you all.
>>>>
>>>>
>>>> Sincerely,
>>>>
>>>> Du Jinfang (Seven) - Technical Operations/VoIP Manager
>>>> Jonathan Palley - CTO
>>>> Idapted Ltd.
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Tue Aug  4 21:41:20 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 5 Aug 2009 12:41:20 +0800
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <2d8777c00908042133o6e2ffd33g37da13392ac6ccb@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
	<86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>
	<86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>
	<2d8777c00908042133o6e2ffd33g37da13392ac6ccb@mail.gmail.com>
Message-ID: <23f91030908042141h9f79d61w86599122677670@mail.gmail.com>

And I  added this on the wiki page:
mod_conference and mod_fifo: We also use FreeSWITCH in our office
environment as a PBX for call center and customer service connected with
VoIP and PSTN(openzap) gateways. It is integrated into our CRM system
naturally and just made sales process, business logic and world wide
conference much more simpler and easier.

:)

7

2009/8/5 Jonathan Palley <jpalley at idapted.com>

> Diego -   Already done.  See the bottom of the page (we linked to another
> page because of its length)!
>
> :)
> Jonathan Palley
> Idapted Ltd.
>
>
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From diego.viola at gmail.com  Tue Aug  4 22:11:24 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 5 Aug 2009 01:11:24 -0400
Subject: [Freeswitch-users] In the spirit of ClueCon: Our FreeSWITCH
	Story
In-Reply-To: <23f91030908042141h9f79d61w86599122677670@mail.gmail.com>
References: <23f91030908040017l7eb2667fy3361430acdd2daeb@mail.gmail.com>
	<5a8712120908040704h69d6c1f6ia9095561422bde32@mail.gmail.com>
	<86a32abc0908041754r5a8a5498oe221b7c87c5da3a1@mail.gmail.com>
	<86a32abc0908041756r24bca00fu34827df496110a2c@mail.gmail.com>
	<2d8777c00908042133o6e2ffd33g37da13392ac6ccb@mail.gmail.com>
	<23f91030908042141h9f79d61w86599122677670@mail.gmail.com>
Message-ID: <86a32abc0908042211g2e023d1fgd6f8eb25630561d9@mail.gmail.com>

Very cool, thanks guys, you make FreeSWITCH even better =D

Your story rocks!

On Wed, Aug 5, 2009 at 12:41 AM, Seven Du<dujinfang at gmail.com> wrote:
> And I ?added this on the wiki page:
> mod_conference and mod_fifo: We also use FreeSWITCH in our office
> environment as a PBX for call center and customer service connected with
> VoIP and PSTN(openzap) gateways. It is integrated into our CRM system
> naturally and just made sales process, business logic and world wide
> conference much more simpler and easier.
> :)
> 7
>
> 2009/8/5 Jonathan Palley <jpalley at idapted.com>
>>
>> Diego -
>> ??Already done. ?See the bottom of the page (we linked to another page
>> because of?its?length)!
>> :)
>> Jonathan?Palley
>> Idapted?Ltd.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From velu.technical at gmail.com  Tue Aug  4 23:14:55 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Wed, 5 Aug 2009 11:44:55 +0530
Subject: [Freeswitch-users] execute function in ESL.pm module is not working
Message-ID: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>

Dear All,
   I registered alarm signal in my Perl server program.
   If ALARM signal occurred I execute the following statement in signal
handler.
    "$conn->execute("playback",$sound_path."voicemail/vm-goodbye.wav")"
  The above statement didn't play that wave file. But before generating the
ALARM signal it worked.

What is the problem?

Please help me in this problem....

Also Is there any idea to do timeout for DTMF digits?

Thanks...

Regards,
K.Velusamy
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From brad.tuan at gmail.com  Wed Aug  5 03:26:01 2009
From: brad.tuan at gmail.com (Brad Tuan)
Date: Wed, 5 Aug 2009 18:26:01 +0800
Subject: [Freeswitch-users] How to change the contact when fs sending
	REGISTER??
Message-ID: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>

As title ,I know how to do when sending INVITE

but how to do it when fs sending REGISTER??

For example , when gateway registering , the contact is
gw+abcd at XXX.XXX.XXX.XXX ,

how to change it to *abcd at XXX.XXX.XXX.XXX??*
**
*Please help*
**
**
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From gregt at cgicommunications.com  Wed Aug  5 05:54:04 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Wed, 5 Aug 2009 08:54:04 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>
	<E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>
	<6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>
Message-ID: <FE05AA09-CA4B-40E0-BE43-626667FFEDA9@cgicommunications.com>

Thanks so much, William. This gives me a great start.
--
Greg



On Aug 4, 2009, at 7:04 PM, William Suffill wrote:

> I wrote some notes on this but have yet to wiki it.
>
> example of an outbound socket connection where the call is answered, a
> variable is set then perhaps play one of the pre-installed files and
> hangup.
> ivrd
>
> fs_ivrd comes with freeswitch. It being a small daemon just invokes
> the script defined in a variable and passes data from it via STDIN/OUT
>
> Since this is an outbound socket connections it needs to be defined in
> the dialplan.
>
>
>
> Ex:
>
> <extension name="outbound-socket">
>
>    <condition field="destination_number" expression="^55(522)$">
>
>        <action application="set"
> data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
>
>        <action application="socket" data="127.0.0.1:8084 async full"/>
>
>    </condition>
>
>  </extension>
>
>
>
> The above dialplan sample would invoke ivr-demo.php when 55522 is
> called as long as fs_ivrd is running. To start fs_ivrd:
>
> /usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 8004
>
>
>
> It takes 2 arguments -h for hostname and -p for port.
>
>
>
> PHP Code
>
> #!/usr/bin/php -q
>
> <?php
>
>
>
>    // set a couple of things so we dont kill the system
>
>    ob_implicit_flush(true);
>
>    set_time_limit(30);
>
>
>
>
>
>    // Open stdin so we can read the AGI data in
>
>    $in = fopen("php://stdin", "r");
>
>    // Connect
>
>    echo "connect\n\n";
>
>    // Answer
>
>    echo "sendmsg\n";
>
>    echo "call-command: execute\n";
>
>    echo "execute-app-name: answer\n\n";
>
>
>
>    // Play a prompt
>
>    echo "sendmsg\n";
>
>    echo "call-command: execute\n";
>
>    echo "execute-app-name: playback\n";
>
>    echo "execute-app-arg:
> /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr- 
> welcome_to_freeswitch.wav\n\n";
>
>
>
>    // Wait
>
>    sleep(5);
>
>
>
>   // Hangup
>
>   echo "sendmsg\n";
>
>   echo "call-command: hangup\n\n";
>
>
>
> fclose($in);
>
>
>
> ?>
>
>
>
> ivrd will call this script for each call. All itdoes is answer the
> channel tell FreeSWITCH to play the ?welcome to freeswitch? prompt.
> Since the script is now controlling all call flow I needed to add a
> wait or it would send the hangup immediately before the prompt was
> played.
>
> Some improvements possible but that's 1 way to do it. It would be
> possible to do the socket directly in PHP but fs_ivrd is a nice option
> too.
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at dialyourleads.com  Mon Aug  3 12:23:04 2009
From: mike at dialyourleads.com (Michael Frager)
Date: Mon, 3 Aug 2009 15:23:04 -0400
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
Message-ID: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>

Hello,

I'm in the process of moving my VOIP application from Asterisk to
FreeSWITCH.

I was wondering if it is possible to emulate the call announcement feature
that is available on Asterisk.

On Asterisk it looks like this, with the "A(...)" parameter:

Dial(SIP/15555551212|180|A(connecttone1))

Note that this announcement is only played for the called party, the calling
party does NOT hear the tone.

I'm guessing this can be done with FreeSWITCH.

Does anyone know how I might accomplish this?


Thanks in advance,

-Mike Fragre
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From david.nembrot at sogeti.com  Wed Aug  5 01:18:30 2009
From: david.nembrot at sogeti.com (David Nembrot)
Date: Wed,  5 Aug 2009 10:18:30 +0200
Subject: [Freeswitch-users] Qualify IM comm. across two distinct SIP domains
Message-ID: <20090805101830.bn9e3vzjms08oskg@mail.sogeti.com>

Hi everybody,

? ?I've just configured two Freeswitch servers (FS#1 and FS#2) to?
enable SIP communications between their two distinct SIP domains. The?
fact is that the IP telephony is up & running across these two?
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory?
way, my guess is that I'm missing something in the config files in?
order to enable IM services throughout the two domains.. Since they?
are in different networks, it seems reasonable for example to force?
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my?
question:

? ?How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

? ?or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.
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-------------- next part --------------


   Hi everybody,

   I've just configured two Freeswitch servers (FS#1 and FS#2) to  
enable SIP communications between their two distinct SIP domains. The  
fact is that the IP telephony is up & running across these two  
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory  
way, my guess is that I'm missing something in the config files in  
order to enable IM services throughout the two domains.. Since they  
are in different networks, it seems reasonable for example to force  
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my  
question:

   How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

   or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.


-------------- next part --------------
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From enno.egbert at googlemail.com  Wed Aug  5 01:52:20 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Wed, 5 Aug 2009 01:52:20 -0700 (PDT)
Subject: [Freeswitch-users]  TLS/SRTP with Innovaphone IP200
Message-ID: <24823167.post@talk.nabble.com>


Hello,

i have a problem using a innovaphone ip200 with freeswitch and tls/srtp. The
freeswitch certificate is in the trust list of the phone and it works with
tls for incomming calls. But outgoing calls were rejected to the mailbox.
The freeswitch configuration is ok, because it works with a snom 320.

Who can help me to confugure the IP200?

Thanks

NOx
-- 
View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24823167.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From ovh at 5ca.com  Wed Aug  5 04:22:10 2009
From: ovh at 5ca.com (Otto)
Date: Wed, 5 Aug 2009 08:22:10 -0300
Subject: [Freeswitch-users] Looking for some FreeSWITCH job
Message-ID: <6B7F51E2C75FB1458C416CFEAD0E822F5D9E5B@ml.5CA.INT>

Diego,

 

You can contact me.

 

Br.

 

Otto van Haaren

www.5ca.com <http://www.5ca.com/> 

ovh at 5ca.com

 

________________________________

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Diego Viola
Sent: dinsdag 4 augustus 2009 23:22
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Looking for some FreeSWITCH job

 

Hi,

I'm currently looking for some FS jobs, I really need one, I'm currently
unemployed and looking for some serious FreeSWITCH jobs.

Anyone?

P.S: I also do FS and web development with any language, PHP, Ruby, etc.
anything really.

Thanks,

Diego

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From david.nembrot at sogeti.com  Wed Aug  5 05:32:10 2009
From: david.nembrot at sogeti.com (David Nembrot)
Date: Wed,  5 Aug 2009 14:32:10 +0200
Subject: [Freeswitch-users] Qualify IM comm. across two distinct SIP domains
Message-ID: <20090805143210.um73qlhsgck80kko@mail.sogeti.com>

Hi everybody,

? ?I've just configured two Freeswitch servers (FS#1 and FS#2) to?
enable SIP communications between their two distinct SIP domains. The?
fact is that the IP telephony is up & running across these two?
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory?
way, my guess is that I'm missing something in the config files in?
order to enable IM services throughout the two domains.. Since they?
are in different networks, it seems reasonable for example to force?
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my?
question:

? ?How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

? ?or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.
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-------------- next part --------------
Hi everybody,

? ?I've just configured two Freeswitch servers (FS#1 and FS#2) to?
enable SIP communications between their two distinct SIP domains. The?
fact is that the IP telephony is up & running across these two?
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory?
way, my guess is that I'm missing something in the config files in?
order to enable IM services throughout the two domains.. Since they?
are in different networks, it seems reasonable for example to force?
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my?
question:

? ?How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

? ?or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.
-------------- next part --------------
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-------------- next part --------------


   Hi everybody,

   I've just configured two Freeswitch servers (FS#1 and FS#2) to  
enable SIP communications between their two distinct SIP domains. The  
fact is that the IP telephony is up & running across these two  
domains, but the IM doesn't seem to be operational...

Basically the dialplan for telephony has been ticked in a satisfactory  
way, my guess is that I'm missing something in the config files in  
order to enable IM services throughout the two domains.. Since they  
are in different networks, it seems reasonable for example to force  
the IM comm. get through the FS#1 so to reach FS#2 domain... hence my  
question:

   How can I shape up a kind of "IM dialplan" on my servers ?
(i.e. set up the routing aspects for IM too)

   or How can I get this IM stuff configured differently?

Thanx & Regards,
David N.


-------------- next part --------------
[Pi??ce jointe retir??e??: type d'origine de la pi??ce jointe: "text/html", nom: "Version_HTML_du_message"]

From dujinfang at gmail.com  Wed Aug  5 06:15:32 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 5 Aug 2009 21:15:32 +0800
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
In-Reply-To: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
References: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
Message-ID: <EC77D164-BAAD-48FA-A3C4-6BFE05CC2E8F@gmail.com>

I think you can check the loopback endpoint or inline dialplan.

On Aug 4, 2009, at 3:23 AM, Michael Frager wrote:
> Hello,
>
> I'm in the process of moving my VOIP application from Asterisk to  
> FreeSWITCH.
>
> I was wondering if it is possible to emulate the call announcement  
> feature that is available on Asterisk.
>
> On Asterisk it looks like this, with the "A(...)" parameter:
>
> Dial(SIP/15555551212|180|A(connecttone1))
>
> Note that this announcement is only played for the called party, the  
> calling party does NOT hear the tone.
>
> I'm guessing this can be done with FreeSWITCH.
>
> Does anyone know how I might accomplish this?
>
>
> Thanks in advance,
>
> -Mike Fragre
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From pjintheusa at gmail.com  Wed Aug  5 06:20:16 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 5 Aug 2009 09:20:16 -0400
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
In-Reply-To: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
References: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
Message-ID: <367751820908050620t40566a84y8422a132142e3c34@mail.gmail.com>

Mike,

I am not familiar with Asterisk so I am not 100% sure this is what you are
looking for, but check out
http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation

It may contain what you need

Phillip Jones


On Mon, Aug 3, 2009 at 3:23 PM, Michael Frager <mike at dialyourleads.com>wrote:

> Hello,
>
> I'm in the process of moving my VOIP application from Asterisk to
> FreeSWITCH.
>
> I was wondering if it is possible to emulate the call announcement feature
> that is available on Asterisk.
>
> On Asterisk it looks like this, with the "A(...)" parameter:
>
> Dial(SIP/15555551212|180|A(connecttone1))
>
> Note that this announcement is only played for the called party, the
> calling party does NOT hear the tone.
>
> I'm guessing this can be done with FreeSWITCH.
>
> Does anyone know how I might accomplish this?
>
>
> Thanks in advance,
>
> -Mike Fragre
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Aug  5 06:22:52 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 08:22:52 -0500
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <24823167.post@talk.nabble.com>
References: <24823167.post@talk.nabble.com>
Message-ID: <DF06C882-055F-488D-B860-46643AF9B22B@freeswitch.org>

Can you ship me a phone to test with?  That's usually the missing  
element when testing this stuff is I just can't afford to buy every  
phone to test with.

/b

On Aug 5, 2009, at 3:52 AM, NOx-WHV wrote:

>
> Hello,
>
> i have a problem using a innovaphone ip200 with freeswitch and tls/ 
> srtp. The
> freeswitch certificate is in the trust list of the phone and it  
> works with
> tls for incomming calls. But outgoing calls were rejected to the  
> mailbox.
> The freeswitch configuration is ok, because it works with a snom 320.
>
> Who can help me to confugure the IP200?
>
> Thanks
>
> NOx
> -- 




From brian at freeswitch.org  Wed Aug  5 06:23:48 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 08:23:48 -0500
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
In-Reply-To: <367751820908050620t40566a84y8422a132142e3c34@mail.gmail.com>
References: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
	<367751820908050620t40566a84y8422a132142e3c34@mail.gmail.com>
Message-ID: <488F9396-69AF-4A50-943D-DD554F946741@freeswitch.org>

That is exactly what he's lookin for.

/b

On Aug 5, 2009, at 8:20 AM, Phillip Jones wrote:

> Mike,
>
> I am not familiar with Asterisk so I am not 100% sure this is what  
> you are looking for, but check out http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
>
> It may contain what you need
>
> Phillip Jones

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From msc at freeswitch.org  Wed Aug  5 06:40:52 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 08:40:52 -0500
Subject: [Freeswitch-users] FreeSWITCH code to emulate Asterisk dialing
	announcement
In-Reply-To: <488F9396-69AF-4A50-943D-DD554F946741@freeswitch.org>
References: <d80cc7bd0908031223l3a37cdeaoce4304d55242e60@mail.gmail.com>
	<367751820908050620t40566a84y8422a132142e3c34@mail.gmail.com>
	<488F9396-69AF-4A50-943D-DD554F946741@freeswitch.org>
Message-ID: <87f2f3b90908050640x5ec18d4ag417f38e31108b301@mail.gmail.com>

Could someone please add this to the FreeSWITCH wiki's Rosetta Stone page?
Thanks!-MC

On Wed, Aug 5, 2009 at 8:23 AM, Brian West <brian at freeswitch.org> wrote:

> That is exactly what he's lookin for.
> /b
>
> On Aug 5, 2009, at 8:20 AM, Phillip Jones wrote:
>
> Mike,
>
> I am not familiar with Asterisk so I am not 100% sure this is what you are
> looking for, but check out
> http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation
>
> It may contain what you need
>
> Phillip Jones
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From max.bridgewater at gmail.com  Wed Aug  5 06:56:39 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Wed, 5 Aug 2009 09:56:39 -0400
Subject: [Freeswitch-users] How to distinguish between the different type of
	call rejections from Javascript?
Message-ID: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>

Hi,

Say i originate a call to a mobile phone and the call fails. There are many
possible reasons: congestion, user busy, call rejected by user, etc. Is
there a way i can get the failure code from Javascript?
Thanks,
Max.
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From dujinfang at gmail.com  Wed Aug  5 07:02:50 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 5 Aug 2009 22:02:50 +0800
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
Message-ID: <D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>

variable_originate_disposition

On Aug 5, 2009, at 9:56 PM, Max Bridgewater wrote:
> Hi,
>
> Say i originate a call to a mobile phone and the call fails. There  
> are many possible reasons: congestion, user busy, call rejected by  
> user, etc. Is there a way i can get the failure code from Javascript?
> Thanks,
> Max.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jim at evolutiontel.net  Wed Aug  5 07:07:50 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Thu, 6 Aug 2009 00:07:50 +1000
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <24823167.post@talk.nabble.com>
References: <24823167.post@talk.nabble.com>
Message-ID: <ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>

Hi NOx,

Can you clarify the direction of the calls.  When you say outgoing do
you mean a call is terminating to the ip200?

I have been down a similar path while testing Eyebeam.  If the
terminating phone sets an option to only accept secure calls and FS
does not send Secure Descriptions in the INVITE, Eyebeam would respond
with 415 response code and the call would fail.  Depending on your
diaplan this could send your call to voicemail.

To fix it I added the following code to dialplan.
<action application="set" data="continue_on_fail=79"/>
<bridge blah blah blah>
<action application="set" data="bypass_media=false"/>
<action application="set" data="proxy_media=true"/>
<action application="set" data="ringback=$${uk-ring}"/>
<action application="pre_answer"/>
<action application="export" data="sip_secure_media=true"/>
<bridge blah blah blah>

The continue on fail captures the 415 response code forces the call to
continue to the next bridge while sip_secure_media forces the second
invite to include security descriptors.  The rest was required because
I did not want to proxy media if the call was not secure, obviously if
the call is secure on a point to point basis FS will have to proxy the
media and this was the only way I could find for it to work.

Hope this helps.

Regards,


On Wed, Aug 5, 2009 at 6:52 PM, NOx-WHV<enno.egbert at googlemail.com> wrote:
>
> Hello,
>
> i have a problem using a innovaphone ip200 with freeswitch and tls/srtp. The
> freeswitch certificate is in the trust list of the phone and it works with
> tls for incomming calls. But outgoing calls were rejected to the mailbox.
> The freeswitch configuration is ok, because it works with a snom 320.
>
> Who can help me to confugure the IP200?
>
> Thanks
>
> NOx
> --
> View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24823167.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net



From gregt at cgicommunications.com  Wed Aug  5 07:40:31 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Wed, 5 Aug 2009 10:40:31 -0400
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
Message-ID: <63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>

Trying to make phpmod and it fails with this:

/usr/bin/ld: cannot find -laspell
collect2: ld returned 1 exit status
make[1]: *** [ESL.so] Error 1

I do have php-devel on this Centos 5.2 machine. Any ideas?
--
Greg
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From brian at freeswitch.org  Wed Aug  5 07:46:51 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 09:46:51 -0500
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
Message-ID: <2B197B9C-9854-4888-8726-0402C2F0689A@freeswitch.org>

install aspell-devel

also Please don't hijack threads... please click new message and start  
a new thread.

Thanks,
Brian

On Aug 5, 2009, at 9:40 AM, Greg Thoen wrote:

> Trying to make phpmod and it fails with this:
>
> /usr/bin/ld: cannot find -laspell
> collect2: ld returned 1 exit status
> make[1]: *** [ESL.so] Error 1
>
> I do have php-devel on this Centos 5.2 machine. Any ideas?
> --
> Greg

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From william.suffill at gmail.com  Wed Aug  5 07:47:38 2009
From: william.suffill at gmail.com (William Suffill)
Date: Wed, 5 Aug 2009 10:47:38 -0400
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca> 
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com> 
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca> 
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com> 
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
Message-ID: <6b65470d0908050747r46da5b14q6ac3f955ad110e71@mail.gmail.com>

You don't have the dev library for aspell

yum install aspell-devel

should do the tick.

-- W



From msc at freeswitch.org  Wed Aug  5 07:54:45 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 5 Aug 2009 09:54:45 -0500
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>
References: <24823167.post@talk.nabble.com>
	<ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>
Message-ID: <95500CDE-C8A9-4F37-958D-43F27B5A6B3F@freeswitch.org>

Jim,

Just curious - could you document this use case on the wiki? Maybe you  
could create a page describing the setup and then link to it from the  
TLS page.

Thanks,
MC

Sent from my iPhone

On Aug 5, 2009, at 9:07 AM, Jim Burke <jim at evolutiontel.net> wrote:

> Hi NOx,
>
> Can you clarify the direction of the calls.  When you say outgoing do
> you mean a call is terminating to the ip200?
>
> I have been down a similar path while testing Eyebeam.  If the
> terminating phone sets an option to only accept secure calls and FS
> does not send Secure Descriptions in the INVITE, Eyebeam would respond
> with 415 response code and the call would fail.  Depending on your
> diaplan this could send your call to voicemail.
>
> To fix it I added the following code to dialplan.
> <action application="set" data="continue_on_fail=79"/>
> <bridge blah blah blah>
> <action application="set" data="bypass_media=false"/>
> <action application="set" data="proxy_media=true"/>
> <action application="set" data="ringback=$${uk-ring}"/>
> <action application="pre_answer"/>
> <action application="export" data="sip_secure_media=true"/>
> <bridge blah blah blah>
>
> The continue on fail captures the 415 response code forces the call to
> continue to the next bridge while sip_secure_media forces the second
> invite to include security descriptors.  The rest was required because
> I did not want to proxy media if the call was not secure, obviously if
> the call is secure on a point to point basis FS will have to proxy the
> media and this was the only way I could find for it to work.
>
> Hope this helps.
>
> Regards,
>
>
> On Wed, Aug 5, 2009 at 6:52 PM, NOx-WHV<enno.egbert at googlemail.com>  
> wrote:
>>
>> Hello,
>>
>> i have a problem using a innovaphone ip200 with freeswitch and tls/ 
>> srtp. The
>> freeswitch certificate is in the trust list of the phone and it  
>> works with
>> tls for incomming calls. But outgoing calls were rejected to the  
>> mailbox.
>> The freeswitch configuration is ok, because it works with a snom 320.
>>
>> Who can help me to confugure the IP200?
>>
>> Thanks
>>
>> NOx
>> --
>> View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24823167.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> -- 
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From helmut.kuper at ewetel.de  Wed Aug  5 07:56:30 2009
From: helmut.kuper at ewetel.de (Helmut Kuper)
Date: Wed, 05 Aug 2009 16:56:30 +0200
Subject: [Freeswitch-users] pocketsphinx
In-Reply-To: <87f2f3b90907311145p6c8d5907mc4545c710e9605fb@mail.gmail.com>
References: <4A5739DE.1080800@ewetel.de> <4A72EF33.4070404@ewetel.de>
	<87f2f3b90907311145p6c8d5907mc4545c710e9605fb@mail.gmail.com>
Message-ID: <4A799D9E.2020103@ewetel.de>

-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1

Hi Michael,

today I put my documentation on FS wiki describing the steps to get a
8kHz sample rate acoustic model basing on voxforge's data for german
language. It's not complete, yet.

You can found it here: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx

regards
Helmut


On 31.07.2009 20:45, Michael Collins wrote:
> Helmut,
> 
> Your hard work is appreciated. Like Brian said, we'd all be interested
> in knowing more. Please feel free to put this on the wiki or see me off
> list and we'll discuss further how to document it for the good of the FS
> community.
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.7 (MingW32)

iD8DBQFKeZ2e4tZeNddg3dwRAlwuAKCXK6b/f3J7tRmcev0/EPAUFGZBbgCfXMQW
B8MAREKeR82dTFnYyFeutig=
=1/Jr
-----END PGP SIGNATURE-----



From woodydickson at gmail.com  Wed Aug  5 08:05:20 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 5 Aug 2009 23:05:20 +0800
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
Message-ID: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>

Hi,

I want to implement a module where freeSWITCH would try to bridge to an
extension and if the bridging operation fails, my module can use the hangup
code to determine the next cause of action.

With switch_caller_extension_add_application(session, extension, "bridge",
"sofia/gateway/mygw/1232323);, if there is an error ( 503 received for
instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or the
module's APP) and go on to the next action.  Is there anyway to control it
so that freeSWITCH would remain to be within the module's APP funtion and
continue executing the code after switch_call_extension_add_application,
when let's say a 4XX or 5XX or CANCEL ( from originator) is received?

I have tried it and found that if the bridging is successful, freeSWITCH
would continue executing the code after
switch_caller_extension_add_application, but if an error is received, then
it would just move on to the next action.

Does anyone know how to deal with this problem?

Woody
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From gregt at cgicommunications.com  Wed Aug  5 08:21:41 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Wed, 5 Aug 2009 11:21:41 -0400
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <2B197B9C-9854-4888-8726-0402C2F0689A@freeswitch.org>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
	<2B197B9C-9854-4888-8726-0402C2F0689A@freeswitch.org>
Message-ID: <BFC0A2A3-F0D5-40DC-8E1B-AA2FB2FFB372@cgicommunications.com>

Oops, what thread did I hijack? I just sent a new email message to freeswitch-users at lists.freeswitch.org
What should I have done?
--
Greg


On Aug 5, 2009, at 10:46 AM, Brian West wrote:

> install aspell-devel
>
> also Please don't hijack threads... please click new message and  
> start a new thread.
>
> Thanks,
> Brian
>
> On Aug 5, 2009, at 9:40 AM, Greg Thoen wrote:
>
>> Trying to make phpmod and it fails with this:
>>
>> /usr/bin/ld: cannot find -laspell
>> collect2: ld returned 1 exit status
>> make[1]: *** [ESL.so] Error 1
>>
>> I do have php-devel on this Centos 5.2 machine. Any ideas?
>> --
>> Greg
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Wed Aug  5 08:39:31 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 10:39:31 -0500
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <BFC0A2A3-F0D5-40DC-8E1B-AA2FB2FFB372@cgicommunications.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
	<2B197B9C-9854-4888-8726-0402C2F0689A@freeswitch.org>
	<BFC0A2A3-F0D5-40DC-8E1B-AA2FB2FFB372@cgicommunications.com>
Message-ID: <44BA0B61-3971-479B-816C-5501035DB28D@freeswitch.org>

Re: [Freeswitch-users] ESL Wrapper



It happens when you click reply... change the subject and the body...  
thats how you hijack a thread.  :)  Happens to the best of us.



/b





On Aug 5, 2009, at 10:21 AM, Greg Thoen wrote:

> Oops, what thread did I hijack? I just sent a new email message to freeswitch-users at lists.freeswitch.org
> What should I have done?
> --
> Greg
>
>
> On Aug 5, 2009, at 10:46 AM, Brian West wrote:
>
>> install aspell-devel
>>
>> also Please don't hijack threads... please click new message and  
>> start a new thread.
>>
>> Thanks,
>> Brian
>>
>> On Aug 5, 2009, at 9:40 AM, Greg Thoen wrote:
>>
>>> Trying to make phpmod and it fails with this:
>>>
>>> /usr/bin/ld: cannot find -laspell
>>> collect2: ld returned 1 exit status
>>> make[1]: *** [ESL.so] Error 1
>>>
>>> I do have php-devel on this Centos 5.2 machine. Any ideas?
>>> --
>>> Greg
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jmesquita at gmail.com  Wed Aug  5 09:15:30 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 5 Aug 2009 13:15:30 -0300
Subject: [Freeswitch-users] How to change the contact when fs sending
	REGISTER??
In-Reply-To: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>
References: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>
Message-ID: <5a8712120908050915k2f414a37vd651f9323114cdbf@mail.gmail.com>

Add the following line to the gw definition:

<param name="extension-in-contact" value="true"/>

jmesquita

On Wed, Aug 5, 2009 at 7:26 AM, Brad Tuan <brad.tuan at gmail.com> wrote:

> As title ,I know how to do when sending INVITE
>
> but how to do it when fs sending REGISTER??
>
> For example , when gateway registering , the contact is
> gw+abcd at XXX.XXX.XXX.XXX ,
>
> how to change it to *abcd at XXX.XXX.XXX.XXX??*
> **
> *Please help*
> **
> **
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at gmail.com  Wed Aug  5 09:23:26 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 5 Aug 2009 13:23:26 -0300
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
Message-ID: <5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>

My guess is that you will receive a message here:

switch_status_t channel_receive_message(switch_core_session_t *session,
switch_core_session_message_t *msg)

The problem here is that you don't have the exact SIP code but there is a
clear relationship between the codes and the messages you receive on the
channel, so I am guessing that is all the same.

Hope this helps.

jmesquita

On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> I want to implement a module where freeSWITCH would try to bridge to an
> extension and if the bridging operation fails, my module can use the hangup
> code to determine the next cause of action.
>
> With switch_caller_extension_add_application(session, extension, "bridge",
> "sofia/gateway/mygw/1232323);, if there is an error ( 503 received for
> instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or the
> module's APP) and go on to the next action.  Is there anyway to control it
> so that freeSWITCH would remain to be within the module's APP funtion and
> continue executing the code after switch_call_extension_add_application,
> when let's say a 4XX or 5XX or CANCEL ( from originator) is received?
>
> I have tried it and found that if the bridging is successful, freeSWITCH
> would continue executing the code after
> switch_caller_extension_add_application, but if an error is received, then
> it would just move on to the next action.
>
> Does anyone know how to deal with this problem?
>
> Woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Wed Aug  5 09:34:50 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 5 Aug 2009 11:34:50 -0500
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
Message-ID: <E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>

The hangup cause will be in the originate_disposition channel variable  
on the A-leg.

sip_term_status will contain the sip code and  
proto_specific_hangup_cause will contain sip:<code>.


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 5-Aug-09 um 11:23 AM schrieb Jo?o Mesquita:

> My guess is that you will receive a message here:
>
> switch_status_t channel_receive_message(switch_core_session_t  
> *session, switch_core_session_message_t *msg)
>
> The problem here is that you don't have the exact SIP code but there  
> is a clear relationship between the codes and the messages you  
> receive on the channel, so I am guessing that is all the same.
>
> Hope this helps.
>
> jmesquita
>
> On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson  
> <woodydickson at gmail.com> wrote:
> Hi,
>
> I want to implement a module where freeSWITCH would try to bridge to  
> an extension and if the bridging operation fails, my module can use  
> the hangup code to determine the next cause of action.
>
> With switch_caller_extension_add_application(session, extension,  
> "bridge", "sofia/gateway/mygw/1232323);, if there is an error ( 503  
> received for instance ) in the outgoing INVITE, freeSWITCH would  
> leave my module ( or the module's APP) and go on to the next  
> action.  Is there anyway to control it so that freeSWITCH would  
> remain to be within the module's APP funtion and continue executing  
> the code after switch_call_extension_add_application, when let's say  
> a 4XX or 5XX or CANCEL ( from originator) is received?
>
> I have tried it and found that if the bridging is successful,  
> freeSWITCH would continue executing the code after  
> switch_caller_extension_add_application, but if an error is  
> received, then it would just move on to the next action.
>
> Does anyone know how to deal with this problem?
>
> Woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mayamatakeshi at gmail.com  Wed Aug  5 09:35:38 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 6 Aug 2009 01:35:38 +0900
Subject: [Freeswitch-users] Monitoring On-Hold/Off-Hold
Message-ID: <15b9404e0908050935m62a6581bxec3f3055e013753f@mail.gmail.com>

Hello,
I'm using mod_event_socket to monitor FS.
I'm using "events plain ALL' and I get lots of channel events. But
curiously, when some channel puts the call on-hold/off-hold, I don't
get any notification. Is it possible to get these events? Am I missing
some setting?

regards,
takeshi



From gregt at cgicommunications.com  Wed Aug  5 10:31:23 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Wed, 5 Aug 2009 13:31:23 -0400
Subject: [Freeswitch-users] Best practice for inbound calls with scripting
Message-ID: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>

Hi. I am setting up a large inbound only system with multiple DIDs  
coming in; each call is processed fairly intensively with db lookups,  
wav files played, pocketsphinx is used, wav files recorded, etc.

Before I get too far down one path, I was wondering if anyone had any  
insight into the best, most scaleable way to do this out of the  
several methods I can do:

1	Call comes in
	dialplan calls specific javascript based on DID
	javascript uses ODBC to pull info from local mysql db
	call is handled in javascript

2	Call comes in
	dialplan calls specific javascript based on DID
	javascript uses CURL to get info from local mysql db
	call is handled in javascript

3	Call comes in
	php socket is listening for call
	php script runs, pulling info from mysql db
	call is handled in php using esl.php

4	Call comes in
	dialplan calls specific lua script  based on DID
	lua uses luasql.mysql to get info from local mysql db
	call is handled in lua using lua api

5	Call comes in
	xml_curl is used for dynamic dialplan
	js called, continues like #1

I know that they will all do essentially the same thing. But once I go  
down the path, I don't want to find out that the way I chose chokes  
with 50 simultaneous inbound calls on different DIDs. Any comments  
would be appreciated. Thanks.
--
Greg


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From jmesquita at gmail.com  Wed Aug  5 10:55:37 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 5 Aug 2009 14:55:37 -0300
Subject: [Freeswitch-users] Monitoring On-Hold/Off-Hold
In-Reply-To: <15b9404e0908050935m62a6581bxec3f3055e013753f@mail.gmail.com>
References: <15b9404e0908050935m62a6581bxec3f3055e013753f@mail.gmail.com>
Message-ID: <5a8712120908051055k95f01d4x61ef7b7dff80298a@mail.gmail.com>

I only see one way out of this. If you manage presence, an event like the
following is sent:

Event-Name: PRESENCE_IN
Core-UUID: 189b12c0-7fb0-11de-b0bc-37eec03ad00f
FreeSWITCH-Hostname: cl-t146-421cl
FreeSWITCH-IPv4: XXXXXX
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-05%2013%3A42%3A24
Event-Date-GMT: Wed,%2005%20Aug%202009%2017%3A42%3A24%20GMT
Event-Date-Timestamp: 1249494144628132
Event-Calling-File: switch_channel.c
Event-Calling-Function: switch_channel_presence
Event-Calling-Line-Number: 472
Channel-State: CS_HIBERNATE
Channel-State-Number: 8
Channel-Name: XXXXX
Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f
Call-Direction: inbound
Presence-Call-Direction: inbound
Answer-State: answered
Caller-Username: 1000
Caller-Dialplan: XML
Caller-Caller-ID-Name: Mesquita
Caller-Caller-ID-Number: 1000
Caller-Network-Addr: XXXXX
Caller-Destination-Number: 1005
Caller-Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f
Caller-Source: mod_sofia
Caller-Context: XXXXX
Caller-Channel-Name: XXXXX
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249494132128119
Caller-Channel-Created-Time: 1249494132128119
Caller-Channel-Answered-Time: 1249494139500129
Caller-Channel-Progress-Time: 1249494132368119
Caller-Channel-Progress-Media-Time: 0
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Other-Leg-Username: 1000
Other-Leg-Dialplan: XML
Other-Leg-Caller-ID-Name: Joao%20Mesquita
Other-Leg-Caller-ID-Number: 1000
Other-Leg-Network-Addr: 190.2.41.65
Other-Leg-Destination-Number:
sip%3A1005%40192.168.0.106%3A4559%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1005%2540190.2.41.65%253A4559
Other-Leg-Unique-ID: 4e7622ac-81e7-11de-b0bc-37eec03ad00f
Other-Leg-Source: mod_sofia
Other-Leg-Context: XXXXX
Other-Leg-Channel-Name: XXXXXX
Other-Leg-Screen-Bit: true
Other-Leg-Privacy-Hide-Name: false
Other-Leg-Privacy-Hide-Number: false
proto: src/switch_channel.c
login: src/switch_channel.c
from: XXXXXX
rpid: unknown
status: hold
event_type: presence
alt_event_type: dialog
event_count: 3

Content-Length: 543
Content-Type: text/event-plain

Other than that, I think it can be patched. I will take a look at it.

Guys, should this be patched on the state machine itself or on the mod_sofia
channel_receive_message?

jmesquita


On Wed, Aug 5, 2009 at 1:35 PM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> Hello,
> I'm using mod_event_socket to monitor FS.
> I'm using "events plain ALL' and I get lots of channel events. But
> curiously, when some channel puts the call on-hold/off-hold, I don't
> get any notification. Is it possible to get these events? Am I missing
> some setting?
>
> regards,
> takeshi
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dule.maillist at gmail.com  Wed Aug  5 10:59:38 2009
From: dule.maillist at gmail.com (Dan Le)
Date: Wed, 5 Aug 2009 13:59:38 -0400
Subject: [Freeswitch-users] Best practice for inbound calls with
	scripting
In-Reply-To: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
References: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
Message-ID: <914fc92a0908051059na35f58ar3f102ef4d80e0027@mail.gmail.com>

One common recommendation is to use lua over js, since it's lighter-weight,
using less resources.
Dan

On Wed, Aug 5, 2009 at 1:31 PM, Greg Thoen <gregt at cgicommunications.com>wrote:

> Hi. I am setting up a large inbound only system with multiple DIDs coming
> in; each call is processed fairly intensively with db lookups, wav files
> played, pocketsphinx is used, wav files recorded, etc.
> Before I get too far down one path, I was wondering if anyone had any
> insight into the best, most scaleable way to do this out of the several
> methods I can do:
> 1 Call comes in
> dialplan calls specific javascript based on DID
> javascript uses ODBC to pull info from local mysql db
> call is handled in javascript
>
> 2 Call comes in
> dialplan calls specific javascript based on DID
> javascript uses CURL to get info from local mysql db
> call is handled in javascript
>
> 3 Call comes in
> php socket is listening for call
> php script runs, pulling info from mysql db
> call is handled in php using esl.php
>
> 4 Call comes in
> dialplan calls specific lua script  based on DID
> lua uses luasql.mysql to get info from local mysql db
> call is handled in lua using lua api
>
> 5 Call comes in
> xml_curl is used for dynamic dialplan
> js called, continues like #1
>
> I know that they will all do
> essentially the same thing. But once I go down the path, I don't want to find out that the way I chose chokes with 50 simultaneous inbound calls on different DIDs. Any comments would be appreciated. Thanks.
>
> --
>
> Greg
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From technical at ttnc.co.uk  Wed Aug  5 11:20:46 2009
From: technical at ttnc.co.uk (TTNC - Adnan Barakat)
Date: Wed, 05 Aug 2009 19:20:46 +0100
Subject: [Freeswitch-users] Best practice for inbound calls with
	scripting
In-Reply-To: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
References: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
Message-ID: <4A79CD7E.6010806@ttnc.co.uk>

Greg Thoen wrote:
> 4 Call comes in
> dialplan calls specific lua script  based on DID
> lua uses luasql.mysql to get info from local mysql db
> call is handled in lua using lua api
This is the option we chose in our set-up (although mysql is on a remote 
server), currently we have over 25k DIDs running on FreeSWITCH via this 
set-up and everything is working wonderfully.

We were using option 5 previously, but found 4 to be much better in general.

Hope this helps.


Adnan



From edpimentl at gmail.com  Wed Aug  5 11:48:16 2009
From: edpimentl at gmail.com (EdPimentl)
Date: Wed, 5 Aug 2009 14:48:16 -0400
Subject: [Freeswitch-users] Best practice for inbound calls with
	scripting
In-Reply-To: <914fc92a0908051059na35f58ar3f102ef4d80e0027@mail.gmail.com>
References: <7AC88983-9003-42CF-8015-E7B6CFC69979@cgicommunications.com>
	<914fc92a0908051059na35f58ar3f102ef4d80e0027@mail.gmail.com>
Message-ID: <9dc4a1670908051148g38958616j57c2a7eebf9f607f@mail.gmail.com>

Here are other scenarios to add

Call request comes in via SMS
Call request comes in via XMPP
Call request comes in via EMAIL
Call request comes in via TWITTER/FACEBOOK/WAVE/

Best regards,
-E
Gpro.ws
edpimentl [SKype ]



On Wed, Aug 5, 2009 at 1:59 PM, Dan Le <dule.maillist at gmail.com> wrote:

> One common recommendation is to use lua over js, since it's lighter-weight,
> using less resources.
> Dan
>
> On Wed, Aug 5, 2009 at 1:31 PM, Greg Thoen <gregt at cgicommunications.com>wrote:
>
>> Hi. I am setting up a large inbound only system with multiple DIDs coming
>> in; each call is processed fairly intensively with db lookups, wav files
>> played, pocketsphinx is used, wav files recorded, etc.
>> Before I get too far down one path, I was wondering if anyone had any
>> insight into the best, most scaleable way to do this out of the several
>> methods I can do:
>> 1 Call comes in
>>  dialplan calls specific javascript based on DID
>> javascript uses ODBC to pull info from local mysql db
>> call is handled in javascript
>>
>> 2 Call comes in
>> dialplan calls specific javascript based on DID
>> javascript uses CURL to get info from local mysql db
>>  call is handled in javascript
>>
>> 3 Call comes in
>> php socket is listening for call
>>  php script runs, pulling info from mysql db
>> call is handled in php using esl.php
>>
>> 4 Call comes in
>>  dialplan calls specific lua script  based on DID
>> lua uses luasql.mysql to get info from local mysql db
>> call is handled in lua using lua api
>>
>> 5 Call comes in
>> xml_curl is used for dynamic dialplan
>> js called, continues like #1
>>
>> I know that they will all do
>> essentially the same thing. But once I go down the path, I don't want to find out that the way I chose chokes with 50 simultaneous inbound calls on different DIDs. Any comments would be appreciated. Thanks.
>>
>> --
>>
>> Greg
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kjoseph.us at gmail.com  Wed Aug  5 13:24:50 2009
From: kjoseph.us at gmail.com (Joseph Khoury)
Date: Wed, 5 Aug 2009 13:24:50 -0700
Subject: [Freeswitch-users] Looking for some FreeSWITCH job
In-Reply-To: <6B7F51E2C75FB1458C416CFEAD0E822F5D9E5B@ml.5CA.INT>
References: <6B7F51E2C75FB1458C416CFEAD0E822F5D9E5B@ml.5CA.INT>
Message-ID: <7d1481c30908051324j5b806cb6y527162037470fead@mail.gmail.com>

Diego,
Contact me also.

Joseph

www.alosmart.com
(myname) at alosmart.com

On Wed, Aug 5, 2009 at 4:22 AM, Otto <ovh at 5ca.com> wrote:

>  Diego,
>
>
>
> You can contact me.
>
>
>
> Br.
>
>
>
> Otto van Haaren
>
> www.5ca.com
>
> ovh at 5ca.com
>
>
>  ------------------------------
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
> *Sent:* dinsdag 4 augustus 2009 23:22
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* [Freeswitch-users] Looking for some FreeSWITCH job
>
>
>
> Hi,
>
> I'm currently looking for some FS jobs, I really need one, I'm currently
> unemployed and looking for some serious FreeSWITCH jobs.
>
> Anyone?
>
> P.S: I also do FS and web development with any language, PHP, Ruby, etc.
> anything really.
>
> Thanks,
>
> Diego
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tomabroad at gmail.com  Wed Aug  5 11:46:37 2009
From: tomabroad at gmail.com (tom)
Date: Wed, 5 Aug 2009 14:46:37 -0400
Subject: [Freeswitch-users] q: fresh installation: Couldnt open
	/usr/local/freeswitch/conf/freeswitch.xml
Message-ID: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>

hi just installed freeswitch via svn.
- bootstrap
- configure
- make install
- ./freeswitch

gives me:
acerdebian:/usr/local/freeswitch/bin# ./freeswitch
Error: stacksize 4194303 is too large: run ulimit -s 240 or run ./freeswitch
-waste.
auto-adjusting stack size for optimal performance....
2009-08-05 14:41:16.908170 [INFO] switch_event.c:565 Activate Eventing
Engine.
2009-08-05 14:41:16.910182 [DEBUG] switch_event.c:553 Create event dispatch
thread 0
2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open
/usr/local/freeswitch/conf/freeswitch.xml (No such file or directory)
Cannot Initialize [Cannot Open log directory or XML Root!]


bump - help

thx
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From mattdfong at gmail.com  Wed Aug  5 13:52:13 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Wed, 5 Aug 2009 13:52:13 -0700
Subject: [Freeswitch-users] q: fresh installation: Couldnt open
	/usr/local/freeswitch/conf/freeswitch.xml
In-Reply-To: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
References: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
Message-ID: <4256bf830908051352t26753d57nbeb6ba27f2a5f77a@mail.gmail.com>

Does the file exist at /usr/local/freeswitch/conf/freeswitch.xml? does the
user you are executing freeswitch as have permission to read the file?
--matt
hello hunter - hosted predictive dialer & voice broadcasting
http://www.hellohunter.com

On Wed, Aug 5, 2009 at 11:46 AM, tom <tomabroad at gmail.com> wrote:

> hi just installed freeswitch via svn.
> - bootstrap
> - configure
> - make install
> - ./freeswitch
>
> gives me:
> acerdebian:/usr/local/freeswitch/bin# ./freeswitch
> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
> ./freeswitch -waste.
> auto-adjusting stack size for optimal performance....
> 2009-08-05 14:41:16.908170 [INFO] switch_event.c:565 Activate Eventing
> Engine.
> 2009-08-05 14:41:16.910182 [DEBUG] switch_event.c:553 Create event dispatch
> thread 0
> 2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt
> open /usr/local/freeswitch/conf/freeswitch.xml (No such file or directory)

Cannot Initialize [Cannot Open log directory or XML Root!]
>
>
> bump - help
>
> thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From sprice at gmail.com  Wed Aug  5 13:53:16 2009
From: sprice at gmail.com (SP)
Date: Wed, 5 Aug 2009 15:53:16 -0500
Subject: [Freeswitch-users] q: fresh installation: Couldnt open
	/usr/local/freeswitch/conf/freeswitch.xml
In-Reply-To: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
References: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
Message-ID: <7e2ac3270908051353m33c3f158r4bd7ec2d415f7a7a@mail.gmail.com>

2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open
/usr/local/freeswitch/conf/freeswitch.xml *(No such file or directory)*

On Wed, Aug 5, 2009 at 13:46, tom<tomabroad at gmail.com> wrote:
> hi just installed freeswitch via svn.
> - bootstrap
> - configure
> - make install
> - ./freeswitch
>
> gives me:
> acerdebian:/usr/local/freeswitch/bin# ./freeswitch
> Error: stacksize 4194303 is too large: run ulimit -s 240 or run
./freeswitch
> -waste.
> auto-adjusting stack size for optimal performance....
> 2009-08-05 14:41:16.908170 [INFO] switch_event.c:565 Activate Eventing
> Engine.
> 2009-08-05 14:41:16.910182 [DEBUG] switch_event.c:553 Create event
dispatch
> thread 0
> 2009-08-05 14:41:16.910763 [ERR] switch_xml.c:1282 Couldnt open
> /usr/local/freeswitch/conf/freeswitch.xml (No such file or directory)
> Cannot Initialize [Cannot Open log directory or XML Root!]
>
>
> bump - help
>
> thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Shannon
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From raffaele.p.guidi at gmail.com  Wed Aug  5 13:54:46 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Wed, 5 Aug 2009 22:54:46 +0200
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
Message-ID: <b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>

interesting! what values can contain "variable_originate_disposition"? And
can I set them manually in a script to reject a call simulating user busy or
call rejected? A lua example?
Thanks,
   Raffaele

On Wed, Aug 5, 2009 at 16:02, Seven Du <dujinfang at gmail.com> wrote:

> variable_originate_disposition
>
> On Aug 5, 2009, at 9:56 PM, Max Bridgewater wrote:
> > Hi,
> >
> > Say i originate a call to a mobile phone and the call fails. There
> > are many possible reasons: congestion, user busy, call rejected by
> > user, etc. Is there a way i can get the failure code from Javascript?
> > Thanks,
> > Max.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Aug  5 13:59:07 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 15:59:07 -0500
Subject: [Freeswitch-users] How to change the contact when fs sending
	REGISTER??
In-Reply-To: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>
References: <f2179ba0908050326x2881d55fn44c501e91bfdccf2@mail.gmail.com>
Message-ID: <978FA06A-8591-44DF-BC08-5CAC2668F6ED@freeswitch.org>

My first question is why would you have to change it?  :P

/b

On Aug 5, 2009, at 5:26 AM, Brad Tuan wrote:

> As title ,I know how to do when sending INVITE
>
> but how to do it when fs sending REGISTER??
>
> For example , when gateway registering , the contact is gw+abcd at XXX.XXX.XXX.XXX 
>  ,
>
> how to change it to abcd at XXX.XXX.XXX.XXX??
>
> Please help
>

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From gmaruzz at celliax.org  Wed Aug  5 14:00:17 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 5 Aug 2009 23:00:17 +0200
Subject: [Freeswitch-users] freepbx for freeswitch
Message-ID: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>

Yay!

http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-future

Darren Schreiber has made the announcement and is doinng a
presentation of FreePBX V3 right now at www.cluecon.com.



From tomabroad at gmail.com  Wed Aug  5 14:09:37 2009
From: tomabroad at gmail.com (tom)
Date: Wed, 5 Aug 2009 17:09:37 -0400
Subject: [Freeswitch-users] q: fresh installation: Couldnt open
	/usr/local/freeswitch/conf/freeswitch.xml
In-Reply-To: <7e2ac3270908051353m33c3f158r4bd7ec2d415f7a7a@mail.gmail.com>
References: <6f7c60c40908051146r6f819bd7g4127d85899f887a6@mail.gmail.com>
	<7e2ac3270908051353m33c3f158r4bd7ec2d415f7a7a@mail.gmail.com>
Message-ID: <6f7c60c40908051409h57277282m6f6e94af47b6d75d@mail.gmail.com>

solved via "make samples"

thx tom
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From nik.middleton at noblesolutions.co.uk  Wed Aug  5 14:24:47 2009
From: nik.middleton at noblesolutions.co.uk (Nik Middleton)
Date: Wed, 5 Aug 2009 22:24:47 +0100
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
Message-ID: <A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>

I'd heard rumours that this was going to happen and it's great news and
good news for FS as well.  With a user friendly front end, FS is sure to
fly.  I have no doubt that this will be the first of many.

Regards,

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Giovanni Maruzzelli
Sent: 05 August 2009 22:00
To: freeswitch-users at lists.freeswitch.org;
freeswitch-dev at lists.freeswitch.org
Subject: [Freeswitch-users] freepbx for freeswitch

Yay!

http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
ure

Darren Schreiber has made the announcement and is doinng a
presentation of FreePBX V3 right now at www.cluecon.com.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From msc at freeswitch.org  Wed Aug  5 14:35:30 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 16:35:30 -0500
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
Message-ID: <87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>

Of course, don't forget this:


For those who want a commercial solution built upon FreeSWITCH: You've got
it!
-MC

On Wed, Aug 5, 2009 at 4:24 PM, Nik Middleton <
nik.middleton at noblesolutions.co.uk> wrote:

> I'd heard rumours that this was going to happen and it's great news and
> good news for FS as well.  With a user friendly front end, FS is sure to
> fly.  I have no doubt that this will be the first of many.
>
> Regards,
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Giovanni Maruzzelli
> Sent: 05 August 2009 22:00
> To: freeswitch-users at lists.freeswitch.org;
> freeswitch-dev at lists.freeswitch.org
> Subject: [Freeswitch-users] freepbx for freeswitch
>
> Yay!
>
> http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
> ure
>
> Darren Schreiber has made the announcement and is doinng a
> presentation of FreePBX V3 right now at www.cluecon.com.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Wed Aug  5 14:49:49 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 5 Aug 2009 17:49:49 -0400
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
	<87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>
Message-ID: <86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>

Yay for FreePBX =D

Does Cudatel uses FreeSWITCH as the engine?

On Wed, Aug 5, 2009 at 5:35 PM, Michael Collins<msc at freeswitch.org> wrote:
> Of course, don't forget this:
> 
>
> For those who want a commercial solution built upon FreeSWITCH: You've got
> it!
> -MC
>
> On Wed, Aug 5, 2009 at 4:24 PM, Nik Middleton
> <nik.middleton at noblesolutions.co.uk> wrote:
>>
>> I'd heard rumours that this was going to happen and it's great news and
>> good news for FS as well. ?With a user friendly front end, FS is sure to
>> fly. ?I have no doubt that this will be the first of many.
>>
>> Regards,
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Giovanni Maruzzelli
>> Sent: 05 August 2009 22:00
>> To: freeswitch-users at lists.freeswitch.org;
>> freeswitch-dev at lists.freeswitch.org
>> Subject: [Freeswitch-users] freepbx for freeswitch
>>
>> Yay!
>>
>> http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
>> ure
>>
>> Darren Schreiber has made the announcement and is doinng a
>> presentation of FreePBX V3 right now at www.cluecon.com.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From sprice at gmail.com  Wed Aug  5 14:59:31 2009
From: sprice at gmail.com (SP)
Date: Wed, 5 Aug 2009 16:59:31 -0500
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
	<87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>
	<86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>
Message-ID: <7e2ac3270908051459l16db1152k8eb6734add1957a3@mail.gmail.com>

would it be pimped if it didn't?

On Wed, Aug 5, 2009 at 16:49, Diego Viola<diego.viola at gmail.com> wrote:
> Yay for FreePBX =D
>
> Does Cudatel uses FreeSWITCH as the engine?
>
> On Wed, Aug 5, 2009 at 5:35 PM, Michael Collins<msc at freeswitch.org> wrote:
>> Of course, don't forget this:
>> 
>>
>> For those who want a commercial solution built upon FreeSWITCH: You've got
>> it!
>> -MC
>>
>> On Wed, Aug 5, 2009 at 4:24 PM, Nik Middleton
>> <nik.middleton at noblesolutions.co.uk> wrote:
>>>
>>> I'd heard rumours that this was going to happen and it's great news and
>>> good news for FS as well. ?With a user friendly front end, FS is sure to
>>> fly. ?I have no doubt that this will be the first of many.
>>>
>>> Regards,
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Giovanni Maruzzelli
>>> Sent: 05 August 2009 22:00
>>> To: freeswitch-users at lists.freeswitch.org;
>>> freeswitch-dev at lists.freeswitch.org
>>> Subject: [Freeswitch-users] freepbx for freeswitch
>>>
>>> Yay!
>>>
>>> http://freepbx.org/news/2009-08-04/freepbx-v3-come-help-us-shape-the-fut
>>> ure
>>>
>>> Darren Schreiber has made the announcement and is doinng a
>>> presentation of FreePBX V3 right now at www.cluecon.com.
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shannon



From msc at freeswitch.org  Wed Aug  5 15:00:06 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 17:00:06 -0500
Subject: [Freeswitch-users] freepbx for freeswitch
In-Reply-To: <86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>
References: <7b197bef0908051400o67159234s8e3968f8831e7ca8@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147498@sun.Noble.local>
	<87f2f3b90908051435n5f075bedpfb4362cc58b6480f@mail.gmail.com>
	<86a32abc0908051449i40cb8831p54da158a8d5b4832@mail.gmail.com>
Message-ID: <87f2f3b90908051500u59e5f6ack4402818e226df7a7@mail.gmail.com>

On Wed, Aug 5, 2009 at 4:49 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Yay for FreePBX =D
>
> Does Cudatel uses FreeSWITCH as the engine?


Yes it does!
-MC
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From msc at freeswitch.org  Wed Aug  5 15:02:28 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 17:02:28 -0500
Subject: [Freeswitch-users] pocketsphinx
In-Reply-To: <4A799D9E.2020103@ewetel.de>
References: <4A5739DE.1080800@ewetel.de> <4A72EF33.4070404@ewetel.de>
	<87f2f3b90907311145p6c8d5907mc4545c710e9605fb@mail.gmail.com>
	<4A799D9E.2020103@ewetel.de>
Message-ID: <87f2f3b90908051502o3943cbcdy35c91cd11174d743@mail.gmail.com>

On Wed, Aug 5, 2009 at 9:56 AM, Helmut Kuper <helmut.kuper at ewetel.de> wrote:

> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Hi Michael,
>
> today I put my documentation on FS wiki describing the steps to get a
> 8kHz sample rate acoustic model basing on voxforge's data for german
> language. It's not complete, yet.
>
> You can found it here: http://wiki.freeswitch.org/wiki/Mod_pocketsphinx
>
> regards
> Helmut
>

Thanks!
I'd like to ask the community members who are interested in ASR and
PocketSphinx to please review Helmut's page and add to it as well as offer
feedback.
-MC
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From msc at freeswitch.org  Wed Aug  5 15:04:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 17:04:54 -0500
Subject: [Freeswitch-users] Fwd: Need Help In IVR
In-Reply-To: <7aa29e790908032247h2a93e838m1ac5fd3f8a3946c7@mail.gmail.com>
References: <7aa29e790908030012rcce847fu8320d833d2c85530@mail.gmail.com>
	<7aa29e790908032247h2a93e838m1ac5fd3f8a3946c7@mail.gmail.com>
Message-ID: <87f2f3b90908051504i70dd6e6ag7bc5fe3fa0200dd2@mail.gmail.com>

Could you give us an update on what you have so far? How about you put your
dialplan and perl script into a pastebin so that we can get a good frame of
reference?
Thanks,
MC

On Tue, Aug 4, 2009 at 12:47 AM, Thangappan.M <thangappan143 at gmail.com>wrote:

> Can you please help me?
>
>
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From msc at freeswitch.org  Wed Aug  5 15:16:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 17:16:41 -0500
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
Message-ID: <87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>

On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> interesting! what values can contain "variable_originate_disposition"? And
> can I set them manually in a script to reject a call simulating user busy or
> call rejected? A lua example?
> Thanks,
>    Raffaele
>

Start here:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup

And note the link to the hangup causes. As far as Lua, I'm not sure there's
a good reason to do it there. Could you give us pseudo code example of what
you're thinking of doing?
-MC
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From raffaele.p.guidi at gmail.com  Wed Aug  5 16:19:10 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Thu, 6 Aug 2009 01:19:10 +0200
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
Message-ID: <b0a8792f0908051619s66caac7na886720273396bdd@mail.gmail.com>

Actually I was reading that page, right now. I wrote a small lua script that
simulates a call with random wait time before answering, randomly not
answering at all and saying things for a random times once answered. This
would be useful for testing purposes simulating load, letting call center
operators try scripts against "fake" numbers with a "realistic" behaviour
and eventually to test and debug an automated dialer.
The script is almost ready (can contribute it should you find it useful), I
used it today to simulate load on my windows laptop with 50 concurrent calls
and peaks of 20/30 simultaneous calls connected (cpu was below 3%). I only
miss some use cases such as some CALL_REJECTED, USER_BUSY, NO_ANSWER

On Thu, Aug 6, 2009 at 00:16, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> interesting! what values can contain "variable_originate_disposition"? And
>> can I set them manually in a script to reject a call simulating user busy or
>> call rejected? A lua example?
>> Thanks,
>>    Raffaele
>>
>
> Start here:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>
> And note the link to the hangup causes. As far as Lua, I'm not sure there's
> a good reason to do it there. Could you give us pseudo code example of what
> you're thinking of doing?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Wed Aug  5 16:23:22 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 5 Aug 2009 18:23:22 -0500
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
Message-ID: <121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>

The bigger problem is some end points won't hang up the call in a  
consistent manner... some phones say user_rejected or user_busy when  
you reject the call with the reject button.

/b

On Aug 5, 2009, at 5:16 PM, Michael Collins wrote:

>
>
> On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <raffaele.p.guidi at gmail.com 
> > wrote:
> interesting! what values can contain  
> "variable_originate_disposition"? And can I set them manually in a  
> script to reject a call simulating user busy or call rejected? A lua  
> example?
>
> Thanks,
>    Raffaele
>
> Start here:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>
> And note the link to the hangup causes. As far as Lua, I'm not sure  
> there's a good reason to do it there. Could you give us pseudo code  
> example of what you're thinking of doing?
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nicolas at medularis.com  Wed Aug  5 16:41:47 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Wed, 5 Aug 2009 19:41:47 -0400
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
Message-ID: <1b46b4e80908051641y170b8d07g98ac29f69250f2ce@mail.gmail.com>

I'm bridging 2 calls in a javascript file, I originate the first call and
then execute a bridge with an origination string for the second call. If I
hangup the first call while trying to make the second call, I get this on
the console:

2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL

But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
NORMAL_CLEARING. And the variable_originate_disposition has a value of
"failure". Where can I get the detail of knowing the call/bridge failed
because of 'ORIGINATOR_CANCEL' as reported through the console?

Thanks!

Nicolas


 the value of variable_originate_disposition at the events level and when I
have an origination failure due to 'ORIGINATOR_CANCEL

On Wed, Aug 5, 2009 at 6:16 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> interesting! what values can contain "variable_originate_disposition"? And
>> can I set them manually in a script to reject a call simulating user busy or
>> call rejected? A lua example?
>> Thanks,
>>    Raffaele
>>
>
> Start here:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>
> And note the link to the hangup causes. As far as Lua, I'm not sure there's
> a good reason to do it there. Could you give us pseudo code example of what
> you're thinking of doing?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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>
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From raffaele.p.guidi at gmail.com  Wed Aug  5 16:44:11 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Thu, 6 Aug 2009 01:44:11 +0200
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
	<121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
Message-ID: <b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>

Well, I would randomly insert all of those cases to make it more
realistic... only thing I cannot manage to issue USER_BUSY from lua (and
neither from the dialplan, actually).

<anti-action application="respond" data="407" /> (407 or 486 or whatever...)


doesn't behave as I expected and neither

<action application="hangup" data="407" /> (407 or 486 or USER_BUSY or
whatever...)


and I cannot find a a session:reject(<hangupcause>) method in lua.

Can you give me a hint?

On Thu, Aug 6, 2009 at 01:23, Brian West <brian at freeswitch.org> wrote:

> The bigger problem is some end points won't hang up the call in a
> consistent manner... some phones say user_rejected or user_busy when you
> reject the call with the reject button.
> /b
>
> On Aug 5, 2009, at 5:16 PM, Michael Collins wrote:
>
>
>
> On Wed, Aug 5, 2009 at 3:54 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> interesting! what values can contain "variable_originate_disposition"? And
>> can I set them manually in a script to reject a call simulating user busy or
>> call rejected? A lua example?
>> Thanks,
>>    Raffaele
>>
>
> Start here:
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_hangup
>
> And note the link to the hangup causes. As far as Lua, I'm not sure there's
> a good reason to do it there. Could you give us pseudo code example of what
> you're thinking of doing?
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From woodydickson at gmail.com  Wed Aug  5 17:20:51 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Thu, 6 Aug 2009 08:20:51 +0800
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
	<E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
Message-ID: <a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>

Hi,

The problem is that I need freeswitch to continue executing the code after
switch_status_t channel_receive_message even when it gets error SIP code
from the destination.  Is that possible?

I know if I set up another action after my module in the dialplan.xml, I can
catch that.

But I would like the code to execute within the route that I have.  Is that
doable?

Woody

On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> The hangup cause will be in the originate_disposition channel
> variable on the A-leg.
> sip_term_status will contain the sip code and proto_specific_hangup_cause
> will contain sip:<code>.
>
>
>   Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
>  Am 5-Aug-09 um 11:23 AM schrieb Jo?o Mesquita:
>
> My guess is that you will receive a message here:
>
> switch_status_t channel_receive_message(switch_core_session_t *session,
> switch_core_session_message_t *msg)
>
> The problem here is that you don't have the exact SIP code but there is a
> clear relationship between the codes and the messages you receive on the
> channel, so I am guessing that is all the same.
>
> Hope this helps.
>
> jmesquita
>
> On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson <woodydickson at gmail.com>wrote:
>
>> Hi,
>>
>> I want to implement a module where freeSWITCH would try to bridge to an
>> extension and if the bridging operation fails, my module can use the hangup
>> code to determine the next cause of action.
>>
>> With switch_caller_extension_add_application(session, extension, "bridge",
>> "sofia/gateway/mygw/1232323);, if there is an error ( 503 received for
>> instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or the
>> module's APP) and go on to the next action.  Is there anyway to control it
>> so that freeSWITCH would remain to be within the module's APP funtion and
>> continue executing the code after switch_call_extension_add_application,
>> when let's say a 4XX or 5XX or CANCEL ( from originator) is received?
>>
>> I have tried it and found that if the bridging is successful, freeSWITCH
>> would continue executing the code after
>> switch_caller_extension_add_application, but if an error is received, then
>> it would just move on to the next action.
>>
>> Does anyone know how to deal with this problem?
>>
>> Woody
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Wed Aug  5 17:36:56 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 5 Aug 2009 19:36:56 -0500
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
	<E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
	<a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>
Message-ID: <9521EE86-37FF-4989-8F58-F61E5110E5C5@avgs.ca>

Hi,

You can set the "continue_on_fail" variable to true (or to the hangup  
causes you want it to ignore) and it'll keep executing whats queued.
For receive_message, unless you hook the session thats being created  
as a B-leg, you won't get anything relevant.
Also set hangup_after_bridge=true if you want to stop failing over  
when it worked.

Im curious, what are you coding? you can transfer the call in the  
dialplan without having to do all this manual queuing in C, thats why  
the routing state and dialplan modules exist. If you need to pull data  
from somewhere you can fill in channel variables that you can  
reference in the dialplan.

/*!
   \brief Transfer an existing session to another location
   \param session the session to transfer
   \param extension the new extension
   \param dialplan the new dialplan (OPTIONAL, may be NULL)
   \param context the new context (OPTIONAL, may be NULL)
*/
SWITCH_DECLARE(switch_status_t) switch_ivr_session_transfer(_In_  
switch_core_session_t *session, const char *extension, const char  
*dialplan,
															const char *context);



Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




Am 5-Aug-09 um 7:20 PM schrieb Woody Dickson:

> Hi,
>
> The problem is that I need freeswitch to continue executing the code  
> after switch_status_t channel_receive_message even when it gets  
> error SIP code from the destination.  Is that possible?
>
> I know if I set up another action after my module in the  
> dialplan.xml, I can catch that.
>
> But I would like the code to execute within the route that I have.   
> Is that doable?
>
> Woody
>
> On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> The hangup cause will be in the originate_disposition channel  
> variable on the A-leg.
>
> sip_term_status will contain the sip code and  
> proto_specific_hangup_cause will contain sip:<code>.
>
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> Am 5-Aug-09 um 11:23 AM schrieb Jo?o Mesquita:
>
>> My guess is that you will receive a message here:
>>
>> switch_status_t channel_receive_message(switch_core_session_t  
>> *session, switch_core_session_message_t *msg)
>>
>> The problem here is that you don't have the exact SIP code but  
>> there is a clear relationship between the codes and the messages  
>> you receive on the channel, so I am guessing that is all the same.
>>
>> Hope this helps.
>>
>> jmesquita
>>
>> On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson <woodydickson at gmail.com 
>> > wrote:
>> Hi,
>>
>> I want to implement a module where freeSWITCH would try to bridge  
>> to an extension and if the bridging operation fails, my module can  
>> use the hangup code to determine the next cause of action.
>>
>> With switch_caller_extension_add_application(session, extension,  
>> "bridge", "sofia/gateway/mygw/1232323);, if there is an error ( 503  
>> received for instance ) in the outgoing INVITE, freeSWITCH would  
>> leave my module ( or the module's APP) and go on to the next  
>> action.  Is there anyway to control it so that freeSWITCH would  
>> remain to be within the module's APP funtion and continue executing  
>> the code after switch_call_extension_add_application, when let's  
>> say a 4XX or 5XX or CANCEL ( from originator) is received?
>>
>> I have tried it and found that if the bridging is successful,  
>> freeSWITCH would continue executing the code after  
>> switch_caller_extension_add_application, but if an error is  
>> received, then it would just move on to the next action.
>>
>> Does anyone know how to deal with this problem?
>>
>> Woody
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From vladrodionov at gmail.com  Wed Aug  5 16:30:09 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Wed, 5 Aug 2009 16:30:09 -0700
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to configure?)
Message-ID: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>

Hi, everybody

This is a newbie question: Suppose I have XX (variable dynamic number) DIDs
assigned to one sip trunk (from VOIP provider ABC ). All calls coming from
VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is it
possible in FS? If yes, how everything should be configuered? Dialplan, sip
gateway? One more question: suppose it is doeable as I hope then how can I
get in my script CalleeID (not a CallerID)? Basicaly,

I want to acomplish the following:

1. Avoid re-configuring FS every time I got new bunch of DIDs
assigned/released from/to my Voip provider.
2. Have a way of extracting CalleeID in my script.

TIA,

Vladimir Rodionov
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From dujinfang at gmail.com  Wed Aug  5 18:26:59 2009
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 6 Aug 2009 09:26:59 +0800
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
References: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
Message-ID: <23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>

mod_easyroute?

2009/8/6 Vladimir Rodionov <vladrodionov at gmail.com>

> Hi, everybody
>
> This is a newbie question: Suppose I have XX (variable dynamic number) DIDs
> assigned to one sip trunk (from VOIP provider ABC ). All calls coming from
> VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is it
> possible in FS? If yes, how everything should be configuered? Dialplan, sip
> gateway? One more question: suppose it is doeable as I hope then how can I
> get in my script CalleeID (not a CallerID)? Basicaly,
>
> I want to acomplish the following:
>
> 1. Avoid re-configuring FS every time I got new bunch of DIDs
> assigned/released from/to my Voip provider.
> 2. Have a way of extracting CalleeID in my script.
>
> TIA,
>
> Vladimir Rodionov
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vladrodionov at gmail.com  Wed Aug  5 18:57:41 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Wed, 5 Aug 2009 18:57:41 -0700
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>
References: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
	<23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>
Message-ID: <3c233920908051857t5037d169r6bc4555b53cd015c@mail.gmail.com>

No, it is more like static routing. I need my *script program* be invoked
when somebody dial in. That is it. One script for all inbound DIDs. Suppose
I have thousand of them.  I think I know how to accomplish this but I am not
sure yet.

in my dialplan I need to define:

<!-- Launch a JavaScript application if dialed in-->
   <extension name="ProviderABC">
    <condition field="source" expression="mod_sofia"/>
    *<condition field="destination_number" expression="^1NXXNXXXXXX$">*
     <action application="javascript"
data="/usr/local/freeswitch/scripts/myapp.js"/>
    </condition>
   </extension>


In provider configuration:

<gateway name="voicepulse">
           <!--/// account username *required* ///-->
           <param name="username" value="your-username"/>
           <!--/// auth realm: *optional* same as gateway name, if blank ///-->
           <param name="realm" value="nyc.voicepulse.com"/>
           <!--/// account password *required* ///-->
           <param name="password" value="your-password"/>
           <!--/// extension for inbound calls: *optional* same as
username, if blank ///-->
*           * *<param name="extension" value="1NXXNXXXXXX"/> *
           <!--/// proxy host: *optional* same as realm, if blank ///-->
           <param name="proxy" value="nyc.voicepulse.com"/>
           <!--/// expire in seconds: *optional* 3600, if blank ///-->
           <param name="expire-seconds" value="600"/>

	   <param name="register" value="true"/>
         </gateway>


Something like this, yes? I can use regular expressions in
destination_number?

Q: There is object Session in JavaScript, Lua. Is Session.destination ==
destination_number from incoming call? It is not clear for me from what I
have read so far.

TIA,

-Vladimir Rodionov

On Wed, Aug 5, 2009 at 6:26 PM, Seven Du <dujinfang at gmail.com> wrote:

> mod_easyroute?
>
> 2009/8/6 Vladimir Rodionov <vladrodionov at gmail.com>
>
>> Hi, everybody
>>
>> This is a newbie question: Suppose I have XX (variable dynamic number)
>> DIDs assigned to one sip trunk (from VOIP provider ABC ). All calls coming
>> from VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is
>> it possible in FS? If yes, how everything should be configuered? Dialplan,
>> sip gateway? One more question: suppose it is doeable as I hope then how can
>> I get in my script CalleeID (not a CallerID)? Basicaly,
>>
>> I want to acomplish the following:
>>
>> 1. Avoid re-configuring FS every time I got new bunch of DIDs
>> assigned/released from/to my Voip provider.
>> 2. Have a way of extracting CalleeID in my script.
>>
>> TIA,
>>
>> Vladimir Rodionov
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Aug  5 19:03:26 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 21:03:26 -0500
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
	<121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
	<b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>
Message-ID: <87f2f3b90908051903y16395a62uf23feec9b3fdceab@mail.gmail.com>

On Wed, Aug 5, 2009 at 6:44 PM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> Well, I would randomly insert all of those cases to make it more
> realistic... only thing I cannot manage to issue USER_BUSY from lua (and
> neither from the dialplan, actually).
>
> <anti-action application="respond" data="407" /> (407 or 486 or
> whatever...)
>
>
> doesn't behave as I expected and neither
>
> <action application="hangup" data="407" /> (407 or 486 or USER_BUSY or
> whatever...)
>
>
> and I cannot find a a session:reject(<hangupcause>) method in lua.
>
> Can you give me a hint?
>

You can execute pretty much any dialplan app with the session:execute
command:
http://wiki.freeswitch.org/wiki/Lua#session:execute

Try something like:
session:execute("hangup","USER_BUSY");

-MC
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From msc at freeswitch.org  Wed Aug  5 20:40:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 5 Aug 2009 22:40:31 -0500
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <3c233920908051857t5037d169r6bc4555b53cd015c@mail.gmail.com>
References: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
	<23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>
	<3c233920908051857t5037d169r6bc4555b53cd015c@mail.gmail.com>
Message-ID: <87f2f3b90908052040s21545229qba24937bd2b14540@mail.gmail.com>

On Wed, Aug 5, 2009 at 8:57 PM, Vladimir Rodionov <vladrodionov at gmail.com>wrote:

> No, it is more like static routing. I need my *script program* be invoked
> when somebody dial in. That is it. One script for all inbound DIDs. Suppose
> I have thousand of them.  I think I know how to accomplish this but I am not
> sure yet.
>

Have the external profile be used only for provider ABC, or define a new
profile. Then in the profile have the calls go to a specific context. You
could have something like this in the sip profile definition:

<param name="context" value="abc_calls"/>

Then create a dialplan context called "abc_calls" that handles all inbound
calls. Create a file in conf/dialplan/ called abc_calls.xml:

<include>
  <context name="abc_calls">
    <extension name="abc_calls">
      <condition field="destination_number" expression="^(.*)$">
        <action application="lua" data="myscript.lua"/>
      </condition>
    </extension>
  </context>
</include>

Essentially you're just creating a SIP profile and a dialplan context that
are servicing your VoIP provider. You can add other profiles/contexts for
other providers if need be.

Let us know how it goes...
-MC
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From pete at privateconnect.com  Wed Aug  5 20:45:30 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Wed, 05 Aug 2009 20:45:30 -0700
Subject: [Freeswitch-users]
 =?utf-8?q?Multiple_DIDs_per_SIP_trunk_=28how_t?=
 =?utf-8?q?o_configure=3F=29?=
Message-ID: <20090805204530.2ad02225396a31c9de30536f2e338977.43419788fb.wbe@email04.secureserver.net>

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From woodydickson at gmail.com  Wed Aug  5 20:54:28 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Thu, 6 Aug 2009 11:54:28 +0800
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <9521EE86-37FF-4989-8F58-F61E5110E5C5@avgs.ca>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
	<E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
	<a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>
	<9521EE86-37FF-4989-8F58-F61E5110E5C5@avgs.ca>
Message-ID: <a44688a0908052054j405c1707j382b3d6ccac8f8a9@mail.gmail.com>

Hi,

In my module, I will collect a list of available failover route that I can
use to failover to whenever a particular error is received.  However, these
available routes has different condition and the condition changes every
half a minute.  Therefore, I need to catch the hangup cause after bridge,
and then figure out the next workable available route based on the latest
condition setting.

It seems like this is only prossible to be done within a C module.  Any
suggestion will be greatly appreciated.

Woody

On Thu, Aug 6, 2009 at 8:36 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
> You can set the "continue_on_fail" variable to true (or to the hangup
> causes you want it to ignore) and it'll keep executing whats queued. For
> receive_message, unless you hook the session thats being created as a B-leg,
> you won't get anything relevant.
> Also set hangup_after_bridge=true if you want to stop failing over when it
> worked.
>
> Im curious, what are you coding? you can transfer the call in the dialplan
> without having to do all this manual queuing in C, thats why the routing
> state and dialplan modules exist. If you need to pull data from somewhere
> you can fill in channel variables that you can reference in the dialplan.
>
> /*!
>   \brief Transfer an existing session to another location
>   \param session the session to transfer
>   \param extension the new extension
>   \param dialplan the new dialplan (OPTIONAL, may be NULL)
>   \param context the new context (OPTIONAL, may be NULL)
> */
> SWITCH_DECLARE(switch_status_t) switch_ivr_session_transfer(_In_
> switch_core_session_t *session, const char *extension, const char *dialplan,
> const char *context);
>
>
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> Am 5-Aug-09 um 7:20 PM schrieb Woody Dickson:
>
> Hi,
>
> The problem is that I need freeswitch to continue executing the code after
> switch_status_t channel_receive_message even when it gets error SIP code
> from the destination.  Is that possible?
>
> I know if I set up another action after my module in the dialplan.xml, I
> can catch that.
>
> But I would like the code to execute within the route that I have.  Is that
> doable?
>
> Woody
>
> On Thu, Aug 6, 2009 at 12:34 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
>> The hangup cause will be in the originate_disposition channel
>> variable on the A-leg.
>> sip_term_status will contain the sip code and proto_specific_hangup_cause
>> will contain sip:<code>.
>>
>>
>>   Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>>  Am 5-Aug-09 um 11:23 AM schrieb Jo?o Mesquita:
>>
>> My guess is that you will receive a message here:
>>
>> switch_status_t channel_receive_message(switch_core_session_t *session,
>> switch_core_session_message_t *msg)
>>
>> The problem here is that you don't have the exact SIP code but there is a
>> clear relationship between the codes and the messages you receive on the
>> channel, so I am guessing that is all the same.
>>
>> Hope this helps.
>>
>> jmesquita
>>
>> On Wed, Aug 5, 2009 at 12:05 PM, Woody Dickson <woodydickson at gmail.com>wrote:
>>
>>> Hi,
>>>
>>> I want to implement a module where freeSWITCH would try to bridge to an
>>> extension and if the bridging operation fails, my module can use the hangup
>>> code to determine the next cause of action.
>>>
>>> With switch_caller_extension_add_application(session, extension,
>>> "bridge", "sofia/gateway/mygw/1232323);, if there is an error ( 503 received
>>> for instance ) in the outgoing INVITE, freeSWITCH would leave my module ( or
>>> the module's APP) and go on to the next action.  Is there anyway to control
>>> it so that freeSWITCH would remain to be within the module's APP funtion and
>>> continue executing the code after switch_call_extension_add_application,
>>> when let's say a 4XX or 5XX or CANCEL ( from originator) is received?
>>>
>>> I have tried it and found that if the bridging is successful, freeSWITCH
>>> would continue executing the code after
>>> switch_caller_extension_add_application, but if an error is received, then
>>> it would just move on to the next action.
>>>
>>> Does anyone know how to deal with this problem?
>>>
>>> Woody
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From velu.technical at gmail.com  Wed Aug  5 21:38:25 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Thu, 6 Aug 2009 10:08:25 +0530
Subject: [Freeswitch-users] Fwd: execute function in ESL.pm module is not
	working
In-Reply-To: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>
References: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>
Message-ID: <1452e2980908052138jed5d676oe5ce73b0b4b0e37e@mail.gmail.com>

Please any one help for this problem..

---------- Forwarded message ----------
From: velusamy velu <velu.technical at gmail.com>
Date: Wed, Aug 5, 2009 at 11:44 AM
Subject: execute function in ESL.pm module is not working
To: freeswitch-users at lists.freeswitch.org


Dear All,
   I registered alarm signal in my Perl server program.
   If ALARM signal occurred I execute the following statement in signal
handler.
    "$conn->execute("playback",$sound_path."voicemail/vm-goodbye.wav")"
  The above statement didn't play that wave file. But before generating the
ALARM signal it worked.

What is the problem?

Please help me in this problem....

Also Is there any idea to do timeout for DTMF digits?

Thanks...

Regards,
K.Velusamy
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From msc at freeswitch.org  Wed Aug  5 22:54:17 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Aug 2009 00:54:17 -0500
Subject: [Freeswitch-users] Fwd: execute function in ESL.pm module is
	not working
In-Reply-To: <1452e2980908052138jed5d676oe5ce73b0b4b0e37e@mail.gmail.com>
References: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>
	<1452e2980908052138jed5d676oe5ce73b0b4b0e37e@mail.gmail.com>
Message-ID: <87f2f3b90908052254l3ecc7fa0ybc92d87c587a9b0d@mail.gmail.com>

On Wed, Aug 5, 2009 at 11:38 PM, velusamy velu <velu.technical at gmail.com>wrote:

> Please any one help for this problem..
>
>
Sorry for the delay but many of the FreeSWITCH experts are at ClueCon right
now so we'll ask for your patience... in the meantime could you pastebin
your script and your dialplan entry so that we can take a look at them?

Thanks,
MC
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From mayamatakeshi at gmail.com  Thu Aug  6 00:03:06 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Thu, 6 Aug 2009 16:03:06 +0900
Subject: [Freeswitch-users] Monitoring On-Hold/Off-Hold
In-Reply-To: <5a8712120908051055k95f01d4x61ef7b7dff80298a@mail.gmail.com>
References: <15b9404e0908050935m62a6581bxec3f3055e013753f@mail.gmail.com>
	<5a8712120908051055k95f01d4x61ef7b7dff80298a@mail.gmail.com>
Message-ID: <15b9404e0908060003s2eb17434q1de769fdde444ad2@mail.gmail.com>

2009/8/6 Jo?o Mesquita <jmesquita at gmail.com>:
> I only see one way out of this. If you manage presence, an event like the
> following is sent:
>
> Event-Name: PRESENCE_IN
> Core-UUID: 189b12c0-7fb0-11de-b0bc-37eec03ad00f
> FreeSWITCH-Hostname: cl-t146-421cl
> FreeSWITCH-IPv4: XXXXXX
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-08-05%2013%3A42%3A24
> Event-Date-GMT: Wed,%2005%20Aug%202009%2017%3A42%3A24%20GMT
> Event-Date-Timestamp: 1249494144628132
> Event-Calling-File: switch_channel.c
> Event-Calling-Function: switch_channel_presence
> Event-Calling-Line-Number: 472
> Channel-State: CS_HIBERNATE
> Channel-State-Number: 8
> Channel-Name: XXXXX
> Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f
> Call-Direction: inbound
> Presence-Call-Direction: inbound
> Answer-State: answered
> Caller-Username: 1000
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: Mesquita
> Caller-Caller-ID-Number: 1000
> Caller-Network-Addr: XXXXX
> Caller-Destination-Number: 1005
> Caller-Unique-ID: 4e73668e-81e7-11de-b0bc-37eec03ad00f
> Caller-Source: mod_sofia
> Caller-Context: XXXXX
> Caller-Channel-Name: XXXXX
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1249494132128119
> Caller-Channel-Created-Time: 1249494132128119
> Caller-Channel-Answered-Time: 1249494139500129
> Caller-Channel-Progress-Time: 1249494132368119
> Caller-Channel-Progress-Media-Time: 0
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
> Other-Leg-Username: 1000
> Other-Leg-Dialplan: XML
> Other-Leg-Caller-ID-Name: Joao%20Mesquita
> Other-Leg-Caller-ID-Number: 1000
> Other-Leg-Network-Addr: 190.2.41.65
> Other-Leg-Destination-Number:
> sip%3A1005%40192.168.0.106%3A4559%3Bfs_nat%3Dyes%3Bfs_path%3Dsip%253A1005%2540190.2.41.65%253A4559
> Other-Leg-Unique-ID: 4e7622ac-81e7-11de-b0bc-37eec03ad00f
> Other-Leg-Source: mod_sofia
> Other-Leg-Context: XXXXX
> Other-Leg-Channel-Name: XXXXXX
> Other-Leg-Screen-Bit: true
> Other-Leg-Privacy-Hide-Name: false
> Other-Leg-Privacy-Hide-Number: false
> proto: src/switch_channel.c
> login: src/switch_channel.c
> from: XXXXXX
> rpid: unknown
> status: hold
> event_type: presence
> alt_event_type: dialog
> event_count: 3
>
> Content-Length: 543
> Content-Type: text/event-plain
>
> Other than that, I think it can be patched. I will take a look at it.

Thanks, that would be the best.

Just in case someone else needs this:
I have also tried to watch for
CHANNEL_EXECUTE/CHANNEL_EXECUTE_COMPLETE with Application set to
playback and some indication of MOH in the Application-Data header.
That would work but:
- they will be fired continuously if you set hold_music=some_file
- they will not be fired if you set hold_music=silence (of course)



From enno.egbert at googlemail.com  Thu Aug  6 02:09:24 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Thu, 6 Aug 2009 02:09:24 -0700 (PDT)
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>
References: <24823167.post@talk.nabble.com>
	<ddad80610908050707y6ab49f53pe64101d5090e376f@mail.gmail.com>
Message-ID: <24841050.post@talk.nabble.com>


Hi Jim,

yes! It?s possible to call the IP200 full encrypted for example from a SNOM
or phonerlite. But when i try to call the SNOM from a innovaphone, the call
fails and i only hear the mailbox.

To modify the dialplan i am not so sure how i works. I don?t have any
experience of configure freeswitch or working with xml files. :confused:

In my dialplan i just modify a few lines. If you want, you can have a look
on the file in the attachment. 

http://www.nabble.com/file/p24841050/default.xml default.xml 

Thanks for your help. =)

NOx


Jim Burke-2 wrote:
> 
> Hi NOx,
> 
> Can you clarify the direction of the calls.  When you say outgoing do
> you mean a call is terminating to the ip200?
> 
> I have been down a similar path while testing Eyebeam.  If the
> terminating phone sets an option to only accept secure calls and FS
> does not send Secure Descriptions in the INVITE, Eyebeam would respond
> with 415 response code and the call would fail.  Depending on your
> diaplan this could send your call to voicemail.
> 
> To fix it I added the following code to dialplan.
> <action application="set" data="continue_on_fail=79"/>
> <bridge blah blah blah>
> <action application="set" data="bypass_media=false"/>
> <action application="set" data="proxy_media=true"/>
> <action application="set" data="ringback=$${uk-ring}"/>
> <action application="pre_answer"/>
> <action application="export" data="sip_secure_media=true"/>
> <bridge blah blah blah>
> 
> The continue on fail captures the 415 response code forces the call to
> continue to the next bridge while sip_secure_media forces the second
> invite to include security descriptors.  The rest was required because
> I did not want to proxy media if the call was not secure, obviously if
> the call is secure on a point to point basis FS will have to proxy the
> media and this was the only way I could find for it to work.
> 
> Hope this helps.
> 
> Regards,
> 
> 
> On Wed, Aug 5, 2009 at 6:52 PM, NOx-WHV<enno.egbert at googlemail.com> wrote:
>>
>> Hello,
>>
>> i have a problem using a innovaphone ip200 with freeswitch and tls/srtp.
>> The
>> freeswitch certificate is in the trust list of the phone and it works
>> with
>> tls for incomming calls. But outgoing calls were rejected to the mailbox.
>> The freeswitch configuration is ok, because it works with a snom 320.
>>
>> Who can help me to confugure the IP200?
>>
>> Thanks
>>
>> NOx
>> --
>> View this message in context:
>> http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24823167.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> 
> -- 
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24841050.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From juanbackson at gmail.com  Thu Aug  6 02:24:03 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 6 Aug 2009 17:24:03 +0800
Subject: [Freeswitch-users] Question about dynamic registration
In-Reply-To: <9FE0F6B8-C4BF-4820-8CC2-6825C5EE8422@freeswitch.org>
References: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>
	<9FE0F6B8-C4BF-4820-8CC2-6825C5EE8422@freeswitch.org>
Message-ID: <27c25bc40908060224l38ad47fdje17065b13647905d@mail.gmail.com>

Hi,

Is there a sample module that I can take a look at on how to do that?
I don't understand how to get the registration request and how to pass back
auth result to freeswitch.

JB

On Mon, Aug 3, 2009 at 8:42 PM, Brian West <brian at freeswitch.org> wrote:

> You could build your own module to do it how ever you please.  But
> forking a script every time to auth is not very scalable.
>
> /b
>
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From markmorreny at gmail.com  Thu Aug  6 02:26:53 2009
From: markmorreny at gmail.com (mark morreny)
Date: Thu, 6 Aug 2009 17:26:53 +0800
Subject: [Freeswitch-users] question about latest version of mod_limit
Message-ID: <20ad6b920908060226t2dcf532aobab23dc0299d0f05@mail.gmail.com>

Hello,

I have the following setup in the dialplan.  Then, I fire up sipp to send
5calls/s and I expect to get limit-pass=false in most of the INFO output.
However, I am getting all "limit-pass=pass".

Does anyone know what is wrong with my dialplan?


 <context name="internal">

  <extension name="test-limit">

    <condition field="destination_number" expression="^9(.*)$">

        <action application="limit_hash_execute" data="192.168.1.102 6000
1/1 transfer to-next" />
        <action application="set" data="limit-pass=false"/>
        <action application="info" />
        <action application="hangup" />

    </condition>

  </extension>
    <extension name="to-next">
        <condition field="destination_number" expression="^to-next$">
        <action application="set" data="limit-pass=true"/>
        <action application="info" />
        <action application="hangup" />



    </condition>
    </extension>
</context>
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From enno.egbert at googlemail.com  Thu Aug  6 02:52:01 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Thu, 6 Aug 2009 02:52:01 -0700 (PDT)
Subject: [Freeswitch-users] TLS/SRTP with Innovaphone IP200
In-Reply-To: <DF06C882-055F-488D-B860-46643AF9B22B@freeswitch.org>
References: <24823167.post@talk.nabble.com>
	<DF06C882-055F-488D-B860-46643AF9B22B@freeswitch.org>
Message-ID: <24841078.post@talk.nabble.com>


Hi Brian, 

where i have to ship the phone? Maybe i can ship it to you. But i have first
to ask the owner, because i also borrow it.

Thanks for your help

NOx
 

Brian West-3 wrote:
> 
> Can you ship me a phone to test with?  That's usually the missing  
> element when testing this stuff is I just can't afford to buy every  
> phone to test with.
> 
> /b
> 
> On Aug 5, 2009, at 3:52 AM, NOx-WHV wrote:
> 
>>
>> Hello,
>>
>> i have a problem using a innovaphone ip200 with freeswitch and tls/ 
>> srtp. The
>> freeswitch certificate is in the trust list of the phone and it  
>> works with
>> tls for incomming calls. But outgoing calls were rejected to the  
>> mailbox.
>> The freeswitch configuration is ok, because it works with a snom 320.
>>
>> Who can help me to confugure the IP200?
>>
>> Thanks
>>
>> NOx
>> -- 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/TLS-SRTP-with-Innovaphone-IP200-tp24823167p24841078.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From thangappan143 at gmail.com  Thu Aug  6 02:55:56 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Thu, 6 Aug 2009 15:25:56 +0530
Subject: [Freeswitch-users] IVR on Freeswitch
Message-ID: <7aa29e790908060255p36577d41hffe05ccc380c2540@mail.gmail.com>

Dear all,

   I am in the process of implementing IVR in Perl using event bound socket
on FreeSWITCH. I want to use all the functionality of IVR in my
implementation.

   I have seen the XML MACRO (default implementation) in the ivr.conf .xml
and demo/en/demo-ivr.xml .In that file I don't want to handle the
inter-digit time out and response timeout and all.I can just configure the
seconds.It will automatically works specified in the  <menu> tag.

I want to specify the menu definitions how they have specified in the XML
field in the ivr.conf.xml
Is there any way in Perl to do that?

I want to handle timeout,interdigittimeout  in Perl.Please help me?





-- 
Regards,
Thangappan.M
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From dujinfang at gmail.com  Thu Aug  6 02:58:59 2009
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 6 Aug 2009 17:58:59 +0800
Subject: [Freeswitch-users] A few questions about lua
Message-ID: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>

ALL-
I have a few questions when scripting lua. According to wiki, it is possible
to run looping forever lua scripts through start-up config or luarun.

1) Will the lua script stop when unload mod_lua? I experienced core dump
when unload mod_lua while there was a running lua script. Reported on jira.

2) How to stop a forever running lua script?  I stop it by listening a
CUSTOM event fired elsewhere. See code below. Is there any standard way like
luastop ?

3) Any way to show how many running lua scripts? luashow ?

4) It seems cannot get the lua script name in a lua script, I made a patch
to jira by assign it to the argv[0].

5) Seems that only EventConsumer("all") working.
EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work. Any
idea to this?

Thanks a lot.



code example:

con = freeswitch.EventConsumer("all");

argv[0] = "test.lua"

freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "] Starting
=====\n");

local all_events = 0


for e in (function() return con:pop(1) end) do
  -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
   all_events = all_events + 1;
freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
 event_name = e:getHeader("Event-Name") or ""
event_subclass = e:getHeader("Event-Subclass") or ""
 if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..
"]---Exiting------\n")
  break
end

end
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From dome at tel.co.th  Thu Aug  6 03:48:58 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 6 Aug 2009 17:48:58 +0700
Subject: [Freeswitch-users] Numeric Value Ranges Expressions in dialplan
Message-ID: <8ccbff060908060348l75ee1062ua6145da2d6f7c4e9@mail.gmail.com>

Dear sir,

    Is posible to check numeric range in dialplan (expression).
   example i got balance vaiable from somewhere and want to check > 0
or not before call bridge application.
   ( I don't want to call scripts)

Best regards.

Dome C.



From jason at jasonjgw.net  Thu Aug  6 04:19:04 2009
From: jason at jasonjgw.net (Jason White)
Date: Thu, 6 Aug 2009 21:19:04 +1000
Subject: [Freeswitch-users] Numeric Value Ranges Expressions in dialplan
In-Reply-To: <8ccbff060908060348l75ee1062ua6145da2d6f7c4e9@mail.gmail.com>
References: <8ccbff060908060348l75ee1062ua6145da2d6f7c4e9@mail.gmail.com>
Message-ID: <20090806111903.GB17479@jdc.jasonjgw.net>

Dome Charoenyost <dome at tel.co.th> wrote:
>     Is posible to check numeric range in dialplan (expression).
>    example i got balance vaiable from somewhere and want to check > 0
> or not before call bridge application.
>    ( I don't want to call scripts)

Can you write a regular expression to match it?

^[1-9]\d*$
for example, might be a good start to identify non-zero integers.




From saeedahmad1981 at gmail.com  Thu Aug  6 05:05:48 2009
From: saeedahmad1981 at gmail.com (Saeed Ahmed)
Date: Thu, 6 Aug 2009 14:05:48 +0200
Subject: [Freeswitch-users] SVN error
Message-ID: <E47E8DBFC76D4E769B7BBF60561CAA27@saeedlaptop>

Hi,

While doing 'make current' or 'svn up' I am getting following errors:

svn: REPORT request failed on '/svn/!svn/vcc/default'

svn: Can't find a temporary directory: Internal error

 

- Saeed

 

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From raffaele.p.guidi at gmail.com  Thu Aug  6 05:49:19 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Thu, 6 Aug 2009 14:49:19 +0200
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <87f2f3b90908051903y16395a62uf23feec9b3fdceab@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
	<121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
	<b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>
	<87f2f3b90908051903y16395a62uf23feec9b3fdceab@mail.gmail.com>
Message-ID: <b0a8792f0908060549v40196e6fm6da7f95f47193bb4@mail.gmail.com>

Done, it (of course, thanks) worked smoothly. I've published the example on
the wiki.
http://wiki.freeswitch.org/wiki/Fakecall_responder (and linked in mod_lua
samples)

Regards,
   Raffaele

On Thu, Aug 6, 2009 at 04:03, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 6:44 PM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> Well, I would randomly insert all of those cases to make it more
>> realistic... only thing I cannot manage to issue USER_BUSY from lua (and
>> neither from the dialplan, actually).
>>
>> <anti-action application="respond" data="407" /> (407 or 486 or
>> whatever...)
>>
>>
>> doesn't behave as I expected and neither
>>
>> <action application="hangup" data="407" /> (407 or 486 or USER_BUSY or
>> whatever...)
>>
>>
>> and I cannot find a a session:reject(<hangupcause>) method in lua.
>>
>> Can you give me a hint?
>>
>
> You can execute pretty much any dialplan app with the session:execute
> command:
> http://wiki.freeswitch.org/wiki/Lua#session:execute
>
> Try something like:
> session:execute("hangup","USER_BUSY");
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From elihay at savion.huji.ac.il  Thu Aug  6 03:47:00 2009
From: elihay at savion.huji.ac.il (Eli Hayun)
Date: Thu, 06 Aug 2009 13:47:00 +0300
Subject: [Freeswitch-users] A few questions about lua
In-Reply-To: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
References: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
Message-ID: <1249555620.5449.1.camel@eli-desktop>

Hi
I dont know about events so much but I cannot see variable "e" is
setting

event_name = e:getHeader("Event-Name") or "" 
event_subclass = e:getHeader("Event-Subclass") or "" 

regurds
Eli

On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:

> ALL-
> 
> 
> 
> I have a few questions when scripting lua. According to wiki, it is
> possible to run looping forever lua scripts through start-up config or
> luarun. 
> 
> 
> 1) Will the lua script stop when unload mod_lua? I experienced core
> dump when unload mod_lua while there was a running lua script.
> Reported on jira.
> 
> 
> 2) How to stop a forever running lua script?  I stop it by listening a
> CUSTOM event fired elsewhere. See code below. Is there any standard
> way like luastop ?
> 
> 
> 3) Any way to show how many running lua scripts? luashow ?
> 
> 
> 4) It seems cannot get the lua script name in a lua script, I made a
> patch to jira by assign it to the argv[0].
> 
> 
> 5) Seems that only EventConsumer("all") working.
> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work.
> Any idea to this?
> 
> 
> Thanks a lot.
> 
> 
> 
> 
>                                                
> code example:
> 
> 
> con = freeswitch.EventConsumer("all");      
> 
> 
> argv[0] = "test.lua"
>                        
> freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "]
> Starting =====\n");
>                         
> local all_events = 0
> 
>                                      
> for e in (function() return con:pop(1) end) do
>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
>    all_events = all_events + 1;
> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
> 
> event_name = e:getHeader("Event-Name") or ""
> event_subclass = e:getHeader("Event-Subclass") or ""
> 
> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
>  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..
> "]---Exiting------\n")
>  break
> end
> 
> 
> end 
> 
> 
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From lakindia89 at gmail.com  Thu Aug  6 03:47:20 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Thu, 6 Aug 2009 16:17:20 +0530
Subject: [Freeswitch-users] Error while creating object
Message-ID: <7d79b3930908060347xe5be545yfeeafad761aba274@mail.gmail.com>

Hi all,
Greets.

I am in the process of controlling the freeswitch with perl.
I have read about mod_perl and I wrote some scripts to test which works
fine.
Yesterday I tried to access the digit_set function.
So I create an object for the freeswitch::DTMF.
But it reported the following error.

2009-08-06 15:53:46 [ERR] mod_perl.c:69 Perl_safe_eval() [require
'/usr/local/freeswitch/conf/test.pl';]
No matching function for overloaded 'new_DTMF' at
/usr/local/freeswitch/perl/freeswitch.pm line 197.
Compilation failed in require at (eval 2) line 1.

Here is my code.

#!/usr/bin/perl
use strict;
use freeswitch;
our $session;
$session->execute("bridge","user/1010");
my $sess=&freeswitch::DTMF::new;
return 1;

The bridge is working fine. But while creating the object it said error.

Can any one explain why this happens and how can I correct it?
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From jim.page at redmatter.com  Thu Aug  6 04:24:15 2009
From: jim.page at redmatter.com (Jim Page)
Date: Thu, 6 Aug 2009 12:24:15 +0100
Subject: [Freeswitch-users] CURL directory issue
Message-ID: <E0C2263D8B20E14390DF8AA03E1F99A0C506DCB1@sha-exch9.shared.ifeltd.com>

Afternoon All

I wonder if someone (perhaps even the illustrious intralanman) could help me out with a problem I am experiencing with a CURL directory.

In the interests of understanding how the mechanism works, I am using a super-braindead php script to return info about a specific set of users. I plan to move to something more sophisticated once the proof of concept is complete, possibly based on intralanman's scripts.

The basic problem is that all works fine (boot, register, voicemail etc), except that user's <variables> seem not to be being read correctly, eg 'toll_allow' and 'user_context'. Here's a typical user XML message I am returning:

<document type="freeswitch/xml">
  <section name="directory">
   <domain name="pbx.redmatter.com">
    <params>
      <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
    </params>

    <variables>
      <variable name="record_stereo" value="true"/>
      <variable name="default_gateway" value="$${default_provider}"/>
      <variable name="default_areacode" value="$${default_areacode}"/>
      <variable name="transfer_fallback_extension" value="operator"/>
    </variables>

    <groups>
      <group name="default">
        <users>            <user id="1009" cidr="172.30.99.0/24">
               <params>
                 <param name="password" value="$${default_password}"/>
                 <param name="vm-password" value="1009"/>
                 <param name="vm-email-all-messages" value="true"/>
                 <param name="vm-attach-file" value="true"/>
                 <param name="vm-mailto" value="1009 at redmatter.com"/>
                 <param name="vm-keep-local-after-email" value="true"/>
                 <param name="vm_message_ext" value="mp3"/>
               </params>
               <variables>
                 <variable name="toll_allow" value="domestic,international,local"/>
                 <variable name="accountcode" value="1009"/>
                 <variable name="user_context" value="default"/>
                 <variable name="effective_caller_id_name" value="Extension 1009"/>
                 <variable name="effective_caller_id_number" value="1009"/>
                 <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
                 <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
                 <variable name="callgroup" value="techsupport"/>
               </variables>
             </user>        </users>
      </group>
    </groups>
   </domain>
 </section>
</document>

I return this kind of message in all cases except the (sip_auth_method=="REGISTER") request message where I return

<document type="freeswitch/xml">
  <section name="directory">
    <domain name="pbx.redmatter.com">
      <user id="1007">
        <params>
          <param name="password" value="1234"/>
        </params>
      </user>
    </domain>
  </section>
</document>

Also it's probably worth mentioning that I have removed all trace of xml from conf/directory and I don't believe there is a conflict happening there. 

The phones register correctly. The trouble is they don't operate on the correct dialplan context (I fixed that by hardcoding the internal gateway to dialplan default), but the 'toll_allow' variable is now not working so that outbound calls fail, which is what made me think that the user variables are being ignored.

Freeswitch version is 1.0.4, built by me and running on a dell 1950 running Centos 5.3 x86_64. HTTP application is running on standard Centos 5.3 apache/php.

Any ideas gratefully and humbly received.

All the best
Jim



From kevin at johnnyvoip.com  Thu Aug  6 08:48:08 2009
From: kevin at johnnyvoip.com (Kevin Green)
Date: Thu, 6 Aug 2009 11:48:08 -0400
Subject: [Freeswitch-users] CURL directory issue
In-Reply-To: <E0C2263D8B20E14390DF8AA03E1F99A0C506DCB1@sha-exch9.shared.ifeltd.com>
References: <E0C2263D8B20E14390DF8AA03E1F99A0C506DCB1@sha-exch9.shared.ifeltd.com>
Message-ID: <ab57a3e80908060848x64bbc808r162273eb4cbcd363@mail.gmail.com>

Try returning the full information on the register. It may be that the
variables are read onto the user profile upon registration and since you are
only supplying a dumbed down version for registration the variables aren't
being read and cached.

Regards,
   Kevin Green

On Thu, Aug 6, 2009 at 7:24 AM, Jim Page <jim.page at redmatter.com> wrote:

> Afternoon All
>
> I wonder if someone (perhaps even the illustrious intralanman) could help
> me out with a problem I am experiencing with a CURL directory.
>
> In the interests of understanding how the mechanism works, I am using a
> super-braindead php script to return info about a specific set of users. I
> plan to move to something more sophisticated once the proof of concept is
> complete, possibly based on intralanman's scripts.
>
> The basic problem is that all works fine (boot, register, voicemail etc),
> except that user's <variables> seem not to be being read correctly, eg
> 'toll_allow' and 'user_context'. Here's a typical user XML message I am
> returning:
>
> <document type="freeswitch/xml">
>  <section name="directory">
>   <domain name="pbx.redmatter.com">
>    <params>
>      <param name="dial-string" value="{presence_id=${dialed_user}@
> ${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>    </params>
>
>    <variables>
>      <variable name="record_stereo" value="true"/>
>      <variable name="default_gateway" value="$${default_provider}"/>
>      <variable name="default_areacode" value="$${default_areacode}"/>
>      <variable name="transfer_fallback_extension" value="operator"/>
>    </variables>
>
>    <groups>
>      <group name="default">
>        <users>            <user id="1009" cidr="172.30.99.0/24">
>               <params>
>                 <param name="password" value="$${default_password}"/>
>                 <param name="vm-password" value="1009"/>
>                 <param name="vm-email-all-messages" value="true"/>
>                 <param name="vm-attach-file" value="true"/>
>                 <param name="vm-mailto" value="1009 at redmatter.com"/>
>                 <param name="vm-keep-local-after-email" value="true"/>
>                 <param name="vm_message_ext" value="mp3"/>
>               </params>
>               <variables>
>                 <variable name="toll_allow"
> value="domestic,international,local"/>
>                 <variable name="accountcode" value="1009"/>
>                 <variable name="user_context" value="default"/>
>                 <variable name="effective_caller_id_name" value="Extension
> 1009"/>
>                 <variable name="effective_caller_id_number" value="1009"/>
>                 <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>                 <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>                 <variable name="callgroup" value="techsupport"/>
>               </variables>
>             </user>        </users>
>      </group>
>    </groups>
>   </domain>
>  </section>
> </document>
>
> I return this kind of message in all cases except the
> (sip_auth_method=="REGISTER") request message where I return
>
> <document type="freeswitch/xml">
>  <section name="directory">
>    <domain name="pbx.redmatter.com">
>      <user id="1007">
>        <params>
>          <param name="password" value="1234"/>
>        </params>
>      </user>
>    </domain>
>  </section>
> </document>
>
> Also it's probably worth mentioning that I have removed all trace of xml
> from conf/directory and I don't believe there is a conflict happening there.
>
> The phones register correctly. The trouble is they don't operate on the
> correct dialplan context (I fixed that by hardcoding the internal gateway to
> dialplan default), but the 'toll_allow' variable is now not working so that
> outbound calls fail, which is what made me think that the user variables are
> being ignored.
>
> Freeswitch version is 1.0.4, built by me and running on a dell 1950 running
> Centos 5.3 x86_64. HTTP application is running on standard Centos 5.3
> apache/php.
>
> Any ideas gratefully and humbly received.
>
> All the best
> Jim
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dujinfang at gmail.com  Thu Aug  6 08:52:07 2009
From: dujinfang at gmail.com (Seven Du)
Date: Thu, 6 Aug 2009 23:52:07 +0800
Subject: [Freeswitch-users] A few questions about lua
In-Reply-To: <1249555620.5449.1.camel@eli-desktop>
References: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
	<1249555620.5449.1.camel@eli-desktop>
Message-ID: <EBFA6F4D-07D0-4C6D-ABDF-8A8FF3976FE2@gmail.com>

for e in (function() return con:pop(1) end) do

btw, the script works.

Thanks.
On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote:
> Hi
> I dont know about events so much but I cannot see variable "e" is  
> setting
>
> event_name = e:getHeader("Event-Name") or ""
> event_subclass = e:getHeader("Event-Subclass") or ""
>
> regurds
> Eli
>
> On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:
>> ALL-
>>
>>
>> I have a few questions when scripting lua. According to wiki, it is  
>> possible to run looping forever lua scripts through start-up config  
>> or luarun.
>>
>>
>> 1) Will the lua script stop when unload mod_lua? I experienced core  
>> dump when unload mod_lua while there was a running lua script.  
>> Reported on jira.
>>
>>
>> 2) How to stop a forever running lua script?  I stop it by  
>> listening a CUSTOM event fired elsewhere. See code below. Is there  
>> any standard way like luastop ?
>>
>>
>> 3) Any way to show how many running lua scripts? luashow ?
>>
>>
>> 4) It seems cannot get the lua script name in a lua script, I made  
>> a patch to jira by assign it to the argv[0].
>>
>>
>> 5) Seems that only EventConsumer("all") working.  
>> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to  
>> work. Any idea to this?
>>
>>
>> Thanks a lot.
>>
>>
>>
>>
>>
>> code example:
>>
>>
>> con = freeswitch.EventConsumer("all");
>>
>>
>> argv[0] = "test.lua"
>>
>> freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "]  
>> Starting =====\n");
>>
>> local all_events = 0
>>
>> for e in (function() return con:pop(1) end) do
>>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
>>    all_events = all_events + 1;
>> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
>>
>> event_name = e:getHeader("Event-Name") or ""
>> event_subclass = e:getHeader("Event-Subclass") or ""
>>
>> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
>>  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..  
>> "]---Exiting------\n")
>>  break
>> end
>>
>>
>> end
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From ivan at myrvold.org  Thu Aug  6 08:53:35 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Thu, 6 Aug 2009 17:53:35 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
Message-ID: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>

Is skypiax now working on Mac OS X in Freeswitch?

Ivan



From brian at freeswitch.org  Thu Aug  6 08:55:37 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 6 Aug 2009 10:55:37 -0500
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
Message-ID: <1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>

I'm not sure about that one.... I haven't tried lately because the API  
differs on the Mac last I looked at it.

/b

On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:

> Is skypiax now working on Mac OS X in Freeswitch?
>
> Ivan




From vladrodionov at gmail.com  Thu Aug  6 08:59:36 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Thu, 6 Aug 2009 08:59:36 -0700
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <20090805204530.2ad02225396a31c9de30536f2e338977.43419788fb.wbe@email04.secureserver.net>
References: <20090805204530.2ad02225396a31c9de30536f2e338977.43419788fb.wbe@email04.secureserver.net>
Message-ID: <3c233920908060859q1b8bf67dkbe5750591446fd6@mail.gmail.com>

Pete,

Thank you for script. I can not find find channel variables rdnis,
sip_to_user and all others which start with "sb" on wiki page

http://wiki.freeswitch.org/wiki/Channel_Variables

Are they undocumented?

-Vladimir Rodionov


On Wed, Aug 5, 2009 at 8:45 PM, Pete Mueller <pete at privateconnect.com>wrote:

> Disclaimer: I'm not familiar with all the mods of FS, There may be one that
> does this already.  There are probably many ways to do this, I am just
> offering one that works well for me.
>
> Item #1 - Findout the callee #.   "destination_number" can be set to
> several different things based on the gateway configuration (forced override
> with an extension) and may or may not start with a "+" so the example below
> may not work.  To make matters worse, different gateways set fields
> differently when they hand off the call.  The most reliable I've found is
> "rdnis" or "sip_to_user" , however if you know you are going to stay with
> one gateway, you can relay on the oddities of the way they are configured.
> I had to write something relatively generic, so I moved all processing to a
> script (see #3 below)
>
> Item #2 - Find the caller ID. This is located in "caller_id_number", but
> remember in your processing that caller ID may be "anonymous", "restricted",
> "unknown" or some other word when dealing with blocked/private numbers.  You
> cannot looks for just numbers.
>
> Item #3 - Routing.  As I mentioned I have 100s of numbers across many
> gateways, so I needed a way to route the calls to the right places AND know
> which gateway the call came in on, so I can bridge the call out the same
> gateway.  I handled this by creating a small DB table (using postgreSQL) and
> connecting using LUA and luasql.  The table has three fields: number,
> gateway, and extension to route to.  In my public.xml I list all the places
> a call can be routed to and the last entry is a unconditional transfer to
> the "switchboard" script.  The switchboard script matches "rdnis" and
> "sip_to_user" to find the callee and then performs a lookup for the
> extension to route to.
>
> If you would like a copy of my switchboard script I can provide it to you
> in a PM.
> -pete
>
>  -------- Original Message --------
> Subject: Re: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
> configure?)
> From: Vladimir Rodionov <vladrodionov at gmail.com>
> Date: Wed, August 05, 2009 6:57 pm
> To: freeswitch-users at lists.freeswitch.org
>
> No, it is more like static routing. I need my *script program* be invoked
> when somebody dial in. That is it. One script for all inbound DIDs. Suppose
> I have thousand of them.  I think I know how to accomplish this but I am not
> sure yet.
>
> in my dialplan I need to define:
>
> <!-- Launch a JavaScript application if dialed in-->
>    <extension name="ProviderABC">
>     <condition field="source" expression="mod_sofia"/>
>     *<condition field="destination_number" expression="^1NXXNXXXXXX$">*
>
>      <action application="javascript" data="/usr/local/freeswitch/scripts/myapp.js"/>
>     </condition>
>    </extension>
>
>
> In provider configuration:
>
> <gateway name="voicepulse">
>            <!--/// account username *required* ///-->
>            <param name="username" value="your-username"/>
>            <!--/// auth realm: *optional* same as gateway name, if blank ///-->
>
>            <param name="realm" value="nyc.voicepulse.com"/>
>            <!--/// account password *required* ///-->
>            <param name="password" value="your-password"/>
>
>            <!--/// extension for inbound calls: *optional* same as username, if blank ///-->
> *           * *<param name="extension" value="1NXXNXXXXXX"/> *
>            <!--/// proxy host: *optional* same as realm, if blank ///-->
>
>            <param name="proxy" value="nyc.voicepulse.com"/>
>            <!--/// expire in seconds: *optional* 3600, if blank ///-->
>            <param name="expire-seconds" value="600"/>
>
> 	   <param name="register" value="true"/>
>          </gateway>
>
>
> Something like this, yes? I can use regular expressions in
> destination_number?
>
> Q: There is object Session in JavaScript, Lua. Is Session.destination ==
> destination_number from incoming call? It is not clear for me from what I
> have read so far.
>
> TIA,
>
> -Vladimir Rodionov
>
> On Wed, Aug 5, 2009 at 6:26 PM, Seven Du <dujinfang at gmail.com> wrote:
>
>> mod_easyroute?
>>
>> 2009/8/6 Vladimir Rodionov <vladrodionov at gmail.com>
>>
>>>  Hi, everybody
>>>
>>> This is a newbie question: Suppose I have XX (variable dynamic number)
>>> DIDs assigned to one sip trunk (from VOIP provider ABC ). All calls coming
>>> from VOIP provider ABC MUST be routed to the same lua/js/whatever script. Is
>>> it possible in FS? If yes, how everything should be configuered? Dialplan,
>>> sip gateway? One more question: suppose it is doeable as I hope then how can
>>> I get in my script CalleeID (not a CallerID)? Basicaly,
>>>
>>> I want to acomplish the following:
>>>
>>> 1. Avoid re-configuring FS every time I got new bunch of DIDs
>>> assigned/released from/to my Voip provider.
>>> 2. Have a way of extracting CalleeID in my script.
>>>
>>> TIA,
>>>
>>> Vladimir Rodionov
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From vladrodionov at gmail.com  Thu Aug  6 09:00:36 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Thu, 6 Aug 2009 09:00:36 -0700
Subject: [Freeswitch-users] Multiple DIDs per SIP trunk (how to
	configure?)
In-Reply-To: <87f2f3b90908052040s21545229qba24937bd2b14540@mail.gmail.com>
References: <3c233920908051630j18990b64qaa561f7f889f3616@mail.gmail.com>
	<23f91030908051826w260eec2blef46778b3b44baa1@mail.gmail.com>
	<3c233920908051857t5037d169r6bc4555b53cd015c@mail.gmail.com>
	<87f2f3b90908052040s21545229qba24937bd2b14540@mail.gmail.com>
Message-ID: <3c233920908060900j33cf4cb3hf50ca4452d1b245e@mail.gmail.com>

Thanks, I will give it it a try and let you know.

On Wed, Aug 5, 2009 at 8:40 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 8:57 PM, Vladimir Rodionov <vladrodionov at gmail.com>wrote:
>
>> No, it is more like static routing. I need my *script program* be invoked
>> when somebody dial in. That is it. One script for all inbound DIDs. Suppose
>> I have thousand of them.  I think I know how to accomplish this but I am not
>> sure yet.
>>
>
> Have the external profile be used only for provider ABC, or define a new
> profile. Then in the profile have the calls go to a specific context. You
> could have something like this in the sip profile definition:
>
> <param name="context" value="abc_calls"/>
>
> Then create a dialplan context called "abc_calls" that handles all inbound
> calls. Create a file in conf/dialplan/ called abc_calls.xml:
>
> <include>
>   <context name="abc_calls">
>     <extension name="abc_calls">
>       <condition field="destination_number" expression="^(.*)$">
>         <action application="lua" data="myscript.lua"/>
>       </condition>
>     </extension>
>   </context>
> </include>
>
> Essentially you're just creating a SIP profile and a dialplan context that
> are servicing your VoIP provider. You can add other profiles/contexts for
> other providers if need be.
>
> Let us know how it goes...
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Thu Aug  6 09:37:25 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 6 Aug 2009 18:37:25 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org> 
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
Message-ID: <7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>

No, it needs implementation of the message pump between the module and
the Skype API.

It's probably kind of trivial, if no other problems I'm not aware of.

I do not have a Mac to implement it, tough :-(.

-giovanni





Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org> wrote:
> I'm not sure about that one.... I haven't tried lately because the API
> differs on the Mac last I looked at it.
>
> /b
>
> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>
>> Is skypiax now working on Mac OS X in Freeswitch?
>>
>> Ivan
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From nicolas at medularis.com  Thu Aug  6 09:38:34 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 6 Aug 2009 12:38:34 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
Message-ID: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>

I'm bridging 2 calls in a javascript file, I originate the first call and
then execute a bridge with an origination string for the second call. If I
hangup the first call while trying to make the second call, I get this on
the console:

2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL

But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
NORMAL_CLEARING. And the variable_originate_disposition has a value of
"failure". Where can I get the detail of the call/bridge failure due to
'ORIGINATOR_CANCEL' as reported through the console?

Thanks!

Nicolas
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From msc at freeswitch.org  Thu Aug  6 09:45:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Aug 2009 11:45:33 -0500
Subject: [Freeswitch-users] How to distinguish between the different
	type of call rejections from Javascript?
In-Reply-To: <b0a8792f0908060549v40196e6fm6da7f95f47193bb4@mail.gmail.com>
References: <c55d46950908050656k785d982fpc041e4e0e4ed6107@mail.gmail.com>
	<D71A445D-79C1-40DF-B38E-88F9E668CDEC@gmail.com>
	<b0a8792f0908051354t43c5d7eboacc8b586a4aa5aa@mail.gmail.com>
	<87f2f3b90908051516q5b7b7dai776e31bebaf69b9b@mail.gmail.com>
	<121B3CA8-92B2-448A-A284-471CF1C2FA1D@freeswitch.org>
	<b0a8792f0908051644r17b642cau193ecf7a012bc766@mail.gmail.com>
	<87f2f3b90908051903y16395a62uf23feec9b3fdceab@mail.gmail.com>
	<b0a8792f0908060549v40196e6fm6da7f95f47193bb4@mail.gmail.com>
Message-ID: <87f2f3b90908060945y5cb7e842le9c9ba02bc3c303b@mail.gmail.com>

On Thu, Aug 6, 2009 at 7:49 AM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> Done, it (of course, thanks) worked smoothly. I've published the example on
> the wiki.
> http://wiki.freeswitch.org/wiki/Fakecall_responder (and linked in mod_lua
> samples)
>
> Regards,
>    Raffaele
>

Thanks for paying the wiki tax! We appreciate it when folks document their
knowledge. Please let me know if you have any wiki questions in the future.
-MC
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From vkozak at abisoft.spb.ru  Thu Aug  6 10:01:09 2009
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Thu, 6 Aug 2009 21:01:09 +0400
Subject: [Freeswitch-users] FreeSwitch doesn't play music on hold forbriged
	channel
Message-ID: <EC1055E8A337451F9F5ADC7414BDFDD9@abisoft.biz>


The scenario is the following:
FS User A dial an extension
Extention opens outbound socket channel to my application
My application bridges the call to FS User B
The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id
The application sends hold to the bridged channel using SendMsg with Other-leg-unique-id
User B is placed on hold but no music on hold is played to the caller (User A)


I have outbound socket channel and the following sequence of commands/event:
listening on [any] 8084 ...
connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250] 34000
connect

myevents

SendMsg
call-command: execute
execute-app-name: bridge
execute-app-arg:user/1000 at uat.agent.starpoundtech.net

Channel-Username: 1001
Channel-Dialplan: XML
Channel-Caller-ID-Name: 1001
Channel-Caller-ID-Number: 1001
Channel-Network-Addr: 172.26.10.39
Channel-Destination-Number: 6666
Channel-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Channel-Source: mod_sofia
Channel-Context: default
Channel-Channel-Name: sofia/internal/1001%40172.26.200.250
Channel-Profile-Index: 1
Channel-Profile-Created-Time: 1249142681680114
Channel-Channel-Created-Time: 1249142681680114
Channel-Channel-Answered-Time: 0
Channel-Channel-Progress-Time: 0
Channel-Channel-Progress-Media-Time: 1249142681809352
Channel-Channel-Hangup-Time: 0
Channel-Channel-Transfer-Time: 0
Channel-Screen-Bit: true
Channel-Privacy-Hide-Name: false
Channel-Privacy-Hide-Number: false
Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/internal/1001%40172.26.200.250
Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Call-Direction: inbound
Answer-State: early
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Network-Addr: 172.26.10.39
Caller-Destination-Number: 6666
Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249142681680114
Caller-Channel-Created-Time: 1249142681680114
Caller-Channel-Answered-Time: 0
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 1249142681809352
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
variable_sip_received_ip: 172.26.10.39
variable_sip_received_port: 13488
variable_sip_via_protocol: udp
variable_sip_authorized: true
variable_sip_mailbox: 1001
variable_sip_auth_username: 1001
variable_sip_auth_realm: 172.26.200.250
variable_mailbox: 1001
variable_toll_allow: domestic,international,local
variable_accountcode: 1001
variable_user_context: default
variable_effective_caller_id_name: Extension%201001
variable_effective_caller_id_number: 1001
variable_outbound_caller_id_name: StarPound%20FreeSWITCH
variable_outbound_caller_id_number: 0000000000
variable_callgroup: techsupport
variable_sip_from_user: 1001
variable_sip_from_uri: 1001%40172.26.200.250
variable_sip_from_host: 172.26.200.250
variable_sip_from_user_stripped: 1001
variable_sip_from_tag: bd11f93c
variable_sofia_profile_name: internal
variable_sip_req_user: 6666
variable_sip_req_uri: 6666%40172.26.200.250
variable_sip_req_host: 172.26.200.250
variable_sip_to_user: 6666
variable_sip_to_uri: 6666%40172.26.200.250
variable_sip_to_host: 172.26.200.250
variable_sip_contact_user: 1001
variable_sip_contact_port: 13488
variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
variable_sip_contact_host: 172.26.10.39
variable_channel_name: sofia/internal/1001%40172.26.200.250
variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
variable_sip_via_host: 172.26.10.39
variable_sip_via_port: 13488
variable_sip_via_rport: 13488
variable_max_forwards: 70
variable_presence_id: 1001%40172.26.200.250
variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_remote_media_ip: 172.26.10.39
variable_remote_media_port: 29826
variable_read_codec: PCMU
variable_read_rate: 8000
variable_write_codec: PCMU
variable_write_rate: 8000
variable_use_profile: nat
variable_record_stereo: true
variable_transfer_fallback_extension: operator
variable_numbering_plan: US
variable_default_areacode: 918
variable_default_gateway: example.com
variable_user_name: default
variable_domain_name: 172.26.200.250
variable_current_application_data: 172.26.200.251%3A8084%20async%20full
variable_current_application: socket
variable_socket_host: 172.26.200.251
variable_local_media_ip: 172.26.200.250
variable_local_media_port: 29370
variable_endpoint_disposition: EARLY%20MEDIA
variable_sip_nat_detected: true
Content-Type: command/reply
Reply-Text: %2BOK%0A
Socket-Mode: async
Control: full

Content-Type: command/reply
Reply-Text: +OK Events Enabled

Content-Type: command/reply
Reply-Text: +OK

Content-Length: 1541
Content-Type: text/event-plain

Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/internal/1001%40172.26.200.250
Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Call-Direction: inbound
Answer-State: early
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Network-Addr: 172.26.10.39
Caller-Destination-Number: 6666
Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249142681680114
Caller-Channel-Created-Time: 1249142681680114
Caller-Channel-Answered-Time: 0
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 1249142681809352
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Application: bridge
Application-Data: user/1000%40uat.agent.starpoundtech.net
Event-Name: CHANNEL_EXECUTE
Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
FreeSWITCH-Hostname: centos4-4-vm
FreeSWITCH-IPv4: 172.26.200.250
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-01%2020%3A04%3A51
Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A51%20GMT
Event-Date-Timestamp: 1249142691754598
Event-Calling-File: switch_core_session.c
Event-Calling-Function: switch_core_session_exec
Event-Calling-Line-Number: 1333

Content-Length: 5242
Content-Type: text/event-plain

Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/internal/1001%40172.26.200.250
Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Call-Direction: inbound
Answer-State: answered
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Network-Addr: 172.26.10.39
Caller-Destination-Number: 6666
Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249142681680114
Caller-Channel-Created-Time: 1249142681680114
Caller-Channel-Answered-Time: 1249142692414509
Caller-Channel-Progress-Time: 1249142691898434
Caller-Channel-Progress-Media-Time: 1249142681809352
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Other-Leg-Username: 1001
Other-Leg-Dialplan: XML
Other-Leg-Caller-ID-Name: Extension%201001
Other-Leg-Caller-ID-Number: 1001
Other-Leg-Network-Addr: 172.26.10.39
Other-Leg-Destination-Number: sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
Other-Leg-Source: mod_sofia
Other-Leg-Context: default
Other-Leg-Channel-Name: sofia/internal/sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
Other-Leg-Screen-Bit: true
Other-Leg-Privacy-Hide-Name: false
Other-Leg-Privacy-Hide-Number: false
variable_sip_received_ip: 172.26.10.39
variable_sip_received_port: 13488
variable_sip_via_protocol: udp
variable_sip_authorized: true
variable_sip_mailbox: 1001
variable_sip_auth_username: 1001
variable_sip_auth_realm: 172.26.200.250
variable_mailbox: 1001
variable_toll_allow: domestic,international,local
variable_accountcode: 1001
variable_user_context: default
variable_effective_caller_id_name: Extension%201001
variable_effective_caller_id_number: 1001
variable_outbound_caller_id_name: StarPound%20FreeSWITCH
variable_outbound_caller_id_number: 0000000000
variable_callgroup: techsupport
variable_sip_from_user: 1001
variable_sip_from_uri: 1001%40172.26.200.250
variable_sip_from_host: 172.26.200.250
variable_sip_from_user_stripped: 1001
variable_sip_from_tag: bd11f93c
variable_sofia_profile_name: internal
variable_sip_req_user: 6666
variable_sip_req_uri: 6666%40172.26.200.250
variable_sip_req_host: 172.26.200.250
variable_sip_to_user: 6666
variable_sip_to_uri: 6666%40172.26.200.250
variable_sip_to_host: 172.26.200.250
variable_sip_contact_user: 1001
variable_sip_contact_port: 13488
variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
variable_sip_contact_host: 172.26.10.39
variable_channel_name: sofia/internal/1001%40172.26.200.250
variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
variable_sip_via_host: 172.26.10.39
variable_sip_via_port: 13488
variable_sip_via_rport: 13488
variable_max_forwards: 70
variable_presence_id: 1001%40172.26.200.250
variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_remote_media_ip: 172.26.10.39
variable_remote_media_port: 29826
variable_read_codec: PCMU
variable_read_rate: 8000
variable_write_codec: PCMU
variable_write_rate: 8000
variable_use_profile: nat
variable_record_stereo: true
variable_transfer_fallback_extension: operator
variable_numbering_plan: US
variable_default_areacode: 918
variable_default_gateway: example.com
variable_user_name: default
variable_domain_name: 172.26.200.250
variable_socket_host: 172.26.200.251
variable_local_media_ip: 172.26.200.250
variable_local_media_port: 29370
variable_endpoint_disposition: EARLY%20MEDIA
variable_current_application_data: user/1000%40uat.agent.starpoundtech.net
variable_current_application: bridge
variable_dialed_user: 1000
variable_dialed_domain: uat.agent.starpoundtech.net
variable_originate_disposition: failure
variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
variable_sip_redirect_contact_user_0: 1000
variable_sip_redirect_contact_host_0: 172.26.10.39
variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_sip_nat_detected: true
Event-Name: CHANNEL_ANSWER
Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
FreeSWITCH-Hostname: centos4-4-vm
FreeSWITCH-IPv4: 172.26.200.250
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-01%2020%3A04%3A52
Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
Event-Date-Timestamp: 1249142692414509
Event-Calling-File: switch_channel.c
Event-Calling-Function: switch_channel_perform_mark_answered
Event-Calling-Line-Number: 1776

Content-Length: 5233
Content-Type: text/event-plain

Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/internal/1001%40172.26.200.250
Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Call-Direction: inbound
Answer-State: answered
Channel-Read-Codec-Name: PCMU
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMU
Channel-Write-Codec-Rate: 8000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Network-Addr: 172.26.10.39
Caller-Destination-Number: 6666
Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1249142681680114
Caller-Channel-Created-Time: 1249142681680114
Caller-Channel-Answered-Time: 1249142692414509
Caller-Channel-Progress-Time: 1249142691898434
Caller-Channel-Progress-Media-Time: 1249142681809352
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: true
Caller-Privacy-Hide-Name: false
Caller-Privacy-Hide-Number: false
Other-Leg-Username: 1001
Other-Leg-Dialplan: XML
Other-Leg-Caller-ID-Name: Extension%201001
Other-Leg-Caller-ID-Number: 1001
Other-Leg-Network-Addr: 172.26.10.39
Other-Leg-Destination-Number: sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
Other-Leg-Source: mod_sofia
Other-Leg-Context: default
Other-Leg-Channel-Name: sofia/internal/sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
Other-Leg-Screen-Bit: true
Other-Leg-Privacy-Hide-Name: false
Other-Leg-Privacy-Hide-Number: false
variable_sip_received_ip: 172.26.10.39
variable_sip_received_port: 13488
variable_sip_via_protocol: udp
variable_sip_authorized: true
variable_sip_mailbox: 1001
variable_sip_auth_username: 1001
variable_sip_auth_realm: 172.26.200.250
variable_mailbox: 1001
variable_toll_allow: domestic,international,local
variable_accountcode: 1001
variable_user_context: default
variable_effective_caller_id_name: Extension%201001
variable_effective_caller_id_number: 1001
variable_outbound_caller_id_name: StarPound%20FreeSWITCH
variable_outbound_caller_id_number: 0000000000
variable_callgroup: techsupport
variable_sip_from_user: 1001
variable_sip_from_uri: 1001%40172.26.200.250
variable_sip_from_host: 172.26.200.250
variable_sip_from_user_stripped: 1001
variable_sip_from_tag: bd11f93c
variable_sofia_profile_name: internal
variable_sip_req_user: 6666
variable_sip_req_uri: 6666%40172.26.200.250
variable_sip_req_host: 172.26.200.250
variable_sip_to_user: 6666
variable_sip_to_uri: 6666%40172.26.200.250
variable_sip_to_host: 172.26.200.250
variable_sip_contact_user: 1001
variable_sip_contact_port: 13488
variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
variable_sip_contact_host: 172.26.10.39
variable_channel_name: sofia/internal/1001%40172.26.200.250
variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
variable_sip_via_host: 172.26.10.39
variable_sip_via_port: 13488
variable_sip_via_rport: 13488
variable_max_forwards: 70
variable_presence_id: 1001%40172.26.200.250
variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_remote_media_ip: 172.26.10.39
variable_remote_media_port: 29826
variable_read_codec: PCMU
variable_read_rate: 8000
variable_write_codec: PCMU
variable_write_rate: 8000
variable_use_profile: nat
variable_record_stereo: true
variable_transfer_fallback_extension: operator
variable_numbering_plan: US
variable_default_areacode: 918
variable_default_gateway: example.com
variable_user_name: default
variable_domain_name: 172.26.200.250
variable_socket_host: 172.26.200.251
variable_local_media_ip: 172.26.200.250
variable_local_media_port: 29370
variable_current_application_data: user/1000%40uat.agent.starpoundtech.net
variable_current_application: bridge
variable_dialed_user: 1000
variable_dialed_domain: uat.agent.starpoundtech.net
variable_sip_redirect_contact_user_0: 1000
variable_sip_redirect_contact_host_0: 172.26.10.39
variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
variable_sip_nat_detected: true
variable_endpoint_disposition: ANSWER
variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
variable_originate_disposition: SUCCESS
Event-Name: CHANNEL_BRIDGE
Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
FreeSWITCH-Hostname: centos4-4-vm
FreeSWITCH-IPv4: 172.26.200.250
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-01%2020%3A04%3A52
Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
Event-Date-Timestamp: 1249142692414509
Event-Calling-File: switch_ivr_bridge.c
Event-Calling-Function: switch_ivr_multi_threaded_bridge
Event-Calling-Line-Number: 828

SendMsg 94b59a38-57c4-4703-9c6e-9985d832d119
call-command: execute
execute-app-name: hold

Content-Type: command/reply
Reply-Text: +OK




- I don't see the variable hold_music ... did you remove it? 

- I didn't. Moreover, I tried to set it explicitly using api uuid_setvar.
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From msc at freeswitch.org  Thu Aug  6 10:30:09 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 6 Aug 2009 12:30:09 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 Release Announcement
Message-ID: <87f2f3b90908061030m1afd5adajcc90e31b6a10b5f6@mail.gmail.com>

We are happy to announce the official release of FreeSWITCH 1.0.4! Please
visit this link <http://digg.com/d3zt5X> to Digg and read the story, and
then spread the word!

Thanks for being such a great community!
-Michael
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From mattdfong at gmail.com  Thu Aug  6 11:25:32 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Thu, 6 Aug 2009 11:25:32 -0700
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
Message-ID: <4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>

Hi Nicolas,
do you have a copy of the .js code you can paste. I would guess tho, that
ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to
false. Just a guess tho.

Hangup causes can be found here:
http://wiki.freeswitch.org/wiki/Hangup_causes

<http://wiki.freeswitch.org/wiki/Hangup_causes>--matt
hello hunter - hosted predictive dialer & voice broadcasting
http://www.hellohunter.com


On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com>wrote:

> I'm bridging 2 calls in a javascript file, I originate the first call and
> then execute a bridge with an origination string for the second call. If I
> hangup the first call while trying to make the second call, I get this on
> the console:
>
> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/005622170039 [KILL]
> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/005622170039 [BREAK]
> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
> Cause: ORIGINATOR_CANCEL
>
> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
> NORMAL_CLEARING. And the variable_originate_disposition has a value of
> "failure". Where can I get the detail of the call/bridge failure due to
> 'ORIGINATOR_CANCEL' as reported through the console?
>
> Thanks!
>
> Nicolas
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nicolas at medularis.com  Thu Aug  6 12:45:01 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Thu, 6 Aug 2009 15:45:01 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
	<4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
Message-ID: <1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>

Hi Matt,

Actually I'm explicitly setting hangup_after_bridge to true, think setting
it to false would help? I'm going to try that.

Here's the JS code:
(Note: session.getVariable() doesn't work, FS complains saying it is not a
function, also tried self.session.getVariable() - that's what the wiki says
- and FS complains that self does not exist)

----------------
var uuid = argv[0]; // Call identifier
var dialstr1 = argv[1]; // Dial string obtained from previous call to LCR
var dialstr2 = argv[2]; // Dial string obtained from previous call to LCR
var greeting_snd = "/var/audio/alert.wav";

console_log("notice", "*********** STARTING C2C Call ***********\n");
timeout = 30;

console_log("notice", "*********** DIALING "+dialstr1+" ***********\n");

//var stUsRing = session.getVariable("us-ring");  // This doesn't work,
self.session.getVariable doesn't work either
var stUsRing = "%(2000,4000,440,480)";

// Create new_session
new_session = new Session(originate_str1);
console_log("notice", "*********** Leg1: " + new_session.cause + "
***********\n");

if (new_session.ready()) {
        // log to the console
        console_log("notice", "*********** Leg1 ("+dialstr1+") CONNECTED!
***********\n");
        console_log("notice", "*********** Playing greeting sound:
"+greeting_snd+" ***********\n");

        new_session.execute("sleep", 100);
        new_session.execute("playback", greeting_snd);

        // Originate second call and bridge
    originate_str2 =
"{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;

        // Create new_session
        new_session.execute("bridge", originate_str2);
        console_log("notice", "*********** Leg2: " + new_session.cause + "
***********\n");

        if (new_session.ready()) {
                console_log("notice", "*********** Leg2 ("+dialstr2+")
CONNECTED! ***********\n");
        }
}

exit();
----------------

Thanks!


Nicolas


On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong <mattdfong at gmail.com> wrote:

> Hi Nicolas,
> do you have a copy of the .js code you can paste. I would guess tho, that
> ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to
> false. Just a guess tho.
>
> Hangup causes can be found here:
> http://wiki.freeswitch.org/wiki/Hangup_causes
>
> <http://wiki.freeswitch.org/wiki/Hangup_causes> --matt
> hello hunter - hosted predictive dialer & voice broadcasting
> http://www.hellohunter.com
>
>
> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>
>> I'm bridging 2 calls in a javascript file, I originate the first call and
>> then execute a bridge with an origination string for the second call. If I
>> hangup the first call while trying to make the second call, I get this on
>> the console:
>>
>> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
>> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
>> sofia/external/005622170039 [KILL]
>> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/external/005622170039 [BREAK]
>> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
>> Cause: ORIGINATOR_CANCEL
>>
>> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
>> NORMAL_CLEARING. And the variable_originate_disposition has a value of
>> "failure". Where can I get the detail of the call/bridge failure due to
>> 'ORIGINATOR_CANCEL' as reported through the console?
>>
>> Thanks!
>>
>> Nicolas
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jim.page at redmatter.com  Thu Aug  6 14:55:08 2009
From: jim.page at redmatter.com (Jim Page)
Date: Thu, 6 Aug 2009 22:55:08 +0100
Subject: [Freeswitch-users] CURL directory issue
In-Reply-To: <ab57a3e80908060848x64bbc808r162273eb4cbcd363@mail.gmail.com>
References: <E0C2263D8B20E14390DF8AA03E1F99A0C506DCB1@sha-exch9.shared.ifeltd.com>
	<ab57a3e80908060848x64bbc808r162273eb4cbcd363@mail.gmail.com>
Message-ID: <C45A122C-C1C0-4267-9028-C192CED52ED4@redmatter.com>

Spot on. Many thanks!
Jim

Sent from my iPhone

On 6 Aug 2009, at 18:02, "Kevin Green" <kevin at johnnyvoip.com<mailto:kevin at johnnyvoip.com>> wrote:

Try returning the full information on the register. It may be that the variables are read onto the user profile upon registration and since you are only supplying a dumbed down version for registration the variables aren't being read and cached.

Regards,
   Kevin Green

On Thu, Aug 6, 2009 at 7:24 AM, Jim Page <<mailto:jim.page at redmatter.com>jim.page at redmatter.com<mailto:jim.page at redmatter.com>> wrote:
Afternoon All

I wonder if someone (perhaps even the illustrious intralanman) could help me out with a problem I am experiencing with a CURL directory.

In the interests of understanding how the mechanism works, I am using a super-braindead php script to return info about a specific set of users. I plan to move to something more sophisticated once the proof of concept is complete, possibly based on intralanman's scripts.

The basic problem is that all works fine (boot, register, voicemail etc), except that user's <variables> seem not to be being read correctly, eg 'toll_allow' and 'user_context'. Here's a typical user XML message I am returning:

<document type="freeswitch/xml">
 <section name="directory">
  <domain name="<http://pbx.redmatter.com>pbx.redmatter.com<http://pbx.redmatter.com>">
   <params>
     <param name="dial-string" value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
   </params>

   <variables>
     <variable name="record_stereo" value="true"/>
     <variable name="default_gateway" value="$${default_provider}"/>
     <variable name="default_areacode" value="$${default_areacode}"/>
     <variable name="transfer_fallback_extension" value="operator"/>
   </variables>

   <groups>
     <group name="default">
       <users>            <user id="1009" cidr="172.30.99.0/24<http://172.30.99.0/24>">
              <params>
                <param name="password" value="$${default_password}"/>
                <param name="vm-password" value="1009"/>
                <param name="vm-email-all-messages" value="true"/>
                <param name="vm-attach-file" value="true"/>
                <param name="vm-mailto" value="<mailto:1009 at redmatter.com>1009 at redmatter.com<mailto:1009 at redmatter.com>"/>
                <param name="vm-keep-local-after-email" value="true"/>
                <param name="vm_message_ext" value="mp3"/>
              </params>
              <variables>
                <variable name="toll_allow" value="domestic,international,local"/>
                <variable name="accountcode" value="1009"/>
                <variable name="user_context" value="default"/>
                <variable name="effective_caller_id_name" value="Extension 1009"/>
                <variable name="effective_caller_id_number" value="1009"/>
                <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/>
                <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/>
                <variable name="callgroup" value="techsupport"/>
              </variables>
            </user>        </users>
     </group>
   </groups>
  </domain>
 </section>
</document>

I return this kind of message in all cases except the (sip_auth_method=="REGISTER") request message where I return

<document type="freeswitch/xml">
 <section name="directory">
   <domain name="<http://pbx.redmatter.com>pbx.redmatter.com<http://pbx.redmatter.com>">
     <user id="1007">
       <params>
         <param name="password" value="1234"/>
       </params>
     </user>
   </domain>
 </section>
</document>

Also it's probably worth mentioning that I have removed all trace of xml from conf/directory and I don't believe there is a conflict happening there.

The phones register correctly. The trouble is they don't operate on the correct dialplan context (I fixed that by hardcoding the internal gateway to dialplan default), but the 'toll_allow' variable is now not working so that outbound calls fail, which is what made me think that the user variables are being ignored.

Freeswitch version is 1.0.4, built by me and running on a dell 1950 running Centos 5.3 x86_64. HTTP application is running on standard Centos 5.3 apache/php.

Any ideas gratefully and humbly received.

All the best
Jim

_______________________________________________
FreeSWITCH-users mailing list
<mailto:FreeSWITCH-users at lists.freeswitch.org>FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
<http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>http://lists.freeswitch.org/mailman/options/freeswitch-users
<http://www.freeswitch.org>http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From Nick.Lemberger at lkfd.net  Thu Aug  6 16:19:03 2009
From: Nick.Lemberger at lkfd.net (Nick Lemberger)
Date: Thu, 06 Aug 2009 18:19:03 -0500
Subject: [Freeswitch-users] Lua Script Return Value & mod_xmlrpc
Message-ID: <4A7B1EC4.2C9A.00FE.0@lkfd.net>

Is it possible to have a LUA script return something to the client when accessed via the XML RPC gateway & luarun?

ie: access the url: http://FSip:8080/api/luarun?myscript.lua and have the script return a value?

-Nick




From pete at privateconnect.com  Thu Aug  6 16:35:30 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Thu, 06 Aug 2009 16:35:30 -0700
Subject: [Freeswitch-users]
	=?utf-8?q?Lua_Script_Return_Value_=26_mod=5Fxm?= =?utf-8?q?lrpc?=
Message-ID: <20090806163530.2ad02225396a31c9de30536f2e338977.1282ed1d44.wbe@email04.secureserver.net>

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From raffaele.p.guidi at gmail.com  Thu Aug  6 17:23:59 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Fri, 7 Aug 2009 02:23:59 +0200
Subject: [Freeswitch-users] A few questions about lua
In-Reply-To: <EBFA6F4D-07D0-4C6D-ABDF-8A8FF3976FE2@gmail.com>
References: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
	<1249555620.5449.1.camel@eli-desktop>
	<EBFA6F4D-07D0-4C6D-ABDF-8A8FF3976FE2@gmail.com>
Message-ID: <b0a8792f0908061723q4fe2a24cobea858addbf4e07@mail.gmail.com>

>> 5) Seems that only EventConsumer("all") working.
EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work. Any
idea to this?

isn't it CHANNEL_HAN*G*UP? Is the G missing only in the email or in the
code, too?

On Thu, Aug 6, 2009 at 17:52, Seven Du <dujinfang at gmail.com> wrote:

> for e in (function() return con:pop(1) end) do
>
> btw, the script works.
>
> Thanks.
> On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote:
> > Hi
> > I dont know about events so much but I cannot see variable "e" is
> > setting
> >
> > event_name = e:getHeader("Event-Name") or ""
> > event_subclass = e:getHeader("Event-Subclass") or ""
> >
> > regurds
> > Eli
> >
> > On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:
> >> ALL-
> >>
> >>
> >> I have a few questions when scripting lua. According to wiki, it is
> >> possible to run looping forever lua scripts through start-up config
> >> or luarun.
> >>
> >>
> >> 1) Will the lua script stop when unload mod_lua? I experienced core
> >> dump when unload mod_lua while there was a running lua script.
> >> Reported on jira.
> >>
> >>
> >> 2) How to stop a forever running lua script?  I stop it by
> >> listening a CUSTOM event fired elsewhere. See code below. Is there
> >> any standard way like luastop ?
> >>
> >>
> >> 3) Any way to show how many running lua scripts? luashow ?
> >>
> >>
> >> 4) It seems cannot get the lua script name in a lua script, I made
> >> a patch to jira by assign it to the argv[0].
> >>
> >>
> >> 5) Seems that only EventConsumer("all") working.
> >> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to
> >> work. Any idea to this?
> >>
> >>
> >> Thanks a lot.
> >>
> >>
> >>
> >>
> >>
> >> code example:
> >>
> >>
> >> con = freeswitch.EventConsumer("all");
> >>
> >>
> >> argv[0] = "test.lua"
> >>
> >> freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "]
> >> Starting =====\n");
> >>
> >> local all_events = 0
> >>
> >> for e in (function() return con:pop(1) end) do
> >>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
> >>    all_events = all_events + 1;
> >> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
> >>
> >> event_name = e:getHeader("Event-Name") or ""
> >> event_subclass = e:getHeader("Event-Subclass") or ""
> >>
> >> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
> >>  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..
> >> "]---Exiting------\n")
> >>  break
> >> end
> >>
> >>
> >> end
> >>
> >>
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vladrodionov at gmail.com  Thu Aug  6 17:55:55 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Thu, 6 Aug 2009 17:55:55 -0700
Subject: [Freeswitch-users] Lua on Windows and additional modules
Message-ID: <3c233920908061755h5d17aa0as3fb24743215a8298@mail.gmail.com>

Good evening,
This is newbie question.

The FreeSWITCH lua module does not support sockets and sql out of box that
is why
I just installed LuaBinaries (including socket, sql modules). My dev
environment is Win XP not Linux/Unix.

I am trying to understand what will happen when lua_module get this:

require "socket" or
require "luasql.mysql"
?

How does lua_module look up additional lua modules on Windows platform?

Do I have to set some env variables?

TIA
-Vladimir Rodionov
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From dujinfang at gmail.com  Thu Aug  6 20:45:16 2009
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 7 Aug 2009 11:45:16 +0800
Subject: [Freeswitch-users] A few questions about lua
In-Reply-To: <b0a8792f0908061723q4fe2a24cobea858addbf4e07@mail.gmail.com>
References: <23f91030908060258p50bdfac9w32f5f1b435d1689b@mail.gmail.com>
	<1249555620.5449.1.camel@eli-desktop>
	<EBFA6F4D-07D0-4C6D-ABDF-8A8FF3976FE2@gmail.com>
	<b0a8792f0908061723q4fe2a24cobea858addbf4e07@mail.gmail.com>
Message-ID: <23f91030908062045s6ebc35efne6f5ce999f085506@mail.gmail.com>

Sorry it's a typo. I read the code, it works not like in event socket. So,
only works with one event.
either
EventConsumer("all")
or
EventConsumer("CUSTOM", "lua::stop");


Thank you.

2009/8/7 Raffaele P. Guidi <raffaele.p.guidi at gmail.com>

> >> 5) Seems that only EventConsumer("all") working.
> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to work. Any
> idea to this?
>
> isn't it CHANNEL_HAN*G*UP? Is the G missing only in the email or in the
> code, too?
>
> On Thu, Aug 6, 2009 at 17:52, Seven Du <dujinfang at gmail.com> wrote:
>
>> for e in (function() return con:pop(1) end) do
>>
>> btw, the script works.
>>
>> Thanks.
>> On Aug 6, 2009, at 6:47 PM, Eli Hayun wrote:
>> > Hi
>> > I dont know about events so much but I cannot see variable "e" is
>> > setting
>> >
>> > event_name = e:getHeader("Event-Name") or ""
>> > event_subclass = e:getHeader("Event-Subclass") or ""
>> >
>> > regurds
>> > Eli
>> >
>> > On Thu, 2009-08-06 at 12:58 +0300, Seven Du wrote:
>> >> ALL-
>> >>
>> >>
>> >> I have a few questions when scripting lua. According to wiki, it is
>> >> possible to run looping forever lua scripts through start-up config
>> >> or luarun.
>> >>
>> >>
>> >> 1) Will the lua script stop when unload mod_lua? I experienced core
>> >> dump when unload mod_lua while there was a running lua script.
>> >> Reported on jira.
>> >>
>> >>
>> >> 2) How to stop a forever running lua script?  I stop it by
>> >> listening a CUSTOM event fired elsewhere. See code below. Is there
>> >> any standard way like luastop ?
>> >>
>> >>
>> >> 3) Any way to show how many running lua scripts? luashow ?
>> >>
>> >>
>> >> 4) It seems cannot get the lua script name in a lua script, I made
>> >> a patch to jira by assign it to the argv[0].
>> >>
>> >>
>> >> 5) Seems that only EventConsumer("all") working.
>> >> EventConsumer("CHANNEL_HANUP CUSTOM lua::stop") doesn't seem to
>> >> work. Any idea to this?
>> >>
>> >>
>> >> Thanks a lot.
>> >>
>> >>
>> >>
>> >>
>> >>
>> >> code example:
>> >>
>> >>
>> >> con = freeswitch.EventConsumer("all");
>> >>
>> >>
>> >> argv[0] = "test.lua"
>> >>
>> >> freeswitch.consoleLog("info", "==== Lua Script [" .. argv[0] .. "]
>> >> Starting =====\n");
>> >>
>> >> local all_events = 0
>> >>
>> >> for e in (function() return con:pop(1) end) do
>> >>   -- freeswitch.consoleLog("info", "event\n" .. e:serialize("xml"));
>> >>    all_events = all_events + 1;
>> >> freeswitch.consoleLog("info", "all_events: " .. all_events .. "\n")
>> >>
>> >> event_name = e:getHeader("Event-Name") or ""
>> >> event_subclass = e:getHeader("Event-Subclass") or ""
>> >>
>> >> if (event_name == "CUSTOM" and event_subclass == "lua::stop") then
>> >>  freeswitch.consoleLog("info", "-----lua Script [" .. argv[0] ..
>> >> "]---Exiting------\n")
>> >>  break
>> >> end
>> >>
>> >>
>> >> end
>> >>
>> >>
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Thu Aug  6 21:07:00 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Fri, 7 Aug 2009 11:07:00 +0700
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 Release Announcement
In-Reply-To: <87f2f3b90908061030m1afd5adajcc90e31b6a10b5f6@mail.gmail.com>
References: <87f2f3b90908061030m1afd5adajcc90e31b6a10b5f6@mail.gmail.com>
Message-ID: <8ccbff060908062107p1f21fd8dp8c631d7a627520f8@mail.gmail.com>

Good News..

2009/8/7 Michael Collins <msc at freeswitch.org>:
> We are happy to announce the official release of FreeSWITCH 1.0.4! Please
> visit this link to Digg and read the story, and then spread the word!
>
> Thanks for being such a great community!
> -Michael
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From pete at privateconnect.com  Thu Aug  6 21:59:34 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Thu, 06 Aug 2009 21:59:34 -0700
Subject: [Freeswitch-users] Lua on Windows and additional modules
Message-ID: <20090806215934.2ad02225396a31c9de30536f2e338977.122e91f14a.wbe@email04.secureserver.net>

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From velu.technical at gmail.com  Thu Aug  6 22:06:21 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Fri, 7 Aug 2009 10:36:21 +0530
Subject: [Freeswitch-users] Fwd: execute function in ESL.pm module is
	not working
In-Reply-To: <87f2f3b90908052254l3ecc7fa0ybc92d87c587a9b0d@mail.gmail.com>
References: <1452e2980908042314g750a309cvff7bce9fed9f87eb@mail.gmail.com>
	<1452e2980908052138jed5d676oe5ce73b0b4b0e37e@mail.gmail.com>
	<87f2f3b90908052254l3ecc7fa0ybc92d87c587a9b0d@mail.gmail.com>
Message-ID: <1452e2980908062206s3b33f867if569a10fff078673@mail.gmail.com>

Dear Expert,
   Thanks for you reply....

My Perl Script is,
use strict;
use warnings;

#---------------------------------------------------------------------------
# Event socket library.
# Socket programming
# printing the data structures
# Using posix parametered functions.
#---------------------------------------------------------------------------
use lib('/root/freeswitch-1.0.3/libs/esl/perl/');
require ESL;
use IO::Socket::INET;
use Data::Dumper qw(Dumper);
use POSIX;
use Config::IniFiles;


# Global variables to store the socket connection and eneterd DTM digits.
my ($conn,$digit);
$digit='';

#Registering the ALARM signal.
$SIG{ALRM}=\&sub_alr;

# When alarm signal occurs call the play_digit function
sub sub_alr {
        print "IN Sigalarm---\n";
        &play_digit;
        return ;
}       # ----------  end of subroutine sub_alr  ----------



# Play the voice files for menu.
sub play(){
                $conn->execute("playback","ivr/ivr-please.wav");
                $conn->execute("playback","ivr/ivr-enter_ext.wav");
}


sub play_digit {
        print "In Play Digit....\n";
        my      ( $par1 )       = $digit; #$digit is global variable

        print "Eneterd Digits=",$digit,"\n";
        ################################################################
        # Here what is my problem the execute function is not working  #
        ################################################################
        $conn->execute("phrase", "spell,$par1");
        return ;
}       # ----------  end of subroutine play_digit  ----------


#---------------------------------------------------------------------------
# IP address and port of the server.
# Sound path file.
#---------------------------------------------------------------------------
my $ip = "192.168.1.222"; my $port = '5057';
my $sound_path = "/usr/local/freeswitch/sounds/en/us/callie/";


# Creating a socket
my $sock = new IO::Socket::INET (
        LocalHost => $ip,
        LocalPort => $port,
        Proto     => 'tcp',
        Listen    => 1,
        Reuse     => 1
);
# Checking the error.
die "Cannot create a socket:$!\n" unless $sock;


for(;;){

        my $new_socket = $sock->accept();
        print "Current Process Id:".POSIX::getpid()."\n";
        my $pid = fork();

        if($pid){
                close($new_socket);
                next;
        }

        print "Child Process Id:".POSIX::getpid()."\n";

        my $fd = fileno($new_socket);
        print "File Number:$fd\n";

        # Create a conenction with Event socket library.
        $conn = new ESL::ESLconnection($fd);

        # Getting the connection informations and values of the variables.
        my $info = $conn->getInfo();

        # Getting the caller id and print the statement.
        my $caller_id =$info->getHeader("caller-caller-id-number");
        printf  "Connected from %s\n", $caller_id;

        # Receive the events from only in this switch.
        $conn->sendRecv("myevents");

        # Answer the call.
        $conn->execute("answer");

        # playback the welcome message.
        $conn->setEventLock("true");

$conn->execute("playback",$sound_path."ivr/ivr-welcome_to_freeswitch.wav");
        $conn->execute("sleep", "1000");

        &play;


        alarm(10);
        while($conn->connected()){

                # Receive the event
                my $event = $conn->recvEvent();

                # Check the event is received
                if($event){
                        # Get the event name  and print it.
                        my $name = $event->getHeader("event-name");
                        print "EVENT:[$name]\n";

                        # If the event name is DTMF then print the  enterted
digit.
                        if($name eq 'DTMF'){
                                my $digi = $event->getHeader("dtmf-digit");
                                # Here concatenate the eneterd digits
                                $digit.=$digi;
                        }
                }
        }
        # Kill the child process.
        print "Disconnected:$caller_id\n"; kill 9,POSIX::getpid();
}


My dial plan is,
<!-- Testing IVR -->
    <extension name="Test">
      <condition field="destination_number" expression="^(200)$">
            <action application="socket" data="192.168.1.222:5057 async
full"/>
      </condition>
    </extension>

The output of the Script is,

Current Process Id:2906
Child Process Id:2908
File Number:4
Connected from 1000
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_ANSWER]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[CHANNEL_EXECUTE]
EVENT:[CHANNEL_EXECUTE_COMPLETE]
EVENT:[DTMF]
EVENT:[DTMF]
EVENT:[DTMF]
EVENT:[DTMF]
IN Sigalarm---
In Play Digit....
Eneterd Digits=7485
Disconnected:1000

When alarm signal generated, it prints digits but it won't execute the
"execute" function..

Please  any one give suggestions where I made wrong...

Thanks...

Regards,
Velusamy.


On Thu, Aug 6, 2009 at 11:24 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 5, 2009 at 11:38 PM, velusamy velu <velu.technical at gmail.com>wrote:
>
>> Please any one help for this problem..
>>
>>
> Sorry for the delay but many of the FreeSWITCH experts are at ClueCon right
> now so we'll ask for your patience... in the meantime could you pastebin
> your script and your dialplan entry so that we can take a look at them?
>
> Thanks,
> MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ryder86 at googlemail.com  Fri Aug  7 00:36:16 2009
From: ryder86 at googlemail.com (Artem Vasiliev)
Date: Fri, 7 Aug 2009 11:36:16 +0400
Subject: [Freeswitch-users] Softphone control
Message-ID: <cb6be45b0908070036x5063615ev93e2526427905a23@mail.gmail.com>

Hi

I have FreeSwitch and external application, which communicates to it via
event socket - listens for events for certain number and gives some
commands.
Is it possible for this application to control client softphones, for
example, make them answer or hold, using the event socket or other
FreeSwitch capabilities?
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From dujinfang at gmail.com  Fri Aug  7 01:10:17 2009
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 7 Aug 2009 16:10:17 +0800
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <cb6be45b0908070036x5063615ev93e2526427905a23@mail.gmail.com>
References: <cb6be45b0908070036x5063615ev93e2526427905a23@mail.gmail.com>
Message-ID: <23f91030908070110j199fc1fcpcce4685318a1e8c9@mail.gmail.com>

You can run FreeSWITCH as a softphone and control it.
http://wiki.freeswitch.org/wiki/Freeswitch_softphone

2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>

> Hi
>
> I have FreeSwitch and external application, which communicates to it via
> event socket - listens for events for certain number and gives some
> commands.
> Is it possible for this application to control client softphones, for
> example, make them answer or hold, using the event socket or other
> FreeSwitch capabilities?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From ryder86 at googlemail.com  Fri Aug  7 04:02:04 2009
From: ryder86 at googlemail.com (Artem Vasiliev)
Date: Fri, 7 Aug 2009 15:02:04 +0400
Subject: [Freeswitch-users] Softphone control
Message-ID: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>

No, I don't want to make softphone from FreeSwitch

I have FS and several users with eyeBeam softphones. I need to control
those eyeBeams

>You can run FreeSWITCH as a softphone and control it.
>http://wiki.freeswitch.org/wiki/Freeswitch_softphone

>2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>

>> Hi
>>
>> I have FreeSwitch and external application, which communicates to it via
>> event socket - listens for events for certain number and gives some
>> commands.
>> Is it possible for this application to control client softphones, for
>> example, make them answer or hold, using the event socket or other
>> FreeSwitch capabilities?
>>



From merul at mac.com  Fri Aug  7 04:17:12 2009
From: merul at mac.com (Merul Patel)
Date: Fri, 07 Aug 2009 12:17:12 +0100
Subject: [Freeswitch-users] /etc/openzap/tones.conf for UK
Message-ID: <599DBC67-02FE-4F9E-9F87-6D1749B81B11@mac.com>

Where can I find a sample tones.conf file for the UK? Am trying to  
configure a USBFXO device for outbound calls.

Thanks in advance,

Merul



From kevin at johnnyvoip.com  Fri Aug  7 08:00:39 2009
From: kevin at johnnyvoip.com (Kevin Green)
Date: Fri, 7 Aug 2009 11:00:39 -0400
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
References: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
Message-ID: <ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>

>From what I am aware you can't use FreeSWITCH to control a softphone
directly though you can make it do things that will have a similar end
result. You could set eyeBeam to auto-answer calls if you want them to
answer right away or orginiate a call that is auto-answered but not bridge
the call until a user on the eyeBeam presses a digit or a socket control
tells it to connect the two ends. You can also use FreeSWITCH to place the
line on hold using event sockets, this will place it on hold in the server
and not directly like placing it on hold in eyeBeam (i.e. the hold button in
eyeBeam likely wont show it as being on hold).

Beyond that if you want to directly control the clients you would need to
look at getting an API access into the eyeBeam client.

I hope this will help.

Regards,
   Kevin Green


On Fri, Aug 7, 2009 at 7:02 AM, Artem Vasiliev <ryder86 at googlemail.com>wrote:

> No, I don't want to make softphone from FreeSwitch
>
> I have FS and several users with eyeBeam softphones. I need to control
> those eyeBeams
>
> >You can run FreeSWITCH as a softphone and control it.
> >http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>
> >2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>
>
> >> Hi
> >>
> >> I have FreeSwitch and external application, which communicates to it via
> >> event socket - listens for events for certain number and gives some
> >> commands.
> >> Is it possible for this application to control client softphones, for
> >> example, make them answer or hold, using the event socket or other
> >> FreeSwitch capabilities?
> >>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From asannucci at gmail.com  Fri Aug  7 08:21:51 2009
From: asannucci at gmail.com (bakko)
Date: Fri, 7 Aug 2009 17:21:51 +0200
Subject: [Freeswitch-users] Spanish Prompts
Message-ID: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>

I'd like to begin record spanish prompts for FS.

Do you know any software/hardware to make it?

Thank you

BR




From nicolas at medularis.com  Fri Aug  7 09:43:29 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 7 Aug 2009 12:43:29 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
	<4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
	<1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>
Message-ID: <1b46b4e80908070943j6e55c125xde064029513b099c@mail.gmail.com>

I changed the script to set hangup_after_bridge to false, but still the same
thing happens, I get this on the console:

2009-08-07 12:27:44.229091 [NOTICE] sofia.c:322 Hangup
sofia/external/00569xxxxxxx [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
2009-08-07 12:27:44.229091 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/00569xxxxxxx [KILL]
2009-08-07 12:27:44.229091 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/00569xxxxxxx [BREAK]
2009-08-07 12:27:44.231471 [NOTICE] switch_ivr_originate.c:1994 Hangup
sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
2009-08-07 12:27:44.231471 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/005622170039 [KILL]
2009-08-07 12:27:44.231471 [DEBUG] switch_core_session.c:932 Send signal
sofia/external/005622170039 [BREAK]
2009-08-07 12:27:44.231471 [DEBUG] switch_ivr_originate.c:2134 Originate
Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
2009-08-07 12:27:44.231471 [DEBUG] switch_core_state_machine.c:398
(sofia/external/00569xxxxxxx) Running State Change CS_HANGUP
2009-08-07 12:27:44.231471 [INFO] mod_dptools.c:2092 Originate Failed.
Cause: ORIGINATOR_CANCEL
2009-08-07 12:27:44.231471 [NOTICE] c2c.js:1 *********** Leg2:
NORMAL_CLEARING ***********


The second to last line comes from the script, and prints the hangup_cause
of he session, instead of getting ORIGINATOR_CANCEL, I'm getting
NORMAL_CLEARING. Where is the ORIGINATOR_CANCEL value set?


Thanks!

Nicolas

On Thu, Aug 6, 2009 at 3:45 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi Matt,
>
> Actually I'm explicitly setting hangup_after_bridge to true, think setting
> it to false would help? I'm going to try that.
>
> Here's the JS code:
> (Note: session.getVariable() doesn't work, FS complains saying it is not a
> function, also tried self.session.getVariable() - that's what the wiki says
> - and FS complains that self does not exist)
>
> ----------------
> var uuid = argv[0]; // Call identifier
> var dialstr1 = argv[1]; // Dial string obtained from previous call to LCR
> var dialstr2 = argv[2]; // Dial string obtained from previous call to LCR
> var greeting_snd = "/var/audio/alert.wav";
>
> console_log("notice", "*********** STARTING C2C Call ***********\n");
> timeout = 30;
>
> console_log("notice", "*********** DIALING "+dialstr1+" ***********\n");
>
> //var stUsRing = session.getVariable("us-ring");  // This doesn't work,
> self.session.getVariable doesn't work either
> var stUsRing = "%(2000,4000,440,480)";
>
> // Create new_session
> new_session = new Session(originate_str1);
> console_log("notice", "*********** Leg1: " + new_session.cause + "
> ***********\n");
>
> if (new_session.ready()) {
>         // log to the console
>         console_log("notice", "*********** Leg1 ("+dialstr1+") CONNECTED!
> ***********\n");
>         console_log("notice", "*********** Playing greeting sound:
> "+greeting_snd+" ***********\n");
>
>         new_session.execute("sleep", 100);
>         new_session.execute("playback", greeting_snd);
>
>         // Originate second call and bridge
>     originate_str2 =
> "{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;
>
>         // Create new_session
>         new_session.execute("bridge", originate_str2);
>         console_log("notice", "*********** Leg2: " + new_session.cause + "
> ***********\n");
>
>         if (new_session.ready()) {
>                 console_log("notice", "*********** Leg2 ("+dialstr2+")
> CONNECTED! ***********\n");
>         }
> }
>
> exit();
> ----------------
>
> Thanks!
>
>
> Nicolas
>
>
>
> On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong <mattdfong at gmail.com> wrote:
>
>> Hi Nicolas,
>> do you have a copy of the .js code you can paste. I would guess tho, that
>> ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to
>> false. Just a guess tho.
>>
>> Hangup causes can be found here:
>> http://wiki.freeswitch.org/wiki/Hangup_causes
>>
>> <http://wiki.freeswitch.org/wiki/Hangup_causes> --matt
>> hello hunter - hosted predictive dialer & voice broadcasting
>> http://www.hellohunter.com
>>
>>
>> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com>wrote:
>>
>>> I'm bridging 2 calls in a javascript file, I originate the first call and
>>> then execute a bridge with an origination string for the second call. If I
>>> hangup the first call while trying to make the second call, I get this on
>>> the console:
>>>
>>> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
>>> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
>>> sofia/external/005622170039 [KILL]
>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal
>>> sofia/external/005622170039 [BREAK]
>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate
>>> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>>> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
>>> Cause: ORIGINATOR_CANCEL
>>>
>>> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see
>>> NORMAL_CLEARING. And the variable_originate_disposition has a value of
>>> "failure". Where can I get the detail of the call/bridge failure due to
>>> 'ORIGINATOR_CANCEL' as reported through the console?
>>>
>>> Thanks!
>>>
>>> Nicolas
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From pjintheusa at gmail.com  Fri Aug  7 10:28:23 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 7 Aug 2009 13:28:23 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <1b46b4e80908070943j6e55c125xde064029513b099c@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
	<4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
	<1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>
	<1b46b4e80908070943j6e55c125xde064029513b099c@mail.gmail.com>
Message-ID: <367751820908071028q7075b710hb0d6eed8c1d4dc54@mail.gmail.com>

What does

bridge_hangup_cause

give you?

On Fri, Aug 7, 2009 at 12:43 PM, Nicolas Brenner <nicolas at medularis.com> wrote:
>
> I changed the script to set hangup_after_bridge to false, but still the same thing happens, I get this on the console:
>
> 2009-08-07 12:27:44.229091 [NOTICE] sofia.c:322 Hangup sofia/external/00569xxxxxxx [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> 2009-08-07 12:27:44.229091 [DEBUG] switch_channel.c:1683 Send signal sofia/external/00569xxxxxxx [KILL]
> 2009-08-07 12:27:44.229091 [DEBUG] switch_core_session.c:932 Send signal sofia/external/00569xxxxxxx [BREAK]
> 2009-08-07 12:27:44.231471 [NOTICE] switch_ivr_originate.c:1994 Hangup sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_channel.c:1683 Send signal sofia/external/005622170039 [KILL]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_core_session.c:932 Send signal sofia/external/005622170039 [BREAK]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_ivr_originate.c:2134 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> 2009-08-07 12:27:44.231471 [DEBUG] switch_core_state_machine.c:398 (sofia/external/00569xxxxxxx) Running State Change CS_HANGUP
> 2009-08-07 12:27:44.231471 [INFO] mod_dptools.c:2092 Originate Failed.? Cause: ORIGINATOR_CANCEL
> 2009-08-07 12:27:44.231471 [NOTICE] c2c.js:1 *********** Leg2: NORMAL_CLEARING ***********
>
>
> The second to last line comes from the script, and prints the hangup_cause of he session, instead of getting ORIGINATOR_CANCEL, I'm getting NORMAL_CLEARING. Where is the ORIGINATOR_CANCEL value set?
>
>
> Thanks!
>
> Nicolas
>
> On Thu, Aug 6, 2009 at 3:45 PM, Nicolas Brenner <nicolas at medularis.com> wrote:
>>
>> Hi Matt,
>>
>> Actually I'm explicitly setting hangup_after_bridge to true, think setting it to false would help? I'm going to try that.
>>
>> Here's the JS code:
>> (Note: session.getVariable() doesn't work, FS complains saying it is not a function, also tried self.session.getVariable() - that's what the wiki says - and FS complains that self does not exist)
>>
>> ----------------
>> var uuid = argv[0]; // Call identifier
>> var dialstr1 = argv[1]; // Dial string obtained from previous call to LCR
>> var dialstr2 = argv[2]; // Dial string obtained from previous call to LCR
>> var greeting_snd = "/var/audio/alert.wav";
>>
>> console_log("notice", "*********** STARTING C2C Call ***********\n");
>> timeout = 30;
>>
>> console_log("notice", "*********** DIALING "+dialstr1+" ***********\n");
>>
>> //var stUsRing = session.getVariable("us-ring");? // This doesn't work, self.session.getVariable doesn't work either
>> var stUsRing = "%(2000,4000,440,480)";
>>
>> // Create new_session
>> new_session = new Session(originate_str1);
>> console_log("notice", "*********** Leg1: " + new_session.cause + " ***********\n");
>>
>> if (new_session.ready()) {
>> ??????? // log to the console
>> ??????? console_log("notice", "*********** Leg1 ("+dialstr1+") CONNECTED! ***********\n");
>> ??????? console_log("notice", "*********** Playing greeting sound: "+greeting_snd+" ***********\n");
>>
>> ??????? new_session.execute("sleep", 100);
>> ??????? new_session.execute("playback", greeting_snd);
>>
>> ??????? // Originate second call and bridge
>> ??? originate_str2 = "{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;
>>
>> ??????? // Create new_session
>> ??????? new_session.execute("bridge", originate_str2);
>> ??????? console_log("notice", "*********** Leg2: " + new_session.cause + " ***********\n");
>>
>> ??????? if (new_session.ready()) {
>> ??????????????? console_log("notice", "*********** Leg2 ("+dialstr2+") CONNECTED! ***********\n");
>> ??????? }
>> }
>>
>> exit();
>> ----------------
>>
>> Thanks!
>>
>>
>> Nicolas
>>
>>
>> On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong <mattdfong at gmail.com> wrote:
>>>
>>> Hi Nicolas,
>>> do you have a copy of the .js code you can paste. I would guess tho, that ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge to false. Just a guess tho.
>>> Hangup causes can be found here:
>>> http://wiki.freeswitch.org/wiki/Hangup_causes
>>> --matt
>>> hello hunter - hosted predictive dialer & voice broadcasting
>>> http://www.hellohunter.com
>>>
>>> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com> wrote:
>>>>
>>>> I'm bridging 2 calls in a javascript file, I originate the first call and then execute a bridge with an origination string for the second call. If I hangup the first call while trying to make the second call, I get this on the console:
>>>>
>>>> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
>>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal sofia/external/005622170039 [KILL]
>>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send signal sofia/external/005622170039 [BREAK]
>>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134 Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
>>>> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.? Cause: ORIGINATOR_CANCEL
>>>>
>>>> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I see NORMAL_CLEARING. And the variable_originate_disposition has a value of "failure". Where can I get the detail of the call/bridge failure due to 'ORIGINATOR_CANCEL' as reported through the console?
>>>>
>>>> Thanks!
>>>>
>>>> Nicolas
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From raffaele.p.guidi at gmail.com  Fri Aug  7 11:50:57 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Fri, 7 Aug 2009 20:50:57 +0200
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>
References: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
	<ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>
Message-ID: <b0a8792f0908071150x68f3e3b5v596bad90bc56d1ad@mail.gmail.com>

Maybe Artem is interested in CTI (computer telephony integration) -
click2dial, opening a url (or statrting a program) on incoming call...?

On Fri, Aug 7, 2009 at 17:00, Kevin Green <kevin at johnnyvoip.com> wrote:

> From what I am aware you can't use FreeSWITCH to control a softphone
> directly though you can make it do things that will have a similar end
> result. You could set eyeBeam to auto-answer calls if you want them to
> answer right away or orginiate a call that is auto-answered but not bridge
> the call until a user on the eyeBeam presses a digit or a socket control
> tells it to connect the two ends. You can also use FreeSWITCH to place the
> line on hold using event sockets, this will place it on hold in the server
> and not directly like placing it on hold in eyeBeam (i.e. the hold button in
> eyeBeam likely wont show it as being on hold).
>
> Beyond that if you want to directly control the clients you would need to
> look at getting an API access into the eyeBeam client.
>
> I hope this will help.
>
> Regards,
>    Kevin Green
>
>
>
> On Fri, Aug 7, 2009 at 7:02 AM, Artem Vasiliev <ryder86 at googlemail.com>wrote:
>
>> No, I don't want to make softphone from FreeSwitch
>>
>> I have FS and several users with eyeBeam softphones. I need to control
>> those eyeBeams
>>
>> >You can run FreeSWITCH as a softphone and control it.
>> >http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>>
>> >2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>
>>
>> >> Hi
>> >>
>> >> I have FreeSwitch and external application, which communicates to it
>> via
>> >> event socket - listens for events for certain number and gives some
>> >> commands.
>> >> Is it possible for this application to control client softphones, for
>> >> example, make them answer or hold, using the event socket or other
>> >> FreeSwitch capabilities?
>> >>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From lfurrea at gmail.com  Fri Aug  7 11:57:55 2009
From: lfurrea at gmail.com (Luis F Urrea)
Date: Fri, 7 Aug 2009 12:57:55 -0600
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
Message-ID: <f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>

On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
edition .

Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for FS
playback.

Here's a guide that has been put together for reference on what to record.

http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml

Regards,



On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:

> I'd like to begin record spanish prompts for FS.
>
> Do you know any software/hardware to make it?
>
> Thank you
>
> BR
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Fri Aug  7 12:10:35 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 7 Aug 2009 15:10:35 -0400
Subject: [Freeswitch-users] Calling multiple destinations with fail over
Message-ID: <367751820908071210y233741fatad5cd451dd64df7@mail.gmail.com>

Hi there,

I am trying to implement a scenario where I can terminate calls to
multiple destinations AND have termination carrier fail over.
Currently I can see how to do one or the other. But not both.

Multiple destinations is easy:

 <action application="bridge"
data="[origination_caller_id_number=1234]sofia/SIP_PROVIDER_1/6095551234,[origination_caller_id_number=55555]sofia/SIP_PROVIDER_1/7325551234"/>

Failover appears to use the same mechanism:

<action application="bridge"
data="sofia/gateway/primary/blah|sofia/gateway/secondary/blah"/>

I can not get my head around how use these together, such that:


- 6095551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 6095551234  through SIP_PROVIDER_2
- Called party does not answer
7325551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 7325551234 through SIP_PROVIDER_2
- Called party answers


This must be a fairly comment requirement so any ideas on what I might
be missing would be very welcome.

Thanks


Phillip Jones



From jmesquita at gmail.com  Fri Aug  7 13:50:32 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 7 Aug 2009 17:50:32 -0300
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <b0a8792f0908071150x68f3e3b5v596bad90bc56d1ad@mail.gmail.com>
References: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
	<ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>
	<b0a8792f0908071150x68f3e3b5v596bad90bc56d1ad@mail.gmail.com>
Message-ID: <5a8712120908071350v2eac4613j7fd53e5158680742@mail.gmail.com>

Stay tuned on fsgui. It will get there really soon.

jmesquita

On Fri, Aug 7, 2009 at 3:50 PM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> Maybe Artem is interested in CTI (computer telephony integration) -
> click2dial, opening a url (or statrting a program) on incoming call...?
>
>
> On Fri, Aug 7, 2009 at 17:00, Kevin Green <kevin at johnnyvoip.com> wrote:
>
>> From what I am aware you can't use FreeSWITCH to control a softphone
>> directly though you can make it do things that will have a similar end
>> result. You could set eyeBeam to auto-answer calls if you want them to
>> answer right away or orginiate a call that is auto-answered but not bridge
>> the call until a user on the eyeBeam presses a digit or a socket control
>> tells it to connect the two ends. You can also use FreeSWITCH to place the
>> line on hold using event sockets, this will place it on hold in the server
>> and not directly like placing it on hold in eyeBeam (i.e. the hold button in
>> eyeBeam likely wont show it as being on hold).
>>
>> Beyond that if you want to directly control the clients you would need to
>> look at getting an API access into the eyeBeam client.
>>
>> I hope this will help.
>>
>> Regards,
>>     Kevin Green
>>
>>
>>
>> On Fri, Aug 7, 2009 at 7:02 AM, Artem Vasiliev <ryder86 at googlemail.com>wrote:
>>
>>> No, I don't want to make softphone from FreeSwitch
>>>
>>> I have FS and several users with eyeBeam softphones. I need to control
>>> those eyeBeams
>>>
>>> >You can run FreeSWITCH as a softphone and control it.
>>> >http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>>>
>>> >2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>
>>>
>>> >> Hi
>>> >>
>>> >> I have FreeSwitch and external application, which communicates to it
>>> via
>>> >> event socket - listens for events for certain number and gives some
>>> >> commands.
>>> >> Is it possible for this application to control client softphones, for
>>> >> example, make them answer or hold, using the event socket or other
>>> >> FreeSwitch capabilities?
>>> >>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From nicolas at medularis.com  Fri Aug  7 14:20:29 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 7 Aug 2009 17:20:29 -0400
Subject: [Freeswitch-users] Which event contains ORIGINATOR_CANCEL?
In-Reply-To: <367751820908071028q7075b710hb0d6eed8c1d4dc54@mail.gmail.com>
References: <1b46b4e80908060938v6d46c689sd91ba46b2cf9af6e@mail.gmail.com>
	<4256bf830908061125n4ae662c1q291d5c1af9774dd2@mail.gmail.com>
	<1b46b4e80908061245w1942c189o3482bed66d18d603@mail.gmail.com>
	<1b46b4e80908070943j6e55c125xde064029513b099c@mail.gmail.com>
	<367751820908071028q7075b710hb0d6eed8c1d4dc54@mail.gmail.com>
Message-ID: <1b46b4e80908071420v42c3f73wfeddacb8af3d7067@mail.gmail.com>

That variable is not available, it is not included with the
CHANNEL_HANGUP_COMPLETE event info. However I discovered that when the
bridge does not work, there are two CHANNEL_HANGUP_COMPLETE events, one for
each leg, nevertheless for some reason the daemon I have watching the events
misses the second leg event, so I was only seeing the result of the first
leg hangup, which is NORMAL_CLEARING, and the second event's hangup_cause is
ORIGINATOR_CANCEL. I don't know why my daemon is missing the event though.
I'll have to dig into this further.


On Fri, Aug 7, 2009 at 1:28 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> What does
>
> bridge_hangup_cause
>
> give you?
>
> On Fri, Aug 7, 2009 at 12:43 PM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
> >
> > I changed the script to set hangup_after_bridge to false, but still the
> same thing happens, I get this on the console:
> >
> > 2009-08-07 12:27:44.229091 [NOTICE] sofia.c:322 Hangup
> sofia/external/00569xxxxxxx [CS_SOFT_EXECUTE] [NORMAL_CLEARING]
> > 2009-08-07 12:27:44.229091 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/00569xxxxxxx [KILL]
> > 2009-08-07 12:27:44.229091 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/00569xxxxxxx [BREAK]
> > 2009-08-07 12:27:44.231471 [NOTICE] switch_ivr_originate.c:1994 Hangup
> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> > 2009-08-07 12:27:44.231471 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/005622170039 [KILL]
> > 2009-08-07 12:27:44.231471 [DEBUG] switch_core_session.c:932 Send signal
> sofia/external/005622170039 [BREAK]
> > 2009-08-07 12:27:44.231471 [DEBUG] switch_ivr_originate.c:2134 Originate
> Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> > 2009-08-07 12:27:44.231471 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/00569xxxxxxx) Running State Change CS_HANGUP
> > 2009-08-07 12:27:44.231471 [INFO] mod_dptools.c:2092 Originate Failed.
> Cause: ORIGINATOR_CANCEL
> > 2009-08-07 12:27:44.231471 [NOTICE] c2c.js:1 *********** Leg2:
> NORMAL_CLEARING ***********
> >
> >
> > The second to last line comes from the script, and prints the
> hangup_cause of he session, instead of getting ORIGINATOR_CANCEL, I'm
> getting NORMAL_CLEARING. Where is the ORIGINATOR_CANCEL value set?
> >
> >
> > Thanks!
> >
> > Nicolas
> >
> > On Thu, Aug 6, 2009 at 3:45 PM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
> >>
> >> Hi Matt,
> >>
> >> Actually I'm explicitly setting hangup_after_bridge to true, think
> setting it to false would help? I'm going to try that.
> >>
> >> Here's the JS code:
> >> (Note: session.getVariable() doesn't work, FS complains saying it is not
> a function, also tried self.session.getVariable() - that's what the wiki
> says - and FS complains that self does not exist)
> >>
> >> ----------------
> >> var uuid = argv[0]; // Call identifier
> >> var dialstr1 = argv[1]; // Dial string obtained from previous call to
> LCR
> >> var dialstr2 = argv[2]; // Dial string obtained from previous call to
> LCR
> >> var greeting_snd = "/var/audio/alert.wav";
> >>
> >> console_log("notice", "*********** STARTING C2C Call ***********\n");
> >> timeout = 30;
> >>
> >> console_log("notice", "*********** DIALING "+dialstr1+" ***********\n");
> >>
> >> //var stUsRing = session.getVariable("us-ring");  // This doesn't work,
> self.session.getVariable doesn't work either
> >> var stUsRing = "%(2000,4000,440,480)";
> >>
> >> // Create new_session
> >> new_session = new Session(originate_str1);
> >> console_log("notice", "*********** Leg1: " + new_session.cause + "
> ***********\n");
> >>
> >> if (new_session.ready()) {
> >>         // log to the console
> >>         console_log("notice", "*********** Leg1 ("+dialstr1+")
> CONNECTED! ***********\n");
> >>         console_log("notice", "*********** Playing greeting sound:
> "+greeting_snd+" ***********\n");
> >>
> >>         new_session.execute("sleep", 100);
> >>         new_session.execute("playback", greeting_snd);
> >>
> >>         // Originate second call and bridge
> >>     originate_str2 =
> "{ignore_early_media=true,originate_timeout="+timeout+",hangup_after_bridge=true,medularis_uuid="+uuid+",c2c_call=true,leg=2}"+dialstr2;
> >>
> >>         // Create new_session
> >>         new_session.execute("bridge", originate_str2);
> >>         console_log("notice", "*********** Leg2: " + new_session.cause +
> " ***********\n");
> >>
> >>         if (new_session.ready()) {
> >>                 console_log("notice", "*********** Leg2 ("+dialstr2+")
> CONNECTED! ***********\n");
> >>         }
> >> }
> >>
> >> exit();
> >> ----------------
> >>
> >> Thanks!
> >>
> >>
> >> Nicolas
> >>
> >>
> >> On Thu, Aug 6, 2009 at 2:25 PM, Matthew Fong <mattdfong at gmail.com>
> wrote:
> >>>
> >>> Hi Nicolas,
> >>> do you have a copy of the .js code you can paste. I would guess tho,
> that ORIGINATOR_CANCLE might be related to not setting hangup_after_bridge
> to false. Just a guess tho.
> >>> Hangup causes can be found here:
> >>> http://wiki.freeswitch.org/wiki/Hangup_causes
> >>> --matt
> >>> hello hunter - hosted predictive dialer & voice broadcasting
> >>> http://www.hellohunter.com
> >>>
> >>> On Thu, Aug 6, 2009 at 9:38 AM, Nicolas Brenner <nicolas at medularis.com>
> wrote:
> >>>>
> >>>> I'm bridging 2 calls in a javascript file, I originate the first call
> and then execute a bridge with an origination string for the second call. If
> I hangup the first call while trying to make the second call, I get this on
> the console:
> >>>>
> >>>> 2009-08-05 16:44:05.69122 [NOTICE] switch_ivr_originate.c:1994 Hangup
> sofia/external/005622170039 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL]
> >>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/005622170039 [KILL]
> >>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/005622170039 [BREAK]
> >>>> 2009-08-05 16:44:05.69122 [DEBUG] switch_ivr_originate.c:2134
> Originate Cancelled by originator termination Cause: 487 [ORIGINATOR_CANCEL]
> >>>> 2009-08-05 16:44:05.69122 [INFO] mod_dptools.c:2092 Originate Failed.
> Cause: ORIGINATOR_CANCEL
> >>>>
> >>>> But if I check hangup_cause in the CHANNEL_HANGUP_COMPLETE event, I
> see NORMAL_CLEARING. And the variable_originate_disposition has a value of
> "failure". Where can I get the detail of the call/bridge failure due to
> 'ORIGINATOR_CANCEL' as reported through the console?
> >>>>
> >>>> Thanks!
> >>>>
> >>>> Nicolas
> >>>>
> >>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>>
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nicolas at medularis.com  Fri Aug  7 15:10:25 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 7 Aug 2009 18:10:25 -0400
Subject: [Freeswitch-users] Error trying to use PHP ESL
Message-ID: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>

Hi,

I'm trying to get started with the ESL using PHP. I compiled the ESL, then
phpmod according to the wiki instructions, but then when I try the examples
in the libs/esl/php dir, they fail saying:

PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
/usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132

Checking ESL.php on line 132, I see there are several different declarations
for the function __construct() with different parameters each, which makes
sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
higher than that or something? What could be the problem?

Thanks!

Nicolas
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From dave at 3c.co.uk  Fri Aug  7 15:54:08 2009
From: dave at 3c.co.uk (David Knell)
Date: Fri, 07 Aug 2009 17:54:08 -0500
Subject: [Freeswitch-users] Cluecon 2009
Message-ID: <1249685648.16901.34.camel@dk-d820>

Just a quick note to say thanks to Cluecon's organisers for putting
together such a useful, informative and packed three days.  I've come
away with a head full of ideas, a bunch of new contacts and a collection
of things to do; I'd thoroughly recommend that anyone interested in IP
telephony blocks out the first week of August 2010, right now..!

Cheers --

Dave

-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk




From gcd at i.ph  Fri Aug  7 16:44:23 2009
From: gcd at i.ph (Nandy Dagondon)
Date: Sat, 8 Aug 2009 07:44:23 +0800
Subject: [Freeswitch-users] /etc/openzap/tones.conf for UK
In-Reply-To: <599DBC67-02FE-4F9E-9F87-6D1749B81B11@mac.com>
References: <599DBC67-02FE-4F9E-9F87-6D1749B81B11@mac.com>
Message-ID: <7d0bfd8c0908071644n2933a91y996745e9f894cf90@mail.gmail.com>

you can create your tones.conf using call progress tones  found at
http://www.3amsystems.com/wireline/tone-search.htm


On Fri, Aug 7, 2009 at 7:17 PM, Merul Patel <merul at mac.com> wrote:

> Where can I find a sample tones.conf file for the UK? Am trying to
> configure a USBFXO device for outbound calls.
>
> Thanks in advance,
>
> Merul
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Fri Aug  7 16:44:41 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Fri, 7 Aug 2009 19:44:41 -0400
Subject: [Freeswitch-users] State of originated call
Message-ID: <c55d46950908071644m7604eb57jf5018c2b378d3255@mail.gmail.com>

Hi,

using javascript, i do originate the call this way:

Session s= new Session(originateStr);

>From this point, is it possible to know what states the call is going
through? In a previous message it was suggested that
variable_originate_disposition would give me the response code. Now, how to
i use this in practice in a script?
How do i for instance retrieve a 180 response code when rining is hapening
on the remote end?

Thanks,
Max.
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From nicolas at medularis.com  Fri Aug  7 17:23:51 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Fri, 7 Aug 2009 20:23:51 -0400
Subject: [Freeswitch-users] Best practices / tips for Event socket daemon
Message-ID: <1b46b4e80908071723m18bf3bcnb2505f0f760b907c@mail.gmail.com>

Hi, I built an event socket daemon that waits for certain events, when it
receives those events, it does some processing and keeps waiting for more
events. The daemon is written on PHP and uses a slightly modified version of
fs_sock.php (from contrib/intralanman/PHP/fs_sock/).

What I am doing / what I want to do: I am generating calls and bridging them
using a JS script. Then the daemon logs the info about the calls and keeps
track of their status in a database.

The problem is: the daemon is apparently missing out on some events, and I
think it is because of the processing/updating on the DB it has to do each
time it "catches" an event on the socket.

My question is: which language would you recommend for the task, and how
would you go about handling events? Should the dameon fork a process for
each event it receives so that it doesn't miss any events? should there be
more than one daemon? ... any tips and recommendations are welcome.

Thanks!

Nicolas
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From rehan at supertec.com  Fri Aug  7 19:21:36 2009
From: rehan at supertec.com (Rehan Ahmed Allahwala)
Date: Sat, 8 Aug 2009 10:21:36 +0800
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <1249685648.16901.34.camel@dk-d820>
References: <1249685648.16901.34.camel@dk-d820>
Message-ID: <865f01c80908071921q13b4f98bxea7426b720a69e83@mail.gmail.com>

Hi all, hope u got ur hands on our mobile pouches from didx.net and
got ur snaps with suzanne for our blogs and new upcoming magazine

Thanks ,

Rehan

On 8/8/09, David Knell <dave at 3c.co.uk> wrote:
> Just a quick note to say thanks to Cluecon's organisers for putting
> together such a useful, informative and packed three days.  I've come
> away with a head full of ideas, a bunch of new contacts and a collection
> of things to do; I'd thoroughly recommend that anyone interested in IP
> telephony blocks out the first week of August 2010, right now..!
>
> Cheers --
>
> Dave
>
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device



From vladrodionov at gmail.com  Fri Aug  7 19:42:38 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Fri, 7 Aug 2009 19:42:38 -0700
Subject: [Freeswitch-users] Best practices / tips for Event socket daemon
In-Reply-To: <1b46b4e80908071723m18bf3bcnb2505f0f760b907c@mail.gmail.com>
References: <1b46b4e80908071723m18bf3bcnb2505f0f760b907c@mail.gmail.com>
Message-ID: <3c233920908071942g522df080g63b2abc31efd6fcc@mail.gmail.com>

Forking process on every incoming event is terrible idea IMO. Threads are
more lightweight than processes.
Can you use threads in PHP? I am not familiar with PHP (Java developer
myself).  I can explain how I would implement it in Java.

There is one SocketReader thread and several Worker threads in a thread
pool.

"SocketReader" thread - reads data (events) from socket. When event arrives
SocketReader checks thread pool, get one Worker (if any) and makes it to
process event. If there no available Workers in a pool then event goes
directly to a EventQueue. When Worker finishes it checks EventQueue and if
there are no events in a queue Worker goes back to thread pool, otherwise it
process event from queue.

-Vladimir Rodionov



On Fri, Aug 7, 2009 at 5:23 PM, Nicolas Brenner <nicolas at medularis.com>wrote:

> Hi, I built an event socket daemon that waits for certain events, when it
> receives those events, it does some processing and keeps waiting for more
> events. The daemon is written on PHP and uses a slightly modified version of
> fs_sock.php (from contrib/intralanman/PHP/fs_sock/).
>
> What I am doing / what I want to do: I am generating calls and bridging
> them using a JS script. Then the daemon logs the info about the calls and
> keeps track of their status in a database.
>
> The problem is: the daemon is apparently missing out on some events, and I
> think it is because of the processing/updating on the DB it has to do each
> time it "catches" an event on the socket.
>
> My question is: which language would you recommend for the task, and how
> would you go about handling events? Should the dameon fork a process for
> each event it receives so that it doesn't miss any events? should there be
> more than one daemon? ... any tips and recommendations are welcome.
>
> Thanks!
>
> Nicolas
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From andrew at hijacked.us  Fri Aug  7 19:56:14 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Fri, 7 Aug 2009 22:56:14 -0400
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
Message-ID: <20090808025613.GA19871@hijacked.us>

On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
> Hi,
> 
> I'm trying to get started with the ESL using PHP. I compiled the ESL, then
> phpmod according to the wiki instructions, but then when I try the examples
> in the libs/esl/php dir, they fail saying:
> 
> PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
> /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132
> 
> Checking ESL.php on line 132, I see there are several different declarations
> for the function __construct() with different parameters each, which makes
> sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
> higher than that or something? What could be the problem?
>

Someone in the IRC channel mentioned this too. I looked at it briefly
and it looks like the latest 'swigall' screwed it up. The original
reporter said he'd file a jira, but you may want to check yourself and
if not make one yourself. In the meantime, the previous version of the
file was reported to work if you really need it.

Andrew



From neffs1 at gmail.com  Fri Aug  7 07:02:13 2009
From: neffs1 at gmail.com (David Kreitschmann)
Date: Fri, 7 Aug 2009 16:02:13 +0200
Subject: [Freeswitch-users] Compact, fanless appliance?
In-Reply-To: <7d0bfd8c0907051621u1c3f553fh96a9df8557952e47@mail.gmail.com>
References: <7d0bfd8c0907041941s75e541dex982fae2195858ea4@mail.gmail.com>
	<C6757EE0.5891C%krice@suspicious.org>
	<7d0bfd8c0907051621u1c3f553fh96a9df8557952e47@mail.gmail.com>
Message-ID: <149B556F-65B6-40F7-802B-92B81447C24F@gmail.com>

I am thinking about using this combination as a router, should also be  
a nice platform for freeswitch

http://www.intel.com/products/desktop/motherboards/D945GSEJT/D945GSEJT-overview.htm
http://www.cartft.com/catalog/il/1058
http://www.cartft.com/catalog/il/1087

energy efficient netbook chipset (most other boards use the desktop  
version), PCI slot, small case, fanless.
you can put the system on an usb stick and just hide it behind the  
front panel. no need for reinstall on the device itself, just put in a  
another usb stick and you're good to go.

or you put in two disks in raid1 for redundancy if you need the disk  
space.

if you don't need PCI you can use this enclosure
http://www.cartft.com/catalog/il/1081


David

Am 06.07.2009 um 01:21 schrieb Nandy Dagondon:

> ok. w/ my apologies. - nandy
>
>
> On Sun, Jul 5, 2009 at 10:49 AM, Ken Rice <krice at suspicious.org>  
> wrote:
> No need to bump these things as this is a mailing list and it annoys  
> quite a few people when you do that
>
>
> From: Nandy Dagondon <gcd at i.ph>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Sun, 5 Jul 2009 10:41:18 +0800
>
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] Compact, fanless appliance?
>
> just bumping this topic.
> -nandy
>
> On Fri, May 8, 2009 at 12:44 AM, Fred-145 <codecomplete at free.fr>  
> wrote:
>
>
> Antonio Gallo wrote:
> > Alix cases are like 6/9 ? from their shop site. I think its easy  
> to find
> > someone who work with aluminium that can make for you custom boxes  
> for
> > like like 6/20 ? at pcs
>
> Unfortunately, none of the PCEngines cases (www.pcengines.ch/order1.php?c=2 
>  <http://www.pcengines.ch/order1.php?c=2> )
>
> allow for a PCI slot, either on top of the mobo, or away from it :-/
>
> I'll see if I can get those from Soekris (http://soekris.eu/shop/cases_en/ 
> )
> allow this, and if I can get a good price for a case + PSU.
>
> Thank you.
> --
> View this message in context: http://www.nabble.com/Compact%2C-fanless-appliance--tp23193738p23430873.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From fraunhofer.lists.freeswitch-001 at traced.net  Thu Aug  6 13:28:19 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Thu, 6 Aug 2009 22:28:19 +0200
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
Message-ID: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>

Hello List, Hello *,

First of all the usual excuses: sorry for the bad english and the long
email, no native speaker and i really tried to make it shorter, but i
guess this would result in even more "check back"s than it already
does :)

we're currently running in a weird "lockup"-scenario in our loadtests.

Our setup is the following:

three freeswitch servers, let's call them A(-leg), M(aster), and
B(-leg) with the goal in mind to initiate calls on M which calls A,
play some file, bridge to B, limit call length and play (different)
prompts to A and B if they exceed that limit.

(Note that A and B work fine, regardless of the amount of load we put on them)
A and B are silly dialplan logic, accepting calls on a certain
extension after a random delay and playing moh. Before calling
playback to a localstream they call a lua script which schedules
hangup somewhere in future (which works flawlessly)

Calls are initiated on M using some hacked up loadgen-script issuing
http requests like
? originate [sofiaSyntaxToExtensionOn_A] 6000
. The 6000 extension on M has the following (xml) dialplan which
essentially does the following:
------
answer()
...playback file...
...set some callerid stuff
set bypass_media
bridge to extension 6009 on B
------
we use "execute_on_answer" on the b-leg to run a script which limits
the length of the call (doesn't matter if it's done via "export
nolocal" or "inlined" into the data part of the bridge application
"{execute_on_answer=lua ...}")

<action application="export" data="nolocal:execute_on_answer=lua
lua/schedule-hangup.lua ${uuid}" />

the lua script "schedula-hangup.lua" does essentially the following:

------
api = freeswitch.API();
local res = api:execute("sched_api", "+10 none lua
lua/c2c-hangup-timeout.lua " .. argv[1]);
------

the 10 seconds are just to speed up the time until it gets stuck.

this is where things start to go wrong. if I comment out the call to
the "schedule-hangup" script, everything works fine, even if it's
under heavy load.

c2c-hangup-timeout.lua does the following:
------------------
local sess = argv[1];
if(sess)
then
? ?freeswitch.consoleLog("INFO", "c2c-hangup-timeout.lua for uuid "
.. sess .. "\n");

? ?api = freeswitch.API();
? ?local stillValid = api:execute("uuid_getvar", sess .. "
Dummy-DoesChannelExists");
? ?if(stillValid:sub(1,4) == "-ERR")
? ?then
? ? ? ?log("session uuid " .. sess .. " disappeared (nothing bad)");
? ?else
? ? ? ?-- this is important!!! Otherwise the aleg get's just hung up!
? ? ? ?api:execute("uuid_media", sess);
? ? ? ?api:execute("uuid_transfer", sess .. " -both timeout");
? ?end
else -- /if(sess)
? ?log("called with nil session?");
end -- /if(sess)

------------------

i guess this needs some explanation:
we get the uuid of the channel as argument in argv[1]. We don't use
? local session = freeswitch.Session(uuid);
since if the channel referenced by "uuid" does not exist any longer,
freeswitch (or the lua bindings) try to interpret the uuid as an
"originate string" and can't figure out how to call that. So we use a
dummy api call to get some channel variable. If the channel does not
exist any longer (A or B already hung up), we get an error message
starting with "-ERR", otherwise the channel still exists (we get
"_unset_" as the value, if it's not set) and we continue by getting
freeswitch back in the media path (uuid_media) and then transferring
both legs to an extension called "timeout" which plays some prompt and
finally calls hangup().

If we don't do the uuid_media call, one of the legs gets hung up when
we transfer them to the extension. This looks like the following on
the console after issuing "uuid_transfer [uuid] -both timeout"
(extensions are not the same as in our loadgen example above)


--------------
2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1334 Hangup (*)
sofia/internal/1000 [CS_HIBERNATE] [BLIND_TRANSFER]
2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer
sofia/internal/1004 at 192.168.179.177:5060 to XML[timeout at default]
2009-07-23 19:57:19.865703 [INFO] mod_dialplan_xml.c:310 Processing
BFR1004->timeout in context default
API CALL [uuid_transfer(73812082-77b1-11de-b9f8-a10bb0eb9f69 -both
timeout)] output:
+OK

2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer (**)
sofia/internal/1000 to XML[timeout at default]
2009-07-23 19:57:19.865703 [NOTICE] switch_core_session.c:1084 Session
60 (sofia/internal/1000) Ended
2009-07-23 19:57:19.865703 [NOTICE] switch_core_session.c:1086 Close
Channel sofia/internal/1000 [CS_DESTROY]
-----------

note that it first does Hangup (denoted by *, no that's not an
asterisk :) on extension 1000 and then tries to Transfer (**) the hung
up channel to the dial plan. this could be the same as in an earlier
post to the list "SIP re-invite / bypass_media // Phillip Jones //
Wed, 01 Jul 2009 13:30:53 -0700)"

This is why we do not directly call sched_transfer() but call a script
in between to do the uuid_media() call. I couldn't figure out how to
call that directly from the xml dialplan and/or how to check if the
channel still exists.

so... after using uuid_media(), both legs are transferred without an
(intermediate|bogus) hangup() call.

This only works fine if we've few concurrent calls. There is no magic
borderline where it starts to refuse work.

Some of the Symptoms are: traffic decreased to zero as no new channels
are successfully brought up, some of the signaling traffic is not
ACKed or OKed, scheduled jobs are not run.

if i read the output of "show channels" correctly, they're all stuck
in different applications like hangup(), some are calling lua but most
of them are in signaling_bridge(). Freeswitch is still responding on
the console and there's almost no load on the machine (no busy polling
or some other kind of running amok).

if i kill one of them using uuid_kill() or kill all of them
using"fsctl hupall" i get "Task was executed late by 866 seconds 12379
sched_api_function (none)" messages and the usual cleanup takes place.
As a quick hack i tried to schedule a uuid_kill() call 20 seconds
after the scheduling call to the lua script but that job is not
executed either.

So what am I doing wrong? Is it some deadlock where uuid_media() and
uuid_transfer() ?are waiting for the other to finish?
Or some other silly simple thing i missed?

Thx in advance

?Benedikt.



From email.list.subscriber at gmail.com  Thu Aug  6 15:03:56 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Thu, 6 Aug 2009 18:03:56 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
Message-ID: <4a7b530a.29578c0a.53a8.0450@mx.google.com>

Hello,

Does anyone have, or know where to get, a pre-compiled copy of
FreeSwitch for Solaris 10/x86?

# uname -a
SunOS hrndvsoi-zm01 5.10 Generic_137112-07 i86pc i386 i86pc


I'm stuck trying to run make/gmake/\/opt/gnu/bin/make:

<snip>
"make" results in:
make: Fatal error: Command failed for target `all-recursive'
Current working directory /home/vmorales/freeswitch
*** Error code 1
make: Fatal error: Command failed for target `all'
</snip>


"gmake" & "/opt/gnu/bin/make" result in:

<snip>
creating libfreeswitch.la
(cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la
libfreeswitch.la)
gcc -I/home/vmorales/freeswitch/src/include
-I/home/vmorales/freeswitch/libs/libteletone/src -fPIC -Werror -g
-ggdb -DPATH_MAX=2048 -g -m32 -I/usr/sfw/include -Wall -std=c99
-pedantic -m32 -o .libs/freeswitch freeswitch-switch.o  -L/usr/sfw/lib
-lm ./.libs/libfreeswitch.so
/home/vmorales/freeswitch/libs/apr/.libs/libapr-1.a
-L/home/vmorales/freeswitch/libs/srtp /usr/sfw/lib/libstdc++.so
libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt -ldl -lnsl -lpthread
libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lcurses -lsocket
-R/home/vmorales/freeswitch-build/lib -R/usr/sfw/lib
Undefined                       first referenced
 symbol                             in file
XML_Parse                           ./.libs/libfreeswitch.so
XML_ParserCreate                    ./.libs/libfreeswitch.so
XML_ErrorString                     ./.libs/libfreeswitch.so
herror                              ./.libs/libfreeswitch.so
XML_SetUserData                     ./.libs/libfreeswitch.so
XML_ParserFree                      ./.libs/libfreeswitch.so
XML_GetErrorCode                    ./.libs/libfreeswitch.so
XML_SetCharacterDataHandler         ./.libs/libfreeswitch.so
XML_SetElementHandler               ./.libs/libfreeswitch.so
ld: fatal: Symbol referencing errors. No output written to
.libs/freeswitch
collect2: ld returned 1 exit status
gmake[2]: *** [freeswitch] Error 1
gmake[1]: *** [all-recursive] Error 1
gmake: *** [all] Error 2
</snip>


Thanks in advance for any information provided.

Vladimir




From alan at chandlerfamily.org.uk  Fri Aug  7 01:18:35 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Fri, 07 Aug 2009 09:18:35 +0100
Subject: [Freeswitch-users] New to Freeswitch - some help needed
Message-ID: <4A7BE35B.8010709@chandlerfamily.org.uk>

I apologize, as my first post to this list, that I ask a detailed set of 
questions, but I have spend some time looking at all the docs and can't 
get what I need to do completely sorted in my head.  I am definitely one 
who likes to UNDERSTAND what is happening rather than follow blank 
recipies, so please bear with me as I try understand all the details. I 
do understand about networking, NAT etc - but I am new to SIP/RTP and in 
particular what I think is a double NAT problem


Firstly - what am I trying to achieve:

I am in the UK and have a small home network behind a D-Link DIR-100 
Router/NAT/Firewall one of those machines, running Debian Lenny, acts as 
my main server for everything (and in an earlier incarnation was the 
firewall/router/nat box too - I only say this is because I had all this 
working using Asterisk a year or so ago, but with this important 
difference in configuration).  Many of the ports on the firewall are 
port forwarded to this machine. I have set Freeswitch up on this server 
to act as a small voip pbx for the home - but MORE IMPORTANTLY - to 
enable my daughter from her house to talk to us.  At my house locally I 
have a Linksys PAP2T two phone SIP box - and that is working with 
Freeswitch's default configuration (I set up to be 1000 and 1001 and 
used all the facilities).  I will later add a Linksys SPA 3102 - 
although I DO NOT intend to use its facility to bridge to the normal 
phone network.

My daughter, living in another house, also has a Nat box (unknown - its 
part of her ADSL modem/router/wireless access point) and also has a 
PAP2T which she will connect to the her network.  This will be her phone.

There is a family relation living in Australia who will load up a 
whatever softphone that we tell him to use.  I expect, but don't know, 
that he will behind a NAT box too.

Later, I have some friends in the USA that I might wish to add it too - 
especially so that we can hold some teleconferences.  They will have a 
mixture of Windows and MACs, and I will need to recommend softphone 
clients for them.

I want to set this up as a small private voice network, so anyone can 
ring anyone else.  I will add fancy facilities such as conferencing and 
voicemail later - I just want to get the basics working first.

Secondly

I installed a stun client on my home machine and ran it against 
stun.freeswitch.org.

It reported:-

Primary: Independent Mapping, Independent Filter, preserves ports, no 
hairpin

But I have no idea what this means - I can't find any clear statement 
via googling for it - how this set of answers maps to the different 
types of NAT that might be required to get this to all work.  CAN 
SOMEONE ENLIGHTEN me please.

Thirdly

I have set up a sip profile called "double nat" from the recipe in the 
wiki.  This defines the SIP port to be 5090.

However, what I DO NOT UNDERSTAND is how the PAP2T box in my daughters 
house will initiate a connection to my server.  Presumably, I have to 
port forward 5090 from the nat box to my server.  IS THAT CORRECT?

I also assume I will have to tell her to use STUN (I believe this is an 
option on the PAP2T)

Fourthly

If I understand SIP correctly, it just initiates the session and the two 
end points then communicate directly via RTP.  What I don't understand 
is how does a session transition from SIP to RTP via the connection set 
up in the the first phase (in terms of passing through the NAT boxes). 
In particular WHICH OF THE TEST RESULTS from my stun client indicate it 
will do the right thing.  (I am going to take a laptop to my daughters 
house with a stun client in to test her network this weekend).

Could someone explain please.

  Fifthly

Is there a recommended SIP softphone with all the right facilities (STUN 
support?)  that works on MAC and WINDOWS (I only use linux myself).

Apologies for the length of this.  I am eager to get the answers so I 
can use an opportunity this weekend to get it working.


-- 
Alan Chandler
http://www.chandlerfamily.org.uk




From alan at chandlerfamily.org.uk  Fri Aug  7 05:19:51 2009
From: alan at chandlerfamily.org.uk (alan at chandlerfamily.org.uk)
Date: Fri, 7 Aug 2009 13:19:51 +0100 (BST)
Subject: [Freeswitch-users] Auto Nat
Message-ID: <9f0cf614f2b2ffb9f7a2edae3bf3d2ff.squirrel@webmail.chandlerfamily.org.uk>

I sent my first e-mail to the list this morning (about 4 hours ago) but it
does not seem to have arrived back, even though I have received other,
later posts.

I have another question related to the first (about how to set everything
up in a double nat environment) - so if I see this and not the other, I
will send the first again.

I am currently running the stable version 1.0.3 of freeswitch.  The wiki
page says that auto-nat is introduced at r 13612.  Is this before or after
that revision? (I don't want to have to download and rebuild the entire
thing if I don't have to).







From alan at chandlerfamily.org.uk  Fri Aug  7 06:17:03 2009
From: alan at chandlerfamily.org.uk (alan at chandlerfamily.org.uk)
Date: Fri, 7 Aug 2009 14:17:03 +0100 (BST)
Subject: [Freeswitch-users] 1.0.4 builds 1.0.3_1 debs
Message-ID: <631613604d8c9862f1cf43f80cb99f94.squirrel@webmail.chandlerfamily.org.uk>

I just downloaded 1.0.4 and build debian packages with it, and it
delivered .deb files names as 1.0.3-1





From demuel at thephinix.org  Fri Aug  7 21:04:44 2009
From: demuel at thephinix.org (demuel at thephinix.org)
Date: Sat, 8 Aug 2009 05:04:44 +0100 (BST)
Subject: [Freeswitch-users] New to Freeswitch - some help needed
In-Reply-To: <4A7BE35B.8010709@chandlerfamily.org.uk>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
Message-ID: <1bfed71a616412ee86da3a7f18b990a0.squirrel@www.thephinix.org>

What a long detailed list of todos. You certainly can't find that kind of
answers in here.

> I apologize, as my first post to this list, that I ask a detailed set of
> questions, but I have spend some time looking at all the docs and can't
> get what I need to do completely sorted in my head.  I am definitely one
> who likes to UNDERSTAND what is happening rather than follow blank
> recipies, so please bear with me as I try understand all the details. I
> do understand about networking, NAT etc - but I am new to SIP/RTP and in
> particular what I think is a double NAT problem
>
>
> Firstly - what am I trying to achieve:
>
> I am in the UK and have a small home network behind a D-Link DIR-100
> Router/NAT/Firewall one of those machines, running Debian Lenny, acts as
> my main server for everything (and in an earlier incarnation was the
> firewall/router/nat box too - I only say this is because I had all this
> working using Asterisk a year or so ago, but with this important
> difference in configuration).  Many of the ports on the firewall are
> port forwarded to this machine. I have set Freeswitch up on this server
> to act as a small voip pbx for the home - but MORE IMPORTANTLY - to
> enable my daughter from her house to talk to us.  At my house locally I
> have a Linksys PAP2T two phone SIP box - and that is working with
> Freeswitch's default configuration (I set up to be 1000 and 1001 and
> used all the facilities).  I will later add a Linksys SPA 3102 -
> although I DO NOT intend to use its facility to bridge to the normal
> phone network.
>
> My daughter, living in another house, also has a Nat box (unknown - its
> part of her ADSL modem/router/wireless access point) and also has a
> PAP2T which she will connect to the her network.  This will be her phone.
>
> There is a family relation living in Australia who will load up a
> whatever softphone that we tell him to use.  I expect, but don't know,
> that he will behind a NAT box too.
>
> Later, I have some friends in the USA that I might wish to add it too -
> especially so that we can hold some teleconferences.  They will have a
> mixture of Windows and MACs, and I will need to recommend softphone
> clients for them.
>
> I want to set this up as a small private voice network, so anyone can
> ring anyone else.  I will add fancy facilities such as conferencing and
> voicemail later - I just want to get the basics working first.
>
> Secondly
>
> I installed a stun client on my home machine and ran it against
> stun.freeswitch.org.
>
> It reported:-
>
> Primary: Independent Mapping, Independent Filter, preserves ports, no
> hairpin
>
> But I have no idea what this means - I can't find any clear statement
> via googling for it - how this set of answers maps to the different
> types of NAT that might be required to get this to all work.  CAN
> SOMEONE ENLIGHTEN me please.
>
> Thirdly
>
> I have set up a sip profile called "double nat" from the recipe in the
> wiki.  This defines the SIP port to be 5090.
>
> However, what I DO NOT UNDERSTAND is how the PAP2T box in my daughters
> house will initiate a connection to my server.  Presumably, I have to
> port forward 5090 from the nat box to my server.  IS THAT CORRECT?
>
> I also assume I will have to tell her to use STUN (I believe this is an
> option on the PAP2T)
>
> Fourthly
>
> If I understand SIP correctly, it just initiates the session and the two
> end points then communicate directly via RTP.  What I don't understand
> is how does a session transition from SIP to RTP via the connection set
> up in the the first phase (in terms of passing through the NAT boxes).
> In particular WHICH OF THE TEST RESULTS from my stun client indicate it
> will do the right thing.  (I am going to take a laptop to my daughters
> house with a stun client in to test her network this weekend).
>
> Could someone explain please.
>
>   Fifthly
>
> Is there a recommended SIP softphone with all the right facilities (STUN
> support?)  that works on MAC and WINDOWS (I only use linux myself).
>
> Apologies for the length of this.  I am eager to get the answers so I
> can use an opportunity this weekend to get it working.
>
>
> --
> Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>





From jason at jasonjgw.net  Fri Aug  7 21:23:42 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 8 Aug 2009 14:23:42 +1000
Subject: [Freeswitch-users] New to Freeswitch - some help needed
In-Reply-To: <4A7BE35B.8010709@chandlerfamily.org.uk>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
Message-ID: <20090808042342.GA3558@jdc.jasonjgw.net>

Alan Chandler <alan at chandlerfamily.org.uk> wrote:
 
> I want to set this up as a small private voice network, so anyone can 
> ring anyone else.  I will add fancy facilities such as conferencing and 
> voicemail later - I just want to get the basics working first.

I have a similar arrangement operating here which involves friends and
colleagues in the U.S., as well as a local VoIP provider that gives me access
to the PSTN.

To eliminate NAT issues, we are using IPv6: each of us has an IPv6 over IPv4
tunnel configured to provide access to the IPv6 Internet. NAT and all the
problems associated with it go away.

Another option, although I don't know how well real-time communication works
in this setting, would be to create a VPN using, for example, OpenVPN so that
the clients and server all appear to be on the same lan.

Alternatively, you could play with port forwarding and FreeSWITCH settings in
an attempt to work around the nat issues - good luck!

I can't answer any questions about MacOS or Windows softphones - there are no
MacOS or Windows machines in my life.




From gcd at i.ph  Fri Aug  7 21:33:37 2009
From: gcd at i.ph (Nandy Dagondon)
Date: Sat, 8 Aug 2009 12:33:37 +0800
Subject: [Freeswitch-users] Auto Nat
In-Reply-To: <9f0cf614f2b2ffb9f7a2edae3bf3d2ff.squirrel@webmail.chandlerfamily.org.uk>
References: <9f0cf614f2b2ffb9f7a2edae3bf3d2ff.squirrel@webmail.chandlerfamily.org.uk>
Message-ID: <7d0bfd8c0908072133p7b7f68aancfabdb78d5fe7954@mail.gmail.com>

r 13612 is after 1.0.3. you better get 1.0.4 recently released.

-nandy

On Fri, Aug 7, 2009 at 8:19 PM, <alan at chandlerfamily.org.uk> wrote:

> I sent my first e-mail to the list this morning (about 4 hours ago) but it
> does not seem to have arrived back, even though I have received other,
> later posts.
>
> I have another question related to the first (about how to set everything
> up in a double nat environment) - so if I see this and not the other, I
> will send the first again.
>
> I am currently running the stable version 1.0.3 of freeswitch.  The wiki
> page says that auto-nat is introduced at r 13612.  Is this before or after
> that revision? (I don't want to have to download and rebuild the entire
> thing if I don't have to).
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Fri Aug  7 21:36:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 00:36:58 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
Message-ID: <84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>

This is not currently a supported platform, it only builds on 64 bit  
right now I think on solaris.

Mike

On Aug 6, 2009, at 6:03 PM, vmorales wrote:

> Hello,
>
> Does anyone have, or know where to get, a pre-compiled copy of
> FreeSwitch for Solaris 10/x86?




From mcampbellsmith at gmail.com  Fri Aug  7 21:39:14 2009
From: mcampbellsmith at gmail.com (Mark Campbell-Smith)
Date: Sat, 8 Aug 2009 14:39:14 +1000
Subject: [Freeswitch-users] New to Freeswitch - some help needed
In-Reply-To: <4A7BE35B.8010709@chandlerfamily.org.uk>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
Message-ID: <33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>

Hi Alan,

I hope you find your answers here as these are the sort of things that
are hard to find on the wiki, which is somewhat outdated in areas.  If
you do find your answers, please post them back here for everyone
else.

I am new to FS also, so my comments below may not be 100% correct!

1. Very similar to what I want to have setup as well.  Do you have a
static IP address at home.  If not, get a dyndns account and setup an
entry there so that your friends/family can register using your dns
name instead of ip address

2. No idea.  Maybe try another stun server?

3. Not sure if double-NAT is needed now with the newer builds of
FreeSwitch.  Download the latest 1.0.4 to be on the safeside and
compile it again!  (I have FS 1.0.4 pre9 and it works I think).  As
long as your clients can register remotely you should be okay. I think
FS can work around most home NATs.  Make sure you have auto-nat set in
your internal.xml file (I think its this one)

4. SIP is the signaling.  RTP is the payload, or voice in your case.
Any transition is done via the SIP signaling.  This is how FS can
transfer calls etc or use the media bypass mode by specifying the IP
address where the RTP should be sent, which does not have to be the
same as the signaling.  Make sure you enable tracing in the
internal.xml file so you can debug the signaling.

You don't need to take a laptop to your daughters to test this.  Use
an internet sip phone like flaphone.com, which works through your web
browser.  This will register with an external IP address exactly like
your daughters and save you time traveling.  Note that sound isn't so
clear for me using this service, but it helps with debugging.

I also would recommend a sip client on windows like Zoiper, or
CounterPath's X-Lite.. both are free.  X-Lite is well known, Zoiper
allows for multiple SIP registrations and comes in a portable version.







On Fri, Aug 7, 2009 at 6:18 PM, Alan Chandler<alan at chandlerfamily.org.uk> wrote:
> I apologize, as my first post to this list, that I ask a detailed set of
> questions, but I have spend some time looking at all the docs and can't
> get what I need to do completely sorted in my head. ?I am definitely one
> who likes to UNDERSTAND what is happening rather than follow blank
> recipies, so please bear with me as I try understand all the details. I
> do understand about networking, NAT etc - but I am new to SIP/RTP and in
> particular what I think is a double NAT problem
>
>
> Firstly - what am I trying to achieve:
>
> I am in the UK and have a small home network behind a D-Link DIR-100
> Router/NAT/Firewall one of those machines, running Debian Lenny, acts as
> my main server for everything (and in an earlier incarnation was the
> firewall/router/nat box too - I only say this is because I had all this
> working using Asterisk a year or so ago, but with this important
> difference in configuration). ?Many of the ports on the firewall are
> port forwarded to this machine. I have set Freeswitch up on this server
> to act as a small voip pbx for the home - but MORE IMPORTANTLY - to
> enable my daughter from her house to talk to us. ?At my house locally I
> have a Linksys PAP2T two phone SIP box - and that is working with
> Freeswitch's default configuration (I set up to be 1000 and 1001 and
> used all the facilities). ?I will later add a Linksys SPA 3102 -
> although I DO NOT intend to use its facility to bridge to the normal
> phone network.
>
> My daughter, living in another house, also has a Nat box (unknown - its
> part of her ADSL modem/router/wireless access point) and also has a
> PAP2T which she will connect to the her network. ?This will be her phone.
>
> There is a family relation living in Australia who will load up a
> whatever softphone that we tell him to use. ?I expect, but don't know,
> that he will behind a NAT box too.
>
> Later, I have some friends in the USA that I might wish to add it too -
> especially so that we can hold some teleconferences. ?They will have a
> mixture of Windows and MACs, and I will need to recommend softphone
> clients for them.
>
> I want to set this up as a small private voice network, so anyone can
> ring anyone else. ?I will add fancy facilities such as conferencing and
> voicemail later - I just want to get the basics working first.
>
> Secondly
>
> I installed a stun client on my home machine and ran it against
> stun.freeswitch.org.
>
> It reported:-
>
> Primary: Independent Mapping, Independent Filter, preserves ports, no
> hairpin
>
> But I have no idea what this means - I can't find any clear statement
> via googling for it - how this set of answers maps to the different
> types of NAT that might be required to get this to all work. ?CAN
> SOMEONE ENLIGHTEN me please.
>
> Thirdly
>
> I have set up a sip profile called "double nat" from the recipe in the
> wiki. ?This defines the SIP port to be 5090.
>
> However, what I DO NOT UNDERSTAND is how the PAP2T box in my daughters
> house will initiate a connection to my server. ?Presumably, I have to
> port forward 5090 from the nat box to my server. ?IS THAT CORRECT?
>
> I also assume I will have to tell her to use STUN (I believe this is an
> option on the PAP2T)
>
> Fourthly
>
> If I understand SIP correctly, it just initiates the session and the two
> end points then communicate directly via RTP. ?What I don't understand
> is how does a session transition from SIP to RTP via the connection set
> up in the the first phase (in terms of passing through the NAT boxes).
> In particular WHICH OF THE TEST RESULTS from my stun client indicate it
> will do the right thing. ?(I am going to take a laptop to my daughters
> house with a stun client in to test her network this weekend).
>
> Could someone explain please.
>
> ?Fifthly
>
> Is there a recommended SIP softphone with all the right facilities (STUN
> support?) ?that works on MAC and WINDOWS (I only use linux myself).
>
> Apologies for the length of this. ?I am eager to get the answers so I
> can use an opportunity this weekend to get it working.
>
>
> --
> Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From dome at tel.co.th  Fri Aug  7 21:57:08 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Sat, 8 Aug 2009 11:57:08 +0700
Subject: [Freeswitch-users] 1.0.4 builds 1.0.3_1 debs
In-Reply-To: <631613604d8c9862f1cf43f80cb99f94.squirrel@webmail.chandlerfamily.org.uk>
References: <631613604d8c9862f1cf43f80cb99f94.squirrel@webmail.chandlerfamily.org.uk>
Message-ID: <8ccbff060908072157u19d0cb1rb48913832c74024@mail.gmail.com>

dch -i

2009/8/7  <alan at chandlerfamily.org.uk>:
> I just downloaded 1.0.4 and build debian packages with it, and it
> delivered .deb files names as 1.0.3-1
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From velu.technical at gmail.com  Fri Aug  7 22:28:27 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Sat, 8 Aug 2009 10:58:27 +0530
Subject: [Freeswitch-users] ESL for Perl
Message-ID: <1452e2980908072228u5e5fc191ufa2cdf002d903396@mail.gmail.com>

Dear all,
   When I do make for Perl ESL libraries in "esl/perl" directory I have got
the following error.

/usr/lib/gcc/i486-linux-gnu/4.1.2/../../../../lib/crt1.o: In function
`_start':
../sysdeps/i386/elf/start.S:115: undefined reference to `main'
esl_wrap.o: In function `_wrap_eslSetLogLevel':
esl_wrap.cpp:(.text+0x1c10): undefined reference to `eslSetLogLevel'
esl_wrap.o: In function `_wrap_ESLconnection_setEventLock':
esl_wrap.cpp:(.text+0x3395): undefined reference to
`ESLconnection::setEventLock(char const*)'
esl_wrap.o: In function `_wrap_ESLconnection_setBlockingExecute':
esl_wrap.cpp:(.text+0x3c65): undefined reference to
`ESLconnection::setBlockingExecute(char const*)'
esl_wrap.o: In function `_wrap_ESLconnection_execute':
esl_wrap.cpp:(.text+0x41ab): undefined reference to
`ESLconnection::execute(char const*, char const*, char const*)'
........

I have understood the when creating soft link ESL.o that error has occurred.

What is the problem?
Is there any dependency to create that link?

please help me.......

Thanks

Regards,
Velusamy
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From mike at jerris.com  Fri Aug  7 22:36:18 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 01:36:18 -0400
Subject: [Freeswitch-users] Outbound Proxy
In-Reply-To: <26f14763-6f40-4468-91f3-e60a46892183@dmmhosting.co.uk>
References: <26f14763-6f40-4468-91f3-e60a46892183@dmmhosting.co.uk>
Message-ID: <3A70DE01-8DEB-489B-9CAF-DA650CD3F2B1@jerris.com>

Please follow up with this on jira so we can make sure this gets  
addressed.

Mike

On Aug 4, 2009, at 5:10 AM, Darren Williams wrote:

> This produces:
>
> send 639 bytes to udp/[62.239.15.140]:5060 at 09:06:29.640488:
>     
> ------------------------------------------------------------------------
>    REGISTER sip:bmnha-01.bt.com SIP/2.0
>    Via: SIP/2.0/UDP  
> 91.121.159.57:5080;rport;branch=z9hG4bKQFFy1Ug5DcU0S
>    Max-Forwards: 70
>    From: <sip: 
> 05061292117 at bmnha-01.bt.com;transport=udp>;tag=NKaDDXrtme3Sr
>    To: <sip:05061292117 at bmnha-01.bt.com;transport=udp>
>    Call-ID: 88de957f-0907-4488-9b19-043ff13391f4
>    CSeq: 118570058 REGISTER
>    Contact: <sip:gw+05061292117 at 91.121.159.57:5080;transport=udp>
>    Expires: 3600
>    User-Agent: THOMSON ST2030 hw5 fw1.56 00-1F-9F-16-4E-99
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO
>    Supported: timer, precondition, path, replaces
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
> recv 726 bytes from udp/[62.239.15.140]:5060 at 09:06:29.656368:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 401 Unauthorized
>    Via: SIP/2.0/UDP  
> 91.121.159.57 
> :5080;received=91.121.159.57;branch=z9hG4bKQFFy1Ug5DcU0S;rport=5080
>    From: <sip: 
> 05061292117 at bmnha-01.bt.com;transport=udp>;tag=NKaDDXrtme3Sr
>    To: <sip: 
> 05061292117 
> @bmnha 
> -01 
> .bt 
> .com 
> ;transport=udp>;tag=SD567ec99-8a92eaf49d827b084b51e364ec68ae70.bc35
>    Call-ID: 88de957f-0907-4488-9b19-043ff13391f4
>    CSeq: 118570058 REGISTER
>    WWW-Authenticate: Digest realm="bmnha-01.bt.com",  
> nonce="4a77fb4b6fe33ffccf77e6bdd6f7a7a397cdfd17", qop="auth"
>    Server: Sip EXpress router (0.9.6 (sparc/solaris))
>    Content-Length: 0
>    Warning: 392 sip:5060 "Noisy feedback tells:  pid=9150  
> req_src_ip=172.20.92.61 req_src_port=5060 in_uri=sip:bmnha-01.bt.com  
> out_uri=sip:bmnha-01.bt.com via_cnt==1"
>
>     
> ------------------------------------------------------------------------
> 2009-08-04 11:06:29.655585 [ERR] sofia_reg.c:1460 05061292117  
> Registration Failed with status Operation has no matching challenge   
> [904]. failure #1 and so on.
>
> Should my side not have responded at this point with another  
> REGISTER with auth information in?

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From mike at jerris.com  Fri Aug  7 22:42:46 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 01:42:46 -0400
Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR -- how to diagnose?
In-Reply-To: <4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
References: <4256bf830908022222j31cfbd79idca9c6b26a8c82b0@mail.gmail.com>
	<177BDBC6-8ACA-400B-82F3-7792C32D9743@avgs.ca>
	<4256bf830908041151l55980e3fifb85c75b87535426@mail.gmail.com>
Message-ID: <3D0B855A-5B2C-4C6F-9306-64B0ECE97091@jerris.com>

http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP

turn the logging all the way up and see what it says.

Mike

On Aug 4, 2009, at 2:51 PM, Matthew Fong wrote:

> Hi Mathieu, thanks for the reply. I enabled sip trace logging and  
> got the logs below, but I am still at a loss at being able to  
> identify the error or reproduce it consistently. The below log  
> indicates to me that my FS server is initiating sending 2 BYE  
> message to my DID provider (didforsale.com). Is there a way I can  
> look further inside FreeSWITCH to see why it is sending this BYE  
> packet?
>
>
> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:44.679208:
> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
> Route: <sip:65.217.40.210:5060;lr>
> Route: <sip:63.110.102.238;lr>
> Max-Forwards: 70
> From: <sip: 
> +1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
> To: <sip: 
> + 
> 1909635XXXX 
> @199.173.100.144:5060;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
> CSeq: 118584736 BYE
> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
> Content-Length: 0
>
>
> sent 882 bytes to udp/[209.216.2.211]:5060 at 17:15:45.182589:
> BYE sip:199.173.100.144:5060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 66.197.142.69:5080;rport;branch=z9hG4bKD0UKrDB508mDD
> Route: <sip:209.216.2.211;ftag=dc7-13c4-2401b7-46dea593-2401b7;lr=on>
> Route: <sip:65.217.40.210:5060;lr>
> Route: <sip:63.110.102.238;lr>
> Max-Forwards: 70
> From: <sip: 
> +1212381XXXX at 63.110.102.238:5060;user=phone>;tag=Ztr5ycrv3QZ1g
> To: <sip: 
> + 
> 1909635XXXX 
> @199.173.100.144:5060;user=phone>;tag=dc7-13c4-2401b7-46dea593-2401b7
> Call-ID: a22bffb89064adc713c42401b78ca6b689e90b32fdbf612c0-0487-7441
> CSeq: 118584736 BYE
> Contact: <sip:+1212381XXXX at 66.197.142.69:5080;transport=udp>
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417M
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO
> Supported: timer, precondition, path, replaces
> Reason: Q.850;cause=25;text="EXCHANGE_ROUTING_ERROR"
> Content-Length: 0
>
> On Sun, Aug 2, 2009 at 11:20 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> Hi,
>
> Digging a bit in mod_sofia releaved that it can be caused by a SIP
> code 482 (loop detected), 483 (too many hops) or 484 (address
> incomplete).
>
> Do a SIP trace to sched more light on what's happening.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca

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From mike at jerris.com  Fri Aug  7 22:46:10 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 01:46:10 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local>
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com>
	<E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com>
	<6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com>
Message-ID: <782241E8-A1A6-4074-8D0A-4D362CF45FA5@jerris.com>

there is some other info on http://wiki.freeswitch.org/wiki/Perl_esl  
about this as well.  could somone try to put together some decent docs  
on fs_ivrd on the wiki as I keep getting asked this stuff privately as  
well.

Mike

On Aug 4, 2009, at 7:04 PM, William Suffill wrote:

> I wrote some notes on this but have yet to wiki it.
>
> example of an outbound socket connection where the call is answered, a
> variable is set then perhaps play one of the pre-installed files and
> hangup.
> ivrd
>
> fs_ivrd comes with freeswitch. It being a small daemon just invokes
> the script defined in a variable and passes data from it via STDIN/OUT
>
> Since this is an outbound socket connections it needs to be defined in
> the dialplan.
>
>
>
> Ex:
>
> <extension name="outbound-socket">
>
>    <condition field="destination_number" expression="^55(522)$">
>
>        <action application="set"
> data="ivr_path=/usr/local/freeswitch/scripts/ivrd-demo.php"/>
>
>        <action application="socket" data="127.0.0.1:8084 async full"/>
>
>    </condition>
>
>  </extension>
>
>
>
> The above dialplan sample would invoke ivr-demo.php when 55522 is
> called as long as fs_ivrd is running. To start fs_ivrd:
>
> /usr/local/freeswitch/bin/fs_ivrd -h 127.0.0.1 -p 8004
>
>
>
> It takes 2 arguments -h for hostname and -p for port.
>
>
>
> PHP Code
>
> #!/usr/bin/php -q
>
> <?php
>
>
>
>    // set a couple of things so we dont kill the system
>
>    ob_implicit_flush(true);
>
>    set_time_limit(30);
>
>
>
>
>
>    // Open stdin so we can read the AGI data in
>
>    $in = fopen("php://stdin", "r");
>
>    // Connect
>
>    echo "connect\n\n";
>
>    // Answer
>
>    echo "sendmsg\n";
>
>    echo "call-command: execute\n";
>
>    echo "execute-app-name: answer\n\n";
>
>
>
>    // Play a prompt
>
>    echo "sendmsg\n";
>
>    echo "call-command: execute\n";
>
>    echo "execute-app-name: playback\n";
>
>    echo "execute-app-arg:
> /usr/local/freeswitch/sounds/en/us/callie/ivr/8000/ivr- 
> welcome_to_freeswitch.wav\n\n";
>
>
>
>    // Wait
>
>    sleep(5);
>
>
>
>   // Hangup
>
>   echo "sendmsg\n";
>
>   echo "call-command: hangup\n\n";
>
>
>
> fclose($in);
>
>
>
> ?>
>
>
>
> ivrd will call this script for each call. All itdoes is answer the
> channel tell FreeSWITCH to play the ?welcome to freeswitch? prompt.
> Since the script is now controlling all call flow I needed to add a
> wait or it would send the hangup immediately before the prompt was
> played.
>
> Some improvements possible but that's 1 way to do it. It would be
> possible to do the socket directly in PHP but fs_ivrd is a nice option
> too.
>
> -- W
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Fri Aug  7 22:49:01 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 01:49:01 -0400
Subject: [Freeswitch-users] phpmod compile error
In-Reply-To: <63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
References: <558004.60211.qm@web59511.mail.ac4.yahoo.com>
	<78A818D3-6C4F-420D-A922-751A69E7E080@avgs.ca>
	<6CE58813-715A-43DB-877B-638B5CE7E6E9@freeswitch.org>
	<87f2f3b90902251342q1e393127ha7fbdbebf6d83dac@mail.gmail.com>
	<1235647534150-2389093.post@n2.nabble.com>
	<A28FFFB7-C42B-403A-8A8A-4CEF49AB9AC2@avgs.ca>
	<1235674261134-2391480.post@n2.nabble.com>
	<191c3a030902261104n560475acqa924491afd07bd42@mail.gmail.com>
	<63B12BA3-374B-4FE1-877E-4B445BC9DA4B@cgicommunications.com>
Message-ID: <179DF0A9-1E96-43EB-976C-FE00365436BC@jerris.com>

Please report this error to centos bug tracker, aspell-devel should be  
a dependency of php-devel.

Mike

On Aug 5, 2009, at 10:40 AM, Greg Thoen wrote:

> Trying to make phpmod and it fails with this:
>
> /usr/bin/ld: cannot find -laspell
> collect2: ld returned 1 exit status
> make[1]: *** [ESL.so] Error 1
>
> I do have php-devel on this Centos 5.2 machine. Any ideas?
> --
> Greg
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Fri Aug  7 22:55:00 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 01:55:00 -0400
Subject: [Freeswitch-users] Question about using
	switch_caller_extension_add_application
In-Reply-To: <a44688a0908052054j405c1707j382b3d6ccac8f8a9@mail.gmail.com>
References: <a44688a0908050805t2e3f0a29oa22e34bf68517d98@mail.gmail.com>
	<5a8712120908050923w2d2048d1rbedc6b22bdeb1814@mail.gmail.com>
	<E4B911F0-6C40-4E55-B2D2-2A9809B0CF0F@avgs.ca>
	<a44688a0908051720m48203679kab3d50519d4534ed@mail.gmail.com>
	<9521EE86-37FF-4989-8F58-F61E5110E5C5@avgs.ca>
	<a44688a0908052054j405c1707j382b3d6ccac8f8a9@mail.gmail.com>
Message-ID: <358E66DE-C5F1-418B-806B-2441E8F42130@jerris.com>

you could probably pull off the same thing with xml_curl for dialplan  
and a simple set of bridge and then transfer in the actions.

Mike

On Aug 5, 2009, at 11:54 PM, Woody Dickson wrote:

> Hi,
>
> In my module, I will collect a list of available failover route that  
> I can use to failover to whenever a particular error is received.   
> However, these available routes has different condition and the  
> condition changes every half a minute.  Therefore, I need to catch  
> the hangup cause after bridge, and then figure out the next workable  
> available route based on the latest condition setting.
>
> It seems like this is only prossible to be done within a C module.   
> Any suggestion will be greatly appreciated.
>
> Woody




From mike at jerris.com  Fri Aug  7 22:57:21 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 01:57:21 -0400
Subject: [Freeswitch-users] Question about dynamic registration
In-Reply-To: <27c25bc40908060224l38ad47fdje17065b13647905d@mail.gmail.com>
References: <27c25bc40908030445k741a2a5dv341d9a4d343b8339@mail.gmail.com>
	<9FE0F6B8-C4BF-4820-8CC2-6825C5EE8422@freeswitch.org>
	<27c25bc40908060224l38ad47fdje17065b13647905d@mail.gmail.com>
Message-ID: <03C39E0E-2303-40EC-9B84-5D0AE499436D@jerris.com>

http://fisheye.freeswitch.org/browse/FreeSWITCH/contrib/cparker/mod_xml_radius

On Aug 6, 2009, at 5:24 AM, Juan Backson wrote:

> Hi,
>
> Is there a sample module that I can take a look at on how to do that?
> I don't understand how to get the registration request and how to  
> pass back auth result to freeswitch.
>
> JB
>
> On Mon, Aug 3, 2009 at 8:42 PM, Brian West <brian at freeswitch.org>  
> wrote:
> You could build your own module to do it how ever you please.  But
> forking a script every time to auth is not very scalable.
>
> /b
> _______________________________________________

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From mike at jerris.com  Fri Aug  7 23:11:12 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 02:11:12 -0400
Subject: [Freeswitch-users] Lua on Windows and additional modules
In-Reply-To: <3c233920908061755h5d17aa0as3fb24743215a8298@mail.gmail.com>
References: <3c233920908061755h5d17aa0as3fb24743215a8298@mail.gmail.com>
Message-ID: <0EE782AE-FC28-4029-AA01-76D87BC08069@jerris.com>

check the sample config files for options to specify these:

http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/autoload_configs/lua.conf.xml?r=10747


On Aug 6, 2009, at 8:55 PM, Vladimir Rodionov wrote:

> Good evening,
> This is newbie question.
>
> The FreeSWITCH lua module does not support sockets and sql out of  
> box that is why
> I just installed LuaBinaries (including socket, sql modules). My dev  
> environment is Win XP not Linux/Unix.
>
> I am trying to understand what will happen when lua_module get this:
>
> require "socket" or
> require "luasql.mysql"
> ?
>
> How does lua_module look up additional lua modules on Windows  
> platform?
>
> Do I have to set some env variables?
>
> TIA
> -Vladimir Rodionov




From merul at mac.com  Sat Aug  8 01:58:04 2009
From: merul at mac.com (Merul Patel)
Date: Sat, 08 Aug 2009 09:58:04 +0100
Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk
	-	Freeswitch equivalent?
Message-ID: <C5CDD462-A4BF-4316-B35C-4892C92DB9F4@mac.com>

Hi,

I came across this thread (http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/010172.html 
) from January, and I'm having a similar problem, in that from the  
moment an incoming analogue call starts ringing, it takes around 5-7  
seconds before the dialplan gets executed.

Tailing the freeswitch log (pastebin'd here: http://pastebin.com/f2a6f7945) 
, it seems to indicate that there is either a problem with an  
unhandled event 17 and/or a delay with the call being put in the  
GET_CALLERID state.

In my case, I'm more concerned about answering the call rather than  
getting the caller id, so I tried setting <param name="enable- 
callerid" value="false"/> in openzap.conf.xml, as per the earlier  
post's author but this still doesn't get the dialplan to execute  
immediately.

I'm running FS 1.0.4 built from the tarball, with zaptel 1.4.12.1 and  
wanpipe 3.5.4.18 in conjunction with a USBFXO device from Sangoma. I'm  
using the out -of-the-box configuration for FS, with the exception of  
one modified public dialplan. My openzap.conf.xml is:

<configuration name="openzap.conf" description="OpenZAP Configuration">
   <settings>
     <param name="debug" value="0"/>
     <!--<param name="hold-music" value="$${moh_uri}"/>-->
     <!--<param name="enable-analog-option" value="call-swap"/>-->
     <!--<param name="enable-analog-option" value="3-way"/>-->
   </settings>
    <pri_spans/>
   <!-- one entry here per openzap span -->
   <analog_spans>
     <span id="1">
       <!--<param name="hold-music" value="$${moh_uri}"/>-->
       <!--<param name="enable-analog-option" value="call-swap"/>-->
       <!--<param name="enable-analog-option" value="3-way"/>-->
       <param name="tonegroup" value="uk"/>
       <param name="digit-timeout" value="2000"/>
       <param name="max-digits" value="11"/>
       <param name="dialplan" value="XML"/>
       <param name="context" value="public"/>
       <param name="enable-callerid" value="true"/>
       <!-- regex to stop dialing when it matches -->
       <!--<param name="dial-regex" value="5555"/>-->
       <!-- regex to stop dialing when it does not match -->
       <!--<param name="fail-dial-regex" value="^5"/>-->
     </span>
   </analog_spans>
</configuration>

Couldn't find any jiras for this issue, so not sure whether it was  
resolved. Any pointers would be welcome.

BR

Merul



From pjintheusa at gmail.com  Sat Aug  8 03:48:51 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 03:48:51 -0700
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
In-Reply-To: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>
References: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>
Message-ID: <367751820908080348p6165b113kc3be468cb32fabb3@mail.gmail.com>

Hi there,

Not sure whether this helps but test this without set bypass_media. In
my setup I have noticed the leg A session ends when bypass_media is
true. Call/bridge continue successfully.

Phillip Jones

On Thu, Aug 6, 2009 at 1:28 PM, Benedikt
Fraunhofer<fraunhofer.lists.freeswitch-001 at traced.net> wrote:
> Hello List, Hello *,
>
> First of all the usual excuses: sorry for the bad english and the long
> email, no native speaker and i really tried to make it shorter, but i
> guess this would result in even more "check back"s than it already
> does :)
>
> we're currently running in a weird "lockup"-scenario in our loadtests.
>
> Our setup is the following:
>
> three freeswitch servers, let's call them A(-leg), M(aster), and
> B(-leg) with the goal in mind to initiate calls on M which calls A,
> play some file, bridge to B, limit call length and play (different)
> prompts to A and B if they exceed that limit.
>
> (Note that A and B work fine, regardless of the amount of load we put on them)
> A and B are silly dialplan logic, accepting calls on a certain
> extension after a random delay and playing moh. Before calling
> playback to a localstream they call a lua script which schedules
> hangup somewhere in future (which works flawlessly)
>
> Calls are initiated on M using some hacked up loadgen-script issuing
> http requests like
> ? originate [sofiaSyntaxToExtensionOn_A] 6000
> . The 6000 extension on M has the following (xml) dialplan which
> essentially does the following:
> ------
> answer()
> ...playback file...
> ...set some callerid stuff
> set bypass_media
> bridge to extension 6009 on B
> ------
> we use "execute_on_answer" on the b-leg to run a script which limits
> the length of the call (doesn't matter if it's done via "export
> nolocal" or "inlined" into the data part of the bridge application
> "{execute_on_answer=lua ...}")
>
> <action application="export" data="nolocal:execute_on_answer=lua
> lua/schedule-hangup.lua ${uuid}" />
>
> the lua script "schedula-hangup.lua" does essentially the following:
>
> ------
> api = freeswitch.API();
> local res = api:execute("sched_api", "+10 none lua
> lua/c2c-hangup-timeout.lua " .. argv[1]);
> ------
>
> the 10 seconds are just to speed up the time until it gets stuck.
>
> this is where things start to go wrong. if I comment out the call to
> the "schedule-hangup" script, everything works fine, even if it's
> under heavy load.
>
> c2c-hangup-timeout.lua does the following:
> ------------------
> local sess = argv[1];
> if(sess)
> then
> ? ?freeswitch.consoleLog("INFO", "c2c-hangup-timeout.lua for uuid "
> .. sess .. "\n");
>
> ? ?api = freeswitch.API();
> ? ?local stillValid = api:execute("uuid_getvar", sess .. "
> Dummy-DoesChannelExists");
> ? ?if(stillValid:sub(1,4) == "-ERR")
> ? ?then
> ? ? ? ?log("session uuid " .. sess .. " disappeared (nothing bad)");
> ? ?else
> ? ? ? ?-- this is important!!! Otherwise the aleg get's just hung up!
> ? ? ? ?api:execute("uuid_media", sess);
> ? ? ? ?api:execute("uuid_transfer", sess .. " -both timeout");
> ? ?end
> else -- /if(sess)
> ? ?log("called with nil session?");
> end -- /if(sess)
>
> ------------------
>
> i guess this needs some explanation:
> we get the uuid of the channel as argument in argv[1]. We don't use
> ? local session = freeswitch.Session(uuid);
> since if the channel referenced by "uuid" does not exist any longer,
> freeswitch (or the lua bindings) try to interpret the uuid as an
> "originate string" and can't figure out how to call that. So we use a
> dummy api call to get some channel variable. If the channel does not
> exist any longer (A or B already hung up), we get an error message
> starting with "-ERR", otherwise the channel still exists (we get
> "_unset_" as the value, if it's not set) and we continue by getting
> freeswitch back in the media path (uuid_media) and then transferring
> both legs to an extension called "timeout" which plays some prompt and
> finally calls hangup().
>
> If we don't do the uuid_media call, one of the legs gets hung up when
> we transfer them to the extension. This looks like the following on
> the console after issuing "uuid_transfer [uuid] -both timeout"
> (extensions are not the same as in our loadgen example above)
>
>
> --------------
> 2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1334 Hangup (*)
> sofia/internal/1000 [CS_HIBERNATE] [BLIND_TRANSFER]
> 2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer
> sofia/internal/1004 at 192.168.179.177:5060 to XML[timeout at default]
> 2009-07-23 19:57:19.865703 [INFO] mod_dialplan_xml.c:310 Processing
> BFR1004->timeout in context default
> API CALL [uuid_transfer(73812082-77b1-11de-b9f8-a10bb0eb9f69 -both
> timeout)] output:
> +OK
>
> 2009-07-23 19:57:19.865703 [NOTICE] switch_ivr.c:1349 Transfer (**)
> sofia/internal/1000 to XML[timeout at default]
> 2009-07-23 19:57:19.865703 [NOTICE] switch_core_session.c:1084 Session
> 60 (sofia/internal/1000) Ended
> 2009-07-23 19:57:19.865703 [NOTICE] switch_core_session.c:1086 Close
> Channel sofia/internal/1000 [CS_DESTROY]
> -----------
>
> note that it first does Hangup (denoted by *, no that's not an
> asterisk :) on extension 1000 and then tries to Transfer (**) the hung
> up channel to the dial plan. this could be the same as in an earlier
> post to the list "SIP re-invite / bypass_media // Phillip Jones //
> Wed, 01 Jul 2009 13:30:53 -0700)"
>
> This is why we do not directly call sched_transfer() but call a script
> in between to do the uuid_media() call. I couldn't figure out how to
> call that directly from the xml dialplan and/or how to check if the
> channel still exists.
>
> so... after using uuid_media(), both legs are transferred without an
> (intermediate|bogus) hangup() call.
>
> This only works fine if we've few concurrent calls. There is no magic
> borderline where it starts to refuse work.
>
> Some of the Symptoms are: traffic decreased to zero as no new channels
> are successfully brought up, some of the signaling traffic is not
> ACKed or OKed, scheduled jobs are not run.
>
> if i read the output of "show channels" correctly, they're all stuck
> in different applications like hangup(), some are calling lua but most
> of them are in signaling_bridge(). Freeswitch is still responding on
> the console and there's almost no load on the machine (no busy polling
> or some other kind of running amok).
>
> if i kill one of them using uuid_kill() or kill all of them
> using"fsctl hupall" i get "Task was executed late by 866 seconds 12379
> sched_api_function (none)" messages and the usual cleanup takes place.
> As a quick hack i tried to schedule a uuid_kill() call 20 seconds
> after the scheduling call to the lua script but that job is not
> executed either.
>
> So what am I doing wrong? Is it some deadlock where uuid_media() and
> uuid_transfer() ?are waiting for the other to finish?
> Or some other silly simple thing i missed?
>
> Thx in advance
>
> ?Benedikt.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From pjintheusa at gmail.com  Sat Aug  8 03:59:56 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 03:59:56 -0700
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
Message-ID: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>

Hi there,

I am trying to implement a scenario where I can terminate calls to
multiple destinations AND have termination carrier fail over.
Currently I can see how to do one or the other. But not both.

Multiple destinations is easy:

 <action application="bridge"
data="sofia/SIP_PROVIDER_1/6095551234,sofia/SIP_PROVIDER_1/7325551234"/>

Failover appears to use the same mechanism:

<action application="bridge"
data="sofia/gateway/primary/blah|sofia/gateway/secondary/blah"/>

I can not get my head around how use these together, such that:


- 6095551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 6095551234  through SIP_PROVIDER_2
- Called party does not answer
7325551234 is dialed through SIP_PROVIDER_1
- if NO_ROUTE_DESTINATION then dial 7325551234 through SIP_PROVIDER_2
- Called party answers


This must be a fairly common requirement so any ideas on what I might
be missing would be very welcome.

Thanks


Phillip Jones



From mike at jerris.com  Sat Aug  8 04:28:22 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 07:28:22 -0400
Subject: [Freeswitch-users] zapata.conf immediate=yes in Asterisk -
	Freeswitch equivalent?
In-Reply-To: <C5CDD462-A4BF-4316-B35C-4892C92DB9F4@mac.com>
References: <C5CDD462-A4BF-4316-B35C-4892C92DB9F4@mac.com>
Message-ID: <E15045EA-3517-4174-B12E-12143BFCE27B@jerris.com>

is this just the timeout waiting for your max digits?

       <param name="max-digits" value="11"/>


On Aug 8, 2009, at 4:58 AM, Merul Patel wrote:

> Hi,
>
> I came across this thread (http://lists.freeswitch.org/pipermail/freeswitch-users/2009-January/010172.html
> ) from January, and I'm having a similar problem, in that from the
> moment an incoming analogue call starts ringing, it takes around 5-7
> seconds before the dialplan gets executed.
>
> Tailing the freeswitch log (pastebin'd here: http://pastebin.com/f2a6f7945)
> , it seems to indicate that there is either a problem with an
> unhandled event 17 and/or a delay with the call being put in the
> GET_CALLERID state.
>
> In my case, I'm more concerned about answering the call rather than
> getting the caller id, so I tried setting <param name="enable-
> callerid" value="false"/> in openzap.conf.xml, as per the earlier
> post's author but this still doesn't get the dialplan to execute
> immediately.
>
> I'm running FS 1.0.4 built from the tarball, with zaptel 1.4.12.1 and
> wanpipe 3.5.4.18 in conjunction with a USBFXO device from Sangoma. I'm
> using the out -of-the-box configuration for FS, with the exception of
> one modified public dialplan. My openzap.conf.xml is:
>
> <configuration name="openzap.conf" description="OpenZAP  
> Configuration">
>   <settings>
>     <param name="debug" value="0"/>
>     <!--<param name="hold-music" value="$${moh_uri}"/>-->
>     <!--<param name="enable-analog-option" value="call-swap"/>-->
>     <!--<param name="enable-analog-option" value="3-way"/>-->
>   </settings>
>    <pri_spans/>
>   <!-- one entry here per openzap span -->
>   <analog_spans>
>     <span id="1">
>       <!--<param name="hold-music" value="$${moh_uri}"/>-->
>       <!--<param name="enable-analog-option" value="call-swap"/>-->
>       <!--<param name="enable-analog-option" value="3-way"/>-->
>       <param name="tonegroup" value="uk"/>
>       <param name="digit-timeout" value="2000"/>
>       <param name="max-digits" value="11"/>
>       <param name="dialplan" value="XML"/>
>       <param name="context" value="public"/>
>       <param name="enable-callerid" value="true"/>
>       <!-- regex to stop dialing when it matches -->
>       <!--<param name="dial-regex" value="5555"/>-->
>       <!-- regex to stop dialing when it does not match -->
>       <!--<param name="fail-dial-regex" value="^5"/>-->
>     </span>
>   </analog_spans>
> </configuration>
>
> Couldn't find any jiras for this issue, so not sure whether it was
> resolved. Any pointers would be welcome.
>
> BR
>
> Merul
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From fraunhofer.lists.freeswitch-001 at traced.net  Sat Aug  8 04:53:05 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Sat, 8 Aug 2009 13:53:05 +0200
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
In-Reply-To: <367751820908080348p6165b113kc3be468cb32fabb3@mail.gmail.com>
References: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>
	<367751820908080348p6165b113kc3be468cb32fabb3@mail.gmail.com>
Message-ID: <a37510170908080453m1e98ae3l544e3d8b301605e1@mail.gmail.com>

Hi Phillip,

2009/8/8 Phillip Jones <pjintheusa at gmail.com>:

> Not sure whether this helps but test this without set bypass_media. In
> my setup I have noticed the leg A session ends when bypass_media is
> true. Call/bridge continue successfully.

thx for that hint. unfortunately we can't do that due to the high
volume we anticipate
and i think i already tried that.

Note that it works in exactly this setup  (with the uuid_media()-call)
if there are only very few calls handled. It just starts to refuse
work once enough jobs are scheduled and not executed.

Cheers
  Benedikt



From frank at impactfax.com  Sat Aug  8 06:10:18 2009
From: frank at impactfax.com (Frank @ Impact)
Date: Sat, 8 Aug 2009 09:10:18 -0400
Subject: [Freeswitch-users] DTMF disable a few secs after call starts
Message-ID: <0B930F23C4A24E3CB0015945EA06C8B7@ws4>

FS is in the media path of an IVR call.
At the moment, the call is ulaw with DTMF in the audio I think coming
into FS and leaving FS.
The call is coming from an Asterisk server and going to an Asterisk
server.
 
Is there a way to disable FS from passing DTMF at some point in the
call?  For example, after 15 seconds, is there a way to get FS to stop
passing DTMF events?
 
Would I have to try to force asterisk to use rfc2833 when sending the
call to FS and when accepting it back from FS?
 
 
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From william.suffill at gmail.com  Sat Aug  8 06:33:49 2009
From: william.suffill at gmail.com (William Suffill)
Date: Sat, 8 Aug 2009 09:33:49 -0400
Subject: [Freeswitch-users] Outbound socket PHP question
In-Reply-To: <782241E8-A1A6-4074-8D0A-4D362CF45FA5@jerris.com>
References: <A2B4F48489739E439D8845296A2FD9C4147458@sun.Noble.local> 
	<87f2f3b90908021535w4cadf78p391425f4d2a03549@mail.gmail.com> 
	<E0DB165C-7EF7-48E7-9C31-10EC39879A32@cgicommunications.com> 
	<6b65470d0908041604n6ad04dd1lbbe82ff83f08116b@mail.gmail.com> 
	<782241E8-A1A6-4074-8D0A-4D362CF45FA5@jerris.com>
Message-ID: <6b65470d0908080633m1b3c15b9me110eb0245a93b0f@mail.gmail.com>

I'll see what I can do. Got a few sections of the PHP ESL to finish
but should be able to
write up fs_ivrd as well.

-- W



From rupa at rupa.com  Sat Aug  8 07:16:37 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 8 Aug 2009 09:16:37 -0500
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
Message-ID: <b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>

While any use of loopback can be considered abuse, that is how I solve this
issue.  I use loopback for each dialed # and then in the dialplan use
mod_lcr to build a dialstring that does failover in order of cost.

So:

 <action application="bridge"
data="sofia/SIP_PROVIDER_1/6095551234,sofia/SIP_PROVIDER_1/7325551234"/>
becomes

 <action application="bridge"
data="loopback/6095551234,loopback/7325551234"/>

In the dialplan you then match on those (just regular 10digit dialing) and
do your normal failover dialstring.  Again, I do this with mod_lcr but that
isn't necessary.

<action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>


On Sat, Aug 8, 2009 at 5:59 AM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi there,
>
> I am trying to implement a scenario where I can terminate calls to
> multiple destinations AND have termination carrier fail over.
> Currently I can see how to do one or the other. But not both.
>
> Multiple destinations is easy:
>
>  <action application="bridge"
> data="sofia/SIP_PROVIDER_1/6095551234,sofia/SIP_PROVIDER_1/7325551234"/>
>
> Failover appears to use the same mechanism:
>
> <action application="bridge"
> data="sofia/gateway/primary/blah|sofia/gateway/secondary/blah"/>
>
> I can not get my head around how use these together, such that:
>
>
> - 6095551234 is dialed through SIP_PROVIDER_1
> - if NO_ROUTE_DESTINATION then dial 6095551234  through SIP_PROVIDER_2
> - Called party does not answer
> 7325551234 is dialed through SIP_PROVIDER_1
> - if NO_ROUTE_DESTINATION then dial 7325551234 through SIP_PROVIDER_2
> - Called party answers
>
>
> This must be a fairly common requirement so any ideas on what I might
> be missing would be very welcome.
>
> Thanks
>
>
> Phillip Jones
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa
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From rupa at rupa.com  Sat Aug  8 07:18:59 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 8 Aug 2009 09:18:59 -0500
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
	<b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
Message-ID: <b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>

On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:

> <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>
>

That of course, should be:

 <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$1..."/>
-- 
-Rupa
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From brian at freeswitch.org  Sat Aug  8 07:26:32 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 8 Aug 2009 09:26:32 -0500
Subject: [Freeswitch-users] ESL for Perl
In-Reply-To: <1452e2980908072228u5e5fc191ufa2cdf002d903396@mail.gmail.com>
References: <1452e2980908072228u5e5fc191ufa2cdf002d903396@mail.gmail.com>
Message-ID: <7A151806-653B-4DDA-84A4-F5EBC25E0D81@freeswitch.org>

You don't do make in the perl directory... You cd .. && 'make permod'  
from the top level dir.

/b

On Aug 8, 2009, at 12:28 AM, velusamy velu wrote:

> Dear all,
>    When I do make for Perl ESL libraries in "esl/perl" directory I  
> have got the following error.
>
> /usr/lib/gcc/i486-linux-gnu/4.1.2/../../../../lib/crt1.o: In  
> function `_start':
> ../sysdeps/i386/elf/start.S:115: undefined reference to `main'
> esl_wrap.o: In function `_wrap_eslSetLogLevel':
> esl_wrap.cpp:(.text+0x1c10): undefined reference to `eslSetLogLevel'
> esl_wrap.o: In function `_wrap_ESLconnection_setEventLock':
> esl_wrap.cpp:(.text+0x3395): undefined reference to  
> `ESLconnection::setEventLock(char const*)'
> esl_wrap.o: In function `_wrap_ESLconnection_setBlockingExecute':
> esl_wrap.cpp:(.text+0x3c65): undefined reference to  
> `ESLconnection::setBlockingExecute(char const*)'
> esl_wrap.o: In function `_wrap_ESLconnection_execute':
> esl_wrap.cpp:(.text+0x41ab): undefined reference to  
> `ESLconnection::execute(char const*, char const*, char const*)'
> ........
>
> I have understood the when creating soft link ESL.o that error has  
> occurred.
>
> What is the problem?
> Is there any dependency to create that link?
>
> please help me.......
>
> Thanks
>
> Regards,
> Velusamy
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From alan at chandlerfamily.org.uk  Sat Aug  8 07:31:02 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Sat, 08 Aug 2009 15:31:02 +0100
Subject: [Freeswitch-users] New to Freeswitch - some help needed
In-Reply-To: <33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
Message-ID: <4A7D8C26.4020408@chandlerfamily.org.uk>

Mark Campbell-Smith wrote:
> Hi Alan,
> 
> I hope you find your answers here as these are the sort of things that
> are hard to find on the wiki, which is somewhat outdated in areas.  If
> you do find your answers, please post them back here for everyone
> else.
> 
> I am new to FS also, so my comments below may not be 100% correct!
> 
> 1. Very similar to what I want to have setup as well.  Do you have a
> static IP address at home.  If not, get a dyndns account and setup an
> entry there so that your friends/family can register using your dns
> name instead of ip address

Thanks for the idea - but I am way ahead on this.

My IP address is allocated by dhcp, but since my router stays switched 
on 24/7, and even powering it off for short periods doesn't change 
things. It has not changed for maybe a year.

So I own my own domain name (chandlerfamily.org.uk) and can point 
home.chandlerfamily.org.uk at my ip address. If it ever changes, I can 
go to my domain name provider and change my dns entry very easily.

You can in fact tell freeswitch to ignore the domain names used by the 
clients with the two parameters in the internal sip profile

     <!--all inbound reg will look in this domain for the users -->
     <param name="force-register-domain" value="$${domain}"/>
     <!--all inbound reg will stored in the db using this domain -->
     <param name="force-register-db-domain" value="$${domain}"/>

(This is from the new 1.0.4 version of freeswitch)

After a discussion on IRC I decided to set $${domain} to 
chandlerfamily.org.uk so theoretically all my phones have addresses 
extn-no at chandlerfamily.org.uk

And in the pap2t box I am using tell it to use chandlerfamily.org.uk as 
its "proxy" and home.chandlerfamily.org.uk as its "outgoing proxy" - for 
phones outside the NAT box.  Inside my home I use the internal name of 
the freeswitch box as the "outgoing proxy".


> 
> 2. No idea.  Maybe try another stun server?

I still don't have any answers to this, but it doesn't appear to be 
important

> 
> 3. Not sure if double-NAT is needed now with the newer builds of
> FreeSwitch.  Download the latest 1.0.4 to be on the safeside and
> compile it again!  (I have FS 1.0.4 pre9 and it works I think).  As
> long as your clients can register remotely you should be okay. I think
> FS can work around most home NATs.  Make sure you have auto-nat set in
> your internal.xml file (I think its this one)
>

I have 1.0.4 installed as of yesterday, and it appears to be working. 
My daughter still has to change the user id's inside her PAP2T box to 
numbers as I could not make the "number-alias" function work - so 
although the sip part appears to work, I have still to find out if the 
voice part does.


> 4. SIP is the signaling.  RTP is the payload, or voice in your case.
> Any transition is done via the SIP signaling.  This is how FS can
> transfer calls etc or use the media bypass mode by specifying the IP
> address where the RTP should be sent, which does not have to be the
> same as the signaling.  Make sure you enable tracing in the
> internal.xml file so you can debug the signaling.
> 
> You don't need to take a laptop to your daughters to test this.  Use
> an internet sip phone like flaphone.com, which works through your web
> browser.  This will register with an external IP address exactly like
> your daughters and save you time traveling.  Note that sound isn't so
> clear for me using this service, but it helps with debugging.
> 
> I also would recommend a sip client on windows like Zoiper, or
> CounterPath's X-Lite.. both are free.  X-Lite is well known, Zoiper
> allows for multiple SIP registrations and comes in a portable version.
> 

I used zoiper before as an iax client - so I'll look again at this


-- 
Alan Chandler
http://www.chandlerfamily.org.uk




From chad at apartmentlines.com  Sat Aug  8 08:15:18 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Sat, 8 Aug 2009 11:15:18 -0400
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase macro is
	playing
In-Reply-To: <4A7D8C26.4020408@chandlerfamily.org.uk>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
Message-ID: <BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>

in a lua script, i've tried using session:setInputCallback() to catch  
DTMF tones while a phrase macro is playing, but it doesn't seem to  
work.  the same callback _does_ catch DTMF when i use  
session:streamFile() to play something.  is there some other way to do  
it?

below is an example of how i'm doing it now.  when voicemail/vm- 
mailbox_full.wav plays, i see key presses being printed to the  
console, but when the phrase plays, no key presses are displayed.

function key_press(session, input_type, data)
	if input_type == "dtmf" then
		freeswitch.consoleLog("info", "Key pressed: " .. data["digit"] ..  
"\n")
	end
end

if session:ready() then
	session:answer()
	session:execute("sleep", "1000")
	session:setInputCallback("key_press", "")
	session:streamFile("voicemail/vm-mailbox_full.wav")
	session:execute("phrase", "voicemail_menu,1:2:3:#")
end



From brian at freeswitch.org  Sat Aug  8 08:59:42 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 8 Aug 2009 10:59:42 -0500
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
Message-ID: <96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>

Examples exist in the scripts/lua folder called callback.lua.  Also  
please DO NOT hijack threads.  You can't click reply, change the  
subject as it messes up the mailing list threading and most mail  
readers will thread it incorrectly.

Thanks,
Brian

On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines wrote:

> in a lua script, i've tried using session:setInputCallback() to catch
> DTMF tones while a phrase macro is playing, but it doesn't seem to
> work.  the same callback _does_ catch DTMF when i use
> session:streamFile() to play something.  is there some other way to do
> it?
>
> below is an example of how i'm doing it now.  when voicemail/vm-
> mailbox_full.wav plays, i see key presses being printed to the
> console, but when the phrase plays, no key presses are displayed.
>
> function key_press(session, input_type, data)
> 	if input_type == "dtmf" then
> 		freeswitch.consoleLog("info", "Key pressed: " .. data["digit"] ..
> "\n")
> 	end
> end
>
> if session:ready() then
> 	session:answer()
> 	session:execute("sleep", "1000")
> 	session:setInputCallback("key_press", "")
> 	session:streamFile("voicemail/vm-mailbox_full.wav")
> 	session:execute("phrase", "voicemail_menu,1:2:3:#")
> end
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From vladrodionov at gmail.com  Sat Aug  8 09:26:05 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Sat, 8 Aug 2009 09:26:05 -0700
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
Message-ID: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>

Good morning,
This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
AS does some logic and transfers call (or forward) out of Voip provider
network to another PSTN number User2.


This is call bridge


    UA1     (PSTN) -                                 ->  UA2 (PSTN)
              -                                            -
                -  (1)                                   -  (4)
                  ->         PSTN Gateway->
                            -                        -
                       (2) -                        - (3)
                          -> FreeSWITCH ->


This is what I want to acomplish
                                    (4)
    UA1     (PSTN) ------------------------------- ->  UA2 (PSTN)
                    -
                       -   (1)
                        ->  PSTN Gateway->
                             -                        -
                       (2)  -                        - (3)
                            -> FreeSWITCH ->


1. Can it be implemented in FreeSWITCH?
2. Does anybody know Voip providers which support out of network call
transfer/forwarding to PSTN?

TIA

-Vladimir Rodionov
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From raffaele.p.guidi at gmail.com  Sat Aug  8 09:50:57 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sat, 8 Aug 2009 18:50:57 +0200
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
Message-ID: <b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>

Brian, I've read 8-9 of your emails asking to not hijack threads in threads
where this it didn't happen (like this). I see this as a new thread with a
single message (plus yours, of course) inside with "[Freeswitch-users] how
to catch DTMF in Lua while a phrase macro is playing" as the subject - and I
think is quite relevant to the message. I suspect you have something wrong
in your email client (mine is gmail) or server and suggest you to give it a
check.

Regards,
   Raffaele

On Sat, Aug 8, 2009 at 17:59, Brian West <brian at freeswitch.org> wrote:

> Examples exist in the scripts/lua folder called callback.lua.  Also
> please DO NOT hijack threads.  You can't click reply, change the
> subject as it messes up the mailing list threading and most mail
> readers will thread it incorrectly.
>
> Thanks,
> Brian
>
> On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines wrote:
>
> > in a lua script, i've tried using session:setInputCallback() to catch
> > DTMF tones while a phrase macro is playing, but it doesn't seem to
> > work.  the same callback _does_ catch DTMF when i use
> > session:streamFile() to play something.  is there some other way to do
> > it?
> >
> > below is an example of how i'm doing it now.  when voicemail/vm-
> > mailbox_full.wav plays, i see key presses being printed to the
> > console, but when the phrase plays, no key presses are displayed.
> >
> > function key_press(session, input_type, data)
> >       if input_type == "dtmf" then
> >               freeswitch.consoleLog("info", "Key pressed: " ..
> data["digit"] ..
> > "\n")
> >       end
> > end
> >
> > if session:ready() then
> >       session:answer()
> >       session:execute("sleep", "1000")
> >       session:setInputCallback("key_press", "")
> >       session:streamFile("voicemail/vm-mailbox_full.wav")
> >       session:execute("phrase", "voicemail_menu,1:2:3:#")
> > end
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From raffaele.p.guidi at gmail.com  Sat Aug  8 09:55:27 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sat, 8 Aug 2009 18:55:27 +0200
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
	<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
Message-ID: <b0a8792f0908080955k150badbcmadf154283c3d2238@mail.gmail.com>

A cut&paste of a gmail search for "label:voip hijack". The subjects seem to
be ok

Chad, Brian, Raffaele (3)
liste.voip

[Freeswitch-users] how to catch DTMF in Lua while a phrase macro is
playing? - ?
please DO NOT *hijack* threads. You can't click reply, change the subject as
it messes up the ?
 18:50Greg .. Brian, Michael (6)
liste.voip

[Freeswitch-users] phpmod compile error? - ? Please don't *hijack* threads...
please click new message and start a new thread. Thanks ?
 Aug 5mashudi, Brian (2)
liste.voip

[Freeswitch-users] video playback on FS? - ? btw don't *hijack* threads
please. thank you in advanced, best regard, mashudi Brian West
brian at freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com ?
 Jun 23Tim .. Michael, Chris (4)
liste.voip

[Freeswitch-users] Routing 911 calls? - ? p.s. Sorry to *hijack* the thread.
--- On Mon, 6/8/09, Michael Collins <msc at freeswitch.org> wrote: From:
Michael Collins <msc at freeswitch.org> Subject: Re ?
 Jun 8Lars .. Brian, Michael (13)
liste.voip

[Freeswitch-users] Total noob question? - ? please do not *hijack* threads..
Click new message and input the address
freeswitch-users at lists.freeswitch.org Thanks, Brian On May 7, 2009, at 8:13
PM, Lars Zeb wrote: I ?
 May 8wchao
liste.voip

[Freeswitch-users] Registration problem with multiple IP phones behind Linux
NAT firewa...? - ? is trying to *hijack* phone A's registration. The packet
capture was taken when running Freeswitch 1.0.4pre4, but I subsequently
upgraded to 1.0.4pre6 and it didn't ?
 May 4Chris, Brian, Anthony (3)
liste.voip

[Freeswitch-users] Audio delay when conferencing? - ? Also don't
*hijack* threads
please! :) /b On Apr 30, 2009, at 3:54 PM, Chris Fowler wrote: I'm using
FreeSWITCH (Build 13168M) and we're having intermittent ?
 Apr 30Chris, Brian (2)
liste.voip

[Freeswitch-users] mod_fifo uuid_transfer into mod_conference audio issue? - ?
PLEASE do not *hijack* threads. You clicked reply.. cleared the body and the
subject and sent the message. DO NOT DO THAT. Please click new message and
input freeswitch ?
 Apr 15Valentin, Brian, Anthony (3)
liste.voip

[Freeswitch-users] Nokia N800? - ? please do not *hijack* threads... you
clicked reply, changed the subject and body which causes it to thread your
message with the original posters thread. So please in the ?
 Apr 1
ConversationsSelect: All, None, Read, Unread, Starred, Unstarred

Regards,
   Raffaele


On Sat, Aug 8, 2009 at 18:50, Raffaele P. Guidi
<raffaele.p.guidi at gmail.com>wrote:

> Brian, I've read 8-9 of your emails asking to not hijack threads in threads
> where this it didn't happen (like this). I see this as a new thread with a
> single message (plus yours, of course) inside with "[Freeswitch-users] how
> to catch DTMF in Lua while a phrase macro is playing" as the subject - and I
> think is quite relevant to the message. I suspect you have something wrong
> in your email client (mine is gmail) or server and suggest you to give it a
> check.
>
> Regards,
>    Raffaele
>
> On Sat, Aug 8, 2009 at 17:59, Brian West <brian at freeswitch.org> wrote:
>
>> Examples exist in the scripts/lua folder called callback.lua.  Also
>> please DO NOT hijack threads.  You can't click reply, change the
>> subject as it messes up the mailing list threading and most mail
>> readers will thread it incorrectly.
>>
>> Thanks,
>> Brian
>>
>> On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines wrote:
>>
>> > in a lua script, i've tried using session:setInputCallback() to catch
>> > DTMF tones while a phrase macro is playing, but it doesn't seem to
>> > work.  the same callback _does_ catch DTMF when i use
>> > session:streamFile() to play something.  is there some other way to do
>> > it?
>> >
>> > below is an example of how i'm doing it now.  when voicemail/vm-
>> > mailbox_full.wav plays, i see key presses being printed to the
>> > console, but when the phrase plays, no key presses are displayed.
>> >
>> > function key_press(session, input_type, data)
>> >       if input_type == "dtmf" then
>> >               freeswitch.consoleLog("info", "Key pressed: " ..
>> data["digit"] ..
>> > "\n")
>> >       end
>> > end
>> >
>> > if session:ready() then
>> >       session:answer()
>> >       session:execute("sleep", "1000")
>> >       session:setInputCallback("key_press", "")
>> >       session:streamFile("voicemail/vm-mailbox_full.wav")
>> >       session:execute("phrase", "voicemail_menu,1:2:3:#")
>> > end
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From chad at apartmentlines.com  Sat Aug  8 10:16:19 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Sat, 8 Aug 2009 13:16:19 -0400
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
Message-ID: <566B33BA-4566-4544-9529-B577D26F6C87@apartmentlines.com>


On Aug 8, 2009, at 11:59 AM, Brian West wrote:

> Examples exist in the scripts/lua folder called callback.lua.

i read this example code, but it doesn't seem to really address my  
question.  if you re-read the code and explanation that i posted  
earlier, the input callback is working when i do something like  
session:streamFile("blah"), but *not* when i do something like  
session:execute("phrase", "blah,args")).  to be clear, the phrase  
plays just fine, but when i press keys during the phrase playback, the  
input callback doesn't get fired.  am i missing something obvious  
here?  you should be able to drop that code in that i provided earlier  
to verify what i'm talking about.

one thing i will add is that i'm using portaudio for making the call  
and sending the DTMF.  i'll try testing this out with a regular sip  
client and see if the same thing occurs...

> Also
> please DO NOT hijack threads.  You can't click reply, change the
> subject as it messes up the mailing list threading and most mail
> readers will thread it incorrectly.

sorry, i've never had that problem before -- i can certainly adjust  :)



From rlucente at teleuco.com  Sat Aug  8 10:25:51 2009
From: rlucente at teleuco.com (Rocco Lucente)
Date: Sat, 8 Aug 2009 19:25:51 +0200
Subject: [Freeswitch-users] question on att_xfer
Message-ID: <2969faac0908081025l41d33ed5r295b86635ca46853@mail.gmail.com>

hi, during a call made with att_xfer, you can resume the user A on ring? Let
me explain better, with bind_meta_app and att_xfer can do this: during a
bridge if B dials *1 (for example), A is parked and B can call C. After C to
answer, B can to dial # and return in conversation with A. B could resume A
while is calling C (for example because I noticed that I've dialed an
incorrect number)?
I tried with execute_on_ring, but without result. Any ideas?
Thank you and good job to all.
-- 
Rocco Lucente
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From pjintheusa at gmail.com  Sat Aug  8 10:40:33 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 10:40:33 -0700
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
In-Reply-To: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
References: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
Message-ID: <367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>

Hi there,

Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
INVITE) and only pass back the media to the network, or pass back
signaling also (SIP REFER)?

I know several suppliers who support SIP re INVITE but none that
support SIP REFER.

Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
and http://wiki.freeswitch.org/wiki/Bypass_Media



On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov at gmail.com> wrote:
> Good morning,
> This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
> through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
> AS does some logic and transfers call (or forward) out of Voip provider
> network to another PSTN number User2.
>
>
> This is call bridge
>
>
> ??? UA1???? (PSTN) -???????????????????????????????? ->? UA2 (PSTN)
> ????????????? -??????????????????????????????????????????? -
> ??????????????? -? (1)?????????????????????????????????? -? (4)
> ????????????????? ->???????? PSTN Gateway->
> ??????????????????????????? -??????????????????????? -
> ?????????????????????? (2) -??????????????????????? - (3)
> ????????????????????????? -> FreeSWITCH ->
>
>
> This is what I want to acomplish
> ??????????????????????????????????? (4)
> ??? UA1???? (PSTN) ------------------------------- ->? UA2 (PSTN)
> ??????????????????? -
> ???????????????????? ? - ? (1)
> ??????????????????????? ->? PSTN Gateway->
> ???????????????????????????? -??????????????????????? -
> ?????????????????????? (2)? -??????????????????????? - (3)
> ??????????????????????????? -> FreeSWITCH ->
>
>
> 1. Can it be implemented in FreeSWITCH?
> 2. Does anybody know Voip providers which support out of network call
> transfer/forwarding to PSTN?
>
> TIA
>
> -Vladimir Rodionov
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

From pjintheusa at gmail.com  Sat Aug  8 10:44:46 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 10:44:46 -0700
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
	<b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
	<b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>
Message-ID: <367751820908081044g67a85684hc34a0b242c67b6a4@mail.gmail.com>

Thanks very much for that - very help.

Why would loopback be considered "abuse"? What would be the downsides
of doing this?

On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker<rupa at rupa.com> wrote:
>
>
> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>
>> <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>
>
> That of course, should be:
>
> ?<action application="bridge" data="sofia/gw1/$1|sofia/gw2/$1..."/>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From stkn at freeswitch.org  Sat Aug  8 11:01:50 2009
From: stkn at freeswitch.org (Stefan Knoblich)
Date: Sat, 08 Aug 2009 20:01:50 +0200
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
 macro is playing
In-Reply-To: <b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>	<4A7D8C26.4020408@chandlerfamily.org.uk>	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
	<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
Message-ID: <4A7DBD8E.4090608@freeswitch.org>

His message was a direct reply to Alan Chandler's mail and Brian's mail client is working perfectly fine
(as are both of my mail clients).

Look at the Message-Id and In-Reply-To headers of the mails if you still think our clients are broken :)


stkn


Raffaele P. Guidi wrote:
> Brian, I've read 8-9 of your emails asking to not hijack threads in
> threads where this it didn't happen (like this). I see this as a new
> thread with a single message (plus yours, of course) inside with
> "[Freeswitch-users] how to catch DTMF in Lua while a phrase macro is
> playing" as the subject - and I think is quite relevant to the message.
> I suspect you have something wrong in your email client (mine is gmail)
> or server and suggest you to give it a check.
> 
> Regards,
>    Raffaele
> 
> On Sat, Aug 8, 2009 at 17:59, Brian West <brian at freeswitch.org
> <mailto:brian at freeswitch.org>> wrote:
> 
>     Examples exist in the scripts/lua folder called callback.lua.  Also
>     please DO NOT hijack threads.  You can't click reply, change the
>     subject as it messes up the mailing list threading and most mail
>     readers will thread it incorrectly.
> 
>     Thanks,
>     Brian
> 
>     On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines wrote:
> 
>     > in a lua script, i've tried using session:setInputCallback() to catch
>     > DTMF tones while a phrase macro is playing, but it doesn't seem to
>     > work.  the same callback _does_ catch DTMF when i use
>     > session:streamFile() to play something.  is there some other way to do
>     > it?
>     >
>     > below is an example of how i'm doing it now.  when voicemail/vm-
>     > mailbox_full.wav plays, i see key presses being printed to the
>     > console, but when the phrase plays, no key presses are displayed.
>     >
>     > function key_press(session, input_type, data)
>     >       if input_type == "dtmf" then
>     >               freeswitch.consoleLog("info", "Key pressed: " ..
>     data["digit"] ..
>     > "\n")
>     >       end
>     > end
>     >
>     > if session:ready() then
>     >       session:answer()
>     >       session:execute("sleep", "1000")
>     >       session:setInputCallback("key_press", "")
>     >       session:streamFile("voicemail/vm-mailbox_full.wav")
>     >       session:execute("phrase", "voicemail_menu,1:2:3:#")
>     > end
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
> 
> 
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
> 
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From william.suffill at gmail.com  Sat Aug  8 11:34:23 2009
From: william.suffill at gmail.com (William Suffill)
Date: Sat, 8 Aug 2009 14:34:23 -0400
Subject: [Freeswitch-users] mod_php needed
In-Reply-To: <191c3a030906150827t2e2e919bw396097bd637a0b91@mail.gmail.com>
References: <4A33E8A0.1070708@xpirio.com>
	<87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com> 
	<A2B4F48489739E439D8845296A2FD9C4147238@sun.Noble.local>
	<4A361708.8020808@xpirio.com> 
	<191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com> 
	<4A3660EB.4070203@xpirio.com>
	<191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com> 
	<6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com> 
	<191c3a030906150827t2e2e919bw396097bd637a0b91@mail.gmail.com>
Message-ID: <6b65470d0908081134x719bbef6r707644d877eee328@mail.gmail.com>

http://wiki.freeswitch.org/wiki/PHP_ESL

Formatting could use a bit of help just exported my working document
to MediaWiki. Hope to add some more information to ESL as I continue
to do more with it.

I'm going to start a page on fs_ivrd as well. Feedback welcome.

-- W

IRC: wsuff on freenode



From vladrodionov at gmail.com  Sat Aug  8 11:39:13 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Sat, 8 Aug 2009 11:39:13 -0700
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
In-Reply-To: <367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>
References: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
	<367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>
Message-ID: <3c233920908081139o769c1310yaee2857d4543da7d@mail.gmail.com>

Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty
sure it is doable because voxeo offers this
option for their Voice XML customers but I am not interested in a hosted
solution at the time - it is quite expensive. As far as I understood, Voip
provider MUST have pstn call transfer feature enabled by telecom provider
(AT&T for example) and this should work fine with SIP.

-Vladimir Rodionov


On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi there,
>
> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
> INVITE) and only pass back the media to the network, or pass back
> signaling also (SIP REFER)?
>
> I know several suppliers who support SIP re INVITE but none that
> support SIP REFER.
>
> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
> and http://wiki.freeswitch.org/wiki/Bypass_Media
>
>
>
> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov at gmail.com>
> wrote:
> > Good morning,
> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
> > through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
> > AS does some logic and transfers call (or forward) out of Voip provider
> > network to another PSTN number User2.
> >
> >
> > This is call bridge
> >
> >
> >     UA1     (PSTN) -                                 ->  UA2 (PSTN)
> >               -                                            -
> >                 -  (1)                                   -  (4)
> >                   ->         PSTN Gateway->
> >                             -                        -
> >                        (2) -                        - (3)
> >                           -> FreeSWITCH ->
> >
> >
> > This is what I want to acomplish
> >                                     (4)
> >     UA1     (PSTN) ------------------------------- ->  UA2 (PSTN)
> >                     -
> >                        -   (1)
> >                         ->  PSTN Gateway->
> >                              -                        -
> >                        (2)  -                        - (3)
> >                             -> FreeSWITCH ->
> >
> >
> > 1. Can it be implemented in FreeSWITCH?
> > 2. Does anybody know Voip providers which support out of network call
> > transfer/forwarding to PSTN?
> >
> > TIA
> >
> > -Vladimir Rodionov
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From raffaele.p.guidi at gmail.com  Sat Aug  8 11:48:28 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sat, 8 Aug 2009 20:48:28 +0200
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <4A7DBD8E.4090608@freeswitch.org>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
	<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
	<4A7DBD8E.4090608@freeswitch.org>
Message-ID: <b0a8792f0908081148l2174443fhcd210e66aec15bf@mail.gmail.com>

Uh, oh, ok. Message-Id and In-Reply-To headers don't mean too much in a
gmail world (I think gmail does a better job than any other mail client).
Well, at least I understood what Brian was talking about, thanks.
Regards,
   Raffaele

On Sat, Aug 8, 2009 at 20:01, Stefan Knoblich <stkn at freeswitch.org> wrote:

> His message was a direct reply to Alan Chandler's mail and Brian's mail
> client is working perfectly fine
> (as are both of my mail clients).
>
> Look at the Message-Id and In-Reply-To headers of the mails if you still
> think our clients are broken :)
>
>
> stkn
>
>
> Raffaele P. Guidi wrote:
> > Brian, I've read 8-9 of your emails asking to not hijack threads in
> > threads where this it didn't happen (like this). I see this as a new
> > thread with a single message (plus yours, of course) inside with
> > "[Freeswitch-users] how to catch DTMF in Lua while a phrase macro is
> > playing" as the subject - and I think is quite relevant to the message.
> > I suspect you have something wrong in your email client (mine is gmail)
> > or server and suggest you to give it a check.
> >
> > Regards,
> >    Raffaele
> >
> > On Sat, Aug 8, 2009 at 17:59, Brian West <brian at freeswitch.org
> > <mailto:brian at freeswitch.org>> wrote:
> >
> >     Examples exist in the scripts/lua folder called callback.lua.  Also
> >     please DO NOT hijack threads.  You can't click reply, change the
> >     subject as it messes up the mailing list threading and most mail
> >     readers will thread it incorrectly.
> >
> >     Thanks,
> >     Brian
> >
> >     On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines wrote:
> >
> >     > in a lua script, i've tried using session:setInputCallback() to
> catch
> >     > DTMF tones while a phrase macro is playing, but it doesn't seem to
> >     > work.  the same callback _does_ catch DTMF when i use
> >     > session:streamFile() to play something.  is there some other way to
> do
> >     > it?
> >     >
> >     > below is an example of how i'm doing it now.  when voicemail/vm-
> >     > mailbox_full.wav plays, i see key presses being printed to the
> >     > console, but when the phrase plays, no key presses are displayed.
> >     >
> >     > function key_press(session, input_type, data)
> >     >       if input_type == "dtmf" then
> >     >               freeswitch.consoleLog("info", "Key pressed: " ..
> >     data["digit"] ..
> >     > "\n")
> >     >       end
> >     > end
> >     >
> >     > if session:ready() then
> >     >       session:answer()
> >     >       session:execute("sleep", "1000")
> >     >       session:setInputCallback("key_press", "")
> >     >       session:streamFile("voicemail/vm-mailbox_full.wav")
> >     >       session:execute("phrase", "voicemail_menu,1:2:3:#")
> >     > end
> >     >
> >     > _______________________________________________
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vladrodionov at gmail.com  Sat Aug  8 11:52:09 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Sat, 8 Aug 2009 11:52:09 -0700
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
In-Reply-To: <3c233920908081139o769c1310yaee2857d4543da7d@mail.gmail.com>
References: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
	<367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>
	<3c233920908081139o769c1310yaee2857d4543da7d@mail.gmail.com>
Message-ID: <3c233920908081152r70acceeoe63cf15f8e1a40dd@mail.gmail.com>

Actually, this is what I need

http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect

Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN
number?

-Vladimir Rodionov


On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov
<vladrodionov at gmail.com>wrote:

> Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty
> sure it is doable because voxeo offers this
> option for their Voice XML customers but I am not interested in a hosted
> solution at the time - it is quite expensive. As far as I understood, Voip
> provider MUST have pstn call transfer feature enabled by telecom provider
> (AT&T for example) and this should work fine with SIP.
>
> -Vladimir Rodionov
>
>
> On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> Hi there,
>>
>> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
>> INVITE) and only pass back the media to the network, or pass back
>> signaling also (SIP REFER)?
>>
>> I know several suppliers who support SIP re INVITE but none that
>> support SIP REFER.
>>
>> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
>> and http://wiki.freeswitch.org/wiki/Bypass_Media
>>
>>
>>
>> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov at gmail.com>
>> wrote:
>> > Good morning,
>> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
>> > through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
>> > AS does some logic and transfers call (or forward) out of Voip provider
>> > network to another PSTN number User2.
>> >
>> >
>> > This is call bridge
>> >
>> >
>> >     UA1     (PSTN) -                                 ->  UA2 (PSTN)
>> >               -                                            -
>> >                 -  (1)                                   -  (4)
>> >                   ->         PSTN Gateway->
>> >                             -                        -
>> >                        (2) -                        - (3)
>> >                           -> FreeSWITCH ->
>> >
>> >
>> > This is what I want to acomplish
>> >                                     (4)
>> >     UA1     (PSTN) ------------------------------- ->  UA2 (PSTN)
>> >                     -
>> >                        -   (1)
>> >                         ->  PSTN Gateway->
>> >                              -                        -
>> >                        (2)  -                        - (3)
>> >                             -> FreeSWITCH ->
>> >
>> >
>> > 1. Can it be implemented in FreeSWITCH?
>> > 2. Does anybody know Voip providers which support out of network call
>> > transfer/forwarding to PSTN?
>> >
>> > TIA
>> >
>> > -Vladimir Rodionov
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From pjintheusa at gmail.com  Sat Aug  8 12:00:39 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 12:00:39 -0700
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
In-Reply-To: <3c233920908081152r70acceeoe63cf15f8e1a40dd@mail.gmail.com>
References: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
	<367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>
	<3c233920908081139o769c1310yaee2857d4543da7d@mail.gmail.com>
	<3c233920908081152r70acceeoe63cf15f8e1a40dd@mail.gmail.com>
Message-ID: <367751820908081200y1ea230c7o12377ffe45bc75eb@mail.gmail.com>

Are your calls coming in on TDM or SIP trunks? Are your calls answered
by FreeSWITCH before you need to redirect them?

On Sat, Aug 8, 2009 at 11:52 AM, Vladimir
Rodionov<vladrodionov at gmail.com> wrote:
> Actually, this is what I need
>
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
>
> Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN
> number?
>
> -Vladimir Rodionov
>
>
> On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <vladrodionov at gmail.com>
> wrote:
>>
>> Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am pretty
>> sure it is doable because voxeo offers this
>> option for their Voice XML customers but I am not interested in a hosted
>> solution at the time - it is quite expensive. As far as I understood, Voip
>> provider MUST have pstn call transfer feature enabled by telecom provider
>> (AT&T for example) and this should work fine with SIP.
>>
>> -Vladimir Rodionov
>>
>>
>> On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa at gmail.com>
>> wrote:
>>>
>>> Hi there,
>>>
>>> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
>>> INVITE) and only pass back the media to the network, or pass back
>>> signaling also (SIP REFER)?
>>>
>>> I know several suppliers who support SIP re INVITE but none that
>>> support SIP REFER.
>>>
>>> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
>>> and http://wiki.freeswitch.org/wiki/Bypass_Media
>>>
>>>
>>>
>>> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov at gmail.com>
>>> wrote:
>>> > Good morning,
>>> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls goes
>>> > through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
>>> > AS does some logic and transfers call (or forward) out of Voip provider
>>> > network to another PSTN number User2.
>>> >
>>> >
>>> > This is call bridge
>>> >
>>> >
>>> > ??? UA1???? (PSTN) -???????????????????????????????? ->? UA2 (PSTN)
>>> > ????????????? -??????????????????????????????????????????? -
>>> > ??????????????? -? (1)?????????????????????????????????? -? (4)
>>> > ????????????????? ->???????? PSTN Gateway->
>>> > ??????????????????????????? -??????????????????????? -
>>> > ?????????????????????? (2) -??????????????????????? - (3)
>>> > ????????????????????????? -> FreeSWITCH ->
>>> >
>>> >
>>> > This is what I want to acomplish
>>> > ??????????????????????????????????? (4)
>>> > ??? UA1???? (PSTN) ------------------------------- ->? UA2 (PSTN)
>>> > ??????????????????? -
>>> > ???????????????????? ? - ? (1)
>>> > ??????????????????????? ->? PSTN Gateway->
>>> > ???????????????????????????? -??????????????????????? -
>>> > ?????????????????????? (2)? -??????????????????????? - (3)
>>> > ??????????????????????????? -> FreeSWITCH ->
>>> >
>>> >
>>> > 1. Can it be implemented in FreeSWITCH?
>>> > 2. Does anybody know Voip providers which support out of network call
>>> > transfer/forwarding to PSTN?
>>> >
>>> > TIA
>>> >
>>> > -Vladimir Rodionov
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From diego.viola at gmail.com  Sat Aug  8 12:06:57 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 8 Aug 2009 15:06:57 -0400
Subject: [Freeswitch-users] mod_php needed
In-Reply-To: <6b65470d0908081134x719bbef6r707644d877eee328@mail.gmail.com>
References: <4A33E8A0.1070708@xpirio.com>
	<87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147238@sun.Noble.local>
	<4A361708.8020808@xpirio.com>
	<191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com>
	<4A3660EB.4070203@xpirio.com>
	<191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com>
	<6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com>
	<191c3a030906150827t2e2e919bw396097bd637a0b91@mail.gmail.com>
	<6b65470d0908081134x719bbef6r707644d877eee328@mail.gmail.com>
Message-ID: <86a32abc0908081206l6932b87bo5b05098c4d2ffe5a@mail.gmail.com>

Damn, looks like we need to learn some formatting =D.

I made some changes on that article to fix the formatting a bit, lets take
care of the wiki as we care about the software :).

On Sat, Aug 8, 2009 at 2:34 PM, William Suffill
<william.suffill at gmail.com>wrote:

> http://wiki.freeswitch.org/wiki/PHP_ESL
>
> Formatting could use a bit of help just exported my working document
> to MediaWiki. Hope to add some more information to ESL as I continue
> to do more with it.
>
> I'm going to start a page on fs_ivrd as well. Feedback welcome.
>
> -- W
>
> IRC: wsuff on freenode
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From chad at apartmentlines.com  Sat Aug  8 12:09:48 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Sat, 8 Aug 2009 15:09:48 -0400
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <b0a8792f0908081148l2174443fhcd210e66aec15bf@mail.gmail.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
	<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
	<4A7DBD8E.4090608@freeswitch.org>
	<b0a8792f0908081148l2174443fhcd210e66aec15bf@mail.gmail.com>
Message-ID: <98727038-1A37-43A4-BFC4-F624451772AF@apartmentlines.com>

does anybody else find it amusing that my thread about Lua scripting  
was hijacked by people talking about hijacking threads??  ;)

/me waits patiently for discussion on the topic


On Aug 8, 2009, at 2:48 PM, Raffaele P. Guidi wrote:

> Uh, oh, ok. Message-Id and In-Reply-To headers don't mean too much  
> in a gmail world (I think gmail does a better job than any other  
> mail client). Well, at least I understood what Brian was talking  
> about, thanks.
>
> Regards,
>    Raffaele
>
> On Sat, Aug 8, 2009 at 20:01, Stefan Knoblich <stkn at freeswitch.org>  
> wrote:
> His message was a direct reply to Alan Chandler's mail and Brian's  
> mail client is working perfectly fine
> (as are both of my mail clients).
>
> Look at the Message-Id and In-Reply-To headers of the mails if you  
> still think our clients are broken :)
>
>
> stkn
>
>
> Raffaele P. Guidi wrote:
> > Brian, I've read 8-9 of your emails asking to not hijack threads in
> > threads where this it didn't happen (like this). I see this as a new
> > thread with a single message (plus yours, of course) inside with
> > "[Freeswitch-users] how to catch DTMF in Lua while a phrase macro is
> > playing" as the subject - and I think is quite relevant to the  
> message.
> > I suspect you have something wrong in your email client (mine is  
> gmail)
> > or server and suggest you to give it a check.
> >
> > Regards,
> >    Raffaele
> >
> > On Sat, Aug 8, 2009 at 17:59, Brian West <brian at freeswitch.org
> > <mailto:brian at freeswitch.org>> wrote:
> >
> >     Examples exist in the scripts/lua folder called callback.lua.   
> Also
> >     please DO NOT hijack threads.  You can't click reply, change the
> >     subject as it messes up the mailing list threading and most mail
> >     readers will thread it incorrectly.
> >
> >     Thanks,
> >     Brian
> >
> >     On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines  
> wrote:
> >
> >     > in a lua script, i've tried using session:setInputCallback()  
> to catch
> >     > DTMF tones while a phrase macro is playing, but it doesn't  
> seem to
> >     > work.  the same callback _does_ catch DTMF when i use
> >     > session:streamFile() to play something.  is there some other  
> way to do
> >     > it?
> >     >
> >     > below is an example of how i'm doing it now.  when voicemail/ 
> vm-
> >     > mailbox_full.wav plays, i see key presses being printed to the
> >     > console, but when the phrase plays, no key presses are  
> displayed.
> >     >
> >     > function key_press(session, input_type, data)
> >     >       if input_type == "dtmf" then
> >     >               freeswitch.consoleLog("info", "Key pressed: " ..
> >     data["digit"] ..
> >     > "\n")
> >     >       end
> >     > end
> >     >
> >     > if session:ready() then
> >     >       session:answer()
> >     >       session:execute("sleep", "1000")
> >     >       session:setInputCallback("key_press", "")
> >     >       session:streamFile("voicemail/vm-mailbox_full.wav")
> >     >       session:execute("phrase", "voicemail_menu,1:2:3:#")
> >     > end
> >     >
> >     > _______________________________________________
> >     > FreeSWITCH-users mailing list
> >     > FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     >
> >     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     > http://www.freeswitch.org
> >
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >  
> ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From vladrodionov at gmail.com  Sat Aug  8 12:12:13 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Sat, 8 Aug 2009 12:12:13 -0700
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
In-Reply-To: <367751820908081200y1ea230c7o12377ffe45bc75eb@mail.gmail.com>
References: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
	<367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>
	<3c233920908081139o769c1310yaee2857d4543da7d@mail.gmail.com>
	<3c233920908081152r70acceeoe63cf15f8e1a40dd@mail.gmail.com>
	<367751820908081200y1ea230c7o12377ffe45bc75eb@mail.gmail.com>
Message-ID: <3c233920908081212s29f6d42ft210755fe50282cba@mail.gmail.com>

A call is coming on SIP trunk. From PSTN.  I does not need to be answered,
actually - I need to do some logic before redirecting call but I can answer
call as well It won't break the app logic.

-Vladimir Rodionov

On Sat, Aug 8, 2009 at 12:00 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Are your calls coming in on TDM or SIP trunks? Are your calls answered
> by FreeSWITCH before you need to redirect them?
>
> On Sat, Aug 8, 2009 at 11:52 AM, Vladimir
> Rodionov<vladrodionov at gmail.com> wrote:
> > Actually, this is what I need
> >
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
> >
> > Will it work with PSTN? Can I redirect incoming PSTN call to another PSTN
> > number?
> >
> > -Vladimir Rodionov
> >
> >
> > On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <
> vladrodionov at gmail.com>
> > wrote:
> >>
> >> Yes, I need SIP-REFER (pass back media and signaling to PSTN). I am
> pretty
> >> sure it is doable because voxeo offers this
> >> option for their Voice XML customers but I am not interested in a hosted
> >> solution at the time - it is quite expensive. As far as I understood,
> Voip
> >> provider MUST have pstn call transfer feature enabled by telecom
> provider
> >> (AT&T for example) and this should work fine with SIP.
> >>
> >> -Vladimir Rodionov
> >>
> >>
> >> On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa at gmail.com>
> >> wrote:
> >>>
> >>> Hi there,
> >>>
> >>> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
> >>> INVITE) and only pass back the media to the network, or pass back
> >>> signaling also (SIP REFER)?
> >>>
> >>> I know several suppliers who support SIP re INVITE but none that
> >>> support SIP REFER.
> >>>
> >>> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
> >>> and http://wiki.freeswitch.org/wiki/Bypass_Media
> >>>
> >>>
> >>>
> >>> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<
> vladrodionov at gmail.com>
> >>> wrote:
> >>> > Good morning,
> >>> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID). Calls
> goes
> >>> > through PSTN Gateway (1) to freeSWITCH application server (AS) (2).
> >>> > AS does some logic and transfers call (or forward) out of Voip
> provider
> >>> > network to another PSTN number User2.
> >>> >
> >>> >
> >>> > This is call bridge
> >>> >
> >>> >
> >>> >     UA1     (PSTN) -                                 ->  UA2 (PSTN)
> >>> >               -                                            -
> >>> >                 -  (1)                                   -  (4)
> >>> >                   ->         PSTN Gateway->
> >>> >                             -                        -
> >>> >                        (2) -                        - (3)
> >>> >                           -> FreeSWITCH ->
> >>> >
> >>> >
> >>> > This is what I want to acomplish
> >>> >                                     (4)
> >>> >     UA1     (PSTN) ------------------------------- ->  UA2 (PSTN)
> >>> >                     -
> >>> >                        -   (1)
> >>> >                         ->  PSTN Gateway->
> >>> >                              -                        -
> >>> >                        (2)  -                        - (3)
> >>> >                             -> FreeSWITCH ->
> >>> >
> >>> >
> >>> > 1. Can it be implemented in FreeSWITCH?
> >>> > 2. Does anybody know Voip providers which support out of network call
> >>> > transfer/forwarding to PSTN?
> >>> >
> >>> > TIA
> >>> >
> >>> > -Vladimir Rodionov
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From vladrodionov at gmail.com  Sat Aug  8 12:21:38 2009
From: vladrodionov at gmail.com (Vladimir Rodionov)
Date: Sat, 8 Aug 2009 12:21:38 -0700
Subject: [Freeswitch-users] Lua on Windows and additional modules
In-Reply-To: <0EE782AE-FC28-4029-AA01-76D87BC08069@jerris.com>
References: <3c233920908061755h5d17aa0as3fb24743215a8298@mail.gmail.com>
	<0EE782AE-FC28-4029-AA01-76D87BC08069@jerris.com>
Message-ID: <3c233920908081221m58acbcdake04a2353b7fbb401@mail.gmail.com>

Thank you all, but I decided to go with ESL and Java.

On Fri, Aug 7, 2009 at 11:11 PM, Michael Jerris <mike at jerris.com> wrote:

> check the sample config files for options to specify these:
>
>
> http://fisheye.freeswitch.org/browse/FreeSWITCH/conf/autoload_configs/lua.conf.xml?r=10747
>
>
> On Aug 6, 2009, at 8:55 PM, Vladimir Rodionov wrote:
>
> > Good evening,
> > This is newbie question.
> >
> > The FreeSWITCH lua module does not support sockets and sql out of
> > box that is why
> > I just installed LuaBinaries (including socket, sql modules). My dev
> > environment is Win XP not Linux/Unix.
> >
> > I am trying to understand what will happen when lua_module get this:
> >
> > require "socket" or
> > require "luasql.mysql"
> > ?
> >
> > How does lua_module look up additional lua modules on Windows
> > platform?
> >
> > Do I have to set some env variables?
> >
> > TIA
> > -Vladimir Rodionov
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From raffaele.p.guidi at gmail.com  Sat Aug  8 12:42:06 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sat, 8 Aug 2009 21:42:06 +0200
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <98727038-1A37-43A4-BFC4-F624451772AF@apartmentlines.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
	<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
	<4A7DBD8E.4090608@freeswitch.org>
	<b0a8792f0908081148l2174443fhcd210e66aec15bf@mail.gmail.com>
	<98727038-1A37-43A4-BFC4-F624451772AF@apartmentlines.com>
Message-ID: <b0a8792f0908081242yfb19d46ofefd68e52145554a@mail.gmail.com>

I did that. And I do find it quite amusing :D

On Sat, Aug 8, 2009 at 21:09, Chad Phillips -- Apartment Lines <
chad at apartmentlines.com> wrote:

> does anybody else find it amusing that my thread about Lua scripting was
> hijacked by people talking about hijacking threads??  ;)
> /me waits patiently for discussion on the topic
>
>
>
> On Aug 8, 2009, at 2:48 PM, Raffaele P. Guidi wrote:
>
> Uh, oh, ok. Message-Id and In-Reply-To headers don't mean too much in a
> gmail world (I think gmail does a better job than any other mail client).
> Well, at least I understood what Brian was talking about, thanks.
> Regards,
>    Raffaele
>
> On Sat, Aug 8, 2009 at 20:01, Stefan Knoblich <stkn at freeswitch.org> wrote:
>
>> His message was a direct reply to Alan Chandler's mail and Brian's mail
>> client is working perfectly fine
>> (as are both of my mail clients).
>>
>> Look at the Message-Id and In-Reply-To headers of the mails if you still
>> think our clients are broken :)
>>
>>
>> stkn
>>
>>
>> Raffaele P. Guidi wrote:
>> > Brian, I've read 8-9 of your emails asking to not hijack threads in
>> > threads where this it didn't happen (like this). I see this as a new
>> > thread with a single message (plus yours, of course) inside with
>> > "[Freeswitch-users] how to catch DTMF in Lua while a phrase macro is
>> > playing" as the subject - and I think is quite relevant to the message.
>> > I suspect you have something wrong in your email client (mine is gmail)
>> > or server and suggest you to give it a check.
>> >
>> > Regards,
>> >    Raffaele
>> >
>> > On Sat, Aug 8, 2009 at 17:59, Brian West <brian at freeswitch.org
>> > <mailto:brian at freeswitch.org>> wrote:
>> >
>> >     Examples exist in the scripts/lua folder called callback.lua.  Also
>> >     please DO NOT hijack threads.  You can't click reply, change the
>> >     subject as it messes up the mailing list threading and most mail
>> >     readers will thread it incorrectly.
>> >
>> >     Thanks,
>> >     Brian
>> >
>> >     On Aug 8, 2009, at 10:15 AM, Chad Phillips -- Apartment Lines wrote:
>> >
>> >     > in a lua script, i've tried using session:setInputCallback() to
>> catch
>> >     > DTMF tones while a phrase macro is playing, but it doesn't seem to
>> >     > work.  the same callback _does_ catch DTMF when i use
>> >     > session:streamFile() to play something.  is there some other way
>> to do
>> >     > it?
>> >     >
>> >     > below is an example of how i'm doing it now.  when voicemail/vm-
>> >     > mailbox_full.wav plays, i see key presses being printed to the
>> >     > console, but when the phrase plays, no key presses are displayed.
>> >     >
>> >     > function key_press(session, input_type, data)
>> >     >       if input_type == "dtmf" then
>> >     >               freeswitch.consoleLog("info", "Key pressed: " ..
>> >     data["digit"] ..
>> >     > "\n")
>> >     >       end
>> >     > end
>> >     >
>> >     > if session:ready() then
>> >     >       session:answer()
>> >     >       session:execute("sleep", "1000")
>> >     >       session:setInputCallback("key_press", "")
>> >     >       session:streamFile("voicemail/vm-mailbox_full.wav")
>> >     >       session:execute("phrase", "voicemail_menu,1:2:3:#")
>> >     > end
>> >     >
>> >     > _______________________________________________
>> >     > FreeSWITCH-users mailing list
>> >     > FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     >
>> >     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >     > http://www.freeswitch.org
>> >
>> >
>> >     _______________________________________________
>> >     FreeSWITCH-users mailing list
>> >     FreeSWITCH-users at lists.freeswitch.org
>> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >     UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >     http://www.freeswitch.org
>> >
>> >
>> >
>> > ------------------------------------------------------------------------
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Sat Aug  8 12:59:10 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 15:59:10 -0400
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
In-Reply-To: <a37510170908080453m1e98ae3l544e3d8b301605e1@mail.gmail.com>
References: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>
	<367751820908080348p6165b113kc3be468cb32fabb3@mail.gmail.com>
	<a37510170908080453m1e98ae3l544e3d8b301605e1@mail.gmail.com>
Message-ID: <1F19675C-E556-4AF0-88C1-B5652D8C635F@jerris.com>

how many does it stop at?  is it the same number each time?

Mike

On Aug 8, 2009, at 7:53 AM, Benedikt Fraunhofer wrote:

> Hi Phillip,
>
> 2009/8/8 Phillip Jones <pjintheusa at gmail.com>:
>
>> Not sure whether this helps but test this without set bypass_media.  
>> In
>> my setup I have noticed the leg A session ends when bypass_media is
>> true. Call/bridge continue successfully.
>
> thx for that hint. unfortunately we can't do that due to the high
> volume we anticipate
> and i think i already tried that.
>
> Note that it works in exactly this setup  (with the uuid_media()-call)
> if there are only very few calls handled. It just starts to refuse
> work once enough jobs are scheduled and not executed.
>
> Cheers
>  Benedikt




From mike at jerris.com  Sat Aug  8 13:00:45 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:00:45 -0400
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <367751820908081044g67a85684hc34a0b242c67b6a4@mail.gmail.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
	<b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
	<b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>
	<367751820908081044g67a85684hc34a0b242c67b6a4@mail.gmail.com>
Message-ID: <F4D7EE24-6B62-4A46-83CB-3E412D3120D5@jerris.com>

loopback ends up using extra threads which we are only able to drop  
later in certain situations so it will decrease your total amount of  
calls you can do if your not careful with them.

Mike

On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:

> Thanks very much for that - very help.
>
> Why would loopback be considered "abuse"? What would be the downsides
> of doing this?
>
> On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker<rupa at rupa.com> wrote:
>>
>>
>> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>
>>> <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>
>>
>> That of course, should be:
>>
>>  <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$1..."/>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Sat Aug  8 13:04:07 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:04:07 -0400
Subject: [Freeswitch-users] Call transfer/forward PSTN-SIP-PSTN
In-Reply-To: <3c233920908081212s29f6d42ft210755fe50282cba@mail.gmail.com>
References: <3c233920908080926i7a1a922aq7855567d09834afd@mail.gmail.com>
	<367751820908081040i1bf12f0bh1c4017041097fd38@mail.gmail.com>
	<3c233920908081139o769c1310yaee2857d4543da7d@mail.gmail.com>
	<3c233920908081152r70acceeoe63cf15f8e1a40dd@mail.gmail.com>
	<367751820908081200y1ea230c7o12377ffe45bc75eb@mail.gmail.com>
	<3c233920908081212s29f6d42ft210755fe50282cba@mail.gmail.com>
Message-ID: <6C03C620-C2BB-4B60-9EE2-4D014736F9CF@jerris.com>

I don't know of any sip carriers who will let you do refer.  you will  
need to find a carrier who supports it.  FreeSWITCH will have no  
problem sending it but I doubt you will find a carrier who will let  
you do it easily.

Mike

On Aug 8, 2009, at 3:12 PM, Vladimir Rodionov wrote:

> A call is coming on SIP trunk. From PSTN.  I does not need to be  
> answered, actually - I need to do some logic before redirecting call  
> but I can answer call as well It won't break the app logic.
>
> -Vladimir Rodionov
>
> On Sat, Aug 8, 2009 at 12:00 PM, Phillip Jones  
> <pjintheusa at gmail.com> wrote:
> Are your calls coming in on TDM or SIP trunks? Are your calls answered
> by FreeSWITCH before you need to redirect them?
>
> On Sat, Aug 8, 2009 at 11:52 AM, Vladimir
> Rodionov<vladrodionov at gmail.com> wrote:
> > Actually, this is what I need
> >
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_redirect
> >
> > Will it work with PSTN? Can I redirect incoming PSTN call to  
> another PSTN
> > number?
> >
> > -Vladimir Rodionov
> >
> >
> > On Sat, Aug 8, 2009 at 11:39 AM, Vladimir Rodionov <vladrodionov at gmail.com 
> >
> > wrote:
> >>
> >> Yes, I need SIP-REFER (pass back media and signaling to PSTN). I  
> am pretty
> >> sure it is doable because voxeo offers this
> >> option for their Voice XML customers but I am not interested in a  
> hosted
> >> solution at the time - it is quite expensive. As far as I  
> understood, Voip
> >> provider MUST have pstn call transfer feature enabled by telecom  
> provider
> >> (AT&T for example) and this should work fine with SIP.
> >>
> >> -Vladimir Rodionov
> >>
> >>
> >> On Sat, Aug 8, 2009 at 10:40 AM, Phillip Jones <pjintheusa at gmail.com 
> >
> >> wrote:
> >>>
> >>> Hi there,
> >>>
> >>> Are you wanting to keep the SIP signaling in FreeSWITCH (SIP re
> >>> INVITE) and only pass back the media to the network, or pass back
> >>> signaling also (SIP REFER)?
> >>>
> >>> I know several suppliers who support SIP re INVITE but none that
> >>> support SIP REFER.
> >>>
> >>> Check out http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_deflect
> >>> and http://wiki.freeswitch.org/wiki/Bypass_Media
> >>>
> >>>
> >>>
> >>> On Sat, Aug 8, 2009 at 9:26 AM, Vladimir Rodionov<vladrodionov at gmail.com 
> >
> >>> wrote:
> >>> > Good morning,
> >>> > This is the scenario: User 1 (PSTN) calls freeSWITCH (DID).  
> Calls goes
> >>> > through PSTN Gateway (1) to freeSWITCH application server (AS)  
> (2).
> >>> > AS does some logic and transfers call (or forward) out of Voip  
> provider
> >>> > network to another PSTN number User2.
> >>> >
> >>> >
> >>> > This is call bridge
> >>> >
> >>> >
> >>> >     UA1     (PSTN) -                                 ->  UA2  
> (PSTN)
> >>> >               -                                            -
> >>> >                 -  (1)                                   -  (4)
> >>> >                   ->         PSTN Gateway->
> >>> >                             -                        -
> >>> >                        (2) -                        - (3)
> >>> >                           -> FreeSWITCH ->
> >>> >
> >>> >
> >>> > This is what I want to acomplish
> >>> >                                     (4)
> >>> >     UA1     (PSTN) ------------------------------- ->  UA2  
> (PSTN)
> >>> >                     -
> >>> >                        -   (1)
> >>> >                         ->  PSTN Gateway->
> >>> >                              -                        -
> >>> >                        (2)  -                        - (3)
> >>> >                             -> FreeSWITCH ->
> >>> >
> >>> >
> >>> > 1. Can it be implemented in FreeSWITCH?
> >>> > 2. Does anybody know Voip providers which support out of  
> network call
> >>> > transfer/forwarding to PSTN?
> >>> >
> >>> > TIA
> >>> >
> >>> > -Vladimir Rodionov
> >>> >
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From mike at jerris.com  Sat Aug  8 13:16:39 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:16:39 -0400
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
Message-ID: <32E54334-B499-4FA7-974B-705D8713A663@jerris.com>

use the SayPhrase method of sessin instead of executing the phrase  
application then your input callback should work as expected.

Mike

On Aug 8, 2009, at 11:15 AM, Chad Phillips -- Apartment Lines wrote:

> in a lua script, i've tried using session:setInputCallback() to catch
> DTMF tones while a phrase macro is playing, but it doesn't seem to
> work.  the same callback _does_ catch DTMF when i use
> session:streamFile() to play something.  is there some other way to do
> it?
>
> below is an example of how i'm doing it now.  when voicemail/vm-
> mailbox_full.wav plays, i see key presses being printed to the
> console, but when the phrase plays, no key presses are displayed.
>
> function key_press(session, input_type, data)
> 	if input_type == "dtmf" then
> 		freeswitch.consoleLog("info", "Key pressed: " .. data["digit"] ..
> "\n")
> 	end
> end
>
> if session:ready() then
> 	session:answer()
> 	session:execute("sleep", "1000")
> 	session:setInputCallback("key_press", "")
> 	session:streamFile("voicemail/vm-mailbox_full.wav")
> 	session:execute("phrase", "voicemail_menu,1:2:3:#")
> end




From mike at jerris.com  Sat Aug  8 13:19:48 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:19:48 -0400
Subject: [Freeswitch-users] FreeSwitch doesn't play music on hold
	forbriged channel
In-Reply-To: <EC1055E8A337451F9F5ADC7414BDFDD9@abisoft.biz>
References: <EC1055E8A337451F9F5ADC7414BDFDD9@abisoft.biz>
Message-ID: <E0DF0981-0AA7-4321-87E5-9AA9E7FFC580@jerris.com>

What do the debug logs on fs say when you try to put the call on hold?

Mike

On Aug 6, 2009, at 1:01 PM, Kozak Vladimir wrote:

>
> The scenario is the following:
> FS User A dial an extension
> Extention opens outbound socket channel to my application
> My application bridges the call to FS User B
> The application check for CHANNEL_BRIDGED event and stores Other-leg- 
> unique-id
> The application sends hold to the bridged channel using SendMsg with  
> Other-leg-unique-id
> User B is placed on hold but no music on hold is played to the  
> caller (User A)
>
>
> I have outbound socket channel and the following sequence of  
> commands/event:
> listening on [any] 8084 ...
> connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru  
> [172.26.200.250] 34000
> connect
>

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From mike at jerris.com  Sat Aug  8 13:21:11 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:21:11 -0400
Subject: [Freeswitch-users] State of originated call
In-Reply-To: <c55d46950908071644m7604eb57jf5018c2b378d3255@mail.gmail.com>
References: <c55d46950908071644m7604eb57jf5018c2b378d3255@mail.gmail.com>
Message-ID: <1E164379-D432-4309-8381-A2932868079C@jerris.com>

typically when these questions are asked we find people really want to  
be doing it the way these signals are automatically passed across.   
Can you describe what your trying to do a bit more?

Mike

On Aug 7, 2009, at 7:44 PM, Max Bridgewater wrote:

> Hi,
>
> using javascript, i do originate the call this way:
>
> Session s= new Session(originateStr);
>
> From this point, is it possible to know what states the call is  
> going through? In a previous message it was suggested that  
> variable_originate_disposition would give me the response code. Now,  
> how to i use this in practice in a script?
> How do i for instance retrieve a 180 response code when rining is  
> hapening on the remote end?
>
> Thanks,
> Max.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From pjintheusa at gmail.com  Sat Aug  8 13:24:45 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 13:24:45 -0700
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <F4D7EE24-6B62-4A46-83CB-3E412D3120D5@jerris.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
	<b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
	<b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>
	<367751820908081044g67a85684hc34a0b242c67b6a4@mail.gmail.com>
	<F4D7EE24-6B62-4A46-83CB-3E412D3120D5@jerris.com>
Message-ID: <367751820908081324r44c7ccf5lfc8ad78c96facb69@mail.gmail.com>

Mike/Rupa ,

Thanks for your help on this. So I am correct that summarizing that
FreeSWITCH does not really support fail over and multiple call
destinations because the same mechanism is used to achieve both? And
that loopback as a solution is possible but not recommended?

Is there any other solution to this? Perhaps a second FS box in the mix?

Phil


On Sat, Aug 8, 2009 at 1:00 PM, Michael Jerris<mike at jerris.com> wrote:
> loopback ends up using extra threads which we are only able to drop
> later in certain situations so it will decrease your total amount of
> calls you can do if your not careful with them.
>
> Mike
>
> On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:
>
>> Thanks very much for that - very help.
>>
>> Why would loopback be considered "abuse"? What would be the downsides
>> of doing this?
>>
>> On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker<rupa at rupa.com> wrote:
>>>
>>>
>>> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker <rupa at rupa.com> wrote:
>>>>
>>>> <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>
>>>
>>> That of course, should be:
>>>
>>> ?<action application="bridge" data="sofia/gw1/$1|sofia/gw2/$1..."/>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mike at jerris.com  Sat Aug  8 13:24:55 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:24:55 -0400
Subject: [Freeswitch-users] DTMF disable a few secs after call starts
In-Reply-To: <0B930F23C4A24E3CB0015945EA06C8B7@ws4>
References: <0B930F23C4A24E3CB0015945EA06C8B7@ws4>
Message-ID: <B1AF3C1E-6235-457A-AFF3-425F24BEB68E@jerris.com>

you would have to write a module in c that hooks the dtmf and throws  
it away.  This would work similar to how bind_meta works.  That being  
said, this is all much easier if your using 2833 as you don't need to  
go to the extra cpu of detecting tones.

Mike

On Aug 8, 2009, at 9:10 AM, Frank @ Impact wrote:

> FS is in the media path of an IVR call.
> At the moment, the call is ulaw with DTMF in the audio I think  
> coming into FS and leaving FS.
> The call is coming from an Asterisk server and going to an Asterisk  
> server.
>
> Is there a way to disable FS from passing DTMF at some point in the  
> call?  For example, after 15 seconds, is there a way to get FS to  
> stop passing DTMF events?
>
> Would I have to try to force asterisk to use rfc2833 when sending  
> the call to FS and when accepting it back from FS?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Sat Aug  8 13:32:12 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 16:32:12 -0400
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <367751820908081324r44c7ccf5lfc8ad78c96facb69@mail.gmail.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
	<b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
	<b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>
	<367751820908081044g67a85684hc34a0b242c67b6a4@mail.gmail.com>
	<F4D7EE24-6B62-4A46-83CB-3E412D3120D5@jerris.com>
	<367751820908081324r44c7ccf5lfc8ad78c96facb69@mail.gmail.com>
Message-ID: <287F078C-1132-413C-B456-A2630082571C@jerris.com>

  I think that summary is totally wrong.  Loopback should be used  
here, and this should work to do what you want, just be aware of what  
that means.

Mike


On Aug 8, 2009, at 4:24 PM, Phillip Jones wrote:

> Mike/Rupa ,
>
> Thanks for your help on this. So I am correct that summarizing that
> FreeSWITCH does not really support fail over and multiple call
> destinations because the same mechanism is used to achieve both? And
> that loopback as a solution is possible but not recommended?
>
> Is there any other solution to this? Perhaps a second FS box in the  
> mix?
>
> Phil
>
>
> On Sat, Aug 8, 2009 at 1:00 PM, Michael Jerris<mike at jerris.com> wrote:
>> loopback ends up using extra threads which we are only able to drop
>> later in certain situations so it will decrease your total amount of
>> calls you can do if your not careful with them.
>>
>> Mike
>>
>> On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:
>>
>>> Thanks very much for that - very help.
>>>
>>> Why would loopback be considered "abuse"? What would be the  
>>> downsides
>>> of doing this?
>>>
>>> On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker<rupa at rupa.com> wrote:
>>>>
>>>>
>>>> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker <rupa at rupa.com>  
>>>> wrote:
>>>>>
>>>>> <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>
>>>>
>>>> That of course, should be:
>>>>
>>>>  <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$1..."/>
>>>> --
>>>> -Rupa




From pjintheusa at gmail.com  Sat Aug  8 13:51:00 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Sat, 8 Aug 2009 13:51:00 -0700
Subject: [Freeswitch-users] Calling Multiple Destinations with Failover
In-Reply-To: <287F078C-1132-413C-B456-A2630082571C@jerris.com>
References: <367751820908080359q1fafc678k7056fb6913bcae42@mail.gmail.com>
	<b39273530908080716p7e2f8d11q174d78f0455f25ff@mail.gmail.com>
	<b39273530908080718t793e6363kb2e3c530b0c08a24@mail.gmail.com>
	<367751820908081044g67a85684hc34a0b242c67b6a4@mail.gmail.com>
	<F4D7EE24-6B62-4A46-83CB-3E412D3120D5@jerris.com>
	<367751820908081324r44c7ccf5lfc8ad78c96facb69@mail.gmail.com>
	<287F078C-1132-413C-B456-A2630082571C@jerris.com>
Message-ID: <367751820908081351pdc18bb9kf97ea308a669f14a@mail.gmail.com>

Ok - that's great. I will build this out - thanks both for your help
on this. Much appreciated.

On Sat, Aug 8, 2009 at 1:32 PM, Michael Jerris<mike at jerris.com> wrote:
> ?I think that summary is totally wrong. ?Loopback should be used
> here, and this should work to do what you want, just be aware of what
> that means.
>
> Mike
>
>
> On Aug 8, 2009, at 4:24 PM, Phillip Jones wrote:
>
>> Mike/Rupa ,
>>
>> Thanks for your help on this. So I am correct that summarizing that
>> FreeSWITCH does not really support fail over and multiple call
>> destinations because the same mechanism is used to achieve both? And
>> that loopback as a solution is possible but not recommended?
>>
>> Is there any other solution to this? Perhaps a second FS box in the
>> mix?
>>
>> Phil
>>
>>
>> On Sat, Aug 8, 2009 at 1:00 PM, Michael Jerris<mike at jerris.com> wrote:
>>> loopback ends up using extra threads which we are only able to drop
>>> later in certain situations so it will decrease your total amount of
>>> calls you can do if your not careful with them.
>>>
>>> Mike
>>>
>>> On Aug 8, 2009, at 1:44 PM, Phillip Jones wrote:
>>>
>>>> Thanks very much for that - very help.
>>>>
>>>> Why would loopback be considered "abuse"? What would be the
>>>> downsides
>>>> of doing this?
>>>>
>>>> On Sat, Aug 8, 2009 at 7:18 AM, Rupa Schomaker<rupa at rupa.com> wrote:
>>>>>
>>>>>
>>>>> On Sat, Aug 8, 2009 at 9:16 AM, Rupa Schomaker <rupa at rupa.com>
>>>>> wrote:
>>>>>>
>>>>>> <action application="bridge" data="sofia/gw1/$1|sofia/gw2/$2..."/>
>>>>>
>>>>> That of course, should be:
>>>>>
>>>>> ?<action application="bridge" data="sofia/gw1/$1|sofia/gw2/$1..."/>
>>>>> --
>>>>> -Rupa
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From fraunhofer.lists.freeswitch-001 at traced.net  Sat Aug  8 14:21:25 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Sat, 8 Aug 2009 23:21:25 +0200
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
In-Reply-To: <1F19675C-E556-4AF0-88C1-B5652D8C635F@jerris.com>
References: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>
	<367751820908080348p6165b113kc3be468cb32fabb3@mail.gmail.com>
	<a37510170908080453m1e98ae3l544e3d8b301605e1@mail.gmail.com>
	<1F19675C-E556-4AF0-88C1-B5652D8C635F@jerris.com>
Message-ID: <a37510170908081421i6e6c9031vaa806e8adf3b5f91@mail.gmail.com>

Hello Mike,

2009/8/8 Michael Jerris <mike at jerris.com>:

> how many does it stop at? ?is it the same number each time?

i tried to express that non-wisdom using the words
"This only works fine if we've few concurrent calls. There is no magic
borderline where it starts to refuse work."

this is surely not the best english, i admit. I just have no exact
"lower-bound" when it suddenly happens. My manual test-calls work only
in the beginning, once the loadgen is on for ~3min even the manual
test-calls do not work. Once it's locked up, it initiates calls
successfully (it rings) but (in my example) M doesn't ACK the answer
event it get's from A.

In my current setup with 100 call-legs i can repeatably make it
happen. Even better (for debugging), it does happen each time.

If someone is interested in the exact setup i can place a tarball
somewhere or - if someone could advise me with some gdb commands he
wants to have -  i'll recompile with debugging symbols and get you
what you want. I'm definitely stuck here.

Thx.

  Beni.



From tayeb.meftah at gmail.com  Sat Aug  8 14:32:29 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 08 Aug 2009 21:32:29 +0000
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk, revamped
In-Reply-To: <7b197bef0907270708h1dd55de2m92b5de869da6404d@mail.gmail.com>
References: <7b197bef0907270708h1dd55de2m92b5de869da6404d@mail.gmail.com>
Message-ID: <4A7DEEED.1020006@gmail.com>

hello,
i'm using mod_skypiax
is working very perfectly in my ubuntu machine, no problem or bug
only remember the featur that i requested,
hide the skype client to the system tray and prev it from poping up
thanks!
Giovanni Maruzzelli wrote:
> Ciao FreeSWITCHers,
>
> please have a look at the much changed wiki page:
> http://wiki.freeswitch.org/wiki/Skypiax_Skype_Endpoint_and_Trunk and
> checkout and test the code in svn.
>
> Much has happened, various bug fixes and features added.
>
> Most notable:
> - multiple instances of the same Skype username on Linux (eg: running
> 20 concurrent channels as "Bob" Skype user)
> - adding and removing interfaces on the fly (patch sent by Muhammad Shahzad)
> - easier creation of Skype clients configuration directory
> - reduced latency
> - better robustness
> - running as Windows Service
> - customized ALSA driver for more devices with less IRQs and context switches
> - custom kernel tickless and 100HZ (eg. solves high load problems in
> CentOS and in virtualization)
> - interactive command line client for prototyping
>
> Also, please note that ALSA drivers version 1.0.20 seems to be much
> more stable in our kind of usage (snd-dummy).
>
> Various other enhancements will come, but in the mean time please give
> feedback on the current svn code (we want to be robust for the 1.0.4
> Release :-) )
>
> See you all at www.cluecon.com, talk on Skypiax August 4th at 4.30 pm !
>
> -giovanni
>
>
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4317 (20090808) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>   



__________ Information from ESET NOD32 Antivirus, version of virus signature database 4317 (20090808) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com





From mike at jerris.com  Sat Aug  8 15:06:09 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 8 Aug 2009 18:06:09 -0400
Subject: [Freeswitch-users] Fwd: strange behaviour with scheduled jobs,
	uuid_transfer hangs up one of the legs if freeswitch is not in the
	media path,
	lua freeswitch.Session(uuid) tries to call out if channel references
	by uuid does not exist any longer
In-Reply-To: <a37510170908081421i6e6c9031vaa806e8adf3b5f91@mail.gmail.com>
References: <a37510170908061328x738c7392k71ca997d095609d4@mail.gmail.com>
	<367751820908080348p6165b113kc3be468cb32fabb3@mail.gmail.com>
	<a37510170908080453m1e98ae3l544e3d8b301605e1@mail.gmail.com>
	<1F19675C-E556-4AF0-88C1-B5652D8C635F@jerris.com>
	<a37510170908081421i6e6c9031vaa806e8adf3b5f91@mail.gmail.com>
Message-ID: <26EDA0B5-7DD8-4CE4-829C-FEDCD3E84775@jerris.com>

please open a bug on jira.freeswitch.org with the details of exactly  
how to re-create this issue

Mike

On Aug 8, 2009, at 5:21 PM, Benedikt Fraunhofer wrote:

> Hello Mike,
>
> 2009/8/8 Michael Jerris <mike at jerris.com>:
>
>> how many does it stop at?  is it the same number each time?
>
> i tried to express that non-wisdom using the words
> "This only works fine if we've few concurrent calls. There is no magic
> borderline where it starts to refuse work."
>
> this is surely not the best english, i admit. I just have no exact
> "lower-bound" when it suddenly happens. My manual test-calls work only
> in the beginning, once the loadgen is on for ~3min even the manual
> test-calls do not work. Once it's locked up, it initiates calls
> successfully (it rings) but (in my example) M doesn't ACK the answer
> event it get's from A.
>
> In my current setup with 100 call-legs i can repeatably make it
> happen. Even better (for debugging), it does happen each time.
>
> If someone is interested in the exact setup i can place a tarball
> somewhere or - if someone could advise me with some gdb commands he
> wants to have -  i'll recompile with debugging symbols and get you
> what you want. I'm definitely stuck here.
>
> Thx.
>
>  Beni.




From mgg at giagnocavo.net  Sat Aug  8 17:17:35 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Sat, 8 Aug 2009 20:17:35 -0400
Subject: [Freeswitch-users] mod_php needed
In-Reply-To: <6b65470d0908081134x719bbef6r707644d877eee328@mail.gmail.com>
References: <4A33E8A0.1070708@xpirio.com>
	<87f2f3b90906131356i594d8d30hb2caec0aca050c48@mail.gmail.com>
	<A2B4F48489739E439D8845296A2FD9C4147238@sun.Noble.local>
	<4A361708.8020808@xpirio.com>
	<191c3a030906150659s45024017v2d30daa07e81157c@mail.gmail.com>
	<4A3660EB.4070203@xpirio.com>
	<191c3a030906150807w4bcfcdet52f35c88651a29b9@mail.gmail.com>
	<6b65470d0906150812u254cdc6eu68d127174f89b5e1@mail.gmail.com>
	<191c3a030906150827t2e2e919bw396097bd637a0b91@mail.gmail.com>
	<6b65470d0908081134x719bbef6r707644d877eee328@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702CC213681@mse17be1.mse17.exchange.ms>

You should be able to use PHP with mod_managed just fine. You'll just need Phalanger:
http://www.codeplex.com/Phalanger
http://www.php-compiler.net/doku.php

I don't have any use for PHP, but if anyone is interested in this, I'd be happy to help out if it doesn't "just work" out of the box. It should work on MS CLR and Mono. It also has CodeDom support, so enabling it for dynamic-compilation/scripting should also be easy.

As an added bonus (apart from the claims of 2x the performance), it even has Visual Studio support, complete with some designers: 
http://www.php-compiler.net/doku.php?id=core%3aphp-in-vs2008

-Michael

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of William Suffill
Sent: Saturday, August 08, 2009 12:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_php needed

http://wiki.freeswitch.org/wiki/PHP_ESL

Formatting could use a bit of help just exported my working document
to MediaWiki. Hope to add some more information to ESL as I continue
to do more with it.

I'm going to start a page on fs_ivrd as well. Feedback welcome.

-- W

IRC: wsuff on freenode

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From brian at freeswitch.org  Sat Aug  8 18:18:01 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 8 Aug 2009 20:18:01 -0500
Subject: [Freeswitch-users] how to catch DTMF in Lua while a phrase
	macro is playing
In-Reply-To: <b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
References: <4A7BE35B.8010709@chandlerfamily.org.uk>
	<33c87fa30908072139m9444779w80f5908fa624e9de@mail.gmail.com>
	<4A7D8C26.4020408@chandlerfamily.org.uk>
	<BC708A39-BA09-4524-A072-92A5E5A870F3@apartmentlines.com>
	<96ADFBB4-A015-417A-B0AD-2C427376E010@freeswitch.org>
	<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de@mail.gmail.com>
Message-ID: <012255F2-64AF-4B14-A063-590162E6DC9A@freeswitch.org>

Yes it happens 100% of the time I say not to do it..

Example this thread started with:

In-Reply-To: <4A7BE35B.8010709 at chandlerfamily.org.uk>
References: <4A7BE35B.8010709 at chandlerfamily.org.uk>


The hijacked thread has:

References: <4A7BE35B.8010709 at chandlerfamily.org.uk>
<33c87fa30908072139m9444779w80f5908fa624e9de at mail.gmail.com>
<4A7D8C26.4020408 at chandlerfamily.org.uk>
<BC708A39-BA09-4524-A072-92A5E5A870F3 at apartmentlines.com>
<96ADFBB4-A015-417A-B0AD-2C427376E010 at freeswitch.org>
<b0a8792f0908080950w107ca4b9jedff8cafc27ed0de at mail.gmail.com>

Which causes it to thread it via the References header.

Hence it was hijacked.

/b


On Aug 8, 2009, at 11:50 AM, Raffaele P. Guidi wrote:

> Brian, I've read 8-9 of your emails asking to not hijack threads in  
> threads where this it didn't happen (like this). I see this as a new  
> thread with a single message (plus yours, of course) inside with  
> "[Freeswitch-users] how to catch DTMF in Lua while a phrase macro is  
> playing" as the subject - and I think is quite relevant to the  
> message. I suspect you have something wrong in your email client  
> (mine is gmail) or server and suggest you to give it a check.
>
> Regards,
>    Raffaele




From ivan at myrvold.org  Sun Aug  9 07:52:14 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Sun, 9 Aug 2009 16:52:14 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
Message-ID: <C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>

I tried to compile mod_skypiax, and am getting problem with X11. On OS  
X Leopard, X11 is installed in /usr/X11/lib/
See below.

What can I do to get past this error?

I can also let you ssh into my machine. Contact me off list in case.

Ivan

making all mod_skypiax
Compiling skypiax_protocol.c...
Compiling mod_skypiax.c...
mkdir .libs
Compiling mod_skypiax.c ...
Creating mod_skypiax.so...
ld: library not found for -lX11
collect2: ld returned 1 exit status
gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/ 
Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/Documents/ 
Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror - 
fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g - 
ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -shared - 
o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/ 
mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/Freeswitch/ 
freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/ 
imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/ 
expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/ 
libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/libiconv.dylib / 
Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr/.libs/ 
libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz - 
lncurses -lX11
make[5]: *** [mod_skypiax.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_skypiax-all] Error 1
make[2]: *** [all-recursive] Error 1


Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:

> No, it needs implementation of the message pump between the module and
> the Skype API.
>
> It's probably kind of trivial, if no other problems I'm not aware of.
>
> I do not have a Mac to implement it, tough :-(.
>
> -giovanni
>
>
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
>
>
>
> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>  
> wrote:
>> I'm not sure about that one.... I haven't tried lately because the  
>> API
>> differs on the Mac last I looked at it.
>>
>> /b
>>
>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>
>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>
>>> Ivan
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From gmaruzz at celliax.org  Sun Aug  9 08:10:57 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 9 Aug 2009 17:10:57 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org> 
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com> 
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
Message-ID: <7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>

Ciao Ivan,

it seems that you do not have the libX11 **development** package installed.

Unfortunately I don't know about OSX, so I cannot help you, but many
on the list know.

BTW: it will probably be of no use to you to compile mod_skypiax on
OSX, because Skype for MACOSX works in another way than Skype for
Linux.
If you know about MacOSX programming, please have a look at
https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
probably be simple enough to add a message pump for MacOSX.

-giovanni




Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org> wrote:
> I tried to compile mod_skypiax, and am getting problem with X11. On OS
> X Leopard, X11 is installed in /usr/X11/lib/
> See below.
>
> What can I do to get past this error?
>
> I can also let you ssh into my machine. Contact me off list in case.
>
> Ivan
>
> making all mod_skypiax
> Compiling skypiax_protocol.c...
> Compiling mod_skypiax.c...
> mkdir .libs
> Compiling mod_skypiax.c ...
> Creating mod_skypiax.so...
> ld: library not found for -lX11
> collect2: ld returned 1 exit status
> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/Documents/
> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -
> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -shared -
> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/
> mod_skypiax.o skypiax_protocol.o ?/Users/imyrvold/Documents/Freeswitch/
> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/
> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/libiconv.dylib /
> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr/.libs/
> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
> lncurses -lX11
> make[5]: *** [mod_skypiax.so] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_skypiax-all] Error 1
> make[2]: *** [all-recursive] Error 1
>
>
> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>
>> No, it needs implementation of the message pump between the module and
>> the Skype API.
>>
>> It's probably kind of trivial, if no other problems I'm not aware of.
>>
>> I do not have a Mac to implement it, tough :-(.
>>
>> -giovanni
>>
>>
>>
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> =========================================
>> www.celliax.org
>> via Pierlombardo 9, 20135 Milano
>> Italy
>> gmaruzz at celliax dot org
>> Cell : +39-347-2665618
>> Fax : +39-02-87390039
>>
>>
>>
>>
>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>> wrote:
>>> I'm not sure about that one.... I haven't tried lately because the
>>> API
>>> differs on the Mac last I looked at it.
>>>
>>> /b
>>>
>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>
>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>
>>>> Ivan
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From dujinfang at gmail.com  Sun Aug  9 08:24:53 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 9 Aug 2009 23:24:53 +0800
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
Message-ID: <6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>


On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
> Ciao Ivan,
>
> it seems that you do not have the libX11 **development** package  
> installed.
>
> Unfortunately I don't know about OSX, so I cannot help you, but many
> on the list know.
> BTW: it will probably be of no use to you to compile mod_skypiax on
> OSX, because Skype for MACOSX works in another way than Skype for
> Linux.

That's right.

> If you know about MacOSX programming, please have a look at
> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
> probably be simple enough to add a message pump for MacOSX.
>
> -giovanni
>
>
Giovanni, I have a Mac and tried to get this work yesterday, but  
haven't got it work. Will try further if I have time. However, I don't  
think it's so useful because I don't know how to run and hence control  
multi-skype instances on Mac.

If someone interested to try this I can check the code into my branch.

>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
>
>
>
> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>  
> wrote:
>> I tried to compile mod_skypiax, and am getting problem with X11. On  
>> OS
>> X Leopard, X11 is installed in /usr/X11/lib/
>> See below.
>>
>> What can I do to get past this error?
>>
>> I can also let you ssh into my machine. Contact me off list in case.
>>
>> Ivan
>>
>> making all mod_skypiax
>> Compiling skypiax_protocol.c...
>> Compiling mod_skypiax.c...
>> mkdir .libs
>> Compiling mod_skypiax.c ...
>> Creating mod_skypiax.so...
>> ld: library not found for -lX11
>> collect2: ld returned 1 exit status
>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/ 
>> Documents/
>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -
>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -shared -
>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/
>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/ 
>> Freeswitch/
>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/
>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/ 
>> libiconv.dylib /
>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/ 
>> apr/.libs/
>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
>> lncurses -lX11
>> make[5]: *** [mod_skypiax.so] Error 1
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_skypiax-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>>
>>
>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>
>>> No, it needs implementation of the message pump between the module  
>>> and
>>> the Skype API.
>>>
>>> It's probably kind of trivial, if no other problems I'm not aware  
>>> of.
>>>
>>> I do not have a Mac to implement it, tough :-(.
>>>
>>> -giovanni
>>>
>>>
>>>
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> =========================================
>>> www.celliax.org
>>> via Pierlombardo 9, 20135 Milano
>>> Italy
>>> gmaruzz at celliax dot org
>>> Cell : +39-347-2665618
>>> Fax : +39-02-87390039
>>>
>>>
>>>
>>>
>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>> wrote:
>>>> I'm not sure about that one.... I haven't tried lately because the
>>>> API
>>>> differs on the Mac last I looked at it.
>>>>
>>>> /b
>>>>
>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>
>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>
>>>>> Ivan
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From ivan at myrvold.org  Sun Aug  9 08:33:09 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Sun, 9 Aug 2009 17:33:09 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
Message-ID: <4F6990B2-E9D0-4EC2-9457-973A643AFDD6@myrvold.org>

Thanks, Giovanni for the pointer!

Yes, I am a Cocoa developer, and have no problem compiling with the  
Skype.framework. But I still have no clue how to include this into the  
skypiax code.
Probably need just a little more hint of how to do it.

Ivan

Den 9. aug.. 2009 kl. 17:10 skrev Giovanni Maruzzelli:

> Ciao Ivan,
>
> it seems that you do not have the libX11 **development** package  
> installed.
>
> Unfortunately I don't know about OSX, so I cannot help you, but many
> on the list know.
>
> BTW: it will probably be of no use to you to compile mod_skypiax on
> OSX, because Skype for MACOSX works in another way than Skype for
> Linux.
> If you know about MacOSX programming, please have a look at
> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
> probably be simple enough to add a message pump for MacOSX.
>
> -giovanni
>
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
>
>
>
> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>  
> wrote:
>> I tried to compile mod_skypiax, and am getting problem with X11. On  
>> OS
>> X Leopard, X11 is installed in /usr/X11/lib/
>> See below.
>>
>> What can I do to get past this error?
>>
>> I can also let you ssh into my machine. Contact me off list in case.
>>
>> Ivan
>>
>> making all mod_skypiax
>> Compiling skypiax_protocol.c...
>> Compiling mod_skypiax.c...
>> mkdir .libs
>> Compiling mod_skypiax.c ...
>> Creating mod_skypiax.so...
>> ld: library not found for -lX11
>> collect2: ld returned 1 exit status
>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/ 
>> Documents/
>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -
>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -shared -
>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/
>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/ 
>> Freeswitch/
>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/
>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/ 
>> libiconv.dylib /
>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/ 
>> apr/.libs/
>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
>> lncurses -lX11
>> make[5]: *** [mod_skypiax.so] Error 1
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_skypiax-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>>
>>
>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>
>>> No, it needs implementation of the message pump between the module  
>>> and
>>> the Skype API.
>>>
>>> It's probably kind of trivial, if no other problems I'm not aware  
>>> of.
>>>
>>> I do not have a Mac to implement it, tough :-(.
>>>
>>> -giovanni
>>>
>>>
>>>
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> =========================================
>>> www.celliax.org
>>> via Pierlombardo 9, 20135 Milano
>>> Italy
>>> gmaruzz at celliax dot org
>>> Cell : +39-347-2665618
>>> Fax : +39-02-87390039
>>>
>>>
>>>
>>>
>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>> wrote:
>>>> I'm not sure about that one.... I haven't tried lately because the
>>>> API
>>>> differs on the Mac last I looked at it.
>>>>
>>>> /b
>>>>
>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>
>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>
>>>>> Ivan
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From ivan at myrvold.org  Sun Aug  9 08:34:37 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Sun, 9 Aug 2009 17:34:37 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
	<6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
Message-ID: <4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>

Yes, I am interested in this, and if you have any source I could have  
a look at it.

Ivan

Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:

>
> On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
>> Ciao Ivan,
>>
>> it seems that you do not have the libX11 **development** package
>> installed.
>>
>> Unfortunately I don't know about OSX, so I cannot help you, but many
>> on the list know.
>> BTW: it will probably be of no use to you to compile mod_skypiax on
>> OSX, because Skype for MACOSX works in another way than Skype for
>> Linux.
>
> That's right.
>
>> If you know about MacOSX programming, please have a look at
>> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
>> probably be simple enough to add a message pump for MacOSX.
>>
>> -giovanni
>>
>>
> Giovanni, I have a Mac and tried to get this work yesterday, but
> haven't got it work. Will try further if I have time. However, I don't
> think it's so useful because I don't know how to run and hence control
> multi-skype instances on Mac.
>
> If someone interested to try this I can check the code into my branch.
>
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> =========================================
>> www.celliax.org
>> via Pierlombardo 9, 20135 Milano
>> Italy
>> gmaruzz at celliax dot org
>> Cell : +39-347-2665618
>> Fax : +39-02-87390039
>>
>>
>>
>>
>> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>
>> wrote:
>>> I tried to compile mod_skypiax, and am getting problem with X11. On
>>> OS
>>> X Leopard, X11 is installed in /usr/X11/lib/
>>> See below.
>>>
>>> What can I do to get past this error?
>>>
>>> I can also let you ssh into my machine. Contact me off list in case.
>>>
>>> Ivan
>>>
>>> making all mod_skypiax
>>> Compiling skypiax_protocol.c...
>>> Compiling mod_skypiax.c...
>>> mkdir .libs
>>> Compiling mod_skypiax.c ...
>>> Creating mod_skypiax.so...
>>> ld: library not found for -lX11
>>> collect2: ld returned 1 exit status
>>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
>>> Documents/
>>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 - 
>>> g -
>>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE - 
>>> shared -
>>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/
>>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/
>>> Freeswitch/
>>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/
>>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
>>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
>>> libiconv.dylib /
>>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
>>> apr/.libs/
>>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
>>> lncurses -lX11
>>> make[5]: *** [mod_skypiax.so] Error 1
>>> make[4]: *** [all] Error 1
>>> make[3]: *** [mod_skypiax-all] Error 1
>>> make[2]: *** [all-recursive] Error 1
>>>
>>>
>>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>>
>>>> No, it needs implementation of the message pump between the module
>>>> and
>>>> the Skype API.
>>>>
>>>> It's probably kind of trivial, if no other problems I'm not aware
>>>> of.
>>>>
>>>> I do not have a Mac to implement it, tough :-(.
>>>>
>>>> -giovanni
>>>>
>>>>
>>>>
>>>>
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> =========================================
>>>> www.celliax.org
>>>> via Pierlombardo 9, 20135 Milano
>>>> Italy
>>>> gmaruzz at celliax dot org
>>>> Cell : +39-347-2665618
>>>> Fax : +39-02-87390039
>>>>
>>>>
>>>>
>>>>
>>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>>> wrote:
>>>>> I'm not sure about that one.... I haven't tried lately because the
>>>>> API
>>>>> differs on the Mac last I looked at it.
>>>>>
>>>>> /b
>>>>>
>>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>>
>>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>>
>>>>>> Ivan
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From gmaruzz at celliax.org  Sun Aug  9 08:52:07 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sun, 9 Aug 2009 17:52:07 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org> 
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com> 
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com> 
	<6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
	<4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
Message-ID: <7b197bef0908090852t7243db6r761ddfce31093ed0@mail.gmail.com>

Seven,
thanks a lot for your effort, please let your stuff be available,
maybe Ivan can make use of it!

Ivan,
in the file src/mod/endpoints/mod_skypiax/skypiax_protocol.c add you
will find #ifdef WIN32 .

it conditional compiles code between WIN32 and linux.

You need to add another #ifdef, so it will compile for OSX.

You will probably be able to use the same pipe mechanism as in Linux
(normal POSIX pipes).
You will for sure need to implement the part that deals with the Skype
API. Maybe it will be not much more than reusing the example code to
interact with the API.

Please, let us know how it goes, and feel *very* free to ask for further info.

-giovanni




Sincerely,

Giovanni Maruzzelli
=========================================
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
gmaruzz at celliax dot org
Cell : +39-347-2665618
Fax : +39-02-87390039




On Sun, Aug 9, 2009 at 5:34 PM, Ivan C Myrvold<ivan at myrvold.org> wrote:
> Yes, I am interested in this, and if you have any source I could have
> a look at it.
>
> Ivan
>
> Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
>
>>
>> On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
>>> Ciao Ivan,
>>>
>>> it seems that you do not have the libX11 **development** package
>>> installed.
>>>
>>> Unfortunately I don't know about OSX, so I cannot help you, but many
>>> on the list know.
>>> BTW: it will probably be of no use to you to compile mod_skypiax on
>>> OSX, because Skype for MACOSX works in another way than Skype for
>>> Linux.
>>
>> That's right.
>>
>>> If you know about MacOSX programming, please have a look at
>>> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
>>> probably be simple enough to add a message pump for MacOSX.
>>>
>>> -giovanni
>>>
>>>
>> Giovanni, I have a Mac and tried to get this work yesterday, but
>> haven't got it work. Will try further if I have time. However, I don't
>> think it's so useful because I don't know how to run and hence control
>> multi-skype instances on Mac.
>>
>> If someone interested to try this I can check the code into my branch.
>>
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> =========================================
>>> www.celliax.org
>>> via Pierlombardo 9, 20135 Milano
>>> Italy
>>> gmaruzz at celliax dot org
>>> Cell : +39-347-2665618
>>> Fax : +39-02-87390039
>>>
>>>
>>>
>>>
>>> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>
>>> wrote:
>>>> I tried to compile mod_skypiax, and am getting problem with X11. On
>>>> OS
>>>> X Leopard, X11 is installed in /usr/X11/lib/
>>>> See below.
>>>>
>>>> What can I do to get past this error?
>>>>
>>>> I can also let you ssh into my machine. Contact me off list in case.
>>>>
>>>> Ivan
>>>>
>>>> making all mod_skypiax
>>>> Compiling skypiax_protocol.c...
>>>> Compiling mod_skypiax.c...
>>>> mkdir .libs
>>>> Compiling mod_skypiax.c ...
>>>> Creating mod_skypiax.so...
>>>> ld: library not found for -lX11
>>>> collect2: ld returned 1 exit status
>>>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>>>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
>>>> Documents/
>>>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>>>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -
>>>> g -
>>>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -
>>>> shared -
>>>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-namespace .libs/
>>>> mod_skypiax.o skypiax_protocol.o ?/Users/imyrvold/Documents/
>>>> Freeswitch/
>>>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>>>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/xml/
>>>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
>>>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
>>>> libiconv.dylib /
>>>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
>>>> apr/.libs/
>>>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
>>>> lncurses -lX11
>>>> make[5]: *** [mod_skypiax.so] Error 1
>>>> make[4]: *** [all] Error 1
>>>> make[3]: *** [mod_skypiax-all] Error 1
>>>> make[2]: *** [all-recursive] Error 1
>>>>
>>>>
>>>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>>>
>>>>> No, it needs implementation of the message pump between the module
>>>>> and
>>>>> the Skype API.
>>>>>
>>>>> It's probably kind of trivial, if no other problems I'm not aware
>>>>> of.
>>>>>
>>>>> I do not have a Mac to implement it, tough :-(.
>>>>>
>>>>> -giovanni
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> =========================================
>>>>> www.celliax.org
>>>>> via Pierlombardo 9, 20135 Milano
>>>>> Italy
>>>>> gmaruzz at celliax dot org
>>>>> Cell : +39-347-2665618
>>>>> Fax : +39-02-87390039
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>>>> wrote:
>>>>>> I'm not sure about that one.... I haven't tried lately because the
>>>>>> API
>>>>>> differs on the Mac last I looked at it.
>>>>>>
>>>>>> /b
>>>>>>
>>>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>>>
>>>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>>>
>>>>>>> Ivan
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From markmorreny at gmail.com  Sun Aug  9 08:57:06 2009
From: markmorreny at gmail.com (mark morreny)
Date: Sun, 9 Aug 2009 23:57:06 +0800
Subject: [Freeswitch-users] Scheduler in module
Message-ID: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>

Hi,

I would like to collect some data in my module which would listen to the
event socket and then a scheduler would kick off every so often to clean up
the data.

Does anyone know what is the best way to implement a scheduler in a mod?

Best Regards,
Mark
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From moises.silva at gmail.com  Sun Aug  9 09:05:47 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Sun, 9 Aug 2009 12:05:47 -0400
Subject: [Freeswitch-users] Scheduler in module
In-Reply-To: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
Message-ID: <c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>

On Sun, Aug 9, 2009 at 11:57 AM, mark morreny <markmorreny at gmail.com> wrote:

> Hi,
>
> I would like to collect some data in my module which would listen to the
> event socket and then a scheduler would kick off every so often to clean up
> the data.
>
> Does anyone know what is the best way to implement a scheduler in a mod?
>
> Best Regards,
> Mark
>

Just use the API that the FreeSWITCH core already has for you. Check
src/include/switch_scheduler.h for details, the comments in there should be
enough for you to learn to use it. If not, you can ask here more specific
questions.
-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
-------------- next part --------------
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From dujinfang at gmail.com  Sun Aug  9 11:02:34 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 10 Aug 2009 02:02:34 +0800
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
	<6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
	<4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
Message-ID: <092B4355-F911-4E37-A5D1-AC923AF9B8FB@gmail.com>

Ivan,

Good to know you are a cocoa dev. Unable to check in code right now,  
will send the diff to you offlist for now.

0) I'm not familiar with Mac dev, just tried my best
1) It doesn't work yet, but should be able to compile, sure you  
already have the Skype framework in place :)
2) if run the skype delegate from a threat, then cannot get event  
callback. e.g. mac_client.c works but mac_client2.c doesn't. Since  
skypiax is running in a thread, we need to figure out this first.
3) it uses Carbon, since I think we only need to low level api, no  
need to bother the complicate of Cocoa.
4) strsep shows some warning on compile, haven't figured out why
5) perhaps you should only add one interface in skypiax.conf.xml
6) do you want to run multi-instances like on Linux?
7) I really not sure if it will work or not :)

Let me know if it helps. I bet you can make it work. Also code will be  
in my branch soon.

7.


On Aug 9, 2009, at 11:34 PM, Ivan C Myrvold wrote:
> Yes, I am interested in this, and if you have any source I could have
> a look at it.
>
> Ivan
>
> Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
>
>>
>> On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
>>> Ciao Ivan,
>>>
>>> it seems that you do not have the libX11 **development** package
>>> installed.
>>>
>>> Unfortunately I don't know about OSX, so I cannot help you, but many
>>> on the list know.
>>> BTW: it will probably be of no use to you to compile mod_skypiax on
>>> OSX, because Skype for MACOSX works in another way than Skype for
>>> Linux.
>>
>> That's right.
>>
>>> If you know about MacOSX programming, please have a look at
>>> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
>>> probably be simple enough to add a message pump for MacOSX.
>>>
>>> -giovanni
>>>
>>>
>> Giovanni, I have a Mac and tried to get this work yesterday, but
>> haven't got it work. Will try further if I have time. However, I  
>> don't
>> think it's so useful because I don't know how to run and hence  
>> control
>> multi-skype instances on Mac.
>>
>> If someone interested to try this I can check the code into my  
>> branch.
>>
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>> =========================================
>>> www.celliax.org
>>> via Pierlombardo 9, 20135 Milano
>>> Italy
>>> gmaruzz at celliax dot org
>>> Cell : +39-347-2665618
>>> Fax : +39-02-87390039
>>>
>>>
>>>
>>>
>>> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>
>>> wrote:
>>>> I tried to compile mod_skypiax, and am getting problem with X11. On
>>>> OS
>>>> X Leopard, X11 is installed in /usr/X11/lib/
>>>> See below.
>>>>
>>>> What can I do to get past this error?
>>>>
>>>> I can also let you ssh into my machine. Contact me off list in  
>>>> case.
>>>>
>>>> Ivan
>>>>
>>>> making all mod_skypiax
>>>> Compiling skypiax_protocol.c...
>>>> Compiling mod_skypiax.c...
>>>> mkdir .libs
>>>> Compiling mod_skypiax.c ...
>>>> Creating mod_skypiax.so...
>>>> ld: library not found for -lX11
>>>> collect2: ld returned 1 exit status
>>>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>>>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
>>>> Documents/
>>>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>>>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -
>>>> g -
>>>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -
>>>> shared -
>>>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat- 
>>>> namespace .libs/
>>>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/
>>>> Freeswitch/
>>>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>>>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/ 
>>>> xml/
>>>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/
>>>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
>>>> libiconv.dylib /
>>>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
>>>> apr/.libs/
>>>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -lz -
>>>> lncurses -lX11
>>>> make[5]: *** [mod_skypiax.so] Error 1
>>>> make[4]: *** [all] Error 1
>>>> make[3]: *** [mod_skypiax-all] Error 1
>>>> make[2]: *** [all-recursive] Error 1
>>>>
>>>>
>>>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>>>
>>>>> No, it needs implementation of the message pump between the module
>>>>> and
>>>>> the Skype API.
>>>>>
>>>>> It's probably kind of trivial, if no other problems I'm not aware
>>>>> of.
>>>>>
>>>>> I do not have a Mac to implement it, tough :-(.
>>>>>
>>>>> -giovanni
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> Sincerely,
>>>>>
>>>>> Giovanni Maruzzelli
>>>>> =========================================
>>>>> www.celliax.org
>>>>> via Pierlombardo 9, 20135 Milano
>>>>> Italy
>>>>> gmaruzz at celliax dot org
>>>>> Cell : +39-347-2665618
>>>>> Fax : +39-02-87390039
>>>>>
>>>>>
>>>>>
>>>>>
>>>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>>>> wrote:
>>>>>> I'm not sure about that one.... I haven't tried lately because  
>>>>>> the
>>>>>> API
>>>>>> differs on the Mac last I looked at it.
>>>>>>
>>>>>> /b
>>>>>>
>>>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>>>
>>>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>>>
>>>>>>> Ivan
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From dujinfang at gmail.com  Sun Aug  9 17:44:32 2009
From: dujinfang at gmail.com (Seven Du)
Date: Mon, 10 Aug 2009 08:44:32 +0800
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <7b197bef0908090852t7243db6r761ddfce31093ed0@mail.gmail.com>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
	<6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
	<4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
	<7b197bef0908090852t7243db6r761ddfce31093ed0@mail.gmail.com>
Message-ID: <09A9F94D-D271-4B70-9BF0-4C9DBB14E5F5@gmail.com>


On Aug 9, 2009, at 11:52 PM, Giovanni Maruzzelli wrote:
> Seven,
> thanks a lot for your effort, please let your stuff be available,
> maybe Ivan can make use of it!
>
svn diff http://svn.freeswitch.org/svn/freeswitch/branches/seven -r  
14473
:14475

When this done I think it's better to split codes into

skypiax_protocol.c
skypiax_protocol_mac.c
skypiax_protocol_linux.c
skypiax_protocol_windows.c

:)

7.
> Ivan,
> in the file src/mod/endpoints/mod_skypiax/skypiax_protocol.c add you
> will find #ifdef WIN32 .
>
> it conditional compiles code between WIN32 and linux.
>
> You need to add another #ifdef, so it will compile for OSX.
>
> You will probably be able to use the same pipe mechanism as in Linux
> (normal POSIX pipes).
> You will for sure need to implement the part that deals with the Skype
> API. Maybe it will be not much more than reusing the example code to
> interact with the API.
>
> Please, let us know how it goes, and feel *very* free to ask for  
> further info.
>
> -giovanni
>
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> =========================================
> www.celliax.org
> via Pierlombardo 9, 20135 Milano
> Italy
> gmaruzz at celliax dot org
> Cell : +39-347-2665618
> Fax : +39-02-87390039
>
>
>
>
> On Sun, Aug 9, 2009 at 5:34 PM, Ivan C Myrvold<ivan at myrvold.org>  
> wrote:
>> Yes, I am interested in this, and if you have any source I could have
>> a look at it.
>>
>> Ivan
>>
>> Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
>>
>>>
>>> On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
>>>> Ciao Ivan,
>>>>
>>>> it seems that you do not have the libX11 **development** package
>>>> installed.
>>>>
>>>> Unfortunately I don't know about OSX, so I cannot help you, but  
>>>> many
>>>> on the list know.
>>>> BTW: it will probably be of no use to you to compile mod_skypiax on
>>>> OSX, because Skype for MACOSX works in another way than Skype for
>>>> Linux.
>>>
>>> That's right.
>>>
>>>> If you know about MacOSX programming, please have a look at
>>>> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
>>>> probably be simple enough to add a message pump for MacOSX.
>>>>
>>>> -giovanni
>>>>
>>>>
>>> Giovanni, I have a Mac and tried to get this work yesterday, but
>>> haven't got it work. Will try further if I have time. However, I  
>>> don't
>>> think it's so useful because I don't know how to run and hence  
>>> control
>>> multi-skype instances on Mac.
>>>
>>> If someone interested to try this I can check the code into my  
>>> branch.
>>>
>>>>
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> =========================================
>>>> www.celliax.org
>>>> via Pierlombardo 9, 20135 Milano
>>>> Italy
>>>> gmaruzz at celliax dot org
>>>> Cell : +39-347-2665618
>>>> Fax : +39-02-87390039
>>>>
>>>>
>>>>
>>>>
>>>> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>
>>>> wrote:
>>>>> I tried to compile mod_skypiax, and am getting problem with X11.  
>>>>> On
>>>>> OS
>>>>> X Leopard, X11 is installed in /usr/X11/lib/
>>>>> See below.
>>>>>
>>>>> What can I do to get past this error?
>>>>>
>>>>> I can also let you ssh into my machine. Contact me off list in  
>>>>> case.
>>>>>
>>>>> Ivan
>>>>>
>>>>> making all mod_skypiax
>>>>> Compiling skypiax_protocol.c...
>>>>> Compiling mod_skypiax.c...
>>>>> mkdir .libs
>>>>> Compiling mod_skypiax.c ...
>>>>> Creating mod_skypiax.so...
>>>>> ld: library not found for -lX11
>>>>> collect2: ld returned 1 exit status
>>>>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>>>>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
>>>>> Documents/
>>>>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>>>>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -
>>>>> g -
>>>>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -
>>>>> shared -
>>>>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat- 
>>>>> namespace .libs/
>>>>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/
>>>>> Freeswitch/
>>>>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>>>>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/ 
>>>>> xml/
>>>>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch. 
>>>>> 09-08-09/
>>>>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
>>>>> libiconv.dylib /
>>>>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
>>>>> apr/.libs/
>>>>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto - 
>>>>> lz -
>>>>> lncurses -lX11
>>>>> make[5]: *** [mod_skypiax.so] Error 1
>>>>> make[4]: *** [all] Error 1
>>>>> make[3]: *** [mod_skypiax-all] Error 1
>>>>> make[2]: *** [all-recursive] Error 1
>>>>>
>>>>>
>>>>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>>>>
>>>>>> No, it needs implementation of the message pump between the  
>>>>>> module
>>>>>> and
>>>>>> the Skype API.
>>>>>>
>>>>>> It's probably kind of trivial, if no other problems I'm not aware
>>>>>> of.
>>>>>>
>>>>>> I do not have a Mac to implement it, tough :-(.
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Sincerely,
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> =========================================
>>>>>> www.celliax.org
>>>>>> via Pierlombardo 9, 20135 Milano
>>>>>> Italy
>>>>>> gmaruzz at celliax dot org
>>>>>> Cell : +39-347-2665618
>>>>>> Fax : +39-02-87390039
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>>>>> wrote:
>>>>>>> I'm not sure about that one.... I haven't tried lately because  
>>>>>>> the
>>>>>>> API
>>>>>>> differs on the Mac last I looked at it.
>>>>>>>
>>>>>>> /b
>>>>>>>
>>>>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>>>>
>>>>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>>>>
>>>>>>>> Ivan
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From godson.g at gmail.com  Sun Aug  9 22:26:23 2009
From: godson.g at gmail.com (Godson Gera)
Date: Mon, 10 Aug 2009 10:56:23 +0530
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
Message-ID: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>

Hi FS Team,


        The files.freeswitch.org is resetting connection since 3 days. As a
result I was not able to download latest release of FS. Got the trunk
version from svn. But still it suffers from the lack of sound files.  When
ever I do 'make uhd-sounds-install' http://files.freeswitch.org resets
connection immediately wget tries 20 times and gives up. Other users on IRC
also reported this issue.

-- 
Thanks & Regards,
Godson Gera
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From ryder86 at googlemail.com  Sun Aug  9 22:44:46 2009
From: ryder86 at googlemail.com (Artem Vasiliev)
Date: Mon, 10 Aug 2009 09:44:46 +0400
Subject: [Freeswitch-users] Softphone control
In-Reply-To: <5a8712120908071350v2eac4613j7fd53e5158680742@mail.gmail.com>
References: <cb6be45b0908070402v404bb94fi22386170220bf4cc@mail.gmail.com>
	<ab57a3e80908070800q1076d8d4j973e8eb518394f5d@mail.gmail.com>
	<b0a8792f0908071150x68f3e3b5v596bad90bc56d1ad@mail.gmail.com>
	<5a8712120908071350v2eac4613j7fd53e5158680742@mail.gmail.com>
Message-ID: <cb6be45b0908092244w3d75dd7i7f0bab93682ffdc2@mail.gmail.com>

>Jo?o Mesquita
Nice thing. I made similar one for Event_Socket (C#).

>Raffaele P. Guidi
Yes, I'm working on CTI. We use a CTI application called WebAgent, but
it's TAPI-based, so we have to create a special DLL to make it work
with FreeSWITCH.

>Kevin Green
Thanks for explanation. It works as you said.
Answer - auto (let it be for now)
Hold/Unhold - via uuid_hold (hold button doesn't work)
Hangup - the only that works properly
Make call - made simple javascript that dials both ends and then bridges them

2009/8/8 Jo?o Mesquita <jmesquita at gmail.com>:
> Stay tuned on fsgui. It will get there really soon.
>
> jmesquita
>
> On Fri, Aug 7, 2009 at 3:50 PM, Raffaele P. Guidi
> <raffaele.p.guidi at gmail.com> wrote:
>>
>> Maybe Artem is interested in CTI (computer telephony integration) -
>> click2dial, opening a url (or statrting a program) on incoming call...?
>>
>> On Fri, Aug 7, 2009 at 17:00, Kevin Green <kevin at johnnyvoip.com> wrote:
>>>
>>> From what I am aware you can't use FreeSWITCH to control a softphone
>>> directly though you can make it do things that will have a similar end
>>> result. You could set eyeBeam to auto-answer calls if you want them to
>>> answer right away or orginiate a call that is auto-answered but not bridge
>>> the call?until a user on the eyeBeam presses a digit or a socket control
>>> tells it to connect the two ends. You can also use FreeSWITCH to place the
>>> line on hold using event sockets, this will place it on hold in the server
>>> and not directly like placing it on hold in eyeBeam (i.e. the hold button in
>>> eyeBeam likely wont show it as being on hold).
>>> Beyond that if you want to directly control the clients you would need to
>>> look at getting an API access into the eyeBeam client.
>>> I hope this will help.
>>> Regards,
>>> ? ?Kevin Green
>>>
>>>
>>> On Fri, Aug 7, 2009 at 7:02 AM, Artem Vasiliev <ryder86 at googlemail.com>
>>> wrote:
>>>>
>>>> No, I don't want to make softphone from FreeSwitch
>>>>
>>>> I have FS and several users with eyeBeam softphones. I need to control
>>>> those eyeBeams
>>>>
>>>> >You can run FreeSWITCH as a softphone and control it.
>>>> >http://wiki.freeswitch.org/wiki/Freeswitch_softphone
>>>>
>>>> >2009/8/7 Artem Vasiliev <ryder86 at googlemail.com>
>>>>
>>>> >> Hi
>>>> >>
>>>> >> I have FreeSwitch and external application, which communicates to it
>>>> >> via
>>>> >> event socket - listens for events for certain number and gives some
>>>> >> commands.
>>>> >> Is it possible for this application to control client softphones, for
>>>> >> example, make them answer or hold, using the event socket or other
>>>> >> FreeSwitch capabilities?



From diego.viola at gmail.com  Sun Aug  9 22:57:00 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 10 Aug 2009 01:57:00 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
Message-ID: <86a32abc0908092257k3cc7695ay3a8017a874bf186a@mail.gmail.com>

I been having this problem also.

On Mon, Aug 10, 2009 at 1:26 AM, Godson Gera <godson.g at gmail.com> wrote:

> Hi FS Team,
>
>
>         The files.freeswitch.org is resetting connection since 3 days. As
> a result I was not able to download latest release of FS. Got the trunk
> version from svn. But still it suffers from the lack of sound files.  When
> ever I do 'make uhd-sounds-install' http://files.freeswitch.org resets
> connection immediately wget tries 20 times and gives up. Other users on IRC
> also reported this issue.
>
> --
> Thanks & Regards,
> Godson Gera
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Sun Aug  9 23:06:05 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Mon, 10 Aug 2009 02:06:05 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908092257k3cc7695ay3a8017a874bf186a@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<86a32abc0908092257k3cc7695ay3a8017a874bf186a@mail.gmail.com>
Message-ID: <86a32abc0908092306k446f5118oa6526769a26fb76c@mail.gmail.com>

http://files-sync.freeswitch.org/

On Mon, Aug 10, 2009 at 1:57 AM, Diego Viola <diego.viola at gmail.com> wrote:

> I been having this problem also.
>
> On Mon, Aug 10, 2009 at 1:26 AM, Godson Gera <godson.g at gmail.com> wrote:
>
>> Hi FS Team,
>>
>>
>>         The files.freeswitch.org is resetting connection since 3 days. As
>> a result I was not able to download latest release of FS. Got the trunk
>> version from svn. But still it suffers from the lack of sound files.  When
>> ever I do 'make uhd-sounds-install' http://files.freeswitch.org resets
>> connection immediately wget tries 20 times and gives up. Other users on IRC
>> also reported this issue.
>>
>> --
>> Thanks & Regards,
>> Godson Gera
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From velu.technical at gmail.com  Sun Aug  9 23:38:38 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Mon, 10 Aug 2009 12:08:38 +0530
Subject: [Freeswitch-users] ALARM signal in esl libraries
Message-ID: <1452e2980908092338j15c4e33cn2cff799bb64464d0@mail.gmail.com>

Dear All,
       I have registered  ALARM signal  in my perl program to handle the
DTMF digit timeout.
When ALARM signal generated  the  connection with  ESL  is  automatically
closed.

I have checked the connection with "connected: function, it returns 0.

Why the connection was closed?
Is there any idea to alive the connection after ALARM signal generation??

Please help me.........

Thanks,
Velusamy
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From lakindia89 at gmail.com  Mon Aug 10 01:00:49 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Mon, 10 Aug 2009 01:00:49 -0700 (PDT)
Subject: [Freeswitch-users] Error while creating object
In-Reply-To: <7d79b3930908060347xe5be545yfeeafad761aba274@mail.gmail.com>
References: <7d79b3930908060347xe5be545yfeeafad761aba274@mail.gmail.com>
Message-ID: <24895716.post@talk.nabble.com>


Can any one please say what I did wrong here?

regards,
Lakshmanan G.


lakshmanan wrote:
> 
> Hi all,
> Greets.
> 
> I am in the process of controlling the freeswitch with perl.
> I have read about mod_perl and I wrote some scripts to test which works
> fine.
> Yesterday I tried to access the digit_set function.
> So I create an object for the freeswitch::DTMF.
> But it reported the following error.
> 
> 2009-08-06 15:53:46 [ERR] mod_perl.c:69 Perl_safe_eval() [require
> '/usr/local/freeswitch/conf/test.pl';]
> No matching function for overloaded 'new_DTMF' at
> /usr/local/freeswitch/perl/freeswitch.pm line 197.
> Compilation failed in require at (eval 2) line 1.
> 
> Here is my code.
> 
> #!/usr/bin/perl
> use strict;
> use freeswitch;
> our $session;
> $session->execute("bridge","user/1010");
> my $sess=&freeswitch::DTMF::new;
> return 1;
> 
> The bridge is working fine. But while creating the object it said error.
> 
> Can any one explain why this happens and how can I correct it?
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/Error-while-creating-object-tp24849065p24895716.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From wangdq.no1 at gmail.com  Sun Aug  9 23:16:39 2009
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Mon, 10 Aug 2009 14:16:39 +0800
Subject: [Freeswitch-users] can freeswitch group_call to gateway extension ?
Message-ID: <ecd05ce30908092316s66051433p506df518942ec93f@mail.gmail.com>

Hello every one :
  I tested FS. and when I use group_call.  But I can't call the extension in
group and not registered in FS.
  Why ? and What  ?
  thanks .
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From michal.bielicki at halo2.pl  Mon Aug 10 02:48:33 2009
From: michal.bielicki at halo2.pl (Michal Bielicki)
Date: Mon, 10 Aug 2009 11:48:33 +0200
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>
Message-ID: <17FB8F3C-0661-4594-912E-9541476DFDAC@halo2.pl>

It builds on both.
Can you tell me how you are trying to build ?

Am 08.08.2009 um 06:36 schrieb Michael Jerris:

> This is not currently a supported platform, it only builds on 64 bit
> right now I think on solaris.
>
> Mike
>
> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>
>> Hello,
>>
>> Does anyone have, or know where to get, a pre-compiled copy of
>> FreeSwitch for Solaris 10/x86?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. info at halokwadrat.de
e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
Hauptgesch?ftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 0000153539

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From vkozak at abisoft.spb.ru  Mon Aug 10 03:00:11 2009
From: vkozak at abisoft.spb.ru (Kozak Vladimir)
Date: Mon, 10 Aug 2009 14:00:11 +0400
Subject: [Freeswitch-users] Fw: FreeSwitch doesn't play music on hold
	forbriged channel
Message-ID: <C5CEF36D42B347B49C82CDE05C3B0E5D@abisoft.biz>

2009-08-12 03:02:03 [DEBUG] switch_core_session.c:706 switch_core_session_queue_private_event() Send signal sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:228 audio_bridge_thread() sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes receive message [UNBRIDGE]
2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
2009-08-12 03:02:03 [DEBUG] switch_ivr.c:540 switch_ivr_parse_event() sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes Command Execute hold()
2009-08-12 03:02:03 [DEBUG] switch_ivr.c:1054 switch_ivr_hold() sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes receive message [HOLD]
2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
2009-08-12 03:02:03 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes entering state [calling]
2009-08-12 03:02:03 [WARNING] switch_ivr_async.c:2239 switch_ivr_broadcast() Channel [sofia/internal/1001 at 172.26.200.250][local_stream://moh] already broadcasting...broadcast aborted
2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:231 audio_bridge_thread() sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes receive message [BRIDGE]
2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523 switch_core_session_perform_receive_message() Send signal sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:233 audio_bridge_thread() Send signal sofia/internal/1001 at 172.26.200.250 [BREAK]
2009-08-12 03:02:03 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state() Channel sofia/internal/sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes entering state [ready]
2009-08-12 03:02:03 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state() Remote SDP:
v=0
o=- 6 3 IN IP4 172.26.10.39
s=CounterPath eyeBeam 1.5
c=IN IP4 172.26.10.39
t=0 0
m=audio 15712 RTP/AVP 0 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=recvonly

2009-08-12 03:02:03 [DEBUG] sofia_glue.c:2483 sofia_glue_negotiate_sdp() Our existing sdp is still good [PCMU 172.26.10.39:15712], let's keep it.
2009-08-12 03:02:03 [DEBUG] sofia_glue.c:2509 sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101

  ----- Original Message ----- 
  From: Michael Jerris 
  To: freeswitch-users at lists.freeswitch.org 
  Sent: Sunday, August 09, 2009 12:19 AM
  Subject: Re: [Freeswitch-users] FreeSwitch doesn't play music on holdforbriged channel


  What do the debug logs on fs say when you try to put the call on hold? 


  Mike


  On Aug 6, 2009, at 1:01 PM, Kozak Vladimir wrote:



    The scenario is the following:
    FS User A dial an extension
    Extention opens outbound socket channel to my application
    My application bridges the call to FS User B
    The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id
    The application sends hold to the bridged channel using SendMsg with Other-leg-unique-id
    User B is placed on hold but no music on hold is played to the caller (User A)


    I have outbound socket channel and the following sequence of commands/event:
    listening on [any] 8084 ...
    connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250] 34000
    connect





------------------------------------------------------------------------------


  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org



  The scenario is the following:
  FS User A dial an extension
  Extention opens outbound socket channel to my application
  My application bridges the call to FS User B
  The application check for CHANNEL_BRIDGED event and stores Other-leg-unique-id
  The application sends hold to the bridged channel using SendMsg with Other-leg-unique-id
  User B is placed on hold but no music on hold is played to the caller (User A)


  I have outbound socket channel and the following sequence of commands/event:
  listening on [any] 8084 ...
  connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250] 34000
  connect

  myevents

  SendMsg
  call-command: execute
  execute-app-name: bridge
  execute-app-arg:user/1000 at uat.agent.starpoundtech.net

  Channel-Username: 1001
  Channel-Dialplan: XML
  Channel-Caller-ID-Name: 1001
  Channel-Caller-ID-Number: 1001
  Channel-Network-Addr: 172.26.10.39
  Channel-Destination-Number: 6666
  Channel-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Channel-Source: mod_sofia
  Channel-Context: default
  Channel-Channel-Name: sofia/internal/1001%40172.26.200.250
  Channel-Profile-Index: 1
  Channel-Profile-Created-Time: 1249142681680114
  Channel-Channel-Created-Time: 1249142681680114
  Channel-Channel-Answered-Time: 0
  Channel-Channel-Progress-Time: 0
  Channel-Channel-Progress-Media-Time: 1249142681809352
  Channel-Channel-Hangup-Time: 0
  Channel-Channel-Transfer-Time: 0
  Channel-Screen-Bit: true
  Channel-Privacy-Hide-Name: false
  Channel-Privacy-Hide-Number: false
  Channel-State: CS_EXECUTE
  Channel-State-Number: 4
  Channel-Name: sofia/internal/1001%40172.26.200.250
  Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Call-Direction: inbound
  Answer-State: early
  Channel-Read-Codec-Name: PCMU
  Channel-Read-Codec-Rate: 8000
  Channel-Write-Codec-Name: PCMU
  Channel-Write-Codec-Rate: 8000
  Caller-Username: 1001
  Caller-Dialplan: XML
  Caller-Caller-ID-Name: 1001
  Caller-Caller-ID-Number: 1001
  Caller-Network-Addr: 172.26.10.39
  Caller-Destination-Number: 6666
  Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Caller-Source: mod_sofia
  Caller-Context: default
  Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
  Caller-Profile-Index: 1
  Caller-Profile-Created-Time: 1249142681680114
  Caller-Channel-Created-Time: 1249142681680114
  Caller-Channel-Answered-Time: 0
  Caller-Channel-Progress-Time: 0
  Caller-Channel-Progress-Media-Time: 1249142681809352
  Caller-Channel-Hangup-Time: 0
  Caller-Channel-Transfer-Time: 0
  Caller-Screen-Bit: true
  Caller-Privacy-Hide-Name: false
  Caller-Privacy-Hide-Number: false
  variable_sip_received_ip: 172.26.10.39
  variable_sip_received_port: 13488
  variable_sip_via_protocol: udp
  variable_sip_authorized: true
  variable_sip_mailbox: 1001
  variable_sip_auth_username: 1001
  variable_sip_auth_realm: 172.26.200.250
  variable_mailbox: 1001
  variable_toll_allow: domestic,international,local
  variable_accountcode: 1001
  variable_user_context: default
  variable_effective_caller_id_name: Extension%201001
  variable_effective_caller_id_number: 1001
  variable_outbound_caller_id_name: StarPound%20FreeSWITCH
  variable_outbound_caller_id_number: 0000000000
  variable_callgroup: techsupport
  variable_sip_from_user: 1001
  variable_sip_from_uri: 1001%40172.26.200.250
  variable_sip_from_host: 172.26.200.250
  variable_sip_from_user_stripped: 1001
  variable_sip_from_tag: bd11f93c
  variable_sofia_profile_name: internal
  variable_sip_req_user: 6666
  variable_sip_req_uri: 6666%40172.26.200.250
  variable_sip_req_host: 172.26.200.250
  variable_sip_to_user: 6666
  variable_sip_to_uri: 6666%40172.26.200.250
  variable_sip_to_host: 172.26.200.250
  variable_sip_contact_user: 1001
  variable_sip_contact_port: 13488
  variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
  variable_sip_contact_host: 172.26.10.39
  variable_channel_name: sofia/internal/1001%40172.26.200.250
  variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
  variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
  variable_sip_via_host: 172.26.10.39
  variable_sip_via_port: 13488
  variable_sip_via_rport: 13488
  variable_max_forwards: 70
  variable_presence_id: 1001%40172.26.200.250
  variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
  variable_remote_media_ip: 172.26.10.39
  variable_remote_media_port: 29826
  variable_read_codec: PCMU
  variable_read_rate: 8000
  variable_write_codec: PCMU
  variable_write_rate: 8000
  variable_use_profile: nat
  variable_record_stereo: true
  variable_transfer_fallback_extension: operator
  variable_numbering_plan: US
  variable_default_areacode: 918
  variable_default_gateway: example.com
  variable_user_name: default
  variable_domain_name: 172.26.200.250
  variable_current_application_data: 172.26.200.251%3A8084%20async%20full
  variable_current_application: socket
  variable_socket_host: 172.26.200.251
  variable_local_media_ip: 172.26.200.250
  variable_local_media_port: 29370
  variable_endpoint_disposition: EARLY%20MEDIA
  variable_sip_nat_detected: true
  Content-Type: command/reply
  Reply-Text: %2BOK%0A
  Socket-Mode: async
  Control: full

  Content-Type: command/reply
  Reply-Text: +OK Events Enabled

  Content-Type: command/reply
  Reply-Text: +OK

  Content-Length: 1541
  Content-Type: text/event-plain

  Channel-State: CS_EXECUTE
  Channel-State-Number: 4
  Channel-Name: sofia/internal/1001%40172.26.200.250
  Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Call-Direction: inbound
  Answer-State: early
  Channel-Read-Codec-Name: PCMU
  Channel-Read-Codec-Rate: 8000
  Channel-Write-Codec-Name: PCMU
  Channel-Write-Codec-Rate: 8000
  Caller-Username: 1001
  Caller-Dialplan: XML
  Caller-Caller-ID-Name: 1001
  Caller-Caller-ID-Number: 1001
  Caller-Network-Addr: 172.26.10.39
  Caller-Destination-Number: 6666
  Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Caller-Source: mod_sofia
  Caller-Context: default
  Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
  Caller-Profile-Index: 1
  Caller-Profile-Created-Time: 1249142681680114
  Caller-Channel-Created-Time: 1249142681680114
  Caller-Channel-Answered-Time: 0
  Caller-Channel-Progress-Time: 0
  Caller-Channel-Progress-Media-Time: 1249142681809352
  Caller-Channel-Hangup-Time: 0
  Caller-Channel-Transfer-Time: 0
  Caller-Screen-Bit: true
  Caller-Privacy-Hide-Name: false
  Caller-Privacy-Hide-Number: false
  Application: bridge
  Application-Data: user/1000%40uat.agent.starpoundtech.net
  Event-Name: CHANNEL_EXECUTE
  Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
  FreeSWITCH-Hostname: centos4-4-vm
  FreeSWITCH-IPv4: 172.26.200.250
  FreeSWITCH-IPv6: %3A%3A1
  Event-Date-Local: 2009-08-01%2020%3A04%3A51
  Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A51%20GMT
  Event-Date-Timestamp: 1249142691754598
  Event-Calling-File: switch_core_session.c
  Event-Calling-Function: switch_core_session_exec
  Event-Calling-Line-Number: 1333

  Content-Length: 5242
  Content-Type: text/event-plain

  Channel-State: CS_EXECUTE
  Channel-State-Number: 4
  Channel-Name: sofia/internal/1001%40172.26.200.250
  Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Call-Direction: inbound
  Answer-State: answered
  Channel-Read-Codec-Name: PCMU
  Channel-Read-Codec-Rate: 8000
  Channel-Write-Codec-Name: PCMU
  Channel-Write-Codec-Rate: 8000
  Caller-Username: 1001
  Caller-Dialplan: XML
  Caller-Caller-ID-Name: 1001
  Caller-Caller-ID-Number: 1001
  Caller-Network-Addr: 172.26.10.39
  Caller-Destination-Number: 6666
  Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Caller-Source: mod_sofia
  Caller-Context: default
  Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
  Caller-Profile-Index: 1
  Caller-Profile-Created-Time: 1249142681680114
  Caller-Channel-Created-Time: 1249142681680114
  Caller-Channel-Answered-Time: 1249142692414509
  Caller-Channel-Progress-Time: 1249142691898434
  Caller-Channel-Progress-Media-Time: 1249142681809352
  Caller-Channel-Hangup-Time: 0
  Caller-Channel-Transfer-Time: 0
  Caller-Screen-Bit: true
  Caller-Privacy-Hide-Name: false
  Caller-Privacy-Hide-Number: false
  Other-Leg-Username: 1001
  Other-Leg-Dialplan: XML
  Other-Leg-Caller-ID-Name: Extension%201001
  Other-Leg-Caller-ID-Number: 1001
  Other-Leg-Network-Addr: 172.26.10.39
  Other-Leg-Destination-Number: sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
  Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
  Other-Leg-Source: mod_sofia
  Other-Leg-Context: default
  Other-Leg-Channel-Name: sofia/internal/sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
  Other-Leg-Screen-Bit: true
  Other-Leg-Privacy-Hide-Name: false
  Other-Leg-Privacy-Hide-Number: false
  variable_sip_received_ip: 172.26.10.39
  variable_sip_received_port: 13488
  variable_sip_via_protocol: udp
  variable_sip_authorized: true
  variable_sip_mailbox: 1001
  variable_sip_auth_username: 1001
  variable_sip_auth_realm: 172.26.200.250
  variable_mailbox: 1001
  variable_toll_allow: domestic,international,local
  variable_accountcode: 1001
  variable_user_context: default
  variable_effective_caller_id_name: Extension%201001
  variable_effective_caller_id_number: 1001
  variable_outbound_caller_id_name: StarPound%20FreeSWITCH
  variable_outbound_caller_id_number: 0000000000
  variable_callgroup: techsupport
  variable_sip_from_user: 1001
  variable_sip_from_uri: 1001%40172.26.200.250
  variable_sip_from_host: 172.26.200.250
  variable_sip_from_user_stripped: 1001
  variable_sip_from_tag: bd11f93c
  variable_sofia_profile_name: internal
  variable_sip_req_user: 6666
  variable_sip_req_uri: 6666%40172.26.200.250
  variable_sip_req_host: 172.26.200.250
  variable_sip_to_user: 6666
  variable_sip_to_uri: 6666%40172.26.200.250
  variable_sip_to_host: 172.26.200.250
  variable_sip_contact_user: 1001
  variable_sip_contact_port: 13488
  variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
  variable_sip_contact_host: 172.26.10.39
  variable_channel_name: sofia/internal/1001%40172.26.200.250
  variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
  variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
  variable_sip_via_host: 172.26.10.39
  variable_sip_via_port: 13488
  variable_sip_via_rport: 13488
  variable_max_forwards: 70
  variable_presence_id: 1001%40172.26.200.250
  variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
  variable_remote_media_ip: 172.26.10.39
  variable_remote_media_port: 29826
  variable_read_codec: PCMU
  variable_read_rate: 8000
  variable_write_codec: PCMU
  variable_write_rate: 8000
  variable_use_profile: nat
  variable_record_stereo: true
  variable_transfer_fallback_extension: operator
  variable_numbering_plan: US
  variable_default_areacode: 918
  variable_default_gateway: example.com
  variable_user_name: default
  variable_domain_name: 172.26.200.250
  variable_socket_host: 172.26.200.251
  variable_local_media_ip: 172.26.200.250
  variable_local_media_port: 29370
  variable_endpoint_disposition: EARLY%20MEDIA
  variable_current_application_data: user/1000%40uat.agent.starpoundtech.net
  variable_current_application: bridge
  variable_dialed_user: 1000
  variable_dialed_domain: uat.agent.starpoundtech.net
  variable_originate_disposition: failure
  variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
  variable_sip_redirect_contact_user_0: 1000
  variable_sip_redirect_contact_host_0: 172.26.10.39
  variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
  variable_sip_nat_detected: true
  Event-Name: CHANNEL_ANSWER
  Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
  FreeSWITCH-Hostname: centos4-4-vm
  FreeSWITCH-IPv4: 172.26.200.250
  FreeSWITCH-IPv6: %3A%3A1
  Event-Date-Local: 2009-08-01%2020%3A04%3A52
  Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
  Event-Date-Timestamp: 1249142692414509
  Event-Calling-File: switch_channel.c
  Event-Calling-Function: switch_channel_perform_mark_answered
  Event-Calling-Line-Number: 1776

  Content-Length: 5233
  Content-Type: text/event-plain

  Channel-State: CS_EXECUTE
  Channel-State-Number: 4
  Channel-Name: sofia/internal/1001%40172.26.200.250
  Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Call-Direction: inbound
  Answer-State: answered
  Channel-Read-Codec-Name: PCMU
  Channel-Read-Codec-Rate: 8000
  Channel-Write-Codec-Name: PCMU
  Channel-Write-Codec-Rate: 8000
  Caller-Username: 1001
  Caller-Dialplan: XML
  Caller-Caller-ID-Name: 1001
  Caller-Caller-ID-Number: 1001
  Caller-Network-Addr: 172.26.10.39
  Caller-Destination-Number: 6666
  Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
  Caller-Source: mod_sofia
  Caller-Context: default
  Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
  Caller-Profile-Index: 1
  Caller-Profile-Created-Time: 1249142681680114
  Caller-Channel-Created-Time: 1249142681680114
  Caller-Channel-Answered-Time: 1249142692414509
  Caller-Channel-Progress-Time: 1249142691898434
  Caller-Channel-Progress-Media-Time: 1249142681809352
  Caller-Channel-Hangup-Time: 0
  Caller-Channel-Transfer-Time: 0
  Caller-Screen-Bit: true
  Caller-Privacy-Hide-Name: false
  Caller-Privacy-Hide-Number: false
  Other-Leg-Username: 1001
  Other-Leg-Dialplan: XML
  Other-Leg-Caller-ID-Name: Extension%201001
  Other-Leg-Caller-ID-Number: 1001
  Other-Leg-Network-Addr: 172.26.10.39
  Other-Leg-Destination-Number: sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
  Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
  Other-Leg-Source: mod_sofia
  Other-Leg-Context: default
  Other-Leg-Channel-Name: sofia/internal/sip%3A1000%40172.26.10.39%3A60152%3Brinstance%3D393ff32df3ec5e33%3Bfs_nat%3Dyes
  Other-Leg-Screen-Bit: true
  Other-Leg-Privacy-Hide-Name: false
  Other-Leg-Privacy-Hide-Number: false
  variable_sip_received_ip: 172.26.10.39
  variable_sip_received_port: 13488
  variable_sip_via_protocol: udp
  variable_sip_authorized: true
  variable_sip_mailbox: 1001
  variable_sip_auth_username: 1001
  variable_sip_auth_realm: 172.26.200.250
  variable_mailbox: 1001
  variable_toll_allow: domestic,international,local
  variable_accountcode: 1001
  variable_user_context: default
  variable_effective_caller_id_name: Extension%201001
  variable_effective_caller_id_number: 1001
  variable_outbound_caller_id_name: StarPound%20FreeSWITCH
  variable_outbound_caller_id_number: 0000000000
  variable_callgroup: techsupport
  variable_sip_from_user: 1001
  variable_sip_from_uri: 1001%40172.26.200.250
  variable_sip_from_host: 172.26.200.250
  variable_sip_from_user_stripped: 1001
  variable_sip_from_tag: bd11f93c
  variable_sofia_profile_name: internal
  variable_sip_req_user: 6666
  variable_sip_req_uri: 6666%40172.26.200.250
  variable_sip_req_host: 172.26.200.250
  variable_sip_to_user: 6666
  variable_sip_to_uri: 6666%40172.26.200.250
  variable_sip_to_host: 172.26.200.250
  variable_sip_contact_user: 1001
  variable_sip_contact_port: 13488
  variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
  variable_sip_contact_host: 172.26.10.39
  variable_channel_name: sofia/internal/1001%40172.26.200.250
  variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
  variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
  variable_sip_via_host: 172.26.10.39
  variable_sip_via_port: 13488
  variable_sip_via_rport: 13488
  variable_max_forwards: 70
  variable_presence_id: 1001%40172.26.200.250
  variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
  variable_remote_media_ip: 172.26.10.39
  variable_remote_media_port: 29826
  variable_read_codec: PCMU
  variable_read_rate: 8000
  variable_write_codec: PCMU
  variable_write_rate: 8000
  variable_use_profile: nat
  variable_record_stereo: true
  variable_transfer_fallback_extension: operator
  variable_numbering_plan: US
  variable_default_areacode: 918
  variable_default_gateway: example.com
  variable_user_name: default
  variable_domain_name: 172.26.200.250
  variable_socket_host: 172.26.200.251
  variable_local_media_ip: 172.26.200.250
  variable_local_media_port: 29370
  variable_current_application_data: user/1000%40uat.agent.starpoundtech.net
  variable_current_application: bridge
  variable_dialed_user: 1000
  variable_dialed_domain: uat.agent.starpoundtech.net
  variable_sip_redirect_contact_user_0: 1000
  variable_sip_redirect_contact_host_0: 172.26.10.39
  variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN%20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac%3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
  variable_sip_nat_detected: true
  variable_endpoint_disposition: ANSWER
  variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
  variable_originate_disposition: SUCCESS
  Event-Name: CHANNEL_BRIDGE
  Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
  FreeSWITCH-Hostname: centos4-4-vm
  FreeSWITCH-IPv4: 172.26.200.250
  FreeSWITCH-IPv6: %3A%3A1
  Event-Date-Local: 2009-08-01%2020%3A04%3A52
  Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
  Event-Date-Timestamp: 1249142692414509
  Event-Calling-File: switch_ivr_bridge.c
  Event-Calling-Function: switch_ivr_multi_threaded_bridge
  Event-Calling-Line-Number: 828

  SendMsg 94b59a38-57c4-4703-9c6e-9985d832d119
  call-command: execute
  execute-app-name: hold

  Content-Type: command/reply
  Reply-Text: +OK




  - I don't see the variable hold_music ... did you remove it? 

  - I didn't. Moreover, I tried to set it explicitly using api uuid_setvar.


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From markmorreny at gmail.com  Mon Aug 10 03:58:04 2009
From: markmorreny at gmail.com (mark morreny)
Date: Mon, 10 Aug 2009 18:58:04 +0800
Subject: [Freeswitch-users] Fwd:  Scheduler in module
In-Reply-To: <c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
Message-ID: <20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>

Hi,

Thank you for pointing out sched_api.

What about if I want to do a recurring schedule. is it possible or I should
just call sched_api another time in the execution of my clean up code.

Thanks,
Mark

---------- Forwarded message ----------
From: Moises Silva <moises.silva at gmail.com>
Date: Mon, Aug 10, 2009 at 12:05 AM
Subject: Re: [Freeswitch-users] Scheduler in module
To: freeswitch-users at lists.freeswitch.org


On Sun, Aug 9, 2009 at 11:57 AM, mark morreny <markmorreny at gmail.com> wrote:

> Hi,
>
> I would like to collect some data in my module which would listen to the
> event socket and then a scheduler would kick off every so often to clean up
> the data.
>
> Does anyone know what is the best way to implement a scheduler in a mod?
>
> Best Regards,
> Mark
>

Just use the API that the FreeSWITCH core already has for you. Check
src/include/switch_scheduler.h for details, the comments in there should be
enough for you to learn to use it. If not, you can ask here more specific
questions.
-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com

_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From mike at jerris.com  Mon Aug 10 05:01:20 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 Aug 2009 08:01:20 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
Message-ID: <266B3B3C-73E6-4264-AFFA-73551794365F@jerris.com>

Anyone experiencing this issue please email me the source ip and the  
address that files.freeswitch.org is resolving as.

Mike

On Aug 10, 2009, at 1:26 AM, Godson Gera <godson.g at gmail.com> wrote:

> Hi FS Team,
>
>
>         The files.freeswitch.org is resetting connection since 3  
> days. As a result I was not able to download latest release of FS.  
> Got the trunk version from svn. But still it suffers from the lack  
> of sound files.  When ever I do 'make uhd-sounds-install' http://files.freeswitch.org 
>  resets connection immediately wget tries 20 times and gives up.  
> Other users on IRC also reported this issue.
>
> -- 
> Thanks & Regards,
> Godson Gera
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Mon Aug 10 05:04:39 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 Aug 2009 08:04:39 -0400
Subject: [Freeswitch-users] can freeswitch group_call to gateway
	extension ?
In-Reply-To: <ecd05ce30908092316s66051433p506df518942ec93f@mail.gmail.com>
References: <ecd05ce30908092316s66051433p506df518942ec93f@mail.gmail.com>
Message-ID: <261A7939-0C0E-48DF-B4D4-9A02D5616898@jerris.com>

Can you rephrase your question?

On Aug 10, 2009, at 2:16 AM, daqiang wang <wangdq.no1 at gmail.com> wrote:

> Hello every one :
>   I tested FS. and when I use group_call.  But I can't call the  
> extension in group and not registered in FS.
>   Why ? and What  ?
>   thanks .
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From t.mahe at telemaque.fr  Mon Aug 10 02:00:26 2009
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Mon, 10 Aug 2009 11:00:26 +0200
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <20090808025613.GA19871@hijacked.us>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
	<20090808025613.GA19871@hijacked.us>
Message-ID: <4A7FE1AA.6030309@telemaque.fr>

Morning guys,

Yes the latest make swigall broke php code generation.

Simple workaround:

edit libs/esl/ESL.i to this content:

----------------------Cut----------------------
%{
#include "esl.h"
#include "esl_oop.h"
%}

%include "esl_oop.h"
---------------------Cut-----------------------

and make reswig

It should be a swig bug, but I'm not sure.

Regards,

Gled

Andrew Thompson a ?crit :
> On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
>   
>> Hi,
>>
>> I'm trying to get started with the ESL using PHP. I compiled the ESL, then
>> phpmod according to the wiki instructions, but then when I try the examples
>> in the libs/esl/php dir, they fail saying:
>>
>> PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
>> /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132
>>
>> Checking ESL.php on line 132, I see there are several different declarations
>> for the function __construct() with different parameters each, which makes
>> sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
>> higher than that or something? What could be the problem?
>>
>>     
>
> Someone in the IRC channel mentioned this too. I looked at it briefly
> and it looks like the latest 'swigall' screwed it up. The original
> reporter said he'd file a jira, but you may want to check yourself and
> if not make one yourself. In the meantime, the previous version of the
> file was reported to work if you really need it.
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
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From mike at jerris.com  Mon Aug 10 05:10:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 Aug 2009 08:10:32 -0400
Subject: [Freeswitch-users] Fw: FreeSwitch doesn't play music on hold
	forbriged channel
In-Reply-To: <C5CEF36D42B347B49C82CDE05C3B0E5D@abisoft.biz>
References: <C5CEF36D42B347B49C82CDE05C3B0E5D@abisoft.biz>
Message-ID: <E1B1030B-D1EA-4EB1-80A7-650EBD3B44D7@jerris.com>

The logs indicate that you are already doing a broadcast to the  
channel.  Can you please come up with the simplest test script to  
reproduce this issue and post to Jira.freeswitch.org

Mike

On Aug 10, 2009, at 6:00 AM, "Kozak Vladimir" <vkozak at abisoft.spb.ru>  
wrote:

> 2009-08-12 03:02:03 [DEBUG] switch_core_session.c:706  
> switch_core_session_queue_private_event() Send signal sofia/internal/ 
> sip:1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> [BREAK]
> 2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:228  
> audio_bridge_thread() sofia/internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> receive message [UNBRIDGE]
> 2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523  
> switch_core_session_perform_receive_message() Send signal sofia/ 
> internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
> 2009-08-12 03:02:03 [DEBUG] switch_ivr.c:540  
> switch_ivr_parse_event()  sofia/internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> Command Execute hold()
> 2009-08-12 03:02:03 [DEBUG] switch_ivr.c:1054 switch_ivr_hold()  
> sofia/internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> receive message [HOLD]
> 2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523  
> switch_core_session_perform_receive_message() Send signal sofia/ 
> internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
> 2009-08-12 03:02:03 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state()  
> Channel sofia/internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> entering state [calling]
> 2009-08-12 03:02:03 [WARNING] switch_ivr_async.c:2239  
> switch_ivr_broadcast() Channel [sofia/internal/1001 at 172.26.200.250] 
> [local_stream://moh] already broadcasting...broadcast aborted
> 2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:231  
> audio_bridge_thread() sofia/internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> receive message [BRIDGE]
> 2009-08-12 03:02:03 [DEBUG] switch_core_session.c:523  
> switch_core_session_perform_receive_message() Send signal sofia/ 
> internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes [BREAK]
> 2009-08-12 03:02:03 [DEBUG] switch_ivr_bridge.c:233  
> audio_bridge_thread() Send signal sofia/internal/1001 at 172.26.200.250  
> [BREAK]
> 2009-08-12 03:02:03 [DEBUG] sofia.c:2728 sofia_handle_sip_i_state()  
> Channel sofia/internal/sip: 
> 1000 at 172.26.10.39:14656;rinstance=01d2259ae2e1949f;fs_nat=yes  
> entering state [ready]
> 2009-08-12 03:02:03 [DEBUG] sofia.c:2732 sofia_handle_sip_i_state()  
> Remote SDP:
> v=0
> o=- 6 3 IN IP4 172.26.10.39
> s=CounterPath eyeBeam 1.5
> c=IN IP4 172.26.10.39
> t=0 0
> m=audio 15712 RTP/AVP 0 101
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=recvonly
>
> 2009-08-12 03:02:03 [DEBUG] sofia_glue.c:2483  
> sofia_glue_negotiate_sdp() Our existing sdp is still good [PCMU 172.26.10.39 
> :15712], let's keep it.
> 2009-08-12 03:02:03 [DEBUG] sofia_glue.c:2509  
> sofia_glue_negotiate_sdp() Set 2833 dtmf payload to 101
>
> ----- Original Message -----
> From: Michael Jerris
> To: freeswitch-users at lists.freeswitch.org
> Sent: Sunday, August 09, 2009 12:19 AM
> Subject: Re: [Freeswitch-users] FreeSwitch doesn't play music on  
> holdforbriged channel
>
> What do the debug logs on fs say when you try to put the call on hold?
>
> Mike
>
> On Aug 6, 2009, at 1:01 PM, Kozak Vladimir wrote:
>
>>
>> The scenario is the following:
>> FS User A dial an extension
>> Extention opens outbound socket channel to my application
>> My application bridges the call to FS User B
>> The application check for CHANNEL_BRIDGED event and stores Other- 
>> leg-unique-id
>> The application sends hold to the bridged channel using SendMsg  
>> with Other-leg-unique-id
>> User B is placed on hold but no music on hold is played to the  
>> caller (User A)
>>
>>
>> I have outbound socket channel and the following sequence of  
>> commands/event:
>> listening on [any] 8084 ...
>> connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250 
>> ] 34000
>> connect
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> The scenario is the following:
> FS User A dial an extension
> Extention opens outbound socket channel to my application
> My application bridges the call to FS User B
> The application check for CHANNEL_BRIDGED event and stores Other-leg- 
> unique-id
> The application sends hold to the bridged channel using SendMsg with  
> Other-leg-unique-id
> User B is placed on hold but no music on hold is played to the  
> caller (User A)
>
>
> I have outbound socket channel and the following sequence of  
> commands/event:
> listening on [any] 8084 ...
> connect to [172.26.200.251] from centos4-4-vm.abisoft.spb.ru [172.26.200.250 
> ] 34000
> connect
>
> myevents
>
> SendMsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg:user/1000 at uat.agent.starpoundtech.net
>
> Channel-Username: 1001
> Channel-Dialplan: XML
> Channel-Caller-ID-Name: 1001
> Channel-Caller-ID-Number: 1001
> Channel-Network-Addr: 172.26.10.39
> Channel-Destination-Number: 6666
> Channel-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Channel-Source: mod_sofia
> Channel-Context: default
> Channel-Channel-Name: sofia/internal/1001%40172.26.200.250
> Channel-Profile-Index: 1
> Channel-Profile-Created-Time: 1249142681680114
> Channel-Channel-Created-Time: 1249142681680114
> Channel-Channel-Answered-Time: 0
> Channel-Channel-Progress-Time: 0
> Channel-Channel-Progress-Media-Time:    1249142681809352
> Channel-Channel-Hangup-Time: 0
> Channel-Channel-Transfer-Time: 0
> Channel-Screen-Bit: true
> Channel-Privacy-Hide-Name: false
> Channel-Privacy-Hide-Number: false
> Channel-State: CS_EXECUTE
> Channel-State-Number: 4
> Channel-Name: sofia/internal/1001%40172.26.200.250
> Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Call-Direction: inbound
> Answer-State: early
> Channel-Read-Codec-Name: PCMU
> Channel-Read-Codec-Rate: 8000
> Channel-Write-Codec-Name: PCMU
> Channel-Write-Codec-Rate: 8000
> Caller-Username: 1001
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: 1001
> Caller-Caller-ID-Number: 1001
> Caller-Network-Addr: 172.26.10.39
> Caller-Destination-Number: 6666
> Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Caller-Source: mod_sofia
> Caller-Context: default
> Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1249142681680114
> Caller-Channel-Created-Time: 1249142681680114
> Caller-Channel-Answered-Time: 0
> Caller-Channel-Progress-Time: 0
> Caller-Channel-Progress-Media-Time: 1249142681809352
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
> variable_sip_received_ip: 172.26.10.39
> variable_sip_received_port: 13488
> variable_sip_via_protocol: udp
> variable_sip_authorized: true
> variable_sip_mailbox: 1001
> variable_sip_auth_username: 1001
> variable_sip_auth_realm: 172.26.200.250
> variable_mailbox: 1001
> variable_toll_allow: domestic,international,local
> variable_accountcode: 1001
> variable_user_context: default
> variable_effective_caller_id_name: Extension%201001
> variable_effective_caller_id_number: 1001
> variable_outbound_caller_id_name: StarPound%20FreeSWITCH
> variable_outbound_caller_id_number: 0000000000
> variable_callgroup: techsupport
> variable_sip_from_user: 1001
> variable_sip_from_uri: 1001%40172.26.200.250
> variable_sip_from_host: 172.26.200.250
> variable_sip_from_user_stripped: 1001
> variable_sip_from_tag: bd11f93c
> variable_sofia_profile_name: internal
> variable_sip_req_user: 6666
> variable_sip_req_uri: 6666%40172.26.200.250
> variable_sip_req_host: 172.26.200.250
> variable_sip_to_user: 6666
> variable_sip_to_uri: 6666%40172.26.200.250
> variable_sip_to_host: 172.26.200.250
> variable_sip_contact_user: 1001
> variable_sip_contact_port: 13488
> variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
> variable_sip_contact_host: 172.26.10.39
> variable_channel_name: sofia/internal/1001%40172.26.200.250
> variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
> variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
> variable_sip_via_host: 172.26.10.39
> variable_sip_via_port: 13488
> variable_sip_via_rport: 13488
> variable_max_forwards: 70
> variable_presence_id: 1001%40172.26.200.250
> variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN 
> %20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN 
> %20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP 
> %200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D 
> %0Aa%3Dfmtp%3A101%200-15%0D%0A
> variable_remote_media_ip: 172.26.10.39
> variable_remote_media_port: 29826
> variable_read_codec: PCMU
> variable_read_rate: 8000
> variable_write_codec: PCMU
> variable_write_rate: 8000
> variable_use_profile: nat
> variable_record_stereo: true
> variable_transfer_fallback_extension: operator
> variable_numbering_plan: US
> variable_default_areacode: 918
> variable_default_gateway: example.com
> variable_user_name: default
> variable_domain_name: 172.26.200.250
> variable_current_application_data: 172.26.200.251%3A8084%20async 
> %20full
> variable_current_application: socket
> variable_socket_host: 172.26.200.251
> variable_local_media_ip: 172.26.200.250
> variable_local_media_port: 29370
> variable_endpoint_disposition: EARLY%20MEDIA
> variable_sip_nat_detected: true
> Content-Type: command/reply
> Reply-Text: %2BOK%0A
> Socket-Mode: async
> Control: full
>
> Content-Type: command/reply
> Reply-Text: +OK Events Enabled
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> Content-Length: 1541
> Content-Type: text/event-plain
>
> Channel-State: CS_EXECUTE
> Channel-State-Number: 4
> Channel-Name: sofia/internal/1001%40172.26.200.250
> Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Call-Direction: inbound
> Answer-State: early
> Channel-Read-Codec-Name: PCMU
> Channel-Read-Codec-Rate: 8000
> Channel-Write-Codec-Name: PCMU
> Channel-Write-Codec-Rate: 8000
> Caller-Username: 1001
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: 1001
> Caller-Caller-ID-Number: 1001
> Caller-Network-Addr: 172.26.10.39
> Caller-Destination-Number: 6666
> Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Caller-Source: mod_sofia
> Caller-Context: default
> Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1249142681680114
> Caller-Channel-Created-Time: 1249142681680114
> Caller-Channel-Answered-Time: 0
> Caller-Channel-Progress-Time: 0
> Caller-Channel-Progress-Media-Time: 1249142681809352
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
> Application: bridge
> Application-Data: user/1000%40uat.agent.starpoundtech.net
> Event-Name: CHANNEL_EXECUTE
> Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
> FreeSWITCH-Hostname: centos4-4-vm
> FreeSWITCH-IPv4: 172.26.200.250
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-08-01%2020%3A04%3A51
> Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A51%20GMT
> Event-Date-Timestamp: 1249142691754598
> Event-Calling-File: switch_core_session.c
> Event-Calling-Function: switch_core_session_exec
> Event-Calling-Line-Number: 1333
>
> Content-Length: 5242
> Content-Type: text/event-plain
>
> Channel-State: CS_EXECUTE
> Channel-State-Number: 4
> Channel-Name: sofia/internal/1001%40172.26.200.250
> Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Call-Direction: inbound
> Answer-State: answered
> Channel-Read-Codec-Name: PCMU
> Channel-Read-Codec-Rate: 8000
> Channel-Write-Codec-Name: PCMU
> Channel-Write-Codec-Rate: 8000
> Caller-Username: 1001
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: 1001
> Caller-Caller-ID-Number: 1001
> Caller-Network-Addr: 172.26.10.39
> Caller-Destination-Number: 6666
> Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Caller-Source: mod_sofia
> Caller-Context: default
> Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1249142681680114
> Caller-Channel-Created-Time: 1249142681680114
> Caller-Channel-Answered-Time: 1249142692414509
> Caller-Channel-Progress-Time: 1249142691898434
> Caller-Channel-Progress-Media-Time: 1249142681809352
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
> Other-Leg-Username: 1001
> Other-Leg-Dialplan: XML
> Other-Leg-Caller-ID-Name: Extension%201001
> Other-Leg-Caller-ID-Number: 1001
> Other-Leg-Network-Addr: 172.26.10.39
> Other-Leg-Destination-Number: sip 
> %3A1000%40172.26.10.39%3A60152%3Brinstance 
> %3D393ff32df3ec5e33%3Bfs_nat%3Dyes
> Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
> Other-Leg-Source: mod_sofia
> Other-Leg-Context: default
> Other-Leg-Channel-Name: sofia/internal/sip 
> %3A1000%40172.26.10.39%3A60152%3Brinstance 
> %3D393ff32df3ec5e33%3Bfs_nat%3Dyes
> Other-Leg-Screen-Bit: true
> Other-Leg-Privacy-Hide-Name: false
> Other-Leg-Privacy-Hide-Number: false
> variable_sip_received_ip: 172.26.10.39
> variable_sip_received_port: 13488
> variable_sip_via_protocol: udp
> variable_sip_authorized: true
> variable_sip_mailbox: 1001
> variable_sip_auth_username: 1001
> variable_sip_auth_realm: 172.26.200.250
> variable_mailbox: 1001
> variable_toll_allow: domestic,international,local
> variable_accountcode: 1001
> variable_user_context: default
> variable_effective_caller_id_name: Extension%201001
> variable_effective_caller_id_number: 1001
> variable_outbound_caller_id_name: StarPound%20FreeSWITCH
> variable_outbound_caller_id_number: 0000000000
> variable_callgroup: techsupport
> variable_sip_from_user: 1001
> variable_sip_from_uri: 1001%40172.26.200.250
> variable_sip_from_host: 172.26.200.250
> variable_sip_from_user_stripped: 1001
> variable_sip_from_tag: bd11f93c
> variable_sofia_profile_name: internal
> variable_sip_req_user: 6666
> variable_sip_req_uri: 6666%40172.26.200.250
> variable_sip_req_host: 172.26.200.250
> variable_sip_to_user: 6666
> variable_sip_to_uri: 6666%40172.26.200.250
> variable_sip_to_host: 172.26.200.250
> variable_sip_contact_user: 1001
> variable_sip_contact_port: 13488
> variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
> variable_sip_contact_host: 172.26.10.39
> variable_channel_name: sofia/internal/1001%40172.26.200.250
> variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
> variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
> variable_sip_via_host: 172.26.10.39
> variable_sip_via_port: 13488
> variable_sip_via_rport: 13488
> variable_max_forwards: 70
> variable_presence_id: 1001%40172.26.200.250
> variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN 
> %20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN 
> %20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP 
> %200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D 
> %0Aa%3Dfmtp%3A101%200-15%0D%0A
> variable_remote_media_ip: 172.26.10.39
> variable_remote_media_port: 29826
> variable_read_codec: PCMU
> variable_read_rate: 8000
> variable_write_codec: PCMU
> variable_write_rate: 8000
> variable_use_profile: nat
> variable_record_stereo: true
> variable_transfer_fallback_extension: operator
> variable_numbering_plan: US
> variable_default_areacode: 918
> variable_default_gateway: example.com
> variable_user_name: default
> variable_domain_name: 172.26.200.250
> variable_socket_host: 172.26.200.251
> variable_local_media_ip: 172.26.200.250
> variable_local_media_port: 29370
> variable_endpoint_disposition: EARLY%20MEDIA
> variable_current_application_data: user/ 
> 1000%40uat.agent.starpoundtech.net
> variable_current_application: bridge
> variable_dialed_user: 1000
> variable_dialed_domain: uat.agent.starpoundtech.net
> variable_originate_disposition: failure
> variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
> variable_sip_redirect_contact_user_0: 1000
> variable_sip_redirect_contact_host_0: 172.26.10.39
> variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN 
> %20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac 
> %3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio 
> %2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap 
> %3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
> variable_sip_nat_detected: true
> Event-Name: CHANNEL_ANSWER
> Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
> FreeSWITCH-Hostname: centos4-4-vm
> FreeSWITCH-IPv4: 172.26.200.250
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-08-01%2020%3A04%3A52
> Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
> Event-Date-Timestamp: 1249142692414509
> Event-Calling-File: switch_channel.c
> Event-Calling-Function: switch_channel_perform_mark_answered
> Event-Calling-Line-Number: 1776
>
> Content-Length: 5233
> Content-Type: text/event-plain
>
> Channel-State: CS_EXECUTE
> Channel-State-Number: 4
> Channel-Name: sofia/internal/1001%40172.26.200.250
> Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Call-Direction: inbound
> Answer-State: answered
> Channel-Read-Codec-Name: PCMU
> Channel-Read-Codec-Rate: 8000
> Channel-Write-Codec-Name: PCMU
> Channel-Write-Codec-Rate: 8000
> Caller-Username: 1001
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: 1001
> Caller-Caller-ID-Number: 1001
> Caller-Network-Addr: 172.26.10.39
> Caller-Destination-Number: 6666
> Caller-Unique-ID: 15826d29-b807-4955-a2f9-038e0b3ee6e2
> Caller-Source: mod_sofia
> Caller-Context: default
> Caller-Channel-Name: sofia/internal/1001%40172.26.200.250
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1249142681680114
> Caller-Channel-Created-Time: 1249142681680114
> Caller-Channel-Answered-Time: 1249142692414509
> Caller-Channel-Progress-Time: 1249142691898434
> Caller-Channel-Progress-Media-Time: 1249142681809352
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: true
> Caller-Privacy-Hide-Name: false
> Caller-Privacy-Hide-Number: false
> Other-Leg-Username: 1001
> Other-Leg-Dialplan: XML
> Other-Leg-Caller-ID-Name: Extension%201001
> Other-Leg-Caller-ID-Number: 1001
> Other-Leg-Network-Addr: 172.26.10.39
> Other-Leg-Destination-Number: sip 
> %3A1000%40172.26.10.39%3A60152%3Brinstance 
> %3D393ff32df3ec5e33%3Bfs_nat%3Dyes
> Other-Leg-Unique-ID: 94b59a38-57c4-4703-9c6e-9985d832d119
> Other-Leg-Source: mod_sofia
> Other-Leg-Context: default
> Other-Leg-Channel-Name: sofia/internal/sip 
> %3A1000%40172.26.10.39%3A60152%3Brinstance 
> %3D393ff32df3ec5e33%3Bfs_nat%3Dyes
> Other-Leg-Screen-Bit: true
> Other-Leg-Privacy-Hide-Name: false
> Other-Leg-Privacy-Hide-Number: false
> variable_sip_received_ip: 172.26.10.39
> variable_sip_received_port: 13488
> variable_sip_via_protocol: udp
> variable_sip_authorized: true
> variable_sip_mailbox: 1001
> variable_sip_auth_username: 1001
> variable_sip_auth_realm: 172.26.200.250
> variable_mailbox: 1001
> variable_toll_allow: domestic,international,local
> variable_accountcode: 1001
> variable_user_context: default
> variable_effective_caller_id_name: Extension%201001
> variable_effective_caller_id_number: 1001
> variable_outbound_caller_id_name: StarPound%20FreeSWITCH
> variable_outbound_caller_id_number: 0000000000
> variable_callgroup: techsupport
> variable_sip_from_user: 1001
> variable_sip_from_uri: 1001%40172.26.200.250
> variable_sip_from_host: 172.26.200.250
> variable_sip_from_user_stripped: 1001
> variable_sip_from_tag: bd11f93c
> variable_sofia_profile_name: internal
> variable_sip_req_user: 6666
> variable_sip_req_uri: 6666%40172.26.200.250
> variable_sip_req_host: 172.26.200.250
> variable_sip_to_user: 6666
> variable_sip_to_uri: 6666%40172.26.200.250
> variable_sip_to_host: 172.26.200.250
> variable_sip_contact_user: 1001
> variable_sip_contact_port: 13488
> variable_sip_contact_uri: 1001%40172.26.10.39%3A13488
> variable_sip_contact_host: 172.26.10.39
> variable_channel_name: sofia/internal/1001%40172.26.200.250
> variable_sip_call_id: ZTYwOGM2NDNmMzA5ZjFmOWRhZGJiNTZkMDEyMjQ4YTc.
> variable_sip_user_agent: X-Lite%20release%201103d%20stamp%2053117
> variable_sip_via_host: 172.26.10.39
> variable_sip_via_port: 13488
> variable_sip_via_rport: 13488
> variable_max_forwards: 70
> variable_presence_id: 1001%40172.26.200.250
> variable_switch_r_sdp: v%3D0%0D%0Ao%3D-%208%202%20IN 
> %20IP4%20172.26.10.39%0D%0As%3DCounterPath%20X-Lite%203.0%0D%0Ac%3DIN 
> %20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio%2029826%20RTP/AVP 
> %200%208%203%20101%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D 
> %0Aa%3Dfmtp%3A101%200-15%0D%0A
> variable_remote_media_ip: 172.26.10.39
> variable_remote_media_port: 29826
> variable_read_codec: PCMU
> variable_read_rate: 8000
> variable_write_codec: PCMU
> variable_write_rate: 8000
> variable_use_profile: nat
> variable_record_stereo: true
> variable_transfer_fallback_extension: operator
> variable_numbering_plan: US
> variable_default_areacode: 918
> variable_default_gateway: example.com
> variable_user_name: default
> variable_domain_name: 172.26.200.250
> variable_socket_host: 172.26.200.251
> variable_local_media_ip: 172.26.200.250
> variable_local_media_port: 29370
> variable_current_application_data: user/ 
> 1000%40uat.agent.starpoundtech.net
> variable_current_application: bridge
> variable_dialed_user: 1000
> variable_dialed_domain: uat.agent.starpoundtech.net
> variable_sip_redirect_contact_user_0: 1000
> variable_sip_redirect_contact_host_0: 172.26.10.39
> variable_switch_m_sdp: v%3D0%0D%0Ao%3D-%201%202%20IN 
> %20IP4%20172.26.10.39%0D%0As%3DCounterPath%20eyeBeam%201.5%0D%0Ac 
> %3DIN%20IP4%20172.26.10.39%0D%0At%3D0%200%0D%0Am%3Daudio 
> %2063944%20RTP/AVP%200%208%203%20101%0D%0Aa%3Drtpmap 
> %3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-15%0D%0A
> variable_sip_nat_detected: true
> variable_endpoint_disposition: ANSWER
> variable_signal_bond: 94b59a38-57c4-4703-9c6e-9985d832d119
> variable_originate_disposition: SUCCESS
> Event-Name: CHANNEL_BRIDGE
> Core-UUID: ffb7a71e-0045-4013-89e2-f8c8ccbcfb4b
> FreeSWITCH-Hostname: centos4-4-vm
> FreeSWITCH-IPv4: 172.26.200.250
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-08-01%2020%3A04%3A52
> Event-Date-GMT: Sat,%2001%20Aug%202009%2016%3A04%3A52%20GMT
> Event-Date-Timestamp: 1249142692414509
> Event-Calling-File: switch_ivr_bridge.c
> Event-Calling-Function: switch_ivr_multi_threaded_bridge
> Event-Calling-Line-Number: 828
>
> SendMsg 94b59a38-57c4-4703-9c6e-9985d832d119
> call-command: execute
> execute-app-name: hold
>
> Content-Type: command/reply
> Reply-Text: +OK
>
>
>
>
> - I don't see the variable hold_music ... did you remove it?
>
> - I didn't. Moreover, I tried to set it explicitly using api  
> uuid_setvar.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> DIV>
> - I don't see the variable hold_music ... did you remove it?
>
> - I didn't. Moreover, I tried to set it explicitly using api  
> uuid_setvar.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> TE>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Mon Aug 10 05:13:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 Aug 2009 08:13:27 -0400
Subject: [Freeswitch-users] Fwd:  Scheduler in module
In-Reply-To: <20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
Message-ID: <985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>

Re schedule is done in your callback, take a look at places that use  
these apis in the code for details.

On Aug 10, 2009, at 6:58 AM, mark morreny <markmorreny at gmail.com> wrote:

> Hi,
>
> Thank you for pointing out sched_api.
>
> What about if I want to do a recurring schedule. is it possible or I  
> should just call sched_api another time in the execution of my clean  
> up code.
>
> Thanks,
> Mark
>
> ---------- Forwarded message ----------
> From: Moises Silva <moises.silva at gmail.com>
> Date: Mon, Aug 10, 2009 at 12:05 AM
> Subject: Re: [Freeswitch-users] Scheduler in module
> To: freeswitch-users at lists.freeswitch.org
>
>
> On Sun, Aug 9, 2009 at 11:57 AM, mark morreny  
> <markmorreny at gmail.com> wrote:
> Hi,
>
> I would like to collect some data in my module which would listen to  
> the event socket and then a scheduler would kick off every so often  
> to clean up the data.
>
> Does anyone know what is the best way to implement a scheduler in a  
> mod?
>
> Best Regards,
> Mark
>
> Just use the API that the FreeSWITCH core already has for you. Check  
> src/include/switch_scheduler.h for details, the comments in there  
> should be enough for you to learn to use it. If not, you can ask  
> here more specific questions.
>
> -- 
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON  
> L3R 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From mike at jerris.com  Mon Aug 10 05:22:29 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 Aug 2009 08:22:29 -0400
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <4A7FE1AA.6030309@telemaque.fr>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
	<20090808025613.GA19871@hijacked.us>
	<4A7FE1AA.6030309@telemaque.fr>
Message-ID: <F123F0B3-9CD9-429B-97EE-83BBA6D74E46@jerris.com>

Can you please post a patch to Jira.freswitch.org and assign it to me.

Mike

On Aug 10, 2009, at 5:00 AM, Tristan Mah? <t.mahe at telemaque.fr> wrote:

> Morning guys,
>
> Yes the latest make swigall broke php code generation.
>
> Simple workaround:
>
> edit libs/esl/ESL.i to this content:
>
> ----------------------Cut----------------------
> %{
> #include "esl.h"
> #include "esl_oop.h"
> %}
>
> %include "esl_oop.h"
> ---------------------Cut-----------------------
>
> and make reswig
>
> It should be a swig bug, but I'm not sure.
>
> Regards,
>
> Gled
>
> Andrew Thompson a ?crit :
>>
>> On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
>>
>>> Hi,
>>>
>>> I'm trying to get started with the ESL using PHP. I compiled the  
>>> ESL, then
>>> phpmod according to the wiki instructions, but then when I try the  
>>> examples
>>> in the libs/esl/php dir, they fail saying:
>>>
>>> PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
>>> /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132
>>>
>>> Checking ESL.php on line 132, I see there are several different  
>>> declarations
>>> for the function __construct() with different parameters each,  
>>> which makes
>>> sense, but doens't work. I am using PHP 5.1.6, is there a required  
>>> minimum
>>> higher than that or something? What could be the problem?
>>>
>>>
>> Someone in the IRC channel mentioned this too. I looked at it briefly
>> and it looks like the latest 'swigall' screwed it up. The original
>> reporter said he'd file a jira, but you may want to check yourself  
>> and
>> if not make one yourself. In the meantime, the previous version of  
>> the
>> file was reported to work if you really need it.
>>
>> Andrew
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From wangdq.no1 at gmail.com  Mon Aug 10 05:26:18 2009
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Mon, 10 Aug 2009 20:26:18 +0800
Subject: [Freeswitch-users] can freeswitch group_call to gateway
	extension ?
In-Reply-To: <261A7939-0C0E-48DF-B4D4-9A02D5616898@jerris.com>
References: <ecd05ce30908092316s66051433p506df518942ec93f@mail.gmail.com>
	<261A7939-0C0E-48DF-B4D4-9A02D5616898@jerris.com>
Message-ID: <ecd05ce30908100526j31f5d5ddl2df548d3da6f7a36@mail.gmail.com>

I mean:
   if the telephone number . is in ghe group . (defined in
/directory/default.xml ) .
  but the telephone number is not registered on the FS . (for example: the
telephone number is a mobile number).
 when I dialed the group_call  extension.
 the mobile can't ring.
 why ?   thanks.


2009/8/10 Michael Jerris <mike at jerris.com>

> Can you rephrase your question?
>
> On Aug 10, 2009, at 2:16 AM, daqiang wang <wangdq.no1 at gmail.com> wrote:
>
> > Hello every one :
> >   I tested FS. and when I use group_call.  But I can't call the
> > extension in group and not registered in FS.
> >   Why ? and What  ?
> >   thanks .
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From t.mahe at telemaque.fr  Mon Aug 10 05:34:33 2009
From: t.mahe at telemaque.fr (=?UTF-8?B?VHJpc3RhbiBNYWjDqQ==?=)
Date: Mon, 10 Aug 2009 14:34:33 +0200
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <F123F0B3-9CD9-429B-97EE-83BBA6D74E46@jerris.com>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>	<20090808025613.GA19871@hijacked.us>	<4A7FE1AA.6030309@telemaque.fr>
	<F123F0B3-9CD9-429B-97EE-83BBA6D74E46@jerris.com>
Message-ID: <4A8013D9.30806@telemaque.fr>

Hello Mike,

No problem to post a patch, but it would break perl/python/etc... as
actually there are some functions defined in ESL.i, wouldn't it ?
I don't know about swig, never used that.

Maybe we could have two swig files instead ? one for generating php and
one for the other languages ?
I'm on IRC if you want to discuss it.

Gled.

Michael Jerris a ?crit :
> Can you please post a patch to Jira.freswitch.org and assign it to me.
>
> Mike
>
> On Aug 10, 2009, at 5:00 AM, Tristan Mah? <t.mahe at telemaque.fr
> <mailto:t.mahe at telemaque.fr>> wrote:
>
>> Morning guys,
>>
>> Yes the latest make swigall broke php code generation.
>>
>> Simple workaround:
>>
>> edit libs/esl/ESL.i to this content:
>>
>> ----------------------Cut----------------------
>> %{
>> #include "esl.h"
>> #include "esl_oop.h"
>> %}
>>
>> %include "esl_oop.h"
>> ---------------------Cut-----------------------
>>
>> and make reswig
>>
>> It should be a swig bug, but I'm not sure.
>>
>> Regards,
>>
>> Gled
>>
>> Andrew Thompson a ?crit :
>>> On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
>>>   
>>>> Hi,
>>>>
>>>> I'm trying to get started with the ESL using PHP. I compiled the ESL, then
>>>> phpmod according to the wiki instructions, but then when I try the examples
>>>> in the libs/esl/php dir, they fail saying:
>>>>
>>>> PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
>>>> /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132
>>>>
>>>> Checking ESL.php on line 132, I see there are several different declarations
>>>> for the function __construct() with different parameters each, which makes
>>>> sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
>>>> higher than that or something? What could be the problem?
>>>>
>>>>     
>>> Someone in the IRC channel mentioned this too. I looked at it briefly
>>> and it looks like the latest 'swigall' screwed it up. The original
>>> reporter said he'd file a jira, but you may want to check yourself and
>>> if not make one yourself. In the meantime, the previous version of the
>>> file was reported to work if you really need it.
>>>
>>> Andrew
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>   
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
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From brian at freeswitch.org  Mon Aug 10 05:53:31 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 07:53:31 -0500
Subject: [Freeswitch-users] Fwd:  Scheduler in module
In-Reply-To: <985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
Message-ID: <5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>

switch_rtp.c has a simple one for the zrtp cache storing.

/b

On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:

> Re schedule is done in your callback, take a look at places that use  
> these apis in the code for details.




From brian at freeswitch.org  Mon Aug 10 06:31:10 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 08:31:10 -0500
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
Message-ID: <80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>

As MikeJ pointed out please report the IP address files.freeswitch.org  
resolves to.  We have that on a content delivery network so that the  
files are closer to you geographically and you can download them  
faster but if you're having an issue I'll need the IP so I can report  
it correctly.

Thanks,
Brian

On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:

> Hi FS Team,
>
>
>         The files.freeswitch.org is resetting connection since 3  
> days. As a result I was not able to download latest release of FS.  
> Got the trunk version from svn. But still it suffers from the lack  
> of sound files.  When ever I do 'make uhd-sounds-install'http://files.freeswitch.org 
>  resets connection immediately wget tries 20 times and gives up.  
> Other users on IRC also reported this issue.
>
> -- 
> Thanks & Regards,
> Godson Gera

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From nicolas at medularis.com  Mon Aug 10 06:39:19 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Mon, 10 Aug 2009 09:39:19 -0400
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <4A8013D9.30806@telemaque.fr>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
	<20090808025613.GA19871@hijacked.us> <4A7FE1AA.6030309@telemaque.fr>
	<F123F0B3-9CD9-429B-97EE-83BBA6D74E46@jerris.com>
	<4A8013D9.30806@telemaque.fr>
Message-ID: <1b46b4e80908100639mb2b6087jbc006f9e420da45e@mail.gmail.com>

Hi, I tried the ESL.i modification and when I do make reswig I get:

make -C php reswig
make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php'
rm -f esl_wrap.* ESL.so php_ESL.* ESL.php
swig -module ESL -php5 -c++ -DMULTIPLICITY -I../src/include -o esl_wrap.cpp
../ESL.i
swig error : Unrecognized option -php5
Use 'swig -help' for available options.
make[1]: *** [esl_wrap.cpp] Error 1
make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php'
make: *** [reswig] Error 2


On Mon, Aug 10, 2009 at 8:34 AM, Tristan Mah? <t.mahe at telemaque.fr> wrote:

>  Hello Mike,
>
> No problem to post a patch, but it would break perl/python/etc... as
> actually there are some functions defined in ESL.i, wouldn't it ?
> I don't know about swig, never used that.
>
> Maybe we could have two swig files instead ? one for generating php and one
> for the other languages ?
> I'm on IRC if you want to discuss it.
>
> Gled.
>
> Michael Jerris a ?crit :
>
> Can you please post a patch to Jira.freswitch.org and assign it to me.
>
>  Mike
>
> On Aug 10, 2009, at 5:00 AM, Tristan Mah? <t.mahe at telemaque.fr> wrote:
>
>  Morning guys,
>
> Yes the latest make swigall broke php code generation.
>
> Simple workaround:
>
> edit libs/esl/ESL.i to this content:
>
> ----------------------Cut----------------------
> %{
> #include "esl.h"
> #include "esl_oop.h"
> %}
>
> %include "esl_oop.h"
> ---------------------Cut-----------------------
>
> and make reswig
>
> It should be a swig bug, but I'm not sure.
>
> Regards,
>
> Gled
>
> Andrew Thompson a ?crit :
>
> On Fri, Aug 07, 2009 at 06:10:25PM -0400, Nicolas Brenner wrote:
>
>
>  Hi,
>
> I'm trying to get started with the ESL using PHP. I compiled the ESL, then
> phpmod according to the wiki instructions, but then when I try the examples
> in the libs/esl/php dir, they fail saying:
>
> PHP Fatal error:  Cannot redeclare ESLconnection::__construct() in
> /usr/local/src/freeswitch/libs/esl/php/ESL.php on line 132
>
> Checking ESL.php on line 132, I see there are several different declarations
> for the function __construct() with different parameters each, which makes
> sense, but doens't work. I am using PHP 5.1.6, is there a required minimum
> higher than that or something? What could be the problem?
>
>
>
>  Someone in the IRC channel mentioned this too. I looked at it briefly
> and it looks like the latest 'swigall' screwed it up. The original
> reporter said he'd file a jira, but you may want to check yourself and
> if not make one yourself. In the meantime, the previous version of the
> file was reported to work if you really need it.
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>     _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Aug 10 06:41:50 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 08:41:50 -0500
Subject: [Freeswitch-users] Error trying to use PHP ESL
In-Reply-To: <1b46b4e80908100639mb2b6087jbc006f9e420da45e@mail.gmail.com>
References: <1b46b4e80908071510s6deeda58h457b1850c767dbc5@mail.gmail.com>
	<20090808025613.GA19871@hijacked.us>
	<4A7FE1AA.6030309@telemaque.fr>
	<F123F0B3-9CD9-429B-97EE-83BBA6D74E46@jerris.com>
	<4A8013D9.30806@telemaque.fr>
	<1b46b4e80908100639mb2b6087jbc006f9e420da45e@mail.gmail.com>
Message-ID: <0DA5E116-7E6E-4A3E-B86D-88329FB2C2D8@freeswitch.org>

If you're removing the newobject lines you're introducing a huge  
memory leak :P  also I'm guessing your swig is older and doesn't have  
php5 support.

/b

On Aug 10, 2009, at 8:39 AM, Nicolas Brenner wrote:

> Hi, I tried the ESL.i modification and when I do make reswig I get:
>
> make -C php reswig
> make[1]: Entering directory `/usr/local/src/freeswitch/libs/esl/php'
> rm -f esl_wrap.* ESL.so php_ESL.* ESL.php
> swig -module ESL -php5 -c++ -DMULTIPLICITY -I../src/include -o  
> esl_wrap.cpp ../ESL.i
> swig error : Unrecognized option -php5
> Use 'swig -help' for available options.
> make[1]: *** [esl_wrap.cpp] Error 1
> make[1]: Leaving directory `/usr/local/src/freeswitch/libs/esl/php'
> make: *** [reswig] Error 2




From ivan at myrvold.org  Mon Aug 10 08:03:36 2009
From: ivan at myrvold.org (Ivan C Myrvold)
Date: Mon, 10 Aug 2009 17:03:36 +0200
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <092B4355-F911-4E37-A5D1-AC923AF9B8FB@gmail.com>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
	<6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
	<4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
	<092B4355-F911-4E37-A5D1-AC923AF9B8FB@gmail.com>
Message-ID: <E6E26B16-4905-49DE-A167-BEFC7A0D1638@myrvold.org>

Seven,
I am afraid I will not be able to help you much with the Carbon code,  
as I am only good at Cocoa programming. You said you chose Carbon  
because you only needed low level API, and that is fair enough, but I  
will also add that you can do the same with only linking to the  
Foundation framework in Cocoa.

I looked a little at the diff file yesterday, and will investigate  
more today, to try to understand how you have done the Skype  
integration to the Freeswitch in the Carbon code.
And I am glad that someone have contributed to get skypiax working in  
OS X. Great work so far!

Ivan

Den 9. aug.. 2009 kl. 20:02 skrev Seven Du:

> Ivan,
>
> Good to know you are a cocoa dev. Unable to check in code right now,
> will send the diff to you offlist for now.
>
> 0) I'm not familiar with Mac dev, just tried my best
> 1) It doesn't work yet, but should be able to compile, sure you
> already have the Skype framework in place :)
> 2) if run the skype delegate from a threat, then cannot get event
> callback. e.g. mac_client.c works but mac_client2.c doesn't. Since
> skypiax is running in a thread, we need to figure out this first.
> 3) it uses Carbon, since I think we only need to low level api, no
> need to bother the complicate of Cocoa.
> 4) strsep shows some warning on compile, haven't figured out why
> 5) perhaps you should only add one interface in skypiax.conf.xml
> 6) do you want to run multi-instances like on Linux?
> 7) I really not sure if it will work or not :)
>
> Let me know if it helps. I bet you can make it work. Also code will be
> in my branch soon.
>
> 7.
>
>
> On Aug 9, 2009, at 11:34 PM, Ivan C Myrvold wrote:
>> Yes, I am interested in this, and if you have any source I could have
>> a look at it.
>>
>> Ivan
>>
>> Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
>>
>>>
>>> On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
>>>> Ciao Ivan,
>>>>
>>>> it seems that you do not have the libX11 **development** package
>>>> installed.
>>>>
>>>> Unfortunately I don't know about OSX, so I cannot help you, but  
>>>> many
>>>> on the list know.
>>>> BTW: it will probably be of no use to you to compile mod_skypiax on
>>>> OSX, because Skype for MACOSX works in another way than Skype for
>>>> Linux.
>>>
>>> That's right.
>>>
>>>> If you know about MacOSX programming, please have a look at
>>>> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
>>>> probably be simple enough to add a message pump for MacOSX.
>>>>
>>>> -giovanni
>>>>
>>>>
>>> Giovanni, I have a Mac and tried to get this work yesterday, but
>>> haven't got it work. Will try further if I have time. However, I
>>> don't
>>> think it's so useful because I don't know how to run and hence
>>> control
>>> multi-skype instances on Mac.
>>>
>>> If someone interested to try this I can check the code into my
>>> branch.
>>>
>>>>
>>>>
>>>> Sincerely,
>>>>
>>>> Giovanni Maruzzelli
>>>> =========================================
>>>> www.celliax.org
>>>> via Pierlombardo 9, 20135 Milano
>>>> Italy
>>>> gmaruzz at celliax dot org
>>>> Cell : +39-347-2665618
>>>> Fax : +39-02-87390039
>>>>
>>>>
>>>>
>>>>
>>>> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>
>>>> wrote:
>>>>> I tried to compile mod_skypiax, and am getting problem with X11.  
>>>>> On
>>>>> OS
>>>>> X Leopard, X11 is installed in /usr/X11/lib/
>>>>> See below.
>>>>>
>>>>> What can I do to get past this error?
>>>>>
>>>>> I can also let you ssh into my machine. Contact me off list in
>>>>> case.
>>>>>
>>>>> Ivan
>>>>>
>>>>> making all mod_skypiax
>>>>> Compiling skypiax_protocol.c...
>>>>> Compiling mod_skypiax.c...
>>>>> mkdir .libs
>>>>> Compiling mod_skypiax.c ...
>>>>> Creating mod_skypiax.so...
>>>>> ld: library not found for -lX11
>>>>> collect2: ld returned 1 exit status
>>>>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
>>>>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
>>>>> Documents/
>>>>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
>>>>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -
>>>>> g -
>>>>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -
>>>>> shared -
>>>>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-
>>>>> namespace .libs/
>>>>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/
>>>>> Freeswitch/
>>>>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
>>>>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/
>>>>> xml/
>>>>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch. 
>>>>> 09-08-09/
>>>>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
>>>>> libiconv.dylib /
>>>>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
>>>>> apr/.libs/
>>>>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto - 
>>>>> lz -
>>>>> lncurses -lX11
>>>>> make[5]: *** [mod_skypiax.so] Error 1
>>>>> make[4]: *** [all] Error 1
>>>>> make[3]: *** [mod_skypiax-all] Error 1
>>>>> make[2]: *** [all-recursive] Error 1
>>>>>
>>>>>
>>>>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
>>>>>
>>>>>> No, it needs implementation of the message pump between the  
>>>>>> module
>>>>>> and
>>>>>> the Skype API.
>>>>>>
>>>>>> It's probably kind of trivial, if no other problems I'm not aware
>>>>>> of.
>>>>>>
>>>>>> I do not have a Mac to implement it, tough :-(.
>>>>>>
>>>>>> -giovanni
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> Sincerely,
>>>>>>
>>>>>> Giovanni Maruzzelli
>>>>>> =========================================
>>>>>> www.celliax.org
>>>>>> via Pierlombardo 9, 20135 Milano
>>>>>> Italy
>>>>>> gmaruzz at celliax dot org
>>>>>> Cell : +39-347-2665618
>>>>>> Fax : +39-02-87390039
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
>>>>>> wrote:
>>>>>>> I'm not sure about that one.... I haven't tried lately because
>>>>>>> the
>>>>>>> API
>>>>>>> differs on the Mac last I looked at it.
>>>>>>>
>>>>>>> /b
>>>>>>>
>>>>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
>>>>>>>
>>>>>>>> Is skypiax now working on Mac OS X in Freeswitch?
>>>>>>>>
>>>>>>>> Ivan
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> FreeSWITCH-users mailing list
>>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>>> http://www.freeswitch.org
>>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>




From email.list.subscriber at gmail.com  Mon Aug 10 08:49:14 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Mon, 10 Aug 2009 11:49:14 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>
Message-ID: <4a804139.1408c00a.6102.38b8@mx.google.com>

Thanks for the response(s):

I ran the "./compile" script with a set PREFIX.  This took a few
attempts with errors before it was able to complete error-free, as I
had to install libtool.

Since then, I have tried running 'make', 'gmake', and
'/opt/gnu/bin/make', but each results with an error.  This is the
error when running 'make' or 'gmake':

<snip>
make: Fatal error: Command failed for target `all-recursive'
Current working directory /home/vmorales/freeswitch-1.0.4
*** Error code 1
make: Fatal error: Command failed for target `all'
</snip>


This is the error when running '/opt/gnu/bin/make':

<snip>
make[5]: *** [mod_amr.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_amr-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               /opt/gnu/bin/make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
</snip>


I re-untar'd before each compile/make attempt.  Let me know if this is
something that I can resolve.

Vladimir


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: Saturday, August 08, 2009 12:37 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

This is not currently a supported platform, it only builds on 64 bit  
right now I think on solaris.

Mike

On Aug 6, 2009, at 6:03 PM, vmorales wrote:

> Hello,
>
> Does anyone have, or know where to get, a pre-compiled copy of
> FreeSwitch for Solaris 10/x86?


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
http://www.freeswitch.org




From david.nembrot at sogeti.com  Mon Aug 10 09:56:47 2009
From: david.nembrot at sogeti.com (David Nembrot)
Date: Mon, 10 Aug 2009 18:56:47 +0200
Subject: [Freeswitch-users] Freeswitch :: VoIP issues using ILBC audio codec
Message-ID: <20090810185647.a3ymeb6d9w804s84@mail.sogeti.com>



   Hello world,

   I've got two FS servers configured as gateways for each other and  
I'm currently testing the telephony. Usinge the ILBC audio codec, I  
figured out that one of the FS servers doesn't forward RTP streams  
correctly to the other server. Here is its status-quo:
INPUT = proper ILBC payload type (97 or 108)
OUTPUT = unknown payload type (97 or 102)

   I've already changed the parameters in internal.xml & external.xml:
<param name="inbound-codec-negotiation" value="greedy"/>
<param name="disable-transcoding" value="true"/>

   When dialing out, I also use the following  
syntax:{absolute_codec_string='GSM,PCMU'}sofia/gateway/mygateway/mynumber

   Is there another thing to do to have proper ILBC streams passing  
through the gateways ?
Thanking y'all in advance ;)

   BR,
David N.

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   Hello world,

   I've got two FS servers configured as gateways for each other and  
I'm currently testing the telephony. Usinge the ILBC audio codec, I  
figured out that one of the FS servers doesn't forward RTP streams  
correctly to the other server. Here is its status-quo:

   INPUT = proper ILBC payload type (97 or 108)
OUTPUT = unknown payload type (97 or 102)

   I've already changed the parameters in internal.xml and also in  
external.xml:
<param name="inbound-codec-negotiation" value="greedy"/>
<param name="disable-transcoding" value="true"/>

   Is there another thing to do to have proper ILBC streams passing  
through the gateways ?
Thanking y'all in advance ;)

   BR,
David N.

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From brian at freeswitch.org  Mon Aug 10 10:03:37 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 12:03:37 -0500
Subject: [Freeswitch-users] Freeswitch :: VoIP issues using ILBC audio
	codec
In-Reply-To: <20090810185647.a3ymeb6d9w804s84@mail.sogeti.com>
References: <20090810185647.a3ymeb6d9w804s84@mail.sogeti.com>
Message-ID: <A0D83BF0-CF9F-41C8-AE0E-1863D7922D36@freeswitch.org>

Well since the codec is in the dynamic range the payload number  
doesn't matter.  I would have to see an RTP trace to see this... you  
must also include all the SIP traffic so wireshark can properly figure  
things out.

/b

On Aug 10, 2009, at 11:56 AM, David Nembrot wrote:

> Hello world,
>
> I've got two FS servers configured as gateways for each other and  
> I'm currently testing the telephony. Usinge the ILBC audio codec, I  
> figured out that one of the FS servers doesn't forward RTP streams  
> correctly to the other server. Here is its status-quo:
> INPUT = proper ILBC payload type (97 or 108)
> OUTPUT = unknown payload type (97 or 102)
>
> I've already changed the parameters in internal.xml & external.xml:
> <param name="inbound-codec-negotiation" value="greedy"/>
> <param name="disable-transcoding" value="true"/>
>
> When dialing out, I also use the following syntax: 
> {absolute_codec_string='GSM,PCMU'}sofia/gateway/mygateway/mynumber
>
> Is there another thing to do to have proper ILBC streams passing  
> through the gateways ?
> Thanking y'all in advance ;)
>
> BR,
> David N.
>
> Hello world,
>
> I've got two FS servers configured as gateways for each other and  
> I'm currently testing the telephony. Usinge the ILBC audio codec, I  
> figured out that one of the FS servers doesn't forward RTP streams  
> correctly to the other server. Here is its status-quo:
>
> INPUT = proper ILBC payload type (97 or 108)
> OUTPUT = unknown payload type (97 or 102)
>
> I've already changed the parameters in internal.xml and also in  
> external.xml:
> <param name="inbound-codec-negotiation" value="greedy"/>
> <param name="disable-transcoding" value="true"/>
>
> Is there another thing to do to have proper ILBC streams passing  
> through the gateways ?
> Thanking y'all in advance ;)
>
> BR,
> David N.
>




From email.list.subscriber at gmail.com  Mon Aug 10 11:14:09 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Mon, 10 Aug 2009 14:14:09 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>
	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>
Message-ID: <4a80632f.1508c00a.4d3c.090d@mx.google.com>

By "./compile" I was referring to "./configure"

Vladimir

-----Original Message-----
From: vmorales [mailto:email.list.subscriber at gmail.com] 
Sent: Monday, August 10, 2009 11:49 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

Thanks for the response(s):

I ran the "./compile" script with a set PREFIX.  This took a few
attempts with errors before it was able to complete error-free, as I
had to install libtool.

Since then, I have tried running 'make', 'gmake', and
'/opt/gnu/bin/make', but each results with an error.  This is the
error when running 'make' or 'gmake':

<snip>
make: Fatal error: Command failed for target `all-recursive'
Current working directory /home/vmorales/freeswitch-1.0.4
*** Error code 1
make: Fatal error: Command failed for target `all'
</snip>


This is the error when running '/opt/gnu/bin/make':

<snip>
make[5]: *** [mod_amr.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_amr-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               /opt/gnu/bin/make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
</snip>


I re-untar'd before each compile/make attempt.  Let me know if this is
something that I can resolve.

Vladimir


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michael Jerris
Sent: Saturday, August 08, 2009 12:37 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

This is not currently a supported platform, it only builds on 64 bit  
right now I think on solaris.

Mike

On Aug 6, 2009, at 6:03 PM, vmorales wrote:

> Hello,
>
> Does anyone have, or know where to get, a pre-compiled copy of
> FreeSwitch for Solaris 10/x86?


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
http://www.freeswitch.org




From pete at privateconnect.com  Mon Aug 10 11:28:06 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Mon, 10 Aug 2009 11:28:06 -0700
Subject: [Freeswitch-users] VoiceMail transcription
Message-ID: <20090810112806.2ad02225396a31c9de30536f2e338977.8f8b5c9b40.wbe@email04.secureserver.net>

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From asannucci at gmail.com  Mon Aug 10 11:43:43 2009
From: asannucci at gmail.com (bakko)
Date: Mon, 10 Aug 2009 20:43:43 +0200
Subject: [Freeswitch-users] Ivr and variables
Message-ID: <8D85DC2E55A74F41AE2190E05A44F028@voztovoice>

I'm tryng to put a variable in a IVR digits line like this:

<entry action="menu-exec-app" digits="${some}" param="transfer 1001 XML 
default"/>

but don't work

Is It possible tu use this solution?

thank you

BR





From dave at 3c.co.uk  Mon Aug 10 11:51:24 2009
From: dave at 3c.co.uk (David Knell)
Date: Mon, 10 Aug 2009 21:51:24 +0300
Subject: [Freeswitch-users] VoiceMail transcription
In-Reply-To: <20090810112806.2ad02225396a31c9de30536f2e338977.8f8b5c9b40.wbe@email04.secureserver.net>
References: <20090810112806.2ad02225396a31c9de30536f2e338977.8f8b5c9b40.wbe@email04.secureserver.net>
Message-ID: <1249930284.20224.34.camel@dk-d820>

Good evening Pete,

The only way to do this is, I'm afraid, to use a human.  We use Amazon's
Mechanical Turk to good effect.

Cheers --

Dave

> Good morning all,
> 
> I realize this is slightly off the FS topic, but I am wondering if
> anyone out there has experience with software packages designed for
> the transcription of voicemails to text.  I've used pocketsphinx  with
> FS to handle IVR menus, but now have the task of figuring out how to
> convert recorded phone conversations (voicemails mostly) to text.
> 
> This does not have to be a real-time process, I can store the audio
> files and process them over time.  This would need to be a software
> (preferable open source) solution.  ASPs like VoiceCloud would not
> work for this application.
> 
> Thanks for any help
> -pete
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk




From max.bridgewater at gmail.com  Mon Aug 10 11:51:36 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 14:51:36 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
Message-ID: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>

Hi get the following messages while running "make" on CentOS 5.3 (Of course
after bootstrap and configure).

os_Linux_x86_64.s: Assembler messages:
os_Linux_x86_64.s:46: Error: bad register name `%rdi)'
os_Linux_x86_64.s:61: Error: bad register name `%rdi)'
os_Linux_x86_64.s:75: Error: bad register name `%rdi)'
os_Linux_x86_64.s:89: Error: bad register name `%rdi)'
make[10]: *** [os_Linux_x86_64.o] Error 1
make[9]: *** [export] Error 2
make[8]: *** [export] Error 2
make[7]: *** [export] Error 2
make[6]: *** [export] Error 2
make[5]: *** [/home/installs/fstrunk/libs/js/libjs.la] Error 2
make[4]: *** [all] Error 1
make[3]: *** [mod_spidermonkey-all] Error 1
make[2]: *** [all-recursive] Error 1

Any idea?

Max.
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From brian at freeswitch.org  Mon Aug 10 11:53:44 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 13:53:44 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
Message-ID: <F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>

Sounds like you have a hardware problem... I just built fine on 5.3 ..  
just make sure you don't do make -j on the first build.

/b

On Aug 10, 2009, at 1:51 PM, Max Bridgewater wrote:

> Hi get the following messages while running "make" on CentOS 5.3 (Of  
> course after bootstrap and configure).
>
> os_Linux_x86_64.s: Assembler messages:
> os_Linux_x86_64.s:46: Error: bad register name `%rdi)'
> os_Linux_x86_64.s:61: Error: bad register name `%rdi)'
> os_Linux_x86_64.s:75: Error: bad register name `%rdi)'
> os_Linux_x86_64.s:89: Error: bad register name `%rdi)'
> make[10]: *** [os_Linux_x86_64.o] Error 1
> make[9]: *** [export] Error 2
> make[8]: *** [export] Error 2
> make[7]: *** [export] Error 2
> make[6]: *** [export] Error 2
> make[5]: *** [/home/installs/fstrunk/libs/js/libjs.la] Error 2
> make[4]: *** [all] Error 1
> make[3]: *** [mod_spidermonkey-all] Error 1
> make[2]: *** [all-recursive] Error 1
>
> Any idea?
>
> Max.

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From pete at privateconnect.com  Mon Aug 10 11:59:49 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Mon, 10 Aug 2009 11:59:49 -0700
Subject: [Freeswitch-users] VoiceMail transcription
Message-ID: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>

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From max.bridgewater at gmail.com  Mon Aug 10 12:02:58 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 15:02:58 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
Message-ID: <c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>

Hmm too bad. It's a new VPS that i purchased. I'm not using make -j and l'm
really sure what to do next.


On Mon, Aug 10, 2009 at 2:53 PM, Brian West <brian at freeswitch.org> wrote:

> Sounds like you have a hardware problem... I just
> built fine on 5.3 .. just make sure you don't do make -j on the first build.
> /b
>
> On Aug 10, 2009, at 1:51 PM, Max Bridgewater wrote:
>
> Hi get the following messages while running "make" on CentOS 5.3 (Of course
> after bootstrap and configure).
>
> os_Linux_x86_64.s: Assembler messages:
> os_Linux_x86_64.s:46: Error: bad register name `%rdi)'
> os_Linux_x86_64.s:61: Error: bad register name `%rdi)'
> os_Linux_x86_64.s:75: Error: bad register name `%rdi)'
> os_Linux_x86_64.s:89: Error: bad register name `%rdi)'
> make[10]: *** [os_Linux_x86_64.o] Error 1
> make[9]: *** [export] Error 2
> make[8]: *** [export] Error 2
> make[7]: *** [export] Error 2
> make[6]: *** [export] Error 2
> make[5]: *** [/home/installs/fstrunk/libs/js/libjs.la] Error 2
> make[4]: *** [all] Error 1
> make[3]: *** [mod_spidermonkey-all] Error 1
> make[2]: *** [all-recursive] Error 1
>
> Any idea?
>
> Max.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Mon Aug 10 12:04:31 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 10 Aug 2009 15:04:31 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
Message-ID: <40A2F6B2-BEA0-423A-A1E3-86EDE0EEC3AC@avgs.ca>

What arch? do a cat /proc/cpuinfo

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 10-Aug-09, at 3:02 PM, Max Bridgewater wrote:

>
> Hmm too bad. It's a new VPS that i purchased. I'm not using make -j  
> and l'm really sure what to do next.
>
>
> On Mon, Aug 10, 2009 at 2:53 PM, Brian West <brian at freeswitch.org>  
> wrote:
> Sounds like you have a hardware problem... I just built fine on  
> 5.3 .. just make sure you don't do make -j on the first build.
>
> /b
>
> On Aug 10, 2009, at 1:51 PM, Max Bridgewater wrote:
>
>> Hi get the following messages while running "make" on CentOS 5.3  
>> (Of course after bootstrap and configure).
>>
>> os_Linux_x86_64.s: Assembler messages:
>> os_Linux_x86_64.s:46: Error: bad register name `%rdi)'
>> os_Linux_x86_64.s:61: Error: bad register name `%rdi)'
>> os_Linux_x86_64.s:75: Error: bad register name `%rdi)'
>> os_Linux_x86_64.s:89: Error: bad register name `%rdi)'
>> make[10]: *** [os_Linux_x86_64.o] Error 1
>> make[9]: *** [export] Error 2
>> make[8]: *** [export] Error 2
>> make[7]: *** [export] Error 2
>> make[6]: *** [export] Error 2
>> make[5]: *** [/home/installs/fstrunk/libs/js/libjs.la] Error 2
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_spidermonkey-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>>
>> Any idea?
>>
>> Max.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Mon Aug 10 12:09:28 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 14:09:28 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
Message-ID: <30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>

Sounds like a 64bit kernel and 32bit userspace maybe?  Just  
guessing... why oh why oh VPS providers have recicarnail interface  
problems.

/b

On Aug 10, 2009, at 2:02 PM, Max Bridgewater wrote:

>
> Hmm too bad. It's a new VPS that i purchased. I'm not using make -j  
> and l'm really sure what to do next.
>




From max.bridgewater at gmail.com  Mon Aug 10 12:20:28 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 15:20:28 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
Message-ID: <c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>

OK, here is the result of cat /proc/cpuinfo. What conclusion can i draw from
this?


processor       : 0
vendor_id       : GenuineIntel
cpu family      : 6
model           : 15
model name      : Intel(R) Core(TM)2 Quad CPU    Q6600  @ 2.40GHz
stepping        : 11
cpu MHz         : 2400.084
cache size      : 4096 KB
physical id     : 0
siblings        : 1
core id         : 0
cpu cores       : 1
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu tsc msr pae mce cx8 apic mca cmov pat pse36 clflush
dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm constant_tsc pni monitor
ds_cpl vmx est tm2 cx16 xtpr lahf_lm
bogomips        : 6005.15
clflush size    : 64
cache_alignment : 64
address sizes   : 36 bits physical, 48 bits virtual
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From brian at freeswitch.org  Mon Aug 10 12:25:41 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 14:25:41 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
Message-ID: <D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>

and uname -a says?

/b

On Aug 10, 2009, at 2:20 PM, Max Bridgewater wrote:

> OK, here is the result of cat /proc/cpuinfo. What conclusion can i  
> draw from this?
>
>
> processor       : 0
> vendor_id       : GenuineIntel
> cpu family      : 6
> model           : 15
> model name      : Intel(R) Core(TM)2 Quad CPU    Q6600  @ 2.40GHz
> stepping        : 11
> cpu MHz         : 2400.084
> cache size      : 4096 KB
> physical id     : 0
> siblings        : 1
> core id         : 0
> cpu cores       : 1
> fpu             : yes
> fpu_exception   : yes
> cpuid level     : 10
> wp              : yes
> flags           : fpu tsc msr pae mce cx8 apic mca cmov pat pse36  
> clflush dts acpi mmx fxsr sse sse2 ss ht tm syscall nx lm  
> constant_tsc pni monitor ds_cpl vmx est tm2 cx16 xtpr lahf_lm
> bogomips        : 6005.15
> clflush size    : 64
> cache_alignment : 64
> address sizes   : 36 bits physical, 48 bits virtual
>




From max.bridgewater at gmail.com  Mon Aug 10 12:31:04 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 15:31:04 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
Message-ID: <c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>

and uname -a says?


Linux miriam 2.6.18-128.1.10.el5xen #1 SMP Thu May 7 11:07:18 EDT 2009
x86_64 x86_64 x86_64 GNU/Linux
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From brian at freeswitch.org  Mon Aug 10 12:32:50 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 14:32:50 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
Message-ID: <06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>

I would call them and make sure they know they have something broken.

/b

On Aug 10, 2009, at 2:31 PM, Max Bridgewater wrote:

>
>
> and uname -a says?
>
> Linux miriam 2.6.18-128.1.10.el5xen #1 SMP Thu May 7 11:07:18 EDT  
> 2009 x86_64 x86_64 x86_64 GNU/Linux
>




From max.bridgewater at gmail.com  Mon Aug 10 12:49:07 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 15:49:07 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
Message-ID: <c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>

I have no idea what is wrong and therefore what to tell them. Without
telling precisely what is wrong it would be easy for them to just categorize
this as an application issue. Any recommendation?

On Mon, Aug 10, 2009 at 3:32 PM, Brian West <brian at freeswitch.org> wrote:

> I would call them and make sure they know they have something broken.
>
> /b
>
> On Aug 10, 2009, at 2:31 PM, Max Bridgewater wrote:
>
> >
> >
> > and uname -a says?
> >
> > Linux miriam 2.6.18-128.1.10.el5xen #1 SMP Thu May 7 11:07:18 EDT
> > 2009 x86_64 x86_64 x86_64 GNU/Linux
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Aug 10 12:51:35 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 14:51:35 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
	<c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
Message-ID: <5E790EFC-8A43-428B-B1BB-C84014D3422F@freeswitch.org>

Just be like everyone else... "Its broken, FIX IT!"  :P  ... Honestly  
I am not sure what to tell them... Could be a Xen bug... never know.

/b

On Aug 10, 2009, at 2:49 PM, Max Bridgewater wrote:

> I have no idea what is wrong and therefore what to tell them.  
> Without telling precisely what is wrong it would be easy for them to  
> just categorize this as an application issue. Any recommendation?
>
> On Mon, Aug 10, 2009 at 3:32 PM, Brian West <brian at freeswitch.org>  
> wrote:
> I would call them and make sure they know they have something broken.
>
> /b

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From mike at jerris.com  Mon Aug 10 13:01:04 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 10 Aug 2009 13:01:04 -0700
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
	<c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
Message-ID: <C41A1A58-090E-4A71-9AA7-FEEFFBDD99D3@jerris.com>

Is this a 32 bit default compiler?  Try CFLAGS=-m64 on the configure  
line.

On Aug 10, 2009, at 12:49 PM, Max Bridgewater  
<max.bridgewater at gmail.com> wrote:

> I have no idea what is wrong and therefore what to tell them.  
> Without telling precisely what is wrong it would be easy for them to  
> just categorize this as an application issue. Any recommendation?
>
> On Mon, Aug 10, 2009 at 3:32 PM, Brian West <brian at freeswitch.org>  
> wrote:
> I would call them and make sure they know they have something broken.
>
> /b
>
> On Aug 10, 2009, at 2:31 PM, Max Bridgewater wrote:
>
> >
> >
> > and uname -a says?
> >
> > Linux miriam 2.6.18-128.1.10.el5xen #1 SMP Thu May 7 11:07:18 EDT
> > 2009 x86_64 x86_64 x86_64 GNU/Linux
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From pjintheusa at gmail.com  Mon Aug 10 13:35:50 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 10 Aug 2009 16:35:50 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
Message-ID: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>

Hi there,

I am attempting to do a simple bridge. Leg A is an inbound DID. Leg B
is terminated through a SIP carrier.

I am unable to use loopback AND bypass_media_after_bridge=true. The
bridge fails.

Here is my three line application:

Session.Answer();

Session.Execute("set", "bypass_media_after_bridge=true");

Session.Execute("bridge", "loopback/6095553828/default");

FreeSWITCH log: http://pastebin.freeswitch.org/9949

I believe the bridge fails as the SIP-REINVITE is issued and fails.

Using loopback and bypass_media_after_bridge=false works fine. As does
using bypass_media_after_bridge=true and sending call directly through
the gateway to which the loopback points.

e.g.  http://pastebin.freeswitch.org/9948

Has anyone else encountered this? Am I missing something?

Any help would be much appreciated!

Thanks


Phillip Jones



From msc at freeswitch.org  Mon Aug 10 15:56:14 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Aug 2009 17:56:14 -0500
Subject: [Freeswitch-users] question about latest version of mod_limit
In-Reply-To: <20ad6b920908060226t2dcf532aobab23dc0299d0f05@mail.gmail.com>
References: <20ad6b920908060226t2dcf532aobab23dc0299d0f05@mail.gmail.com>
Message-ID: <87f2f3b90908101556s6836143j7103124c94bdd9ad@mail.gmail.com>

Possibly the calls are being disconnected too quickly? Try adding a sleep
for 1000ms on each one and see what happens.
-MC

On Thu, Aug 6, 2009 at 4:26 AM, mark morreny <markmorreny at gmail.com> wrote:

> Hello,
>
> I have the following setup in the dialplan.  Then, I fire up sipp to send
> 5calls/s and I expect to get limit-pass=false in most of the INFO output.
> However, I am getting all "limit-pass=pass".
>
> Does anyone know what is wrong with my dialplan?
>
>
>  <context name="internal">
>
>   <extension name="test-limit">
>
>     <condition field="destination_number" expression="^9(.*)$">
>
>         <action application="limit_hash_execute" data="192.168.1.102 6000
> 1/1 transfer to-next" />
>         <action application="set" data="limit-pass=false"/>
>         <action application="info" />
>         <action application="hangup" />
>
>     </condition>
>
>   </extension>
>     <extension name="to-next">
>         <condition field="destination_number" expression="^to-next$">
>         <action application="set" data="limit-pass=true"/>
>         <action application="info" />
>         <action application="hangup" />
>
>
>
>     </condition>
>     </extension>
> </context>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Mon Aug 10 16:00:18 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Mon, 10 Aug 2009 19:00:18 -0400
Subject: [Freeswitch-users] question about latest version of mod_limit
In-Reply-To: <20ad6b920908060226t2dcf532aobab23dc0299d0f05@mail.gmail.com>
References: <20ad6b920908060226t2dcf532aobab23dc0299d0f05@mail.gmail.com>
Message-ID: <3DF62608-6E4E-4322-85EC-5384FEC58A71@avgs.ca>

You are using it wrong.
<action application="limit_hash_execute" data="192.168.1.102 6000 1/1  
bridge sofia/gateway/etc" />

The purpose of limit_hash_execute is to have a resource that is  
released if the command doesnt complete properly.
If you want to transfer when the res. is over-limit, use limit_hash,  
not limit_hash_execute.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 6-Aug-09, at 5:26 AM, mark morreny wrote:

> Hello,
>
> I have the following setup in the dialplan.  Then, I fire up sipp to  
> send 5calls/s and I expect to get limit-pass=false in most of the  
> INFO output.  However, I am getting all "limit-pass=pass".
>
> Does anyone know what is wrong with my dialplan?
>
>
>  <context name="internal">
>
>   <extension name="test-limit">
>
>     <condition field="destination_number" expression="^9(.*)$">
>
>         <action application="limit_hash_execute" data="192.168.1.102  
> 6000 1/1 transfer to-next" />
>         <action application="set" data="limit-pass=false"/>
>         <action application="info" />
>         <action application="hangup" />
>
>     </condition>
>
>   </extension>
>     <extension name="to-next">
>         <condition field="destination_number" expression="^to-next$">
>         <action application="set" data="limit-pass=true"/>
>         <action application="info" />
>         <action application="hangup" />
>
>
>
>     </condition>
>     </extension>
> </context>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From msc at freeswitch.org  Mon Aug 10 16:04:24 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Aug 2009 18:04:24 -0500
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
Message-ID: <87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>

Be sure to hop on IRC and speak with jmesquita because he's been working on
this also. It would be very good to have Spanish-speaking users review the
Spanish phrase_es.xml file for correctness. Also, we need Spanish-speaking
programmers to assist with the mod_say application for Spanish - there are
things that you have to do in Spanish that you don't have to do in English.

Thanks,
MC

On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

> On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
> edition .
>
> Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for
> FS playback.
>
> Here's a guide that has been put together for reference on what to record.
>
> http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml
>
> Regards,
>
>
>
>
> On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:
>
>> I'd like to begin record spanish prompts for FS.
>>
>> Do you know any software/hardware to make it?
>>
>> Thank you
>>
>> BR
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Aug 10 16:29:50 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 10 Aug 2009 18:29:50 -0500
Subject: [Freeswitch-users] Error while creating object
In-Reply-To: <24895716.post@talk.nabble.com>
References: <7d79b3930908060347xe5be545yfeeafad761aba274@mail.gmail.com>
	<24895716.post@talk.nabble.com>
Message-ID: <87f2f3b90908101629p71048eb1vbccd19a19d328fff@mail.gmail.com>

On Mon, Aug 10, 2009 at 3:00 AM, lakshmanan <lakindia89 at gmail.com> wrote:

>
> Can any one please say what I did wrong here?
>

Maybe this instead?
my $sess=&freeswitch::DTMF->new();
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From kevin at kgolding.co.uk  Mon Aug 10 16:36:02 2009
From: kevin at kgolding.co.uk (Kevin Golding)
Date: Tue, 11 Aug 2009 00:36:02 +0100
Subject: [Freeswitch-users] ACL issue
Message-ID: <4A80AEE2.7050306@kgolding.co.uk>

Hello,

Am in the progress of setting up a test freeswitch box. Have got
internal extensions working, along with an outgoing gateway.

My issue is getting the incoming calls.

The error I'm getting is: "2009-08-10 21:47:42 [DEBUG] sofia.c:3785
sofia_handle_sip_i_invite() IP 213.166.5.129 Rejected by acl "domains".
Falling back to Digest auth."

After reading the docs, I believed I needed to add the 213.166.5.129 IP
to the "domains" list, but alas I could not find such a list by default.
It appears I was missing an acl.conf.xml file. I have created a new
acl.conf.xml file, and have put the following in it:

<configuration name="acl.conf" description="Network Lists">
   <network-lists>
     <list name="domains" value="allow">
         <node type="allow" domain="$${domain}"/>
         <node type="allow" domain="213.166.5.128/28"/>
         <node type="allow" domain="87.238.72.128/26"/>
     </list>
   </network-lists>
</configuration>

But I am still getting the same error for an incoming call.

Best regards,
--
Kevin Golding



From brian at freeswitch.org  Mon Aug 10 16:46:20 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 18:46:20 -0500
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <4A80AEE2.7050306@kgolding.co.uk>
References: <4A80AEE2.7050306@kgolding.co.uk>
Message-ID: <7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>

First off edit the profile to use your own custom ACL... not the  
domains one... secondly you'll have to use cidr= instead of domain=.

Domain= will search the user directory building an ACL list from all  
users with the cidr= attribute.

/b


On Aug 10, 2009, at 6:36 PM, Kevin Golding wrote:

> acl.conf.xml file, and have put the following in it:
>
> <configuration name="acl.conf" description="Network Lists">
>   <network-lists>
>     <list name="domains" value="allow">
>         <node type="allow" domain="$${domain}"/>
>         <node type="allow" domain="213.166.5.128/28"/>
>         <node type="allow" domain="87.238.72.128/26"/>
>     </list>
>   </network-lists>
> </configuration>




From max.bridgewater at gmail.com  Mon Aug 10 17:01:27 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 20:01:27 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <C41A1A58-090E-4A71-9AA7-FEEFFBDD99D3@jerris.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
	<c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
	<C41A1A58-090E-4A71-9AA7-FEEFFBDD99D3@jerris.com>
Message-ID: <c55d46950908101701o50d94acle2f183f744d97591@mail.gmail.com>

I suppose it is a 32 bit default compiler. When i compile with the suggested
flag it fails right away. I've attached the log file for your curiosity.
Meanwhil i've submited the issue to the hosting company. But i doubt
anything meaningful or useful will come from there.

Cheers,
Max.


On Mon, Aug 10, 2009 at 4:01 PM, Michael Jerris <mike at jerris.com> wrote:

> Is this a 32 bit default compiler?  Try CFLAGS=-m64 on the configure line.
>
>
> On Aug 10, 2009, at 12:49 PM, Max Bridgewater <max.bridgewater at gmail.com>
> wrote:
>
> I have no idea what is wrong and therefore what to tell them. Without
> telling precisely what is wrong it would be easy for them to just categorize
> this as an application issue. Any recommendation?
>
> On Mon, Aug 10, 2009 at 3:32 PM, Brian West < <brian at freeswitch.org>
> brian at freeswitch.org> wrote:
>
>> I would call them and make sure they know they have something broken.
>>
>> /b
>>
>> On Aug 10, 2009, at 2:31 PM, Max Bridgewater wrote:
>>
>> >
>> >
>> > and uname -a says?
>> >
>> > Linux miriam 2.6.18-128.1.10.el5xen #1 SMP Thu May 7 11:07:18 EDT
>> > 2009 x86_64 x86_64 x86_64 GNU/Linux
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Mon Aug 10 17:06:08 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 19:06:08 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101701o50d94acle2f183f744d97591@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<F190922F-8190-4FD1-B34B-4373EF2C6AED@freeswitch.org>
	<c55d46950908101202q5056b6a2sad483638014cc867@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
	<c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
	<C41A1A58-090E-4A71-9AA7-FEEFFBDD99D3@jerris.com>
	<c55d46950908101701o50d94acle2f183f744d97591@mail.gmail.com>
Message-ID: <228DA52C-066F-4165-A568-DF9688EF4C61@freeswitch.org>

What flags are you using?
/b

On Aug 10, 2009, at 7:01 PM, Max Bridgewater wrote:

> I suppose it is a 32 bit default compiler. When i compile with the  
> suggested flag it fails right away. I've attached the log file for  
> your curiosity. Meanwhil i've submited the issue to the hosting  
> company. But i doubt anything meaningful or useful will come from  
> there.
>
> Cheers,
> Max.
>




From max.bridgewater at gmail.com  Mon Aug 10 17:14:41 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 10 Aug 2009 20:14:41 -0400
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <228DA52C-066F-4165-A568-DF9688EF4C61@freeswitch.org>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
	<c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
	<C41A1A58-090E-4A71-9AA7-FEEFFBDD99D3@jerris.com>
	<c55d46950908101701o50d94acle2f183f744d97591@mail.gmail.com>
	<228DA52C-066F-4165-A568-DF9688EF4C61@freeswitch.org>
Message-ID: <c55d46950908101714x6690cf17o5b81bbc6ea04f7@mail.gmail.com>

CFLAGS=-m64 on the configure line.



On Mon, Aug 10, 2009 at 8:06 PM, Brian West <brian at freeswitch.org> wrote:

> What flags are you using?
> /b
>
> On Aug 10, 2009, at 7:01 PM, Max Bridgewater wrote:
>
> > I suppose it is a 32 bit default compiler. When i compile with the
> > suggested flag it fails right away. I've attached the log file for
> > your curiosity. Meanwhil i've submited the issue to the hosting
> > company. But i doubt anything meaningful or useful will come from
> > there.
> >
> > Cheers,
> > Max.
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Aug 10 17:53:18 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 10 Aug 2009 19:53:18 -0500
Subject: [Freeswitch-users] Unable to make trunk on CentOS 5.3
In-Reply-To: <c55d46950908101714x6690cf17o5b81bbc6ea04f7@mail.gmail.com>
References: <c55d46950908101151n58714142yf212d612dd508b2@mail.gmail.com>
	<30992FFF-4DE8-4C00-BAF0-BE9580491B68@freeswitch.org>
	<c55d46950908101220g55733a80t18426b240fbe227c@mail.gmail.com>
	<D48724FD-8976-452D-AE1B-92A9059357B4@freeswitch.org>
	<c55d46950908101231s698bc914ud90ec7ebf9829@mail.gmail.com>
	<06349D8F-D1A8-4569-A027-076EE775AB10@freeswitch.org>
	<c55d46950908101249y5c8898f6m4a61eaeea52b41fd@mail.gmail.com>
	<C41A1A58-090E-4A71-9AA7-FEEFFBDD99D3@jerris.com>
	<c55d46950908101701o50d94acle2f183f744d97591@mail.gmail.com>
	<228DA52C-066F-4165-A568-DF9688EF4C61@freeswitch.org>
	<c55d46950908101714x6690cf17o5b81bbc6ea04f7@mail.gmail.com>
Message-ID: <580A1245-3E46-49AB-A95F-1F276F59B738@freeswitch.org>

You shouldn't be doing that.

/b

On Aug 10, 2009, at 7:14 PM, Max Bridgewater wrote:

> CFLAGS=-m64 on the configure line.




From jmesquita at gmail.com  Mon Aug 10 20:00:43 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 11 Aug 2009 00:00:43 -0300
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
Message-ID: <5a8712120908102000m421ea27fv371620bf66d69f6b@mail.gmail.com>

Hey! I have recorded a couple of samples and I will patch whatever is needed
to support portuguese(Brazil) and spanish on say. Don't worry, I am on top
of it.

jmesquita

On Mon, Aug 10, 2009 at 8:04 PM, Michael Collins <msc at freeswitch.org> wrote:

> Be sure to hop on IRC and speak with jmesquita because he's been working on
> this also. It would be very good to have Spanish-speaking users review the
> Spanish phrase_es.xml file for correctness. Also, we need Spanish-speaking
> programmers to assist with the mod_say application for Spanish - there are
> things that you have to do in Spanish that you don't have to do in English.
>
> Thanks,
> MC
>
>
> On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>
>> On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
>> edition .
>>
>> Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for
>> FS playback.
>>
>> Here's a guide that has been put together for reference on what to record.
>>
>> http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml
>>
>> Regards,
>>
>>
>>
>>
>> On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:
>>
>>> I'd like to begin record spanish prompts for FS.
>>>
>>> Do you know any software/hardware to make it?
>>>
>>> Thank you
>>>
>>> BR
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Mon Aug 10 21:03:37 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 11 Aug 2009 12:03:37 +0800
Subject: [Freeswitch-users] skypiax on Mac OS X
In-Reply-To: <E6E26B16-4905-49DE-A167-BEFC7A0D1638@myrvold.org>
References: <FB3633A3-B1F9-4C57-A194-2E7AF412169E@myrvold.org>
	<1CE48DB3-7B9F-484D-A66B-37D55A916CAD@freeswitch.org>
	<7b197bef0908060937u3eb82227u831720f48f3a4425@mail.gmail.com>
	<C8B6DBDA-8453-4F6B-B9D4-F327B7BE49CF@myrvold.org>
	<7b197bef0908090810l2050c4e0g850447f716897fac@mail.gmail.com>
	<6A09A3D1-7201-45B3-AFBE-6D04FD0071AC@gmail.com>
	<4073BF16-A89F-47A6-8859-EF1DBF902EAD@myrvold.org>
	<092B4355-F911-4E37-A5D1-AC923AF9B8FB@gmail.com>
	<E6E26B16-4905-49DE-A167-BEFC7A0D1638@myrvold.org>
Message-ID: <23f91030908102103g50f49af4o97dd97ef0df666a4@mail.gmail.com>

2009/8/10 Ivan C Myrvold <ivan at myrvold.org>

> Seven,
> I am afraid I will not be able to help you much with the Carbon code,
> as I am only good at Cocoa programming. You said you chose Carbon
> because you only needed low level API, and that is fair enough, but I
> will also add that you can do the same with only linking to the
> Foundation framework in Cocoa.
>

Link with Cocoa is OK but just need extra hack to the standard Makefile and
not so necessary. And I think Carbon code is more C friendly.
I made a jira to Skype-Dev, hope someone can help us.


>
> I looked a little at the diff file yesterday, and will investigate
> more today, to try to understand how you have done the Skype
> integration to the Freeswitch in the Carbon code.
> And I am glad that someone have contributed to get skypiax working in
> OS X. Great work so far!
>
> Ivan
>
> Den 9. aug.. 2009 kl. 20:02 skrev Seven Du:
>
> > Ivan,
> >
> > Good to know you are a cocoa dev. Unable to check in code right now,
> > will send the diff to you offlist for now.
> >
> > 0) I'm not familiar with Mac dev, just tried my best
> > 1) It doesn't work yet, but should be able to compile, sure you
> > already have the Skype framework in place :)
> > 2) if run the skype delegate from a threat, then cannot get event
> > callback. e.g. mac_client.c works but mac_client2.c doesn't. Since
> > skypiax is running in a thread, we need to figure out this first.
> > 3) it uses Carbon, since I think we only need to low level api, no
> > need to bother the complicate of Cocoa.
> > 4) strsep shows some warning on compile, haven't figured out why
> > 5) perhaps you should only add one interface in skypiax.conf.xml
> > 6) do you want to run multi-instances like on Linux?
> > 7) I really not sure if it will work or not :)
> >
> > Let me know if it helps. I bet you can make it work. Also code will be
> > in my branch soon.
> >
> > 7.
> >
> >
> > On Aug 9, 2009, at 11:34 PM, Ivan C Myrvold wrote:
> >> Yes, I am interested in this, and if you have any source I could have
> >> a look at it.
> >>
> >> Ivan
> >>
> >> Den 9. aug.. 2009 kl. 17:24 skrev Seven Du:
> >>
> >>>
> >>> On Aug 9, 2009, at 11:10 PM, Giovanni Maruzzelli wrote:
> >>>> Ciao Ivan,
> >>>>
> >>>> it seems that you do not have the libX11 **development** package
> >>>> installed.
> >>>>
> >>>> Unfortunately I don't know about OSX, so I cannot help you, but
> >>>> many
> >>>> on the list know.
> >>>> BTW: it will probably be of no use to you to compile mod_skypiax on
> >>>> OSX, because Skype for MACOSX works in another way than Skype for
> >>>> Linux.
> >>>
> >>> That's right.
> >>>
> >>>> If you know about MacOSX programming, please have a look at
> >>>> https://developer.skype.com/Docs/ApiDoc/Skype_API_on_Mac it would
> >>>> probably be simple enough to add a message pump for MacOSX.
> >>>>
> >>>> -giovanni
> >>>>
> >>>>
> >>> Giovanni, I have a Mac and tried to get this work yesterday, but
> >>> haven't got it work. Will try further if I have time. However, I
> >>> don't
> >>> think it's so useful because I don't know how to run and hence
> >>> control
> >>> multi-skype instances on Mac.
> >>>
> >>> If someone interested to try this I can check the code into my
> >>> branch.
> >>>
> >>>>
> >>>>
> >>>> Sincerely,
> >>>>
> >>>> Giovanni Maruzzelli
> >>>> =========================================
> >>>> www.celliax.org
> >>>> via Pierlombardo 9, 20135 Milano
> >>>> Italy
> >>>> gmaruzz at celliax dot org
> >>>> Cell : +39-347-2665618
> >>>> Fax : +39-02-87390039
> >>>>
> >>>>
> >>>>
> >>>>
> >>>> On Sun, Aug 9, 2009 at 4:52 PM, Ivan C Myrvold<ivan at myrvold.org>
> >>>> wrote:
> >>>>> I tried to compile mod_skypiax, and am getting problem with X11.
> >>>>> On
> >>>>> OS
> >>>>> X Leopard, X11 is installed in /usr/X11/lib/
> >>>>> See below.
> >>>>>
> >>>>> What can I do to get past this error?
> >>>>>
> >>>>> I can also let you ssh into my machine. Contact me off list in
> >>>>> case.
> >>>>>
> >>>>> Ivan
> >>>>>
> >>>>> making all mod_skypiax
> >>>>> Compiling skypiax_protocol.c...
> >>>>> Compiling mod_skypiax.c...
> >>>>> mkdir .libs
> >>>>> Compiling mod_skypiax.c ...
> >>>>> Creating mod_skypiax.so...
> >>>>> ld: library not found for -lX11
> >>>>> collect2: ld returned 1 exit status
> >>>>> gcc -DSKYPIAX_SVN_VERSION=\"14471\" -I/Users/imyrvold/Documents/
> >>>>> Freeswitch/freeswitch.09-08-09/src/include -I/Users/imyrvold/
> >>>>> Documents/
> >>>>> Freeswitch/freeswitch.09-08-09/libs/libteletone/src -Werror -
> >>>>> fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -
> >>>>> g -
> >>>>> ggdb -DMACOSX -g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -
> >>>>> shared -
> >>>>> o .libs/mod_skypiax.so -dynamic -bundle -force-flat-
> >>>>> namespace .libs/
> >>>>> mod_skypiax.o skypiax_protocol.o  /Users/imyrvold/Documents/
> >>>>> Freeswitch/
> >>>>> freeswitch.09-08-09/.libs/libfreeswitch.dylib -L/usr/lib -L/Users/
> >>>>> imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/apr-util/
> >>>>> xml/
> >>>>> expat/lib /Users/imyrvold/Documents/Freeswitch/freeswitch.
> >>>>> 09-08-09/
> >>>>> libs/apr-util/xml/expat/lib/.libs/libexpat.a /usr/lib/
> >>>>> libiconv.dylib /
> >>>>> Users/imyrvold/Documents/Freeswitch/freeswitch.09-08-09/libs/
> >>>>> apr/.libs/
> >>>>> libapr-1.a -ldl -lpthread -lm -L/opt/local/lib -lssl -lcrypto -
> >>>>> lz -
> >>>>> lncurses -lX11
> >>>>> make[5]: *** [mod_skypiax.so] Error 1
> >>>>> make[4]: *** [all] Error 1
> >>>>> make[3]: *** [mod_skypiax-all] Error 1
> >>>>> make[2]: *** [all-recursive] Error 1
> >>>>>
> >>>>>
> >>>>> Den 6. aug.. 2009 kl. 18:37 skrev Giovanni Maruzzelli:
> >>>>>
> >>>>>> No, it needs implementation of the message pump between the
> >>>>>> module
> >>>>>> and
> >>>>>> the Skype API.
> >>>>>>
> >>>>>> It's probably kind of trivial, if no other problems I'm not aware
> >>>>>> of.
> >>>>>>
> >>>>>> I do not have a Mac to implement it, tough :-(.
> >>>>>>
> >>>>>> -giovanni
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> Sincerely,
> >>>>>>
> >>>>>> Giovanni Maruzzelli
> >>>>>> =========================================
> >>>>>> www.celliax.org
> >>>>>> via Pierlombardo 9, 20135 Milano
> >>>>>> Italy
> >>>>>> gmaruzz at celliax dot org
> >>>>>> Cell : +39-347-2665618
> >>>>>> Fax : +39-02-87390039
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>>
> >>>>>> On Thu, Aug 6, 2009 at 5:55 PM, Brian West<brian at freeswitch.org>
> >>>>>> wrote:
> >>>>>>> I'm not sure about that one.... I haven't tried lately because
> >>>>>>> the
> >>>>>>> API
> >>>>>>> differs on the Mac last I looked at it.
> >>>>>>>
> >>>>>>> /b
> >>>>>>>
> >>>>>>> On Aug 6, 2009, at 10:53 AM, Ivan C Myrvold wrote:
> >>>>>>>
> >>>>>>>> Is skypiax now working on Mac OS X in Freeswitch?
> >>>>>>>>
> >>>>>>>> Ivan
> >>>>>>>
> >>>>>>>
> >>>>>>> _______________________________________________
> >>>>>>> FreeSWITCH-users mailing list
> >>>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>>> http://www.freeswitch.org
> >>>>>>>
> >>>>>>
> >>>>>> _______________________________________________
> >>>>>> FreeSWITCH-users mailing list
> >>>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>>> http://www.freeswitch.org
> >>>>>>
> >>>>>
> >>>>>
> >>>>> _______________________________________________
> >>>>> FreeSWITCH-users mailing list
> >>>>> FreeSWITCH-users at lists.freeswitch.org
> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>>> http://www.freeswitch.org
> >>>>>
> >>>>
> >>>> _______________________________________________
> >>>> FreeSWITCH-users mailing list
> >>>> FreeSWITCH-users at lists.freeswitch.org
> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>>> http://www.freeswitch.org
> >>>
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From velu.technical at gmail.com  Mon Aug 10 21:29:44 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Tue, 11 Aug 2009 09:59:44 +0530
Subject: [Freeswitch-users] Fwd: ALARM signal in esl libraries
In-Reply-To: <1452e2980908092338j15c4e33cn2cff799bb64464d0@mail.gmail.com>
References: <1452e2980908092338j15c4e33cn2cff799bb64464d0@mail.gmail.com>
Message-ID: <1452e2980908102129g51255d5bt430a3e73bd6f2e99@mail.gmail.com>

Dear Greats,
    Could you please help me to solve this problem?

---------- Forwarded message ----------
From: velusamy velu <velu.technical at gmail.com>
Date: Mon, Aug 10, 2009 at 12:08 PM
Subject: ALARM signal in esl libraries
To: freeswitch-users at lists.freeswitch.org


Dear All,
       I have registered  ALARM signal  in my perl program to handle the
DTMF digit timeout.
When ALARM signal generated  the  connection with  ESL  is  automatically
closed.

I have checked the connection with "connected: function, it returns 0.

Why the connection was closed?
Is there any idea to alive the connection after ALARM signal generation??

Please help me.........

Thanks,
Velusamy
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From godson.g at gmail.com  Mon Aug 10 22:20:33 2009
From: godson.g at gmail.com (Godson Gera)
Date: Tue, 11 Aug 2009 10:50:33 +0530
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
Message-ID: <f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>

Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
name resolution. But the problem is still there even if I use my ISP's  DNS
(mostly because its resolving to the same to IP address) .

On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org> wrote:

> As MikeJ pointed out please report the IP address files.freeswitch.orgresolves to.  We have that on a content delivery network so that the files
> are closer to you geographically and you can download them faster but if
> you're having an issue I'll need the IP so I can report it correctly.
> Thanks,
> Brian
>
> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>
> Hi FS Team,
>
>
>         The files.freeswitch.org is resetting connection since 3 days. As
> a result I was not able to download latest release of FS. Got the trunk
> version from svn. But still it suffers from the lack of sound files.  When
> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
> connection immediately wget tries 20 times and gives up. Other users on IRC
> also reported this issue.
>
> --
> Thanks & Regards,
> Godson Gera
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Thanks & Regards,
Godson Gera
http://godson.in
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From velu.technical at gmail.com  Mon Aug 10 22:27:15 2009
From: velu.technical at gmail.com (velusamy velu)
Date: Tue, 11 Aug 2009 10:57:15 +0530
Subject: [Freeswitch-users] Timeouts in Dial plan
Message-ID: <1452e2980908102227o90aaef2i2f8fd36a2eaeefb6@mail.gmail.com>

Dear All,
      How to handle timeouts in Dialplan?

Thanks,
Velusamy.
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From peter.olsson at visionutveckling.se  Mon Aug 10 23:16:44 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 11 Aug 2009 08:16:44 +0200
Subject: [Freeswitch-users] OpenZAP/Sangoma in Windows
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED9435@cooper>

Hi, I'm trying to evaluate the OpenZAP/Sangoma-support in Windows, using PRI E1 connections.

I'm aware it's not yet to be considered as stable, but I'd still want to try it out some, and also help detecting bugs while I'm at it :)

Anyway, I have a couple of questions:

1. Has anyone tested it in Windows at all? I know the build-files for Visual Studio has only been checked in for a couple of months, so that's why I'm asking.

2. Does anyone have any directions how to configure the driver within Windows? Should I use BitStream or HDLC, and how should the channel groups be configured?

I have built everything correctly, and I've succeeded to open (some of) the channels when I configure each channel in a separate group, but I don't think that's the way it should be done...

Any suggestions welcome!

/Peter



From peter.olsson at visionutveckling.se  Mon Aug 10 23:20:50 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 11 Aug 2009 08:20:50 +0200
Subject: [Freeswitch-users] UniMRCP support for Windows...
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED9436@cooper>

Yet another Windows question from me.. :)

I've seen that vcproj-files for mod_unimrcp has been added to SVN, but they are not yet included in the main FreeSWITCH solution-file.

What's the curent status of this, should it work if I compile it separately, or is it not yet complete for Windows?

/Peter



From diego.viola at gmail.com  Mon Aug 10 23:54:39 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 11 Aug 2009 02:54:39 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
Message-ID: <86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>

The address for me is: 69.174.57.101

On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera <godson.g at gmail.com> wrote:

> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
> name resolution. But the problem is still there even if I use my ISP's  DNS
> (mostly because its resolving to the same to IP address) .
>
> On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org> wrote:
>
>> As MikeJ pointed out please report the IP address files.freeswitch.orgresolves to.  We have that on a content delivery network so that the files
>> are closer to you geographically and you can download them faster but if
>> you're having an issue I'll need the IP so I can report it correctly.
>> Thanks,
>> Brian
>>
>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>
>> Hi FS Team,
>>
>>
>>         The files.freeswitch.org is resetting connection since 3 days. As
>> a result I was not able to download latest release of FS. Got the trunk
>> version from svn. But still it suffers from the lack of sound files.  When
>> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>> connection immediately wget tries 20 times and gives up. Other users on IRC
>> also reported this issue.
>>
>> --
>> Thanks & Regards,
>> Godson Gera
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Thanks & Regards,
> Godson Gera
> http://godson.in
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From maxim.tsvetov at gmail.com  Tue Aug 11 00:10:36 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Tue, 11 Aug 2009 11:10:36 +0400
Subject: [Freeswitch-users] answer command
Message-ID: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>

Hello,

I found in WIKI list of supported commands
http://wiki.freeswitch.org/wiki/Mod_commands.

I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)

That's what Freeswitch replies for this command:

"answer: Command not found!"

?ould you please help?


Regards,
Maxim Tsvetov
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From kevin at kgolding.co.uk  Tue Aug 11 00:38:20 2009
From: kevin at kgolding.co.uk (Kevin Golding)
Date: Tue, 11 Aug 2009 08:38:20 +0100
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>
References: <4A80AEE2.7050306@kgolding.co.uk>
	<7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>
Message-ID: <4A811FEC.7040808@kgolding.co.uk>

Thanks Brian,

Well spotted with the domain/cidr :)

I changed the line <param name="apply-inbound-acl" value="domains"/> in 
the internal.xml with a new value, and changed the list to match but I 
still get the same 'Rejected by acl "domains"' error.

And yes I reloaded the xml. :)

Kevin

Brian West wrote:
> First off edit the profile to use your own custom ACL... not the  
> domains one... secondly you'll have to use cidr= instead of domain=.
> 
> Domain= will search the user directory building an ACL list from all  
> users with the cidr= attribute.
> 
> /b
> 
> 
> On Aug 10, 2009, at 6:36 PM, Kevin Golding wrote:
> 
>> acl.conf.xml file, and have put the following in it:
>>
>> <configuration name="acl.conf" description="Network Lists">
>>   <network-lists>
>>     <list name="domains" value="allow">
>>         <node type="allow" domain="$${domain}"/>
>>         <node type="allow" domain="213.166.5.128/28"/>
>>         <node type="allow" domain="87.238.72.128/26"/>
>>     </list>
>>   </network-lists>
>> </configuration>




From solko at gcdf.pl  Tue Aug 11 00:49:13 2009
From: solko at gcdf.pl (Szymon Olko)
Date: Tue, 11 Aug 2009 09:49:13 +0200
Subject: [Freeswitch-users] answer command
In-Reply-To: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
Message-ID: <4A812279.6080404@gcdf.pl>

Maxim Tsvetov pisze:
> Hello,
> 
> I found in WIKI list of supported commands
> http://wiki.freeswitch.org/wiki/Mod_commands.
> 
> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
> 
> That's what Freeswitch replies for this command:
> 
> "answer: Command not found!"
> 
> ?ould you please help?

Answer is in dialplan commands.

http://wiki.freeswitch.org/wiki/Mod_dptools

Szymon



From lakindia89 at gmail.com  Tue Aug 11 01:59:59 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Tue, 11 Aug 2009 14:29:59 +0530
Subject: [Freeswitch-users] Error while creating object
In-Reply-To: <87f2f3b90908101629p71048eb1vbccd19a19d328fff@mail.gmail.com>
References: <7d79b3930908060347xe5be545yfeeafad761aba274@mail.gmail.com>
	<24895716.post@talk.nabble.com>
	<87f2f3b90908101629p71048eb1vbccd19a19d328fff@mail.gmail.com>
Message-ID: <7d79b3930908110159m1a1602a4tcccfb46e33873fbb@mail.gmail.com>

Thanks for your replay.
I've tried that. But it says following error message.

2009-08-11 14:23:09 [ERR] mod_perl.c:69 Perl_safe_eval() [require
'/usr/local/freeswitch/conf/test.pl';]
Undefined subroutine &freeswitch::DTMF called at
/usr/local/freeswitch/conf/test.pl line 6.Compilation failed in require at
(eval 2) line 1.

Please help me to solve this issue!!!

On Tue, Aug 11, 2009 at 4:59 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Aug 10, 2009 at 3:00 AM, lakshmanan <lakindia89 at gmail.com> wrote:
>
>>
>> Can any one please say what I did wrong here?
>>
>
> Maybe this instead?
> my $sess=&freeswitch::DTMF->new();
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From michal.bielicki at halo2.pl  Tue Aug 11 02:33:06 2009
From: michal.bielicki at halo2.pl (Michal Bielicki)
Date: Tue, 11 Aug 2009 11:33:06 +0200
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4a80632f.1508c00a.4d3c.090d@mx.google.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>
	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>
	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>
	<4a80632f.1508c00a.4d3c.090d@mx.google.com>
Message-ID: <A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>

I'll retst it later today and give you a link with instructions

Am 10.08.2009 um 20:14 schrieb vmorales:

> By "./compile" I was referring to "./configure"
>
> Vladimir
>
> -----Original Message-----
> From: vmorales [mailto:email.list.subscriber at gmail.com]
> Sent: Monday, August 10, 2009 11:49 AM
> To: 'freeswitch-users at lists.freeswitch.org'
> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> Thanks for the response(s):
>
> I ran the "./compile" script with a set PREFIX.  This took a few
> attempts with errors before it was able to complete error-free, as I
> had to install libtool.
>
> Since then, I have tried running 'make', 'gmake', and
> '/opt/gnu/bin/make', but each results with an error.  This is the
> error when running 'make' or 'gmake':
>
> <snip>
> make: Fatal error: Command failed for target `all-recursive'
> Current working directory /home/vmorales/freeswitch-1.0.4
> *** Error code 1
> make: Fatal error: Command failed for target `all'
> </snip>
>
>
> This is the error when running '/opt/gnu/bin/make':
>
> <snip>
> make[5]: *** [mod_amr.so] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_amr-all] Error 1
> make[2]: *** [all-recursive] Error 1
> Making all in build
> +-------- FreeSWITCH Build Complete -----------+
> + FreeSWITCH has been successfully built.      +
> + Install by running:                          +
> +                                              +
> +               /opt/gnu/bin/make install                   +
> +----------------------------------------------+
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> </snip>
>
>
> I re-untar'd before each compile/make attempt.  Let me know if this is
> something that I can resolve.
>
> Vladimir
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Jerris
> Sent: Saturday, August 08, 2009 12:37 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> This is not currently a supported platform, it only builds on 64 bit
> right now I think on solaris.
>
> Mike
>
> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>
>> Hello,
>>
>> Does anyone have, or know where to get, a pre-compiled copy of
>> FreeSwitch for Solaris 10/x86?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. info at halokwadrat.de
e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
Hauptgesch?ftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 0000153539

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From maxim.tsvetov at gmail.com  Tue Aug 11 05:34:19 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Tue, 11 Aug 2009 05:34:19 -0700 (PDT)
Subject: [Freeswitch-users] answer command
In-Reply-To: <4A812279.6080404@gcdf.pl>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl>
Message-ID: <24916765.post@talk.nabble.com>


Thank you.

Is it possible to use this command from command line or via event_socket
interface?



Szymon Olko wrote:
> 
> Maxim Tsvetov pisze:
>> Hello,
>> 
>> I found in WIKI list of supported commands
>> http://wiki.freeswitch.org/wiki/Mod_commands.
>> 
>> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
>> 
>> That's what Freeswitch replies for this command:
>> 
>> "answer: Command not found!"
>> 
>> ?ould you please help?
> 
> Answer is in dialplan commands.
> 
> http://wiki.freeswitch.org/wiki/Mod_dptools
> 
> Szymon
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/answer-command-tp24912812p24916765.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From samu60 at gmail.com  Tue Aug 11 00:49:52 2009
From: samu60 at gmail.com (samuel)
Date: Tue, 11 Aug 2009 09:49:52 +0200
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
Message-ID: <d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>

I've taken a look at phrase_es.xml and would like other spanish native
speakers to check the following (i preferred to write down here my
impressions so others can contribute instead of sending a patch):

1) Centavos

<prompt phrase="Centavos por minuto" filename="cents-per-minute.wav"/>
<prompt phrase="Centavos" filename="cents.wav"/>

probably in european spanish the most apropiate word for cents is "c?ntimo"
but, if i'm not mistaken, in south american spanish is more common to use
"centavos" so this is more a localization issue...as you wish.
Since I live in Spain I would use the following:
<prompt phrase="C?ntimos por minuto" filename="cents-per-minute.wav"/>
<prompt phrase="C?ntimos" filename="cents.wav"/>

2) Minnor stress corrections:
instead of Dolar, you should use the stressed word D?lar:
-<prompt phrase="Dolar" filename="dollar.wav"/>
+<prompt phrase="D?lar" filename="dollar.wav"/>

instead of numero, you should use the stressed word n?mero:
-<prompt phrase="por favor ingrese su numero de usuario, seguido por"
filename="vm-enter_id.wav"/>
+<prompt phrase="por favor ingrese su n?mero de usuario, seguido por"
filename="vm-enter_id.wav"/>

3) en instead of a
-<prompt phrase="La persona a la extensi?n" filename="vm-person.wav"/>
+<prompt phrase="La persona en la extensi?n" filename="vm-person.wav"/>

4) femenin genre of contrase?a changes the word seguido for seguida
-<prompt phrase="por favor ingrese su contrase?a, seguido por"
filename="vm-enter_pass.wav"/>
+<prompt phrase="por favor ingrese su contrase?a, seguida por"
filename="vm-enter_pass.wav"/>

5) I would propose the word "almohadilla" as the translation of pound so I
would change following sentences:
-<prompt phrase="Signo de n?mero" filename="35.wav"/>
+<prompt phrase="Almohadilla" filename="35.wav"/>

-<prompt phrase="Signo de n?mero" filename="pound.wav"/>
+<prompt phrase="Almohadilla" filename="pound.wav"/>

-<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del
signo de n?mero" filename="ivr-enter_ext.wav"/>
+<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido de
almohadilla" filename="ivr-enter_ext.wav"/>

-<prompt phrase="Seguido del signo de n?mero" filename="followed.wav"/>
+<prompt phrase="Seguido de almohadilla" filename="followed.wav"/>

6)changing prepositions:
-<prompt phrase="Un error ha ocurrido, por favor contacte al administrador"
filename="error.wav"/>
+<prompt phrase="Un error ha ocurrido, por favor contacte con el
administrador" filename="error.wav"/>

7)instead of using ingrese, it sounds more natural to use introduzca when
interacting with an IVR asking for inserting data. So:
-<prompt phrase="por favor ingrese su numero de usuario, seguido por"
filename="vm-enter_id.wav"/>
+<prompt phrase="por favor introduzca su n?mero de usuario, seguido por"
filename="vm-enter_id.wav"/>

-<prompt phrase="por favor ingrese su contrase?a, seguido por"
filename="vm-enter_pass.wav"/>
+<prompt phrase="por favor introduzca su contrase?a, seguida por"
filename="vm-enter_pass.wav"/>

-<prompt phrase="Por favor ingrese el pin de la conferencia."
filename="conf-pin.wav"/>
+<prompt phrase="Por favor introduzca el pin de la conferencia."
filename="conf-pin.wav"/>

-<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del
signo de n?mero" filename="ivr-enter_ext.wav"/>
+<prompt phrase="por favor introduzca el n?mero de una extensi?n seguido de
almohadilla" filename="ivr-enter_ext.wav"/>



Comments?

Samuel


2009/8/11 Michael Collins <msc at freeswitch.org>

> Be sure to hop on IRC and speak with jmesquita because he's been working on
> this also. It would be very good to have Spanish-speaking users review the
> Spanish phrase_es.xml file for correctness. Also, we need Spanish-speaking
> programmers to assist with the mod_say application for Spanish - there are
> things that you have to do in Spanish that you don't have to do in English.
>
> Thanks,
> MC
>
>
> On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>
>> On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
>> edition .
>>
>> Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for
>> FS playback.
>>
>> Here's a guide that has been put together for reference on what to record.
>>
>> http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml
>>
>> Regards,
>>
>>
>>
>>
>> On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:
>>
>>> I'd like to begin record spanish prompts for FS.
>>>
>>> Do you know any software/hardware to make it?
>>>
>>> Thank you
>>>
>>> BR
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
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From asannucci at gmail.com  Tue Aug 11 06:01:24 2009
From: asannucci at gmail.com (bakko)
Date: Tue, 11 Aug 2009 15:01:24 +0200
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice><f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com><87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
Message-ID: <E97B464EBED44E2AB7D71362840E1836@voztovoice>

Maybe the best solution is make two spanish translation. 

One for american y one for Espa?a.

I'm italian but i know this little differences beacouse i live in Colombia and i have studied in Spain.

Maybe i can help with italian prompts but i don't understand how.

BR

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From testeador01 at gmail.com  Tue Aug 11 06:09:53 2009
From: testeador01 at gmail.com (Milena)
Date: Tue, 11 Aug 2009 08:09:53 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <24916765.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
Message-ID: <c29b930b0908110609y3c224641x6c2cfa5a263b53e5@mail.gmail.com>

sendmsg
call-command: execute
execute-app-name: answer\n\n

the example is in this wiki page:
http://wiki.freeswitch.org/wiki/Event_socket_outbound
<http://wiki.freeswitch.org/wiki/Event_socket_outbound>
2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>

>
> Thank you.
>
> Is it possible to use this command from command line or via event_socket
> interface?
>
>
>
> Szymon Olko wrote:
> >
> > Maxim Tsvetov pisze:
> >> Hello,
> >>
> >> I found in WIKI list of supported commands
> >> http://wiki.freeswitch.org/wiki/Mod_commands.
> >>
> >> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
> >>
> >> That's what Freeswitch replies for this command:
> >>
> >> "answer: Command not found!"
> >>
> >> ?ould you please help?
> >
> > Answer is in dialplan commands.
> >
> > http://wiki.freeswitch.org/wiki/Mod_dptools
> >
> > Szymon
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/answer-command-tp24912812p24916765.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
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From darren at aleph-com.net  Tue Aug 11 06:18:25 2009
From: darren at aleph-com.net (Darren Wiebe)
Date: Tue, 11 Aug 2009 07:18:25 -0600
Subject: [Freeswitch-users] VoiceMail transcription
In-Reply-To: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
References: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
Message-ID: <4A816FA1.3050701@aleph-com.net>

We've been using mycaption for a number of months and have been very 
happy with the results.  I'd suggest giving them a try.

Darren Wiebe
darren at aleph-com.net

Pete Mueller wrote:
> I apologize, I should have been more clear.  We will be using humans 
> to scan the translated results.  But we are looking for a system to 
> perform the "first pass" on the audio to hopefully help the human type 
> less.
>
> Although the question has been raised if it's faster to have a human 
> just transcribe the whole thing, or fix up what the computer spit 
> out.  If you have any insights on this, that would be great.
>
> -pete
>
>     -------- Original Message --------
>     Subject: Re: [Freeswitch-users] VoiceMail transcription
>     From: David Knell <dave at 3c.co.uk>
>     Date: Mon, August 10, 2009 11:51 am
>     To: freeswitch-users at lists.freeswitch.org
>
>     Good evening Pete,
>
>     The only way to do this is, I'm afraid, to use a human. We use
>     Amazon's
>     Mechanical Turk to good effect.
>
>     Cheers --
>
>     Dave
>
>     > Good morning all,
>     >
>     > I realize this is slightly off the FS topic, but I am wondering if
>     > anyone out there has experience with software packages designed for
>     > the transcription of voicemails to text. I've used pocketsphinx with
>     > FS to handle IVR menus, but now have the task of figuring out how to
>     > convert recorded phone conversations (voicemails mostly) to text.
>     >
>     > This does not have to be a real-time process, I can store the audio
>     > files and process them over time. This would need to be a software
>     > (preferable open source) solution. ASPs like VoiceCloud would not
>     > work for this application.
>     >
>     > Thanks for any help
>     > -pete
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     -- 
>     David Knell, Director, 3C Limited
>     T: +44 20 3298 2000
>     E: dave at 3c.co.uk
>     W: http://www.3c.co.uk
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   




From wangdq.no1 at gmail.com  Tue Aug 11 06:19:13 2009
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Tue, 11 Aug 2009 21:19:13 +0800
Subject: [Freeswitch-users] answer command
In-Reply-To: <24916765.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
Message-ID: <ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>

why not use:
    session:answer()


2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>

>
> Thank you.
>
> Is it possible to use this command from command line or via event_socket
> interface?
>
>
>
> Szymon Olko wrote:
> >
> > Maxim Tsvetov pisze:
> >> Hello,
> >>
> >> I found in WIKI list of supported commands
> >> http://wiki.freeswitch.org/wiki/Mod_commands.
> >>
> >> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
> >>
> >> That's what Freeswitch replies for this command:
> >>
> >> "answer: Command not found!"
> >>
> >> ?ould you please help?
> >
> > Answer is in dialplan commands.
> >
> > http://wiki.freeswitch.org/wiki/Mod_dptools
> >
> > Szymon
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/answer-command-tp24912812p24916765.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From maxim.tsvetov at gmail.com  Tue Aug 11 06:30:40 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Tue, 11 Aug 2009 06:30:40 -0700 (PDT)
Subject: [Freeswitch-users] answer command
In-Reply-To: <c29b930b0908110609y3c224641x6c2cfa5a263b53e5@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<c29b930b0908110609y3c224641x6c2cfa5a263b53e5@mail.gmail.com>
Message-ID: <24917680.post@talk.nabble.com>


I've tried all this command from FS console 
and all of them return "Unknown command".



Milena-6 wrote:
> 
> sendmsg
> call-command: execute
> execute-app-name: answer\n\n
> 
> the example is in this wiki page:
> http://wiki.freeswitch.org/wiki/Event_socket_outbound
> <http://wiki.freeswitch.org/wiki/Event_socket_outbound>
> 2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>
> 
>>
>> Thank you.
>>
>> Is it possible to use this command from command line or via event_socket
>> interface?
>>
>>
>>
>> Szymon Olko wrote:
>> >
>> > Maxim Tsvetov pisze:
>> >> Hello,
>> >>
>> >> I found in WIKI list of supported commands
>> >> http://wiki.freeswitch.org/wiki/Mod_commands.
>> >>
>> >> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
>> >>
>> >> That's what Freeswitch replies for this command:
>> >>
>> >> "answer: Command not found!"
>> >>
>> >> ?ould you please help?
>> >
>> > Answer is in dialplan commands.
>> >
>> > http://wiki.freeswitch.org/wiki/Mod_dptools
>> >
>> > Szymon
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> --
>> View this message in context:
>> http://www.nabble.com/answer-command-tp24912812p24916765.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/answer-command-tp24912812p24917680.html
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From maxim.tsvetov at gmail.com  Tue Aug 11 06:31:32 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Tue, 11 Aug 2009 06:31:32 -0700 (PDT)
Subject: [Freeswitch-users] answer command
In-Reply-To: <ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
Message-ID: <24917706.post@talk.nabble.com>


I've tried all this command from FS console 
and all of them return "Unknown command"


daqiang wang wrote:
> 
> why not use:
>     session:answer()
> 
> 
> 2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>
> 
>>
>> Thank you.
>>
>> Is it possible to use this command from command line or via event_socket
>> interface?
>>
>>
>>
>> Szymon Olko wrote:
>> >
>> > Maxim Tsvetov pisze:
>> >> Hello,
>> >>
>> >> I found in WIKI list of supported commands
>> >> http://wiki.freeswitch.org/wiki/Mod_commands.
>> >>
>> >> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
>> >>
>> >> That's what Freeswitch replies for this command:
>> >>
>> >> "answer: Command not found!"
>> >>
>> >> ?ould you please help?
>> >
>> > Answer is in dialplan commands.
>> >
>> > http://wiki.freeswitch.org/wiki/Mod_dptools
>> >
>> > Szymon
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> --
>> View this message in context:
>> http://www.nabble.com/answer-command-tp24912812p24916765.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/answer-command-tp24912812p24917706.html
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From dave at 3c.co.uk  Tue Aug 11 06:40:14 2009
From: dave at 3c.co.uk (David Knell)
Date: Tue, 11 Aug 2009 16:40:14 +0300
Subject: [Freeswitch-users] VoiceMail transcription
In-Reply-To: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
References: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
Message-ID: <1249998014.20224.82.camel@dk-d820>

Hi Pete,

I'm afraid that the answer's still the same: use a human.  Here's an
article describing the state of the art:
http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/
- the links to previous stories at the bottom provide good background.

--Dave

> I apologize, I should have been more clear.  We will be using humans
> to scan the translated results.  But we are looking for a system to
> perform the "first pass" on the audio to hopefully help the human type
> less.
> 
> Although the question has been raised if it's faster to have a human
> just transcribe the whole thing, or fix up what the computer spit out.
> If you have any insights on this, that would be great.
> 
> -pete
> 
>         -------- Original Message --------
>         Subject: Re: [Freeswitch-users] VoiceMail transcription
>         From: David Knell <dave at 3c.co.uk>
>         Date: Mon, August 10, 2009 11:51 am
>         To: freeswitch-users at lists.freeswitch.org
>         
>         Good evening Pete,
>         
>         The only way to do this is, I'm afraid, to use a human. We use
>         Amazon's
>         Mechanical Turk to good effect.
>         
>         Cheers --
>         
>         Dave
>         
>         > Good morning all,
>         > 
>         > I realize this is slightly off the FS topic, but I am
>         wondering if
>         > anyone out there has experience with software packages
>         designed for
>         > the transcription of voicemails to text. I've used
>         pocketsphinx with
>         > FS to handle IVR menus, but now have the task of figuring
>         out how to
>         > convert recorded phone conversations (voicemails mostly) to
>         text.
>         > 
>         > This does not have to be a real-time process, I can store
>         the audio
>         > files and process them over time. This would need to be a
>         software
>         > (preferable open source) solution. ASPs like VoiceCloud
>         would not
>         > work for this application.
>         > 
>         > Thanks for any help
>         > -pete
>         > _______________________________________________
>         > FreeSWITCH-users mailing list
>         > FreeSWITCH-users at lists.freeswitch.org
>         >
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         >
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         > http://www.freeswitch.org
>         -- 
>         David Knell, Director, 3C Limited
>         T: +44 20 3298 2000
>         E: dave at 3c.co.uk
>         W: http://www.3c.co.uk
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk




From dujinfang at gmail.com  Tue Aug 11 06:40:41 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 11 Aug 2009 21:40:41 +0800
Subject: [Freeswitch-users] answer command
In-Reply-To: <24917706.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
Message-ID: <2AFE75D2-99F3-4A12-BADC-F6753C76166D@gmail.com>

answer only works on outbound event socket. why you don't answer in a  
dialplan? what's scenario you use this?

On Aug 11, 2009, at 9:31 PM, Maxim Tsvetov wrote:
>
> I've tried all this command from FS console
> and all of them return "Unknown command"
>
>
> daqiang wang wrote:
>>
>> why not use:
>>    session:answer()
>>
>>
>> 2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>
>>
>>>
>>> Thank you.
>>>
>>> Is it possible to use this command from command line or via  
>>> event_socket
>>> interface?
>>>
>>>
>>>
>>> Szymon Olko wrote:
>>>>
>>>> Maxim Tsvetov pisze:
>>>>> Hello,
>>>>>
>>>>> I found in WIKI list of supported commands
>>>>> http://wiki.freeswitch.org/wiki/Mod_commands.
>>>>>
>>>>> I need command "answer" but it doesn't exist (Freeswitch 1.04,  
>>>>> Win32)
>>>>>
>>>>> That's what Freeswitch replies for this command:
>>>>>
>>>>> "answer: Command not found!"
>>>>>
>>>>> ?ould you please help?
>>>>
>>>> Answer is in dialplan commands.
>>>>
>>>> http://wiki.freeswitch.org/wiki/Mod_dptools
>>>>
>>>> Szymon
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> --
>>> View this message in context:
>>> http://www.nabble.com/answer-command-tp24912812p24916765.html
>>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://www.nabble.com/answer-command-tp24912812p24917706.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From testeador01 at gmail.com  Tue Aug 11 06:46:01 2009
From: testeador01 at gmail.com (Milena)
Date: Tue, 11 Aug 2009 08:46:01 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <24917706.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
Message-ID: <c29b930b0908110646w470a0ec0mc65cd34e5f04825b@mail.gmail.com>

you can send the "answer" on the console too;this is *how to* use
*sendmsg*from the console:
http://wiki.freeswitch.org/wiki/Event_Socket#sendmsg

<http://wiki.freeswitch.org/wiki/Event_Socket#sendmsg>btw, the previous
command i gave you works just fine on event socket, it will work for you if
you replicate the Netcat example.

2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>

>
> I've tried all this command from FS console
> and all of them return "Unknown command"
>
>
> daqiang wang wrote:
> >
> > why not use:
> >     session:answer()
> >
> >
> > 2009/8/11 Maxim Tsvetov <maxim.tsvetov at gmail.com>
> >
> >>
> >> Thank you.
> >>
> >> Is it possible to use this command from command line or via event_socket
> >> interface?
> >>
> >>
> >>
> >> Szymon Olko wrote:
> >> >
> >> > Maxim Tsvetov pisze:
> >> >> Hello,
> >> >>
> >> >> I found in WIKI list of supported commands
> >> >> http://wiki.freeswitch.org/wiki/Mod_commands.
> >> >>
> >> >> I need command "answer" but it doesn't exist (Freeswitch 1.04, Win32)
> >> >>
> >> >> That's what Freeswitch replies for this command:
> >> >>
> >> >> "answer: Command not found!"
> >> >>
> >> >> ?ould you please help?
> >> >
> >> > Answer is in dialplan commands.
> >> >
> >> > http://wiki.freeswitch.org/wiki/Mod_dptools
> >> >
> >> > Szymon
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> --
> >> View this message in context:
> >> http://www.nabble.com/answer-command-tp24912812p24916765.html
> >> Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> --
> View this message in context:
> http://www.nabble.com/answer-command-tp24912812p24917706.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Aug 11 06:49:53 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 08:49:53 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <24917706.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
Message-ID: <59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>

Its not an api command the docs are wrong and should be fixed.

/b

On Aug 11, 2009, at 8:31 AM, Maxim Tsvetov wrote:

>
> I've tried all this command from FS console
> and all of them return "Unknown command"
>




From pjintheusa at gmail.com  Tue Aug 11 06:59:54 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 11 Aug 2009 09:59:54 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
Message-ID: <367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>

It occurred to me that I better check that this still occurs using lua
and that this is not a mod_manged issue.

The lua script:

session:answer();
session:execute("set", "bypass_media_after_bridge=true");
session:execute("bridge", "loopback/6095553828/default");

does produce the same issue. The bridge fails.

I am wondering whether this is something I should just put straight into jira.

On Mon, Aug 10, 2009 at 4:35 PM, Phillip Jones<pjintheusa at gmail.com> wrote:
> Hi there,
>
> I am attempting to do a simple bridge. Leg A is an inbound DID. Leg B
> is terminated through a SIP carrier.
>
> I am unable to use loopback AND bypass_media_after_bridge=true. The
> bridge fails.
>
> Here is my three line application:
>
> Session.Answer();
>
> Session.Execute("set", "bypass_media_after_bridge=true");
>
> Session.Execute("bridge", "loopback/6095553828/default");
>
> FreeSWITCH log: http://pastebin.freeswitch.org/9949
>
> I believe the bridge fails as the SIP-REINVITE is issued and fails.
>
> Using loopback and bypass_media_after_bridge=false works fine. As does
> using bypass_media_after_bridge=true and sending call directly through
> the gateway to which the loopback points.
>
> e.g. ?http://pastebin.freeswitch.org/9948
>
> Has anyone else encountered this? Am I missing something?
>
> Any help would be much appreciated!
>
> Thanks
>
>
> Phillip Jones
>



From testeador01 at gmail.com  Tue Aug 11 07:05:56 2009
From: testeador01 at gmail.com (Milena)
Date: Tue, 11 Aug 2009 09:05:56 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
Message-ID: <c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>

Hello Brian,

I wanna fix the wiki, but to make sure i got it right, does it only work on
outbound event socket? or is there any other scenario where it would work.

Thank you.


2009/8/11 Brian West <brian at freeswitch.org>

> Its not an api command the docs are wrong and should be fixed.
>
> /b
>
> On Aug 11, 2009, at 8:31 AM, Maxim Tsvetov wrote:
>
> >
> > I've tried all this command from FS console
> > and all of them return "Unknown command"
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Aug 11 07:07:27 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 09:07:27 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
Message-ID: <6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>

loopback requires media you can't be using it in this manner.  There  
is NO reason to use loopback in the first place.  Can you show me what  
you're doing in the dialplan that requires you to use loopback?

We gave you the rope... now you just have to stop from hanging  
yourself!  :P

/b

On Aug 11, 2009, at 8:59 AM, Phillip Jones wrote:

> It occurred to me that I better check that this still occurs using lua
> and that this is not a mod_manged issue.
>
> The lua script:
>
> session:answer();
> session:execute("set", "bypass_media_after_bridge=true");
> session:execute("bridge", "loopback/6095553828/default");
>
> does produce the same issue. The bridge fails.
>
> I am wondering whether this is something I should just put straight  
> into jira.




From email.list.subscriber at gmail.com  Tue Aug 11 07:12:57 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Tue, 11 Aug 2009 10:12:57 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>
	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>
Message-ID: <4a817c27.14b48c0a.5c23.6dfe@mx.google.com>

Would truly appreciate that.

Thanks Michal!


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michal Bielicki
Sent: Tuesday, August 11, 2009 5:33 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

I'll retst it later today and give you a link with instructions

Am 10.08.2009 um 20:14 schrieb vmorales:

> By "./compile" I was referring to "./configure"
>
> Vladimir
>
> -----Original Message-----
> From: vmorales [mailto:email.list.subscriber at gmail.com]
> Sent: Monday, August 10, 2009 11:49 AM
> To: 'freeswitch-users at lists.freeswitch.org'
> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> Thanks for the response(s):
>
> I ran the "./compile" script with a set PREFIX.  This took a few
> attempts with errors before it was able to complete error-free, as I
> had to install libtool.
>
> Since then, I have tried running 'make', 'gmake', and
> '/opt/gnu/bin/make', but each results with an error.  This is the
> error when running 'make' or 'gmake':
>
> <snip>
> make: Fatal error: Command failed for target `all-recursive'
> Current working directory /home/vmorales/freeswitch-1.0.4
> *** Error code 1
> make: Fatal error: Command failed for target `all'
> </snip>
>
>
> This is the error when running '/opt/gnu/bin/make':
>
> <snip>
> make[5]: *** [mod_amr.so] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_amr-all] Error 1
> make[2]: *** [all-recursive] Error 1
> Making all in build
> +-------- FreeSWITCH Build Complete -----------+
> + FreeSWITCH has been successfully built.      +
> + Install by running:                          +
> +                                              +
> +               /opt/gnu/bin/make install                   +
> +----------------------------------------------+
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> </snip>
>
>
> I re-untar'd before each compile/make attempt.  Let me know if this
is
> something that I can resolve.
>
> Vladimir
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Jerris
> Sent: Saturday, August 08, 2009 12:37 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> This is not currently a supported platform, it only builds on 64 bit
> right now I think on solaris.
>
> Mike
>
> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>
>> Hello,
>>
>> Does anyone have, or know where to get, a pre-compiled copy of
>> FreeSwitch for Solaris 10/x86?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
> http://www.freeswitch.org

Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. info at halokwadrat.de
e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
Hauptgesch?ftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 0000153539





From kirk.bateman at gmail.com  Tue Aug 11 07:17:47 2009
From: kirk.bateman at gmail.com (Kirk Bateman)
Date: Tue, 11 Aug 2009 15:17:47 +0100
Subject: [Freeswitch-users] VoiceMail transcription
In-Reply-To: <1249998014.20224.82.camel@dk-d820>
References: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
	<1249998014.20224.82.camel@dk-d820>
Message-ID: <2bee4fc40908110717p43649678ue2b94ecb571bd09@mail.gmail.com>

I'm still interested in getting pocketsphinx to attempt speech recognition
on an audio file.

To be honest, most of the problem is that at 8Khz (mobile phone call rate),
speech detection is NOT very accurate, at 16Khz it IS significantly better.

I'm planning to have a play with the speechtools module and mod_pocketsphinx
etc to try and get an audio file parsed, spare time permitting.

Will let the list know if I get anywhere.

Regards

Kirk Bateman


2009/8/11 David Knell <dave at 3c.co.uk>

> Hi Pete,
>
> I'm afraid that the answer's still the same: use a human.  Here's an
> article describing the state of the art:
> http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/
> - the links to previous stories at the bottom provide good background.
>
> --Dave
>
> > I apologize, I should have been more clear.  We will be using humans
> > to scan the translated results.  But we are looking for a system to
> > perform the "first pass" on the audio to hopefully help the human type
> > less.
> >
> > Although the question has been raised if it's faster to have a human
> > just transcribe the whole thing, or fix up what the computer spit out.
> > If you have any insights on this, that would be great.
> >
> > -pete
> >
> >         -------- Original Message --------
> >         Subject: Re: [Freeswitch-users] VoiceMail transcription
> >         From: David Knell <dave at 3c.co.uk>
> >         Date: Mon, August 10, 2009 11:51 am
> >         To: freeswitch-users at lists.freeswitch.org
> >
> >         Good evening Pete,
> >
> >         The only way to do this is, I'm afraid, to use a human. We use
> >         Amazon's
> >         Mechanical Turk to good effect.
> >
> >         Cheers --
> >
> >         Dave
> >
> >         > Good morning all,
> >         >
> >         > I realize this is slightly off the FS topic, but I am
> >         wondering if
> >         > anyone out there has experience with software packages
> >         designed for
> >         > the transcription of voicemails to text. I've used
> >         pocketsphinx with
> >         > FS to handle IVR menus, but now have the task of figuring
> >         out how to
> >         > convert recorded phone conversations (voicemails mostly) to
> >         text.
> >         >
> >         > This does not have to be a real-time process, I can store
> >         the audio
> >         > files and process them over time. This would need to be a
> >         software
> >         > (preferable open source) solution. ASPs like VoiceCloud
> >         would not
> >         > work for this application.
> >         >
> >         > Thanks for any help
> >         > -pete
> >         > _______________________________________________
> >         > FreeSWITCH-users mailing list
> >         > FreeSWITCH-users at lists.freeswitch.org
> >         >
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         >
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         > http://www.freeswitch.org
> >         --
> >         David Knell, Director, 3C Limited
> >         T: +44 20 3298 2000
> >         E: dave at 3c.co.uk
> >         W: http://www.3c.co.uk
> >
> >
> >         _______________________________________________
> >         FreeSWITCH-users mailing list
> >         FreeSWITCH-users at lists.freeswitch.org
> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >         UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >         http://www.freeswitch.org
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From nicolas at medularis.com  Tue Aug 11 07:27:27 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Tue, 11 Aug 2009 10:27:27 -0400
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
Message-ID: <1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>

I think bakko is right, I have a few observations about the changes proposed
by samuel, and I think it makes a lot of sense to have different
localizations.


On Tue, Aug 11, 2009 at 3:49 AM, samuel <samu60 at gmail.com> wrote:

> I've taken a look at phrase_es.xml and would like other spanish native
> speakers to check the following (i preferred to write down here my
> impressions so others can contribute instead of sending a patch):
>
> 1) Centavos
>
> <prompt phrase="Centavos por minuto" filename="cents-per-minute.wav"/>
> <prompt phrase="Centavos" filename="cents.wav"/>


> probably in european spanish the most apropiate word for cents is "c?ntimo"
> but, if i'm not mistaken, in south american spanish is more common to use
> "centavos" so this is more a localization issue...as you wish.
> Since I live in Spain I would use the following:
> <prompt phrase="C?ntimos por minuto" filename="cents-per-minute.wav"/>
> <prompt phrase="C?ntimos" filename="cents.wav"/>
>

You are right, in South America we use Centavos.


>
> 2) Minnor stress corrections:
> instead of Dolar, you should use the stressed word D?lar:
> -<prompt phrase="Dolar" filename="dollar.wav"/>
> +<prompt phrase="D?lar" filename="dollar.wav"/>
>
> instead of numero, you should use the stressed word n?mero:
> -<prompt phrase="por favor ingrese su numero de usuario, seguido por"
> filename="vm-enter_id.wav"/>
> +<prompt phrase="por favor ingrese su n?mero de usuario, seguido por"
> filename="vm-enter_id.wav"/>
>
> 3) en instead of a
> -<prompt phrase="La persona a la extensi?n" filename="vm-person.wav"/>
> +<prompt phrase="La persona en la extensi?n" filename="vm-person.wav"/>
>
> 4) femenin genre of contrase?a changes the word seguido for seguida
> -<prompt phrase="por favor ingrese su contrase?a, seguido por"
> filename="vm-enter_pass.wav"/>
> +<prompt phrase="por favor ingrese su contrase?a, seguida por"
> filename="vm-enter_pass.wav"/>
>
> 5) I would propose the word "almohadilla" as the translation of pound so I
> would change following sentences:
> -<prompt phrase="Signo de n?mero" filename="35.wav"/>
> +<prompt phrase="Almohadilla" filename="35.wav"/>
>
> -<prompt phrase="Signo de n?mero" filename="pound.wav"/>
> +<prompt phrase="Almohadilla" filename="pound.wav"/>
>

I don't know about the rest of the countries in South America, but at least
in Chile it would me much clearer "Signo de n?mero" instead of Almohadilla
(first time I've heard that name for the pound sign).


>
> -<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del
> signo de n?mero" filename="ivr-enter_ext.wav"/>
> +<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido de
> almohadilla" filename="ivr-enter_ext.wav"/>
>
> -<prompt phrase="Seguido del signo de n?mero" filename="followed.wav"/>
> +<prompt phrase="Seguido de almohadilla" filename="followed.wav"/>
>
> 6)changing prepositions:
> -<prompt phrase="Un error ha ocurrido, por favor contacte al administrador"
> filename="error.wav"/>
> +<prompt phrase="Un error ha ocurrido, por favor contacte con el
> administrador" filename="error.wav"/>
>

Here it would be:
<prompt phrase="Ha ocurrido un error, por favor contacte al administrador"
filename="error.wav"/>


> 7)instead of using ingrese, it sounds more natural to use introduzca when
> interacting with an IVR asking for inserting data. So:
> -<prompt phrase="por favor ingrese su numero de usuario, seguido por"
> filename="vm-enter_id.wav"/>
> +<prompt phrase="por favor introduzca su n?mero de usuario, seguido por"
> filename="vm-enter_id.wav"/>
>
> -<prompt phrase="por favor ingrese su contrase?a, seguido por"
> filename="vm-enter_pass.wav"/>
> +<prompt phrase="por favor introduzca su contrase?a, seguida por"
> filename="vm-enter_pass.wav"/>
>
> -<prompt phrase="Por favor ingrese el pin de la conferencia."
> filename="conf-pin.wav"/>
> +<prompt phrase="Por favor introduzca el pin de la conferencia."
> filename="conf-pin.wav"/>
>
> -<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del
> signo de n?mero" filename="ivr-enter_ext.wav"/>
> +<prompt phrase="por favor introduzca el n?mero de una extensi?n seguido de
> almohadilla" filename="ivr-enter_ext.wav"/>
>
>
>
> Comments?
>

I am not very convinced by the use of stresses in the words, I know it's
"more correct", but sometimes it's a pain, especially if you are working on
a remote server and don't have the right keyboard setting or something like
that. I've also had a few headaches with file encoding, so depending on that
and changes to the encoding you can screw your config files really easily.
For simplification I'd leave all stresses out.


>
> Samuel
>
>
> 2009/8/11 Michael Collins <msc at freeswitch.org>
>
> Be sure to hop on IRC and speak with jmesquita because he's been working on
>> this also. It would be very good to have Spanish-speaking users review the
>> Spanish phrase_es.xml file for correctness. Also, we need Spanish-speaking
>> programmers to assist with the mod_say application for Spanish - there are
>> things that you have to do in Spanish that you don't have to do in English.
>>
>> Thanks,
>> MC
>>
>>
>> On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>>
>>> On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
>>> edition .
>>>
>>> Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for
>>> FS playback.
>>>
>>> Here's a guide that has been put together for reference on what to
>>> record.
>>>
>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml
>>>
>>> Regards,
>>>
>>>
>>>
>>>
>>> On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:
>>>
>>>> I'd like to begin record spanish prompts for FS.
>>>>
>>>> Do you know any software/hardware to make it?
>>>>
>>>> Thank you
>>>>
>>>> BR
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Tue Aug 11 07:30:59 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 09:30:59 -0500
Subject: [Freeswitch-users] Fwd: ALARM signal in esl libraries
In-Reply-To: <1452e2980908102129g51255d5bt430a3e73bd6f2e99@mail.gmail.com>
References: <1452e2980908092338j15c4e33cn2cff799bb64464d0@mail.gmail.com>
	<1452e2980908102129g51255d5bt430a3e73bd6f2e99@mail.gmail.com>
Message-ID: <ADEDE29B-B1C2-4563-A20B-C16225ED7881@freeswitch.org>

Replying again asking for help on the same thing over and over won't  
get you far on this list.  It'll usually get you ignored.  Can you  
explain what you're trying to do better?  Why are you needing an  
alarm? what purpose?

/b

On Aug 10, 2009, at 11:29 PM, velusamy velu wrote:

> Dear Greats,
>     Could you please help me to solve this problem?
>
> ---------- Forwarded message ----------
> From: velusamy velu <velu.technical at gmail.com>
> Date: Mon, Aug 10, 2009 at 12:08 PM
> Subject: ALARM signal in esl libraries
> To: freeswitch-users at lists.freeswitch.org
>
>
> Dear All,
>        I have registered  ALARM signal  in my perl program to handle  
> the DTMF digit timeout.
> When ALARM signal generated  the  connection with  ESL  is   
> automatically  closed.
>
> I have checked the connection with "connected: function, it returns 0.
>
> Why the connection was closed?
> Is there any idea to alive the connection after ALARM signal  
> generation??
>
> Please help me.........
>
> Thanks,
> Velusamy
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From pjintheusa at gmail.com  Tue Aug 11 07:46:55 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 11 Aug 2009 10:46:55 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
Message-ID: <367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>

Thanks for the reply Brian.

I went through this fairly carefully with Mike Jerris - please see
http://www.nabble.com/Calling-Multiple-Destinations-with-Failover-td24877157.html

I am very open to any other mechanism that allows the calling multiple
destinations with carrier failover support. I would have thought this
is a fairly common requirement and is a fundamental requirement in my
application.

As always - your help is appreciated.

On Tue, Aug 11, 2009 at 10:07 AM, Brian West<brian at freeswitch.org> wrote:
> loopback requires media you can't be using it in this manner. ?There
> is NO reason to use loopback in the first place. ?Can you show me what
> you're doing in the dialplan that requires you to use loopback?
>
> We gave you the rope... now you just have to stop from hanging
> yourself! ?:P
>
> /b
>
> On Aug 11, 2009, at 8:59 AM, Phillip Jones wrote:
>
>> It occurred to me that I better check that this still occurs using lua
>> and that this is not a mod_manged issue.
>>
>> The lua script:
>>
>> session:answer();
>> session:execute("set", "bypass_media_after_bridge=true");
>> session:execute("bridge", "loopback/6095553828/default");
>>
>> does produce the same issue. The bridge fails.
>>
>> I am wondering whether this is something I should just put straight
>> into jira.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From samu60 at gmail.com  Tue Aug 11 07:48:36 2009
From: samu60 at gmail.com (samuel)
Date: Tue, 11 Aug 2009 16:48:36 +0200
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
Message-ID: <d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>

I'm also for different spanish localization if it's not too complicated. It
was also for me the first time I see "signo de n?mero" for pound ;)


Samuel.

2009/8/11 Nicolas Brenner <nicolas at medularis.com>

> I think bakko is right, I have a few observations about the changes
> proposed by samuel, and I think it makes a lot of sense to have different
> localizations.
>
>
> On Tue, Aug 11, 2009 at 3:49 AM, samuel <samu60 at gmail.com> wrote:
>
>> I've taken a look at phrase_es.xml and would like other spanish native
>> speakers to check the following (i preferred to write down here my
>> impressions so others can contribute instead of sending a patch):
>>
>> 1) Centavos
>>
>> <prompt phrase="Centavos por minuto" filename="cents-per-minute.wav"/>
>> <prompt phrase="Centavos" filename="cents.wav"/>
>
>
>> probably in european spanish the most apropiate word for cents is
>> "c?ntimo" but, if i'm not mistaken, in south american spanish is more common
>> to use "centavos" so this is more a localization issue...as you wish.
>> Since I live in Spain I would use the following:
>> <prompt phrase="C?ntimos por minuto" filename="cents-per-minute.wav"/>
>> <prompt phrase="C?ntimos" filename="cents.wav"/>
>>
>
> You are right, in South America we use Centavos.
>
>
>>
>> 2) Minnor stress corrections:
>> instead of Dolar, you should use the stressed word D?lar:
>> -<prompt phrase="Dolar" filename="dollar.wav"/>
>> +<prompt phrase="D?lar" filename="dollar.wav"/>
>>
>> instead of numero, you should use the stressed word n?mero:
>> -<prompt phrase="por favor ingrese su numero de usuario, seguido por"
>> filename="vm-enter_id.wav"/>
>> +<prompt phrase="por favor ingrese su n?mero de usuario, seguido por"
>> filename="vm-enter_id.wav"/>
>>
>> 3) en instead of a
>> -<prompt phrase="La persona a la extensi?n" filename="vm-person.wav"/>
>> +<prompt phrase="La persona en la extensi?n" filename="vm-person.wav"/>
>>
>> 4) femenin genre of contrase?a changes the word seguido for seguida
>> -<prompt phrase="por favor ingrese su contrase?a, seguido por"
>> filename="vm-enter_pass.wav"/>
>> +<prompt phrase="por favor ingrese su contrase?a, seguida por"
>> filename="vm-enter_pass.wav"/>
>>
>> 5) I would propose the word "almohadilla" as the translation of pound so I
>> would change following sentences:
>> -<prompt phrase="Signo de n?mero" filename="35.wav"/>
>> +<prompt phrase="Almohadilla" filename="35.wav"/>
>>
>> -<prompt phrase="Signo de n?mero" filename="pound.wav"/>
>> +<prompt phrase="Almohadilla" filename="pound.wav"/>
>>
>
> I don't know about the rest of the countries in South America, but at least
> in Chile it would me much clearer "Signo de n?mero" instead of Almohadilla
> (first time I've heard that name for the pound sign).
>
>
>>
>> -<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del
>> signo de n?mero" filename="ivr-enter_ext.wav"/>
>> +<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido de
>> almohadilla" filename="ivr-enter_ext.wav"/>
>>
>> -<prompt phrase="Seguido del signo de n?mero" filename="followed.wav"/>
>> +<prompt phrase="Seguido de almohadilla" filename="followed.wav"/>
>>
>> 6)changing prepositions:
>> -<prompt phrase="Un error ha ocurrido, por favor contacte al
>> administrador" filename="error.wav"/>
>> +<prompt phrase="Un error ha ocurrido, por favor contacte con el
>> administrador" filename="error.wav"/>
>>
>
> Here it would be:
> <prompt phrase="Ha ocurrido un error, por favor contacte al administrador"
> filename="error.wav"/>
>
>
>> 7)instead of using ingrese, it sounds more natural to use introduzca when
>> interacting with an IVR asking for inserting data. So:
>> -<prompt phrase="por favor ingrese su numero de usuario, seguido por"
>> filename="vm-enter_id.wav"/>
>> +<prompt phrase="por favor introduzca su n?mero de usuario, seguido por"
>> filename="vm-enter_id.wav"/>
>>
>> -<prompt phrase="por favor ingrese su contrase?a, seguido por"
>> filename="vm-enter_pass.wav"/>
>> +<prompt phrase="por favor introduzca su contrase?a, seguida por"
>> filename="vm-enter_pass.wav"/>
>>
>> -<prompt phrase="Por favor ingrese el pin de la conferencia."
>> filename="conf-pin.wav"/>
>> +<prompt phrase="Por favor introduzca el pin de la conferencia."
>> filename="conf-pin.wav"/>
>>
>> -<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del
>> signo de n?mero" filename="ivr-enter_ext.wav"/>
>> +<prompt phrase="por favor introduzca el n?mero de una extensi?n seguido
>> de almohadilla" filename="ivr-enter_ext.wav"/>
>>
>>
>>
>> Comments?
>>
>
> I am not very convinced by the use of stresses in the words, I know it's
> "more correct", but sometimes it's a pain, especially if you are working on
> a remote server and don't have the right keyboard setting or something like
> that. I've also had a few headaches with file encoding, so depending on that
> and changes to the encoding you can screw your config files really easily.
> For simplification I'd leave all stresses out.
>
>
>>
>> Samuel
>>
>>
>> 2009/8/11 Michael Collins <msc at freeswitch.org>
>>
>> Be sure to hop on IRC and speak with jmesquita because he's been working
>>> on this also. It would be very good to have Spanish-speaking users review
>>> the Spanish phrase_es.xml file for correctness. Also, we need
>>> Spanish-speaking programmers to assist with the mod_say application for
>>> Spanish - there are things that you have to do in Spanish that you don't
>>> have to do in English.
>>>
>>> Thanks,
>>> MC
>>>
>>>
>>> On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com> wrote:
>>>
>>>> On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for
>>>> edition .
>>>>
>>>> Record and edit at 48000Hz and then I use sox to downsample to 8000Hz
>>>> for FS playback.
>>>>
>>>> Here's a guide that has been put together for reference on what to
>>>> record.
>>>>
>>>>
>>>> http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml
>>>>
>>>> Regards,
>>>>
>>>>
>>>>
>>>>
>>>> On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:
>>>>
>>>>> I'd like to begin record spanish prompts for FS.
>>>>>
>>>>> Do you know any software/hardware to make it?
>>>>>
>>>>> Thank you
>>>>>
>>>>> BR
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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>
>
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From brian at freeswitch.org  Tue Aug 11 07:58:22 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 09:58:22 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
Message-ID: <D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>

This can all be accomplished with standard bridge lines stacked one on  
top of the other with continue_on_fail set to the list of codes you  
wish to fail over with.  You would also want to set  
hangup_after_bridge=true.  Loopback should NEVER be used in this case.

http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge

/b

On Aug 11, 2009, at 9:46 AM, Phillip Jones wrote:

> Thanks for the reply Brian.
>
> I went through this fairly carefully with Mike Jerris - please see
> http://www.nabble.com/Calling-Multiple-Destinations-with-Failover-td24877157.html
>
> I am very open to any other mechanism that allows the calling multiple
> destinations with carrier failover support. I would have thought this
> is a fairly common requirement and is a fundamental requirement in my
> application.
>
> As always - your help is appreciated.

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From moises.silva at gmail.com  Tue Aug 11 07:59:15 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 11 Aug 2009 10:59:15 -0400
Subject: [Freeswitch-users] OpenZAP/Sangoma in Windows
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED9435@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED9435@cooper>
Message-ID: <c4d05cbe0908110759j2c929796g1ce96f530b7ad681@mail.gmail.com>

On Tue, Aug 11, 2009 at 2:16 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

> Hi, I'm trying to evaluate the OpenZAP/Sangoma-support in Windows, using
> PRI E1 connections.
>

Thanks for testing this :-)

I have been meaning to install FreeSWITCH on Windows but just could not find
the time.


> 1. Has anyone tested it in Windows at all? I know the build-files for
> Visual Studio has only been checked in for a couple of months, so that's why
> I'm asking.
>

The drivers have been tested quite well but not using FreeSWITCH.


> 2. Does anyone have any directions how to configure the driver within
> Windows? Should I use BitStream or HDLC, and how should the channel groups
> be configured?
>

You should use HDLC for the D-channel and Bitstream for the B-Channels.
Typically you would create 2 groups, one with channels 1-23 and the other
with just channel 24. The first group would work in bitstream and
TDM_CHAN_VOICE_API operational mode and the second in HDLC/API mode.

You can find me in #openzap, #freeswitch or #freeswitch-dev as "moy" if you
have more questions.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From dftoro at yahoo.com  Tue Aug 11 08:07:27 2009
From: dftoro at yahoo.com (Diego Toro)
Date: Tue, 11 Aug 2009 08:07:27 -0700 (PDT)
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
Message-ID: <805599.45498.qm@web33502.mail.mud.yahoo.com>

Hi all,
?
I have some suggested code to?support currency and number pronounced in spanish,
?
http://jira.freeswitch.org/browse/MODAPP-317?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel
?

Diego 
http://lacarretade.blogspot.com
?
?


--- On Tue, 8/11/09, Nicolas Brenner <nicolas at medularis.com> wrote:


From: Nicolas Brenner <nicolas at medularis.com>
Subject: Re: [Freeswitch-users] Spanish Prompts
To: freeswitch-users at lists.freeswitch.org
Date: Tuesday, August 11, 2009, 9:27 AM


I think bakko is right, I have a few observations about the changes proposed by samuel, and I think it makes a lot of sense to have different localizations.



On Tue, Aug 11, 2009 at 3:49 AM, samuel <samu60 at gmail.com> wrote:

I've taken a look at phrase_es.xml and would like other spanish native speakers to check the following (i preferred to write down here my impressions so others can contribute instead of sending a patch):

1) Centavos

<prompt phrase="Centavos por minuto" filename="cents-per-minute.wav"/>
<prompt phrase="Centavos" filename="cents.wav"/>?

probably in european spanish the most apropiate word for cents is "c?ntimo" but, if i'm not mistaken, in south american spanish is more common to use "centavos" so this is more a localization issue...as you wish.
Since I live in Spain I would use the following:
<prompt phrase="C?ntimos por minuto" filename="cents-per-minute.wav"/>
<prompt phrase="C?ntimos" filename="cents.wav"/>


You are right, in South America we use Centavos.
?

2) Minnor stress corrections:
instead of Dolar, you should use the stressed word D?lar:
-<prompt phrase="Dolar" filename="dollar.wav"/>
+<prompt phrase="D?lar" filename="dollar.wav"/>

instead of numero, you should use the stressed word n?mero:
-<prompt phrase="por favor ingrese su numero de usuario, seguido por" filename="vm-enter_id.wav"/>
+<prompt phrase="por favor ingrese su n?mero de usuario, seguido por" filename="vm-enter_id.wav"/>

3) en instead of a
-<prompt phrase="La persona a la extensi?n" filename="vm-person.wav"/>
+<prompt phrase="La persona en la extensi?n" filename="vm-person.wav"/>

4) femenin genre of contrase?a changes the word seguido for seguida
-<prompt phrase="por favor ingrese su contrase?a, seguido por" filename="vm-enter_pass.wav"/>
+<prompt phrase="por favor ingrese su contrase?a, seguida por" filename="vm-enter_pass.wav"/>

5) I would propose the word "almohadilla" as the translation of pound so I would change following sentences:
-<prompt phrase="Signo de n?mero" filename="35.wav"/>
+<prompt phrase="Almohadilla" filename="35.wav"/>

-<prompt phrase="Signo de n?mero" filename="pound.wav"/>
+<prompt phrase="Almohadilla" filename="pound.wav"/>


I don't know about the rest of the countries in South America, but at least in Chile it would me much clearer "Signo de n?mero" instead of Almohadilla (first time I've heard that name for the pound sign).
?

-<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del signo de n?mero" filename="ivr-enter_ext.wav"/>
+<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido de almohadilla" filename="ivr-enter_ext.wav"/>

-<prompt phrase="Seguido del signo de n?mero" filename="followed.wav"/>
+<prompt phrase="Seguido de almohadilla" filename="followed.wav"/>

6)changing prepositions:
-<prompt phrase="Un error ha ocurrido, por favor contacte al administrador" filename="error.wav"/>
+<prompt phrase="Un error ha ocurrido, por favor contacte con el administrador" filename="error.wav"/>


Here it would be:
<prompt phrase="Ha ocurrido un error, por favor contacte al administrador" filename="error.wav"/> 



7)instead of using ingrese, it sounds more natural to use introduzca when interacting with an IVR asking for inserting data. So:
-<prompt phrase="por favor ingrese su numero de usuario, seguido por" filename="vm-enter_id.wav"/>
+<prompt phrase="por favor introduzca su n?mero de usuario, seguido por" filename="vm-enter_id.wav"/>

-<prompt phrase="por favor ingrese su contrase?a, seguido por" filename="vm-enter_pass.wav"/>
+<prompt phrase="por favor introduzca su contrase?a, seguida por" filename="vm-enter_pass.wav"/>

-<prompt phrase="Por favor ingrese el pin de la conferencia." filename="conf-pin.wav"/>
+<prompt phrase="Por favor introduzca el pin de la conferencia." filename="conf-pin.wav"/>

-<prompt phrase="por favor ingrese el n?mero de una extensi?n seguido del signo de n?mero" filename="ivr-enter_ext.wav"/>
+<prompt phrase="por favor introduzca el n?mero de una extensi?n seguido de almohadilla" filename="ivr-enter_ext.wav"/>



Comments?


I am not very convinced by the use of stresses in the words, I know it's "more correct", but sometimes it's a pain, especially if you are working on a remote server and don't have the right keyboard setting or something like that. I've also had a few headaches with file encoding, so depending on that and changes to the encoding you can screw your config files really easily. For simplification I'd leave all stresses out.
?

Samuel



2009/8/11 Michael Collins <msc at freeswitch.org>




Be sure to hop on IRC and speak with jmesquita because he's been working on this also. It would be very good to have Spanish-speaking users review the Spanish phrase_es.xml file for correctness. Also, we need Spanish-speaking programmers to assist with the mod_say application for Spanish - there are things that you have to do in Spanish that you don't have to do in English.

Thanks,
MC





On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com> wrote:

On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound for edition .

Record and edit at 48000Hz and then I use sox to downsample to 8000Hz for FS playback.

Here's a guide that has been put together for reference on what to record.

http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml

Regards,







On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:

I'd like to begin record spanish prompts for FS.

Do you know any software/hardware to make it?

Thank you

BR


_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



_______________________________________________
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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-----Inline Attachment Follows-----


_______________________________________________
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From mike at jerris.com  Tue Aug 11 08:38:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 Aug 2009 11:38:27 -0400
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
Message-ID: <1513B1DC-2CD5-484F-B2CD-74B25E5AB822@jerris.com>

Correct me if I am wrong, but can't all these differences be done in a  
different soundset without modifying any code or other config?

Mike

On Aug 11, 2009, at 10:27 AM, Nicolas Brenner wrote:

> I think bakko is right, I have a few observations about the changes  
> proposed by samuel, and I think it makes a lot of sense to have  
> different localizations.
>
>
> On Tue, Aug 11, 2009 at 3:49 AM, samuel <samu60 at gmail.com> wrote:
> I've taken a look at phrase_es.xml and would like other spanish  
> native speakers to check the following (i preferred to write down  
> here my impressions so others can contribute instead of sending a  
> patch):
>
> 1) Centavos
>
> <prompt phrase="Centavos por minuto" filename="cents-per-minute.wav"/>
> <prompt phrase="Centavos" filename="cents.wav"/>
>
> probably in european spanish the most apropiate word for cents is  
> "c?ntimo" but, if i'm not mistaken, in south american spanish is  
> more common to use "centavos" so this is more a localization  
> issue...as you wish.
> Since I live in Spain I would use the following:
> <prompt phrase="C?ntimos por minuto" filename="cents-per-minute.wav"/>
> <prompt phrase="C?ntimos" filename="cents.wav"/>
>
> You are right, in South America we use Centavos.
>
>
> 2) Minnor stress corrections:
> instead of Dolar, you should use the stressed word D?lar:
> -<prompt phrase="Dolar" filename="dollar.wav"/>
> +<prompt phrase="D?lar" filename="dollar.wav"/>
>
> instead of numero, you should use the stressed word n?mero:
> -<prompt phrase="por favor ingrese su numero de usuario, seguido  
> por" filename="vm-enter_id.wav"/>
> +<prompt phrase="por favor ingrese su n?mero de usuario, seguido  
> por" filename="vm-enter_id.wav"/>
>
> 3) en instead of a
> -<prompt phrase="La persona a la extensi?n" filename="vm-person.wav"/>
> +<prompt phrase="La persona en la extensi?n" filename="vm- 
> person.wav"/>
>
> 4) femenin genre of contrase?a changes the word seguido for seguida
> -<prompt phrase="por favor ingrese su contrase?a, seguido por"  
> filename="vm-enter_pass.wav"/>
> +<prompt phrase="por favor ingrese su contrase?a, seguida por"  
> filename="vm-enter_pass.wav"/>
>
> 5) I would propose the word "almohadilla" as the translation of  
> pound so I would change following sentences:
> -<prompt phrase="Signo de n?mero" filename="35.wav"/>
> +<prompt phrase="Almohadilla" filename="35.wav"/>
>
> -<prompt phrase="Signo de n?mero" filename="pound.wav"/>
> +<prompt phrase="Almohadilla" filename="pound.wav"/>
>
> I don't know about the rest of the countries in South America, but  
> at least in Chile it would me much clearer "Signo de n?mero" instead  
> of Almohadilla (first time I've heard that name for the pound sign).
>
>
> -<prompt phrase="por favor ingrese el n?mero de una extensi?n  
> seguido del signo de n?mero" filename="ivr-enter_ext.wav"/>
> +<prompt phrase="por favor ingrese el n?mero de una extensi?n  
> seguido de almohadilla" filename="ivr-enter_ext.wav"/>
>
> -<prompt phrase="Seguido del signo de n?mero"  
> filename="followed.wav"/>
> +<prompt phrase="Seguido de almohadilla" filename="followed.wav"/>
>
> 6)changing prepositions:
> -<prompt phrase="Un error ha ocurrido, por favor contacte al  
> administrador" filename="error.wav"/>
> +<prompt phrase="Un error ha ocurrido, por favor contacte con el  
> administrador" filename="error.wav"/>
>
> Here it would be:
> <prompt phrase="Ha ocurrido un error, por favor contacte al  
> administrador" filename="error.wav"/>
>
>
> 7)instead of using ingrese, it sounds more natural to use introduzca  
> when interacting with an IVR asking for inserting data. So:
> -<prompt phrase="por favor ingrese su numero de usuario, seguido  
> por" filename="vm-enter_id.wav"/>
> +<prompt phrase="por favor introduzca su n?mero de usuario, seguido  
> por" filename="vm-enter_id.wav"/>
>
> -<prompt phrase="por favor ingrese su contrase?a, seguido por"  
> filename="vm-enter_pass.wav"/>
> +<prompt phrase="por favor introduzca su contrase?a, seguida por"  
> filename="vm-enter_pass.wav"/>
>
> -<prompt phrase="Por favor ingrese el pin de la conferencia."  
> filename="conf-pin.wav"/>
> +<prompt phrase="Por favor introduzca el pin de la conferencia."  
> filename="conf-pin.wav"/>
>
> -<prompt phrase="por favor ingrese el n?mero de una extensi?n  
> seguido del signo de n?mero" filename="ivr-enter_ext.wav"/>
> +<prompt phrase="por favor introduzca el n?mero de una extensi?n  
> seguido de almohadilla" filename="ivr-enter_ext.wav"/>
>
>
>
> Comments?
>
> I am not very convinced by the use of stresses in the words, I know  
> it's "more correct", but sometimes it's a pain, especially if you  
> are working on a remote server and don't have the right keyboard  
> setting or something like that. I've also had a few headaches with  
> file encoding, so depending on that and changes to the encoding you  
> can screw your config files really easily. For simplification I'd  
> leave all stresses out.
>
>
> Samuel
>
>
> 2009/8/11 Michael Collins <msc at freeswitch.org>
>
> Be sure to hop on IRC and speak with jmesquita because he's been  
> working on this also. It would be very good to have Spanish-speaking  
> users review the Spanish phrase_es.xml file for correctness. Also,  
> we need Spanish-speaking programmers to assist with the mod_say  
> application for Spanish - there are things that you have to do in  
> Spanish that you don't have to do in English.
>
> Thanks,
> MC
>
>
> On Fri, Aug 7, 2009 at 1:57 PM, Luis F Urrea <lfurrea at gmail.com>  
> wrote:
> On linux I use Jack + Jamin + Ardour GTK2 for the takes and Rezound  
> for edition .
>
> Record and edit at 48000Hz and then I use sox to downsample to  
> 8000Hz for FS playback.
>
> Here's a guide that has been put together for reference on what to  
> record.
>
> http://fisheye.freeswitch.org/browse/FreeSWITCH/docs/phrase/phrase_es.xml
>
> Regards,
>
>
>
>
> On Fri, Aug 7, 2009 at 9:21 AM, bakko <asannucci at gmail.com> wrote:
> I'd like to begin record spanish prompts for FS.
>
> Do you know any software/hardware to make it?
>
> Thank you
>
> BR
>
>

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From pjintheusa at gmail.com  Tue Aug 11 09:08:03 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 11 Aug 2009 12:08:03 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
Message-ID: <367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>

Can you use this method to make simultaneous calls though while
preserving carrier order? i.e. each call must try carrier 1 first then
carrier 2 (because each carrier terminates a different subset of
numbers)

So - if I did NOT want to support failover I would use:

session:execute("bridge",
"sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828");
 << call 6095553828 and 7325553828 at the same time

If I were to use failover and not need multiple destinations then I would use:

session:execute("bridge",
"sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828");
<< call using broadvox first and quest it that fails

I need to combine:

Call 6095553828  and 7325553828 simultaneously and each trying
broadvox then on fail quest.

session:execute("bridge",
"(sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828),(sofia/gateway/broadvox/7325553828,sofia/gateway/quest/7325553828)");

But of course that syntax does not work.

I can not see another syntax that would achieve this?


On Tue, Aug 11, 2009 at 10:58 AM, Brian West<brian at freeswitch.org> wrote:
> This can all be accomplished with standard bridge lines stacked one on top
> of the other with continue_on_fail set to the list of codes you wish to fail
> over with. ?You would also want to set hangup_after_bridge=true. ?Loopback
> should NEVER be used in this case.
> http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail
> http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge
> /b
> On Aug 11, 2009, at 9:46 AM, Phillip Jones wrote:
>
> Thanks for the reply Brian.
>
> I went through this fairly carefully with Mike Jerris - please see
> http://www.nabble.com/Calling-Multiple-Destinations-with-Failover-td24877157.html
>
> I am very open to any other mechanism that allows the calling multiple
> destinations with carrier failover support. I would have thought this
> is a fairly common requirement and is a fundamental requirement in my
> application.
>
> As always - your help is appreciated.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From jmesquita at gmail.com  Tue Aug 11 09:30:59 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 11 Aug 2009 13:30:59 -0300
Subject: [Freeswitch-users] VoiceMail transcription
In-Reply-To: <2bee4fc40908110717p43649678ue2b94ecb571bd09@mail.gmail.com>
References: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
	<1249998014.20224.82.camel@dk-d820>
	<2bee4fc40908110717p43649678ue2b94ecb571bd09@mail.gmail.com>
Message-ID: <5a8712120908110930s74b84bf8x673ca1209fbae965@mail.gmail.com>

I am sorry for the ignorance on the matter, but how does google voice does?
Do they also have humans?

jmesquita

On Tue, Aug 11, 2009 at 11:17 AM, Kirk Bateman <kirk.bateman at gmail.com>wrote:

> I'm still interested in getting pocketsphinx to attempt speech recognition
> on an audio file.
>
> To be honest, most of the problem is that at 8Khz (mobile phone call rate),
> speech detection is NOT very accurate, at 16Khz it IS significantly better.
>
> I'm planning to have a play with the speechtools module and
> mod_pocketsphinx etc to try and get an audio file parsed, spare time
> permitting.
>
> Will let the list know if I get anywhere.
>
> Regards
>
> Kirk Bateman
>
>
> 2009/8/11 David Knell <dave at 3c.co.uk>
>
> Hi Pete,
>>
>> I'm afraid that the answer's still the same: use a human.  Here's an
>> article describing the state of the art:
>> http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/
>> - the links to previous stories at the bottom provide good background.
>>
>> --Dave
>>
>> > I apologize, I should have been more clear.  We will be using humans
>> > to scan the translated results.  But we are looking for a system to
>> > perform the "first pass" on the audio to hopefully help the human type
>> > less.
>> >
>> > Although the question has been raised if it's faster to have a human
>> > just transcribe the whole thing, or fix up what the computer spit out.
>> > If you have any insights on this, that would be great.
>> >
>> > -pete
>> >
>> >         -------- Original Message --------
>> >         Subject: Re: [Freeswitch-users] VoiceMail transcription
>> >         From: David Knell <dave at 3c.co.uk>
>> >         Date: Mon, August 10, 2009 11:51 am
>> >         To: freeswitch-users at lists.freeswitch.org
>> >
>> >         Good evening Pete,
>> >
>> >         The only way to do this is, I'm afraid, to use a human. We use
>> >         Amazon's
>> >         Mechanical Turk to good effect.
>> >
>> >         Cheers --
>> >
>> >         Dave
>> >
>> >         > Good morning all,
>> >         >
>> >         > I realize this is slightly off the FS topic, but I am
>> >         wondering if
>> >         > anyone out there has experience with software packages
>> >         designed for
>> >         > the transcription of voicemails to text. I've used
>> >         pocketsphinx with
>> >         > FS to handle IVR menus, but now have the task of figuring
>> >         out how to
>> >         > convert recorded phone conversations (voicemails mostly) to
>> >         text.
>> >         >
>> >         > This does not have to be a real-time process, I can store
>> >         the audio
>> >         > files and process them over time. This would need to be a
>> >         software
>> >         > (preferable open source) solution. ASPs like VoiceCloud
>> >         would not
>> >         > work for this application.
>> >         >
>> >         > Thanks for any help
>> >         > -pete
>> >         > _______________________________________________
>> >         > FreeSWITCH-users mailing list
>> >         > FreeSWITCH-users at lists.freeswitch.org
>> >         >
>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >         >
>> >         UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >         > http://www.freeswitch.org
>> >         --
>> >         David Knell, Director, 3C Limited
>> >         T: +44 20 3298 2000
>> >         E: dave at 3c.co.uk
>> >         W: http://www.3c.co.uk
>> >
>> >
>> >         _______________________________________________
>> >         FreeSWITCH-users mailing list
>> >         FreeSWITCH-users at lists.freeswitch.org
>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >         UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >         http://www.freeswitch.org
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> --
>> David Knell, Director, 3C Limited
>> T: +44 20 3298 2000
>> E: dave at 3c.co.uk
>> W: http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Aug 11 09:55:26 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 11:55:26 -0500
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <4A811FEC.7040808@kgolding.co.uk>
References: <4A80AEE2.7050306@kgolding.co.uk>
	<7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>
	<4A811FEC.7040808@kgolding.co.uk>
Message-ID: <87f2f3b90908110955s3f67e710je913a34c7cc3055a@mail.gmail.com>

On Tue, Aug 11, 2009 at 2:38 AM, Kevin Golding <kevin at kgolding.co.uk> wrote:

> Thanks Brian,
>
> Well spotted with the domain/cidr :)
>
> I changed the line <param name="apply-inbound-acl" value="domains"/> in
> the internal.xml with a new value, and changed the list to match but I
> still get the same 'Rejected by acl "domains"' error.
>

You need to apply the name of your new ACL in the sip profile. In the file
internal.xml look for this line:
<param name="apply-inbound-acl" value="domains"/>

And change "domains" to whatever your new ACL is named:
<param name="apply-inbound-acl" value="YOUR_NEW_ACL_NAME_HERE"/>

Then restart FS or go to the command line and execute:
reloadacl reloadxml

Should be good after that!
-MC

>
> And yes I reloaded the xml. :)
>
> Kevin
>
> Brian West wrote:
> > First off edit the profile to use your own custom ACL... not the
> > domains one... secondly you'll have to use cidr= instead of domain=.
> >
> > Domain= will search the user directory building an ACL list from all
> > users with the cidr= attribute.
> >
> > /b
> >
> >
> > On Aug 10, 2009, at 6:36 PM, Kevin Golding wrote:
> >
> >> acl.conf.xml file, and have put the following in it:
> >>
> >> <configuration name="acl.conf" description="Network Lists">
> >>   <network-lists>
> >>     <list name="domains" value="allow">
> >>         <node type="allow" domain="$${domain}"/>
> >>         <node type="allow" domain="213.166.5.128/28"/>
> >>         <node type="allow" domain="87.238.72.128/26"/>
> >>     </list>
> >>   </network-lists>
> >> </configuration>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bsnipes at snipes.org  Tue Aug 11 09:56:04 2009
From: bsnipes at snipes.org (bsnipes)
Date: Tue, 11 Aug 2009 09:56:04 -0700 (PDT)
Subject: [Freeswitch-users]  Multiple phones and single extensions
Message-ID: <24921557.post@talk.nabble.com>


I've tried to do this a couple of ways in the past but can't quite seem to
get it right. The scenario is this:

attorney at extension 326 with 2 phones
secretary at extension 327 with 1 phone and a blf that monitors 326

Initially I was simply adding the attorney's extension to the secretary's
phone as a second registration but the issue comes up that the monitor light
for the 326 line lights up as if the attorney is on the phone and to other
people it shows the attorney's extension in use as if he were on the phone. 
Ok says I... I'll change the dialstring and have it ring both his and her
extension so that if she picks up it will be on her extension and it will
still show her with her blf if he is on his 326 extension. That works but
the blf light stays lit a lot after she picks up on a call destined for his
extension.  I use the following dialstring lines:

<action application="set" data="pcaller_id_name=ATTY:${caller_id_name}"/>
<action application="set"
data="dialstring=${sofia_contact(internal/${dialed_ext}@192.168.5.4)},[origination_caller_id_name=${pcaller_id_name}]sofia/default/327"/>

The phones are SNOM 370s and what  I am not sure of is if the light
constantly blinking is a firmware issue on the phone or a notification issue
from FS. I've noticed that the same constant blinking happens with groups
also. If I monitor a phone that is a member of a group and the group gets
called the light for that monitor key and the message light will blink even
if no one is on the phone. Doesn't happen every time but multiple times a
day it does.

Any suggestions?
Brian

-- 
View this message in context: http://www.nabble.com/Multiple-phones-and-single-extensions-tp24921557p24921557.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From msc at freeswitch.org  Tue Aug 11 10:09:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 12:09:51 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
Message-ID: <87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>

On Tue, Aug 11, 2009 at 9:05 AM, Milena <testeador01 at gmail.com> wrote:

>
> Hello Brian,
>
> I wanna fix the wiki, but to make sure i got it right, does it only work on
> outbound event socket? or is there any other scenario where it would work.
>

FYI, Diego Viola fixed the wiki. (Thanks Diego!)
-MC
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From lfurrea at gmail.com  Tue Aug 11 10:11:41 2009
From: lfurrea at gmail.com (Luis F Urrea)
Date: Tue, 11 Aug 2009 11:11:41 -0600
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <1513B1DC-2CD5-484F-B2CD-74B25E5AB822@jerris.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<1513B1DC-2CD5-484F-B2CD-74B25E5AB822@jerris.com>
Message-ID: <f120f4c80908111011h75ced645g74e363bbdabf1d19@mail.gmail.com>

On Tue, Aug 11, 2009 at 9:38 AM, Michael Jerris <mike at jerris.com> wrote:

> Correct me if I am wrong, but can't all these differences be done in a
> different soundset without modifying any code or other config?
>

Particularly the changes in discussion only refer to different ways to say
things.
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From brian at freeswitch.org  Tue Aug 11 10:37:55 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 12:37:55 -0500
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <4A811FEC.7040808@kgolding.co.uk>
References: <4A80AEE2.7050306@kgolding.co.uk>
	<7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>
	<4A811FEC.7040808@kgolding.co.uk>
Message-ID: <0C9B302E-C1A6-4D1F-831B-63FA2AEB8C37@freeswitch.org>

you have to now apply the ACL correctly to the sofia profile.

/b

On Aug 11, 2009, at 2:38 AM, Kevin Golding wrote:

> Thanks Brian,
>
> Well spotted with the domain/cidr :)
>
> I changed the line <param name="apply-inbound-acl" value="domains"/>  
> in
> the internal.xml with a new value, and changed the list to match but I
> still get the same 'Rejected by acl "domains"' error.
>
> And yes I reloaded the xml. :)
>
> Kevin




From msc at freeswitch.org  Tue Aug 11 11:07:23 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 13:07:23 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
Message-ID: <87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>

On Tue, Aug 11, 2009 at 11:08 AM, Phillip Jones <pjintheusa at gmail.com>wrote:

> Can you use this method to make simultaneous calls though while
> preserving carrier order? i.e. each call must try carrier 1 first then
> carrier 2 (because each carrier terminates a different subset of
> numbers)
>
> So - if I did NOT want to support failover I would use:
>
> session:execute("bridge",
> "sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828");
>  << call 6095553828 and 7325553828 at the same time
>
> If I were to use failover and not need multiple destinations then I would
> use:
>
> session:execute("bridge",
> "sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828");
> << call using broadvox first and quest it that fails
>
> I need to combine:
>
> Call 6095553828  and 7325553828 simultaneously and each trying
> broadvox then on fail quest.
>
> session:execute("bridge",
>
> "(sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828),(sofia/gateway/broadvox/7325553828,sofia/gateway/quest/7325553828)");
>
> But of course that syntax does not work.
>
> I can not see another syntax that would achieve this?


Just to make sure I'm reading you correctly I want to clarify...

You are trying to maintain carrier order and also trying to dial
simultaneously. If I understand what you mean, you have two separate
dialstrings that you want to dial simultaneously and if BOTH fail, THEN you
want to dial the third dialstring?

If the above is correct then this dialplan snippet SHOULD work:

<action application="set" data="continue_on_fail=true"/>
<action application="bridge"
data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
<!-- if either of the above dialstrings is successful then this next line
won't execute... -->
<action application="bridge" data="sofia/gateway/quest/7325553828"/>

I also noticed that you're using js in your example... I hope you will
consider using the dialplan instead. The DP is quick and clean. At the very
least, if you really need js for some logic then let the js do the logic and
then transfer back out to the dialplan. That way you will use fewer system
resources and your project can scale more easily.

-MC
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From diego.viola at gmail.com  Tue Aug 11 11:10:42 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 11 Aug 2009 14:10:42 -0400
Subject: [Freeswitch-users] answer command
In-Reply-To: <87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
Message-ID: <86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>

Michael, you're welcome :).

Milena, answer is a mod_dptools command, you can use it from the XML
dialplan or from the event socket outbound. mod_commands API are APIs that
you execute from the socket, event socket inbound, etc. But you can also
execute them from event socket outbound using the "api" command.

I hope that makes sense, correct me if I'm wrong =D.

On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Aug 11, 2009 at 9:05 AM, Milena <testeador01 at gmail.com> wrote:
>
>>
>> Hello Brian,
>>
>> I wanna fix the wiki, but to make sure i got it right, does it only work
>> on outbound event socket? or is there any other scenario where it would
>> work.
>>
>
> FYI, Diego Viola fixed the wiki. (Thanks Diego!)
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Aug 11 11:23:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 13:23:41 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
Message-ID: <87f2f3b90908111123x7316d515lf7afe91e56aad842@mail.gmail.com>

On Tue, Aug 11, 2009 at 1:10 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Michael, you're welcome :).
>
> Milena, answer is a mod_dptools command, you can use it from the XML
> dialplan or from the event socket outbound. mod_commands API are APIs that
> you execute from the socket, event socket inbound, etc. But you can also
> execute them from event socket outbound using the "api" command.
>
> I hope that makes sense, correct me if I'm wrong =D.


A simple way to remember it is like this:

mod_dptools are dialplan tools that go in the dialplan, aka dialplan apps
mod_commands are APIs that are typed at the FS command line

Of course, there's some crossover, but as long as you separate the idea of
an API and an app you'll be fine.
-MC
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From diego.viola at gmail.com  Tue Aug 11 11:24:08 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 11 Aug 2009 14:24:08 -0400
Subject: [Freeswitch-users] answer command
In-Reply-To: <86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
Message-ID: <86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>

I suggest that you learn the differences between mod_commands commands and
mod_dptools applications, and also the interfaces where you can access and
use them.

As said before, mod_dptools is accessible from dialplan, event socket
outbound, etc. and mod_commands is accessible from the CLI, event socket
(inbound/outbound), XML RPC, etc.

That's all described in the wiki I think.

Let us know if you have any questions =D.

On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Michael, you're welcome :).
>
> Milena, answer is a mod_dptools command, you can use it from the XML
> dialplan or from the event socket outbound. mod_commands API are APIs that
> you execute from the socket, event socket inbound, etc. But you can also
> execute them from event socket outbound using the "api" command.
>
> I hope that makes sense, correct me if I'm wrong =D.
>
> On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Tue, Aug 11, 2009 at 9:05 AM, Milena <testeador01 at gmail.com> wrote:
>>
>>>
>>> Hello Brian,
>>>
>>> I wanna fix the wiki, but to make sure i got it right, does it only work
>>> on outbound event socket? or is there any other scenario where it would
>>> work.
>>>
>>
>> FYI, Diego Viola fixed the wiki. (Thanks Diego!)
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From boonedox at gmail.com  Tue Aug 11 11:29:44 2009
From: boonedox at gmail.com (Jeremiah Johnson)
Date: Tue, 11 Aug 2009 12:29:44 -0600
Subject: [Freeswitch-users] hangup_after_bridge=false only works if
	bypass_media=false
Message-ID: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>

This is an integral part of my application.  I need to have FreeSWITCH
outside of the media path as well as be able to do multiple bridges for the
same "A" leg.
/*WORKS*/
<action application="set" data="hangup_after_bridge=false"/>
<action application="bridge"
data="sofia/gateway/${mygateway}/1${client_one}"/>
<action application="bridge"
data="sofia/gateway/${mygateway}/1${client_two}"/>

/*DOES NOT WORK*/
<action application="set" data="hangup_after_bridge=false"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge"
data="sofia/gateway/${mygateway}/1${client_one}"/>
<action application="bridge"
data="sofia/gateway/${mygateway}/1${client_two}"/>

In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the leg for
client_one hangs up.
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From pjintheusa at gmail.com  Tue Aug 11 11:43:36 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 11 Aug 2009 14:43:36 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
Message-ID: <367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>

Hi Michael,

Thanks for trying to help with this.

> <action application="set" data="continue_on_fail=true"/>
> <action application="bridge"
> data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
> <!-- if either of the above dialstrings is successful then this next line
> won't execute... -->
> <action application="bridge" data="sofia/gateway/quest/7325553828"/>

In the dial plan above, consider that target is sitting at their
6095553828 phone. And that broadvox does not route to 6095553828  and
passes me a NO_ROUTE_DESTINATION.

The target will never get the call.

So if I extend this to:

<action application="bridge"  data="sofia/gateway/broadvox/6095553828,
sofia/gateway/broadvox/7325553828"/>
<action application="bridge" data="sofia/gateway/quest/6095553828.
sofia/gateway/quest/7325553828"/>

In this plan you are not calling the numbers simultaneously if the
broadvox is not able to route both calls. I can not see that you can
stack this way and get the ability to call multiple destinations, so I
must be missing something.

"Call blast" or "Followme" is basically the functionality I am trying
to implement. Separately we use multiple carriers. This is fairly
common requirement.

I am using .NET by the way - but point taken.

Any other ideas on how to achieve multiple destinations with carrier
failover would great!

Thanks again.


On Tue, Aug 11, 2009 at 2:07 PM, Michael Collins<msc at freeswitch.org> wrote:
>
>
> On Tue, Aug 11, 2009 at 11:08 AM, Phillip Jones <pjintheusa at gmail.com>
> wrote:
>>
>> Can you use this method to make simultaneous calls though while
>> preserving carrier order? i.e. each call must try carrier 1 first then
>> carrier 2 (because each carrier terminates a different subset of
>> numbers)
>>
>> So - if I did NOT want to support failover I would use:
>>
>> session:execute("bridge",
>> "sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828");
>> ?<< call 6095553828 and 7325553828 at the same time
>>
>> If I were to use failover and not need multiple destinations then I would
>> use:
>>
>> session:execute("bridge",
>> "sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828");
>> << call using broadvox first and quest it that fails
>>
>> I need to combine:
>>
>> Call 6095553828 ?and 7325553828 simultaneously and each trying
>> broadvox then on fail quest.
>>
>> session:execute("bridge",
>>
>> "(sofia/gateway/broadvox/6095553828|sofia/gateway/quest/6095553828),(sofia/gateway/broadvox/7325553828,sofia/gateway/quest/7325553828)");
>>
>> But of course that syntax does not work.
>>
>> I can not see another syntax that would achieve this?
>
> Just to make sure I'm reading you correctly I want to clarify...
>
> You are trying to maintain carrier order and also trying to dial
> simultaneously. If I understand what you mean, you have two separate
> dialstrings that you want to dial simultaneously and if BOTH fail, THEN you
> want to dial the third dialstring?
>
> If the above is correct then this dialplan snippet SHOULD work:
>
> <action application="set" data="continue_on_fail=true"/>
> <action application="bridge"
> data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
> <!-- if either of the above dialstrings is successful then this next line
> won't execute... -->
> <action application="bridge" data="sofia/gateway/quest/7325553828"/>
>
> I also noticed that you're using js in your example... I hope you will
> consider using the dialplan instead. The DP is quick and clean. At the very
> least, if you really need js for some logic then let the js do the logic and
> then transfer back out to the dialplan. That way you will use fewer system
> resources and your project can scale more easily.
>
> -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From pjintheusa at gmail.com  Tue Aug 11 11:50:42 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 11 Aug 2009 14:50:42 -0400
Subject: [Freeswitch-users] hangup_after_bridge=false only works if
	bypass_media=false
In-Reply-To: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
References: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
Message-ID: <367751820908111150s468a1717l20cf409ef8b7f66b@mail.gmail.com>

Hi there,

I the 'does not work' example your media stream is back up at the
carrier as the called party hangs up.

You would have to cause another SIP REINVITE and re capture the media.

See http://wiki.freeswitch.org/wiki/Bypass_Media and the "How to
disable/enable it on the fly?" section as a start.




On Tue, Aug 11, 2009 at 2:29 PM, Jeremiah Johnson<boonedox at gmail.com> wrote:
> This is an integral part of my application. ?I need to have FreeSWITCH
> outside of the media path as well as be able to do multiple bridges for the
> same "A" leg.
> /*WORKS*/
> <action application="set" data="hangup_after_bridge=false"/>
> <action application="bridge"
> data="sofia/gateway/${mygateway}/1${client_one}"/>
> <action application="bridge"
> data="sofia/gateway/${mygateway}/1${client_two}"/>
> /*DOES NOT WORK*/
> <action application="set" data="hangup_after_bridge=false"/>
> <action application="set" data="bypass_media=true"/>
> <action application="bridge"
> data="sofia/gateway/${mygateway}/1${client_one}"/>
> <action application="bridge"
> data="sofia/gateway/${mygateway}/1${client_two}"/>
> In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the leg for
> client_one hangs up.
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From lfurrea at gmail.com  Tue Aug 11 11:55:45 2009
From: lfurrea at gmail.com (Luis F Urrea)
Date: Tue, 11 Aug 2009 12:55:45 -0600
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <f120f4c80908111011h75ced645g74e363bbdabf1d19@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<1513B1DC-2CD5-484F-B2CD-74B25E5AB822@jerris.com>
	<f120f4c80908111011h75ced645g74e363bbdabf1d19@mail.gmail.com>
Message-ID: <f120f4c80908111155y7704c08fi922706487ab5b3d9@mail.gmail.com>

The patch submitted by Diego Toro, however would fix the actual
pronunciation of currency and dates  from 20th - 29th which are not working
with the "english" implementation of mod_say_es.c

In regards the dates pronunciation for voicemail for example we used the
following changes to mod_say_es.c

372,375c412,427
-               say_file("time/day-%d.wav", tm.tm_wday);
-               say_file("time/mon-%d.wav", tm.tm_mon);
-               say_num(tm.tm_mday, SSM_COUNTED);
-               say_num(tm.tm_year + 1900, SSM_PRONOUNCED);
---
+               if (tm.tm_mday == 1) {
+                       say_file("time/day-%d.wav", tm.tm_wday);
+                       say_num(tm.tm_mday, SSM_COUNTED);
+                       say_file("time/de.wav");
+                       say_file("time/mon-%d.wav", tm.tm_mon);
+                       say_file("time/de.wav");
+                       say_num(tm.tm_year + 1900, SSM_PRONOUNCED);
+
+               } else {
+                       say_file("time/day-%d.wav", tm.tm_wday);
+                       say_num(tm.tm_mday, SSM_PRONOUNCED);
+                       say_file("time/de.wav");
+                       say_file("time/mon-%d.wav", tm.tm_mon);
+                       say_file("time/de.wav");
+                       say_num(tm.tm_year + 1900, SSM_PRONOUNCED);
+               }
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From gshfreesw at gmail.com  Tue Aug 11 12:10:33 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Tue, 11 Aug 2009 15:10:33 -0400
Subject: [Freeswitch-users] Call transfer issues
Message-ID: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>

Hello,

Good work guys. I am having good fun using freeswitch so far. Currently, I
am having a serious issue on making a call transfer happen. The scenario is
simple.

1. Caller A arrives on extension 1 and is waiting on a fifo queue.
2. Caller B arrives on  extension 2 and dial plan bridges the call to Caller
C.

Now after a certain period of time, I want to hang up Caller B and put
Caller C in the fifo queue where Caller A is waiting. I know the UUIDs of
the all 3 calls. Naively, I tried the following and failed.

1. I put the bridged call to Caller C on park .
2. I did a "hang up" to caller B.
3. I did "fifo in" for Caller C into fifo queue of caller A.

This did not work. What is the right way to do this? Do I have make sure
uuid_transfer? I am using Event socket to do all of this.

-Shameem.
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From brian at freeswitch.org  Tue Aug 11 12:19:15 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 14:19:15 -0500
Subject: [Freeswitch-users] Call transfer issues
In-Reply-To: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>
References: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>
Message-ID: <D554D137-805E-4924-A43B-3930275451BE@freeswitch.org>

OK I try so hard to follow what you're doing but it makes little sense  
to me.

I'm guessing you'll need to set/export hangup_after_bridge=false to  
prevent the hangup from taking place on B and C!

The other option is to uuid_transfer both B and C to park using the - 
both then hangup on C and then uuid_bridge A and C.  Because once you  
break the bridge on B and C by hanging up on B you left C hanging so  
its naturally going to hangup.

-USAGE: <uuid> [-bleg|-both] <dest-exten> [<dialplan>] [<context>]

/b

On Aug 11, 2009, at 2:10 PM, Shameem Shiek wrote:

> Hello,
>
> Good work guys. I am having good fun using freeswitch so far.  
> Currently, I am having a serious issue on making a call transfer  
> happen. The scenario is simple.
>
> 1. Caller A arrives on extension 1 and is waiting on a fifo queue.
> 2. Caller B arrives on  extension 2 and dial plan bridges the call  
> to Caller C.
>
> Now after a certain period of time, I want to hang up Caller B and  
> put Caller C in the fifo queue where Caller A is waiting. I know the  
> UUIDs of the all 3 calls. Naively, I tried the following and failed.
>
> 1. I put the bridged call to Caller C on park .
> 2. I did a "hang up" to caller B.
> 3. I did "fifo in" for Caller C into fifo queue of caller A.
>
> This did not work. What is the right way to do this? Do I have make  
> sure uuid_transfer? I am using Event socket to do all of this.
>
> -Shameem.
> _______________




From msc at freeswitch.org  Tue Aug 11 12:19:43 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 14:19:43 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
Message-ID: <87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>

On Tue, Aug 11, 2009 at 1:43 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi Michael,
>
> Thanks for trying to help with this.
>
> > <action application="set" data="continue_on_fail=true"/>
> > <action application="bridge"
> >
> data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
> > <!-- if either of the above dialstrings is successful then this next line
> > won't execute... -->
> > <action application="bridge" data="sofia/gateway/quest/7325553828"/>
>
> In the dial plan above, consider that target is sitting at their
> 6095553828 phone. And that broadvox does not route to 6095553828  and
> passes me a NO_ROUTE_DESTINATION.
>
> The target will never get the call.
>
> So if I extend this to:
>
> <action application="bridge"  data="sofia/gateway/broadvox/6095553828,
> sofia/gateway/broadvox/7325553828"/>
> <action application="bridge" data="sofia/gateway/quest/6095553828.
> sofia/gateway/quest/7325553828"/>
>
> In this plan you are not calling the numbers simultaneously if the
> broadvox is not able to route both calls. I can not see that you can
> stack this way and get the ability to call multiple destinations, so I
> must be missing something.


When you say "call multiple destinations" what exactly do you mean? If the
Broadvox is unable to connect either call then the dialplan moves on to the
Quest dial attempts. If that isn't what you need then can you clarify? Under
what condition(s) would the dialplan need to attempt to call via Quest?


>
> "Call blast" or "Followme" is basically the functionality I am trying
> to implement. Separately we use multiple carriers. This is fairly
> common requirement.
>

Yep, makes sense. Could you maybe write it out in pseudo code? Something
like this:
#1 Call phone numbers ABC and XYZ simultaneously via Broadvox
#2 If BOTH ABC AND XYZ fail, then call ABC and XYZ via Quest


>
> I am using .NET by the way - but point taken.
>

My mistake. :)


>
> Any other ideas on how to achieve multiple destinations with carrier
> failover would great!
>

I'm 100% certain that this is possible with FreeSWITCH. You might just need
a different approach, depending upon the circumstances with your carriers.
Try the above suggestions first and let's see what happens. We'll take it
from there.
-MC
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From alan at chandlerfamily.org.uk  Tue Aug 11 12:25:14 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Tue, 11 Aug 2009 20:25:14 +0100
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
Message-ID: <4A81C59A.2020705@chandlerfamily.org.uk>

samuel wrote:
> I'm also for different spanish localization if it's not too complicated. 
> It was also for me the first time I see "signo de n?mero" for pound ;)
> 


I just idly noticed this - so just a comment from a Brit who iteracts 
quite a bit with Americans.

The # symbol in the UK is not called "pound" because (I don't know if 
this will come out on your screens correctly) we use ? for our currency.

I would refer to the # symbol as "hash" or just possibly the "number 
symbol".


-- 
Alan Chandler
http://www.chandlerfamily.org.uk





From gshfreesw at gmail.com  Tue Aug 11 12:33:50 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Tue, 11 Aug 2009 15:33:50 -0400
Subject: [Freeswitch-users] Call transfer issues
In-Reply-To: <D554D137-805E-4924-A43B-3930275451BE@freeswitch.org>
References: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>
	<D554D137-805E-4924-A43B-3930275451BE@freeswitch.org>
Message-ID: <5070fcbd0908111233w6ecefeeex92b1304f04744f23@mail.gmail.com>

Thanks for the explanation. I was trying to find the correct way to do this
as I saw several ways of doing this. I will try to explain how I did on the
asterisk land.

In asterisk land, Caller A will be parked at extension 701 and Caller B
"transfers" the call to extension 701 and and Caller A and C are connected
and Caller B is hung up right after transfer.

In Freeswitch, Can Caller B do a "transfer" and connect caller A and C? How
can I do this? Since Freeswitch does not have a concept of parked
extensions.

Thank you for the help.


On Tue, Aug 11, 2009 at 3:19 PM, Brian West <brian at freeswitch.org> wrote:

> OK I try so hard to follow what you're doing but it makes little sense
> to me.
>
> I'm guessing you'll need to set/export hangup_after_bridge=false to
> prevent the hangup from taking place on B and C!
>
> The other option is to uuid_transfer both B and C to park using the -
> both then hangup on C and then uuid_bridge A and C.  Because once you
> break the bridge on B and C by hanging up on B you left C hanging so
> its naturally going to hangup.
>
> -USAGE: <uuid> [-bleg|-both] <dest-exten> [<dialplan>] [<context>]
>
> /b
>
> On Aug 11, 2009, at 2:10 PM, Shameem Shiek wrote:
>
> > Hello,
> >
> > Good work guys. I am having good fun using freeswitch so far.
> > Currently, I am having a serious issue on making a call transfer
> > happen. The scenario is simple.
> >
> > 1. Caller A arrives on extension 1 and is waiting on a fifo queue.
> > 2. Caller B arrives on  extension 2 and dial plan bridges the call
> > to Caller C.
> >
> > Now after a certain period of time, I want to hang up Caller B and
> > put Caller C in the fifo queue where Caller A is waiting. I know the
> > UUIDs of the all 3 calls. Naively, I tried the following and failed.
> >
> > 1. I put the bridged call to Caller C on park .
> > 2. I did a "hang up" to caller B.
> > 3. I did "fifo in" for Caller C into fifo queue of caller A.
> >
> > This did not work. What is the right way to do this? Do I have make
> > sure uuid_transfer? I am using Event socket to do all of this.
> >
> > -Shameem.
> > _______________
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Aug 11 12:40:14 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 14:40:14 -0500
Subject: [Freeswitch-users] Call transfer issues
In-Reply-To: <5070fcbd0908111233w6ecefeeex92b1304f04744f23@mail.gmail.com>
References: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>
	<D554D137-805E-4924-A43B-3930275451BE@freeswitch.org>
	<5070fcbd0908111233w6ecefeeex92b1304f04744f23@mail.gmail.com>
Message-ID: <55901B44-E104-4829-8B40-81B36A3522CA@freeswitch.org>


On Aug 11, 2009, at 2:33 PM, Shameem Shiek wrote:

> Thanks for the explanation. I was trying to find the correct way to  
> do this as I saw several ways of doing this. I will try to explain  
> how I did on the asterisk land.
>
> In asterisk land, Caller A will be parked at extension 701 and  
> Caller B "transfers" the call to extension 701 and and Caller A and  
> C are connected and Caller B is hung up right after transfer.
>
> In Freeswitch, Can Caller B do a "transfer" and connect caller A and  
> C? How can I do this? Since Freeswitch does not have a concept of  
> parked extensions.

This is where you're wrong... you can park people with the park  
application just transfer them to it... But you have to take off the  
asterisk hat.. I told you in the last email how to do it correctly  
using nothing but API commands.

/b


>
> Thank you for the help.




From mike at jerris.com  Tue Aug 11 12:45:38 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 Aug 2009 15:45:38 -0400
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <4A81C59A.2020705@chandlerfamily.org.uk>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
	<4A81C59A.2020705@chandlerfamily.org.uk>
Message-ID: <06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>

again, this issue should be addressed when you do a sound set for that  
dialect, we are attempting to keep the c code common for all dialects  
within a language, we will see if this works unless anyone can point  
to a place this will not work.

Mike

On Aug 11, 2009, at 3:25 PM, Alan Chandler wrote:

> samuel wrote:
>> I'm also for different spanish localization if it's not too  
>> complicated.
>> It was also for me the first time I see "signo de n?mero" for  
>> pound ;)
>>
>
>
> I just idly noticed this - so just a comment from a Brit who iteracts
> quite a bit with Americans.
>
> The # symbol in the UK is not called "pound" because (I don't know if
> this will come out on your screens correctly) we use ? for our  
> currency.
>
> I would refer to the # symbol as "hash" or just possibly the "number
> symbol".
>
>
> -- 
> Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jmesquita at gmail.com  Tue Aug 11 12:49:31 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 11 Aug 2009 16:49:31 -0300
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
	<4A81C59A.2020705@chandlerfamily.org.uk>
	<06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>
Message-ID: <5a8712120908111249t3e23ceeese1c18d6fef52a227@mail.gmail.com>

Mike, the gender thing will eventually have to change code, I guess. I have
not yet looked at the say code, so I am just imagining here.



On Tue, Aug 11, 2009 at 4:45 PM, Michael Jerris <mike at jerris.com> wrote:

> again, this issue should be addressed when you do a sound set for that
> dialect, we are attempting to keep the c code common for all dialects
> within a language, we will see if this works unless anyone can point
> to a place this will not work.
>
> Mike
>
> On Aug 11, 2009, at 3:25 PM, Alan Chandler wrote:
>
> > samuel wrote:
> >> I'm also for different spanish localization if it's not too
> >> complicated.
> >> It was also for me the first time I see "signo de n?mero" for
> >> pound ;)
> >>
> >
> >
> > I just idly noticed this - so just a comment from a Brit who iteracts
> > quite a bit with Americans.
> >
> > The # symbol in the UK is not called "pound" because (I don't know if
> > this will come out on your screens correctly) we use ? for our
> > currency.
> >
> > I would refer to the # symbol as "hash" or just possibly the "number
> > symbol".
> >
> >
> > --
> > Alan Chandler
> > http://www.chandlerfamily.org.uk
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Aug 11 12:54:12 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 14:54:12 -0500
Subject: [Freeswitch-users] Ivr and variables
In-Reply-To: <8D85DC2E55A74F41AE2190E05A44F028@voztovoice>
References: <8D85DC2E55A74F41AE2190E05A44F028@voztovoice>
Message-ID: <87f2f3b90908111254i4479d3f9m68ab8cce6a62554e@mail.gmail.com>

What value is in the variable ${some} ?
-MC

On Mon, Aug 10, 2009 at 1:43 PM, bakko <asannucci at gmail.com> wrote:

> I'm tryng to put a variable in a IVR digits line like this:
>
> <entry action="menu-exec-app" digits="${some}" param="transfer 1001 XML
> default"/>
>
> but don't work
>
> Is It possible tu use this solution?
>
> thank you
>
> BR
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Tue Aug 11 12:54:16 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 11 Aug 2009 15:54:16 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
Message-ID: <367751820908111254h4f39b8d5hf793e3e9dce78cbb@mail.gmail.com>

Thanks Michael,

pseudo code would look this.

1) Call 6095556666 and 7325556666 simultaneously.
2a) As you dial 6095556666 go to the next carrier if you receive
NO_ROUTE_DESTINATION.
2b) As you dial 7325556666 go to the next carrier if you receive
NO_ROUTE_DESTINATION
3) The first number to answer and accept the call - bridge to leg a.
Drop the other.


Brian maintains that this can be done without loopback and loopback
should never be used. Mike Jerris believes that loop back handles this
and should be used. I am happy for all the help, but I am a bit
confused at this stage.

If this can not be done in the dailplan then I will need to pursue
loopback and see how far I get with that. At the moment it works but
not with bypass_media.

As I said "call followme" and carrier failover are fairly common
requirements so I do feel I am missing something. This must be
implemented elsewhere.

Again - thank you all for your help. You do a great job of getting
getting your heads around all this stuff day in day out.

Phillip Jones


On Tue, Aug 11, 2009 at 3:19 PM, Michael Collins<msc at freeswitch.org> wrote:
>
>
> On Tue, Aug 11, 2009 at 1:43 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
>>
>> Hi Michael,
>>
>> Thanks for trying to help with this.
>>
>> > <action application="set" data="continue_on_fail=true"/>
>> > <action application="bridge"
>> >
>> > data="sofia/gateway/broadvox/6095553828,sofia/gateway/broadvox/7325553828"/>
>> > <!-- if either of the above dialstrings is successful then this next
>> > line
>> > won't execute... -->
>> > <action application="bridge" data="sofia/gateway/quest/7325553828"/>
>>
>> In the dial plan above, consider that target is sitting at their
>> 6095553828 phone. And that broadvox does not route to 6095553828 ?and
>> passes me a NO_ROUTE_DESTINATION.
>>
>> The target will never get the call.
>>
>> So if I extend this to:
>>
>> <action application="bridge" ?data="sofia/gateway/broadvox/6095553828,
>> sofia/gateway/broadvox/7325553828"/>
>> <action application="bridge" data="sofia/gateway/quest/6095553828.
>> sofia/gateway/quest/7325553828"/>
>>
>> In this plan you are not calling the numbers simultaneously if the
>> broadvox is not able to route both calls. I can not see that you can
>> stack this way and get the ability to call multiple destinations, so I
>> must be missing something.
>
> When you say "call multiple destinations" what exactly do you mean? If the
> Broadvox is unable to connect either call then the dialplan moves on to the
> Quest dial attempts. If that isn't what you need then can you clarify? Under
> what condition(s) would the dialplan need to attempt to call via Quest?
>
>>
>>
>> "Call blast" or "Followme" is basically the functionality I am trying
>> to implement. Separately we use multiple carriers. This is fairly
>> common requirement.
>
> Yep, makes sense. Could you maybe write it out in pseudo code? Something
> like this:
> #1 Call phone numbers ABC and XYZ simultaneously via Broadvox
> #2 If BOTH ABC AND XYZ fail, then call ABC and XYZ via Quest
>
>>
>> I am using .NET by the way - but point taken.
>
> My mistake. :)
>
>>
>> Any other ideas on how to achieve multiple destinations with carrier
>> failover would great!
>
> I'm 100% certain that this is possible with FreeSWITCH. You might just need
> a different approach, depending upon the circumstances with your carriers.
> Try the above suggestions first and let's see what happens. We'll take it
> from there.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Tue Aug 11 12:54:53 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 14:54:53 -0500
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <5a8712120908111249t3e23ceeese1c18d6fef52a227@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
	<4A81C59A.2020705@chandlerfamily.org.uk>
	<06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>
	<5a8712120908111249t3e23ceeese1c18d6fef52a227@mail.gmail.com>
Message-ID: <87f2f3b90908111254pf443a35qa306f7da93fb336a@mail.gmail.com>

2009/8/11 Jo?o Mesquita <jmesquita at gmail.com>

> Mike, the gender thing will eventually have to change code, I guess. I have
> not yet looked at the say code, so I am just imagining here.
>

Are there gender differences between dialects of the same language?
-MC
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From dave at 3c.co.uk  Tue Aug 11 12:55:12 2009
From: dave at 3c.co.uk (David Knell)
Date: Tue, 11 Aug 2009 22:55:12 +0300
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
Message-ID: <1250020512.4659.16.camel@dk-d820>

Just to add my $0.02-worth (if you're feeling generous..) - I don't
think that the dialplan is expressive enough to do what's needed here,
and that's where the trouble's coming from.  It's not enormously tricky
to build a generic "dial this set of numbers according to these rules"
service using something hanging off the event socket - there's a writeup
here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
be done on SoftIVR.

To roll something similar yourself using the event socket, you'd need to
map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
and have some way of passing messages around between the threads
handling the different call legs, assuming that you're using one thread
per leg.

--Dave




From brian at freeswitch.org  Tue Aug 11 12:55:51 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 14:55:51 -0500
Subject: [Freeswitch-users] Ivr and variables
In-Reply-To: <87f2f3b90908111254i4479d3f9m68ab8cce6a62554e@mail.gmail.com>
References: <8D85DC2E55A74F41AE2190E05A44F028@voztovoice>
	<87f2f3b90908111254i4479d3f9m68ab8cce6a62554e@mail.gmail.com>
Message-ID: <3332EB9D-AFA8-4D51-835D-9C6D812237C8@freeswitch.org>

What he is wanting is not possible.

/b

On Aug 11, 2009, at 2:54 PM, Michael Collins wrote:

> What value is in the variable ${some} ?
> -MC




From jmesquita at gmail.com  Tue Aug 11 12:56:28 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 11 Aug 2009 16:56:28 -0300
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <87f2f3b90908111254pf443a35qa306f7da93fb336a@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
	<4A81C59A.2020705@chandlerfamily.org.uk>
	<06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>
	<5a8712120908111249t3e23ceeese1c18d6fef52a227@mail.gmail.com>
	<87f2f3b90908111254pf443a35qa306f7da93fb336a@mail.gmail.com>
Message-ID: <5a8712120908111256v4b072587r4c5ead9fa4d4afb3@mail.gmail.com>

Oops, I thought you were saying different languages. Sorry about that.

jmesquita

On Tue, Aug 11, 2009 at 4:54 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> 2009/8/11 Jo?o Mesquita <jmesquita at gmail.com>
>
>> Mike, the gender thing will eventually have to change code, I guess. I
>> have not yet looked at the say code, so I am just imagining here.
>>
>
> Are there gender differences between dialects of the same language?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Tue Aug 11 12:59:47 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 11 Aug 2009 15:59:47 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
Message-ID: <86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>

It resolves fine for me already.

On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola <diego.viola at gmail.com> wrote:

> The address for me is: 69.174.57.101
>
>
> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera <godson.g at gmail.com> wrote:
>
>> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
>> name resolution. But the problem is still there even if I use my ISP's  DNS
>> (mostly because its resolving to the same to IP address) .
>>
>> On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> As MikeJ pointed out please report the IP address files.freeswitch.orgresolves to.  We have that on a content delivery network so that the files
>>> are closer to you geographically and you can download them faster but if
>>> you're having an issue I'll need the IP so I can report it correctly.
>>> Thanks,
>>> Brian
>>>
>>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>>
>>> Hi FS Team,
>>>
>>>
>>>         The files.freeswitch.org is resetting connection since 3 days.
>>> As a result I was not able to download latest release of FS. Got the trunk
>>> version from svn. But still it suffers from the lack of sound files.  When
>>> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>>> connection immediately wget tries 20 times and gives up. Other users on IRC
>>> also reported this issue.
>>>
>>> --
>>> Thanks & Regards,
>>> Godson Gera
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Thanks & Regards,
>> Godson Gera
>> http://godson.in
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From brian at freeswitch.org  Tue Aug 11 13:01:28 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 15:01:28 -0500
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
Message-ID: <8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>

Does it still give you connection refused?

/b

On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:

> It resolves fine for me already.
>
> On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> The address for me is: 69.174.57.101
>
>
> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera <godson.g at gmail.com>  
> wrote:
> Ok here is the address 69.174.57.101 . Most of the time I use  
> OpenDNS for name resolution. But the problem is still there even if  
> I use my ISP's  DNS (mostly because its resolving to the same to IP  
> address) .
>
> On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org>  
> wrote:
> As MikeJ pointed out please report the IP address  
> files.freeswitch.org resolves to.  We have that on a content  
> delivery network so that the files are closer to you geographically  
> and you can download them faster but if you're having an issue I'll  
> need the IP so I can report it correctly.
>
> Thanks,
> Brian
>
> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>
>> Hi FS Team,
>>
>>
>>         The files.freeswitch.org is resetting connection since 3  
>> days. As a result I was not able to download latest release of FS.  
>> Got the trunk version from svn. But still it suffers from the lack  
>> of sound files.  When ever I do 'make uhd-sounds-install'http://files.freeswitch.org 
>>  resets connection immediately wget tries 20 times and gives up.  
>> Other users on IRC also reported this issue.
>>
>> -- 
>> Thanks & Regards,
>> Godson Gera
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Thanks & Regards,
> Godson Gera
> http://godson.in
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From diego.viola at gmail.com  Tue Aug 11 13:03:28 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Tue, 11 Aug 2009 16:03:28 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
	<8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
Message-ID: <86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>

Nope.

On Tue, Aug 11, 2009 at 4:01 PM, Brian West <brian at freeswitch.org> wrote:

> Does it still give you connection refused?
> /b
>
> On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:
>
> It resolves fine for me already.
>
> On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> The address for me is: 69.174.57.101
>>
>>
>> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera <godson.g at gmail.com> wrote:
>>
>>> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS for
>>> name resolution. But the problem is still there even if I use my ISP's  DNS
>>> (mostly because its resolving to the same to IP address) .
>>>
>>> On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org>wrote:
>>>
>>>> As MikeJ pointed out please report the IP address files.freeswitch.orgresolves to.  We have that on a content delivery network so that the files
>>>> are closer to you geographically and you can download them faster but if
>>>> you're having an issue I'll need the IP so I can report it correctly.
>>>> Thanks,
>>>> Brian
>>>>
>>>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>>>
>>>> Hi FS Team,
>>>>
>>>>
>>>>         The files.freeswitch.org is resetting connection since 3 days.
>>>> As a result I was not able to download latest release of FS. Got the trunk
>>>> version from svn. But still it suffers from the lack of sound files.  When
>>>> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>>>> connection immediately wget tries 20 times and gives up. Other users on IRC
>>>> also reported this issue.
>>>>
>>>> --
>>>> Thanks & Regards,
>>>> Godson Gera
>>>>
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>> --
>>> Thanks & Regards,
>>> Godson Gera
>>> http://godson.in
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From boonedox at gmail.com  Tue Aug 11 13:13:40 2009
From: boonedox at gmail.com (Jeremiah Johnson)
Date: Tue, 11 Aug 2009 14:13:40 -0600
Subject: [Freeswitch-users] hangup_after_bridge=false only works if
	bypass_media=false
In-Reply-To: <367751820908111150s468a1717l20cf409ef8b7f66b@mail.gmail.com>
References: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
	<367751820908111150s468a1717l20cf409ef8b7f66b@mail.gmail.com>
Message-ID: <c626f0a50908111313q2844f4abn2b988c699325fc2e@mail.gmail.com>

I guess that makes sense, but that's not the answer I wanted to hear.  I
don't necessarily know when the "B" leg is going to hangup so I may not be
able to do the re-invite beforehand.  Is there some sort of workaround?

On Tue, Aug 11, 2009 at 12:50 PM, Phillip Jones <pjintheusa at gmail.com>wrote:

> Hi there,
>
> I the 'does not work' example your media stream is back up at the
> carrier as the called party hangs up.
>
> You would have to cause another SIP REINVITE and re capture the media.
>
> See http://wiki.freeswitch.org/wiki/Bypass_Media and the "How to
> disable/enable it on the fly?" section as a start.
>
>
>
>
> On Tue, Aug 11, 2009 at 2:29 PM, Jeremiah Johnson<boonedox at gmail.com>
> wrote:
> > This is an integral part of my application.  I need to have FreeSWITCH
> > outside of the media path as well as be able to do multiple bridges for
> the
> > same "A" leg.
> > /*WORKS*/
> > <action application="set" data="hangup_after_bridge=false"/>
> > <action application="bridge"
> > data="sofia/gateway/${mygateway}/1${client_one}"/>
> > <action application="bridge"
> > data="sofia/gateway/${mygateway}/1${client_two}"/>
> > /*DOES NOT WORK*/
> > <action application="set" data="hangup_after_bridge=false"/>
> > <action application="set" data="bypass_media=true"/>
> > <action application="bridge"
> > data="sofia/gateway/${mygateway}/1${client_one}"/>
> > <action application="bridge"
> > data="sofia/gateway/${mygateway}/1${client_two}"/>
> > In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the leg
> for
> > client_one hangs up.
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From boonedox at gmail.com  Tue Aug 11 13:17:47 2009
From: boonedox at gmail.com (Jeremiah Johnson)
Date: Tue, 11 Aug 2009 14:17:47 -0600
Subject: [Freeswitch-users] hangup_after_bridge=false only works if
	bypass_media=false
In-Reply-To: <c626f0a50908111313q2844f4abn2b988c699325fc2e@mail.gmail.com>
References: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
	<367751820908111150s468a1717l20cf409ef8b7f66b@mail.gmail.com>
	<c626f0a50908111313q2844f4abn2b988c699325fc2e@mail.gmail.com>
Message-ID: <c626f0a50908111317v327d4db1sb609eef4ba568da6@mail.gmail.com>

Also, if I understand bypass_media, the SIP signaling is going through
FreeSWITCH and the audio is endpoint to endpoint, so I would think the
manner in which the "B" leg hangup is handled is controlled by FreeSWITCH.
 That is what appears to be happening in the debug.

On Tue, Aug 11, 2009 at 2:13 PM, Jeremiah Johnson <boonedox at gmail.com>wrote:

> I guess that makes sense, but that's not the answer I wanted to hear.  I
> don't necessarily know when the "B" leg is going to hangup so I may not be
> able to do the re-invite beforehand.  Is there some sort of workaround?
>
>
> On Tue, Aug 11, 2009 at 12:50 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> Hi there,
>>
>> I the 'does not work' example your media stream is back up at the
>> carrier as the called party hangs up.
>>
>> You would have to cause another SIP REINVITE and re capture the media.
>>
>> See http://wiki.freeswitch.org/wiki/Bypass_Media and the "How to
>> disable/enable it on the fly?" section as a start.
>>
>>
>>
>>
>> On Tue, Aug 11, 2009 at 2:29 PM, Jeremiah Johnson<boonedox at gmail.com>
>> wrote:
>> > This is an integral part of my application.  I need to have FreeSWITCH
>> > outside of the media path as well as be able to do multiple bridges for
>> the
>> > same "A" leg.
>> > /*WORKS*/
>> > <action application="set" data="hangup_after_bridge=false"/>
>> > <action application="bridge"
>> > data="sofia/gateway/${mygateway}/1${client_one}"/>
>> > <action application="bridge"
>> > data="sofia/gateway/${mygateway}/1${client_two}"/>
>> > /*DOES NOT WORK*/
>> > <action application="set" data="hangup_after_bridge=false"/>
>> > <action application="set" data="bypass_media=true"/>
>> > <action application="bridge"
>> > data="sofia/gateway/${mygateway}/1${client_one}"/>
>> > <action application="bridge"
>> > data="sofia/gateway/${mygateway}/1${client_two}"/>
>> > In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the leg
>> for
>> > client_one hangs up.
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From gshfreesw at gmail.com  Tue Aug 11 13:27:55 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Tue, 11 Aug 2009 16:27:55 -0400
Subject: [Freeswitch-users] Call transfer issues
In-Reply-To: <55901B44-E104-4829-8B40-81B36A3522CA@freeswitch.org>
References: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>
	<D554D137-805E-4924-A43B-3930275451BE@freeswitch.org>
	<5070fcbd0908111233w6ecefeeex92b1304f04744f23@mail.gmail.com>
	<55901B44-E104-4829-8B40-81B36A3522CA@freeswitch.org>
Message-ID: <5070fcbd0908111327i641ebdd6pe6dc389c7707624f@mail.gmail.com>

OK. I am little confused here. You say I can "transfer" to a parked call
using "uuid_transfer" . I do not see any options there where would or how
would I specify parked call in the options.

Can I do:

uuid_transfer <uuid> -both "park"

Where "park" is the destination?


On Tue, Aug 11, 2009 at 3:40 PM, Brian West <brian at freeswitch.org> wrote:

>
> On Aug 11, 2009, at 2:33 PM, Shameem Shiek wrote:
>
> > Thanks for the explanation. I was trying to find the correct way to
> > do this as I saw several ways of doing this. I will try to explain
> > how I did on the asterisk land.
> >
> > In asterisk land, Caller A will be parked at extension 701 and
> > Caller B "transfers" the call to extension 701 and and Caller A and
> > C are connected and Caller B is hung up right after transfer.
> >
> > In Freeswitch, Can Caller B do a "transfer" and connect caller A and
> > C? How can I do this? Since Freeswitch does not have a concept of
> > parked extensions.
>
> This is where you're wrong... you can park people with the park
> application just transfer them to it... But you have to take off the
> asterisk hat.. I told you in the last email how to do it correctly
> using nothing but API commands.
>
> /b
>
>
> >
> > Thank you for the help.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Tue Aug 11 13:34:50 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 15:34:50 -0500
Subject: [Freeswitch-users] Call transfer issues
In-Reply-To: <5070fcbd0908111327i641ebdd6pe6dc389c7707624f@mail.gmail.com>
References: <5070fcbd0908111210k6f687324w3b6c1662e815efdb@mail.gmail.com>
	<D554D137-805E-4924-A43B-3930275451BE@freeswitch.org>
	<5070fcbd0908111233w6ecefeeex92b1304f04744f23@mail.gmail.com>
	<55901B44-E104-4829-8B40-81B36A3522CA@freeswitch.org>
	<5070fcbd0908111327i641ebdd6pe6dc389c7707624f@mail.gmail.com>
Message-ID: <5F3CDC9A-6E29-4EA3-8C8B-BB821A2D6E25@freeswitch.org>

You also have uuid_park... but uuid_transfer to the dialplan were the  
extension calls the park application.

/b

On Aug 11, 2009, at 3:27 PM, Shameem Shiek wrote:

> OK. I am little confused here. You say I can "transfer" to a parked  
> call using "uuid_transfer" . I do not see any options there where  
> would or how would I specify parked call in the options.
>
> Can I do:
>
> uuid_transfer <uuid> -both "park"
>
> Where "park" is the destination?




From mike at jerris.com  Tue Aug 11 13:45:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 Aug 2009 16:45:51 -0400
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <5a8712120908111256v4b072587r4c5ead9fa4d4afb3@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
	<4A81C59A.2020705@chandlerfamily.org.uk>
	<06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>
	<5a8712120908111249t3e23ceeese1c18d6fef52a227@mail.gmail.com>
	<87f2f3b90908111254pf443a35qa306f7da93fb336a@mail.gmail.com>
	<5a8712120908111256v4b072587r4c5ead9fa4d4afb3@mail.gmail.com>
Message-ID: <3D7EF5AC-2204-4B74-87D2-11554E8D27C3@jerris.com>

We have a plan to address this already, I can't recall if we added the  
gender types in code yet.

Mike

On Aug 11, 2009, at 3:56 PM, Jo?o Mesquita wrote:

> Oops, I thought you were saying different languages. Sorry about that.
>
> jmesquita
>
> On Tue, Aug 11, 2009 at 4:54 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> 2009/8/11 Jo?o Mesquita <jmesquita at gmail.com>
>
> Mike, the gender thing will eventually have to change code, I guess.  
> I have not yet looked at the say code, so I am just imagining here.
>
> Are there gender differences between dialects of the same language?
> -MC
>

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From asannucci at gmail.com  Tue Aug 11 14:00:14 2009
From: asannucci at gmail.com (bakko)
Date: Tue, 11 Aug 2009 23:00:14 +0200
Subject: [Freeswitch-users] Ivr and variables
In-Reply-To: <87f2f3b90908111254i4479d3f9m68ab8cce6a62554e@mail.gmail.com>
References: <8D85DC2E55A74F41AE2190E05A44F028@voztovoice>
	<87f2f3b90908111254i4479d3f9m68ab8cce6a62554e@mail.gmail.com>
Message-ID: <1C081CD9FB8642238779D6BCC8D0CCF8@voztovoice>

The value of  ${some} is 1234.

Thank you.

BR

> What value is in the variable ${some} ?
> -MC
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From kevin at kgolding.co.uk  Tue Aug 11 14:21:35 2009
From: kevin at kgolding.co.uk (Kevin Golding)
Date: Tue, 11 Aug 2009 22:21:35 +0100
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <0C9B302E-C1A6-4D1F-831B-63FA2AEB8C37@freeswitch.org>
References: <4A80AEE2.7050306@kgolding.co.uk>	<7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>	<4A811FEC.7040808@kgolding.co.uk>
	<0C9B302E-C1A6-4D1F-831B-63FA2AEB8C37@freeswitch.org>
Message-ID: <4A81E0DF.3060609@kgolding.co.uk>

Sorry, but could you give me a pointer on what this involves or where to 
read up on it please?

Brian West wrote:
> you have to now apply the ACL correctly to the sofia profile.
> 
> /b
> 
> On Aug 11, 2009, at 2:38 AM, Kevin Golding wrote:
> 
>> Thanks Brian,
>>
>> Well spotted with the domain/cidr :)
>>
>> I changed the line <param name="apply-inbound-acl" value="domains"/>  
>> in
>> the internal.xml with a new value, and changed the list to match but I
>> still get the same 'Rejected by acl "domains"' error.
>>
>> And yes I reloaded the xml. :)
>>
>> Kevin
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From brian at freeswitch.org  Tue Aug 11 14:25:36 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 11 Aug 2009 16:25:36 -0500
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <4A81E0DF.3060609@kgolding.co.uk>
References: <4A80AEE2.7050306@kgolding.co.uk>	<7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>	<4A811FEC.7040808@kgolding.co.uk>
	<0C9B302E-C1A6-4D1F-831B-63FA2AEB8C37@freeswitch.org>
	<4A81E0DF.3060609@kgolding.co.uk>
Message-ID: <09C4F1BE-A4F3-466F-A102-5F1F490201CD@freeswitch.org>

I would guess in conf/sip_profiles/internal.xml where it applies the  
domains ACL to the profile... you can change that or add additional  
lines to apply more ACL's to the profile.

/b

On Aug 11, 2009, at 4:21 PM, Kevin Golding wrote:

> Sorry, but could you give me a pointer on what this involves or  
> where to
> read up on it please?
>
> Brian West wrote:
>> you have to now apply the ACL correctly to the sofia profile.
>>
>> /b
>




From msc at freeswitch.org  Tue Aug 11 14:30:48 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 11 Aug 2009 16:30:48 -0500
Subject: [Freeswitch-users] Ivr and variables
In-Reply-To: <1C081CD9FB8642238779D6BCC8D0CCF8@voztovoice>
References: <8D85DC2E55A74F41AE2190E05A44F028@voztovoice>
	<87f2f3b90908111254i4479d3f9m68ab8cce6a62554e@mail.gmail.com>
	<1C081CD9FB8642238779D6BCC8D0CCF8@voztovoice>
Message-ID: <87f2f3b90908111430n2587cceeid8fa10e16a9bb63c@mail.gmail.com>

As Brian mentioned earlier, what you are trying to do is not possible in the
ivr.conf.xml file. Variables are not interpolated in the digits param. If
you need this dynamic functionality you will need to investigate
mod_xml_curl which allows you to create dynamic dialplans, etc.

-MC

On Tue, Aug 11, 2009 at 4:00 PM, bakko <asannucci at gmail.com> wrote:

>  The value of  ${some} is 1234.
>
> Thank you.
>
> BR
>
> > What value is in the variable ${some} ?
> > -MC
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kevin at kgolding.co.uk  Tue Aug 11 14:58:16 2009
From: kevin at kgolding.co.uk (Kevin Golding)
Date: Tue, 11 Aug 2009 22:58:16 +0100
Subject: [Freeswitch-users] ACL issue
In-Reply-To: <09C4F1BE-A4F3-466F-A102-5F1F490201CD@freeswitch.org>
References: <4A80AEE2.7050306@kgolding.co.uk>	<7345DBD4-941C-4CBB-AAA1-85057C6E1D8E@freeswitch.org>	<4A811FEC.7040808@kgolding.co.uk>	<0C9B302E-C1A6-4D1F-831B-63FA2AEB8C37@freeswitch.org>	<4A81E0DF.3060609@kgolding.co.uk>
	<09C4F1BE-A4F3-466F-A102-5F1F490201CD@freeswitch.org>
Message-ID: <4A81E978.1000107@kgolding.co.uk>

Thank you Brian. Have got the ACL issue sorted with your help - thank 
you. See new thread for my next problem :)

kevin

Brian West wrote:
> I would guess in conf/sip_profiles/internal.xml where it applies the  
> domains ACL to the profile... you can change that or add additional  
> lines to apply more ACL's to the profile.
> 
> /b
> 
> On Aug 11, 2009, at 4:21 PM, Kevin Golding wrote:
> 
>> Sorry, but could you give me a pointer on what this involves or  
>> where to
>> read up on it please?
>>
>> Brian West wrote:
>>> you have to now apply the ACL correctly to the sofia profile.
>>>
>>> /b
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From mgg at giagnocavo.net  Tue Aug 11 15:13:05 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Tue, 11 Aug 2009 18:13:05 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <1250020512.4659.16.camel@dk-d820>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110659v346b2f72mf0a09cbec20d9eb0@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>

It's also simple enough to write a plugin in one of the scripting languages to add an app to do exactly what you want...

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Knell
Sent: Tuesday, August 11, 2009 1:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Loopback and bypass_media

Just to add my $0.02-worth (if you're feeling generous..) - I don't
think that the dialplan is expressive enough to do what's needed here,
and that's where the trouble's coming from.  It's not enormously tricky
to build a generic "dial this set of numbers according to these rules"
service using something hanging off the event socket - there's a writeup
here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
be done on SoftIVR.

To roll something similar yourself using the event socket, you'd need to
map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
and have some way of passing messages around between the threads
handling the different call legs, assuming that you're using one thread
per leg.

--Dave


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From diego.viola at gmail.com  Tue Aug 11 21:56:29 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 12 Aug 2009 00:56:29 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
	<8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
	<86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>
Message-ID: <86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>

Resolving files.freeswitch.org... failed: Temporary failure in name
resolution.

Again...

On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Nope.
>
>
> On Tue, Aug 11, 2009 at 4:01 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Does it still give you connection refused?
>> /b
>>
>> On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:
>>
>> It resolves fine for me already.
>>
>> On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>
>>> The address for me is: 69.174.57.101
>>>
>>>
>>> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera <godson.g at gmail.com> wrote:
>>>
>>>> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS
>>>> for name resolution. But the problem is still there even if I use my ISP's
>>>> DNS (mostly because its resolving to the same to IP address) .
>>>>
>>>> On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org>wrote:
>>>>
>>>>> As MikeJ pointed out please report the IP address files.freeswitch.orgresolves to.  We have that on a content delivery network so that the files
>>>>> are closer to you geographically and you can download them faster but if
>>>>> you're having an issue I'll need the IP so I can report it correctly.
>>>>> Thanks,
>>>>> Brian
>>>>>
>>>>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>>>>
>>>>> Hi FS Team,
>>>>>
>>>>>
>>>>>         The files.freeswitch.org is resetting connection since 3 days.
>>>>> As a result I was not able to download latest release of FS. Got the trunk
>>>>> version from svn. But still it suffers from the lack of sound files.  When
>>>>> ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>>>>> connection immediately wget tries 20 times and gives up. Other users on IRC
>>>>> also reported this issue.
>>>>>
>>>>> --
>>>>> Thanks & Regards,
>>>>> Godson Gera
>>>>>
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>> --
>>>> Thanks & Regards,
>>>> Godson Gera
>>>> http://godson.in
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From diego.viola at gmail.com  Tue Aug 11 22:01:12 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 12 Aug 2009 01:01:12 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
	<8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
	<86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>
	<86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
Message-ID: <86a32abc0908112201n1e505d33w74d46a5570104500@mail.gmail.com>

http://files-sync.freeswitch.org/ works fine.

On Wed, Aug 12, 2009 at 12:56 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Resolving files.freeswitch.org... failed: Temporary failure in name
> resolution.
>
> Again...
>
>
> On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola <diego.viola at gmail.com>wrote:
>
>> Nope.
>>
>>
>> On Tue, Aug 11, 2009 at 4:01 PM, Brian West <brian at freeswitch.org> wrote:
>>
>>> Does it still give you connection refused?
>>> /b
>>>
>>> On Aug 11, 2009, at 2:59 PM, Diego Viola wrote:
>>>
>>> It resolves fine for me already.
>>>
>>> On Tue, Aug 11, 2009 at 2:54 AM, Diego Viola <diego.viola at gmail.com>wrote:
>>>
>>>> The address for me is: 69.174.57.101
>>>>
>>>>
>>>> On Tue, Aug 11, 2009 at 1:20 AM, Godson Gera <godson.g at gmail.com>wrote:
>>>>
>>>>> Ok here is the address 69.174.57.101 . Most of the time I use OpenDNS
>>>>> for name resolution. But the problem is still there even if I use my ISP's
>>>>> DNS (mostly because its resolving to the same to IP address) .
>>>>>
>>>>> On Mon, Aug 10, 2009 at 7:01 PM, Brian West <brian at freeswitch.org>wrote:
>>>>>
>>>>>> As MikeJ pointed out please report the IP address
>>>>>> files.freeswitch.org resolves to.  We have that on a content delivery
>>>>>> network so that the files are closer to you geographically and you can
>>>>>> download them faster but if you're having an issue I'll need the IP so I can
>>>>>> report it correctly.
>>>>>> Thanks,
>>>>>> Brian
>>>>>>
>>>>>> On Aug 10, 2009, at 12:26 AM, Godson Gera wrote:
>>>>>>
>>>>>> Hi FS Team,
>>>>>>
>>>>>>
>>>>>>         The files.freeswitch.org is resetting connection since 3
>>>>>> days. As a result I was not able to download latest release of FS. Got the
>>>>>> trunk version from svn. But still it suffers from the lack of sound files.
>>>>>> When ever I do 'make uhd-sounds-install'http://files.freeswitch.org resets
>>>>>> connection immediately wget tries 20 times and gives up. Other users on IRC
>>>>>> also reported this issue.
>>>>>>
>>>>>> --
>>>>>> Thanks & Regards,
>>>>>> Godson Gera
>>>>>>
>>>>>>
>>>>>>
>>>>>> _______________________________________________
>>>>>> FreeSWITCH-users mailing list
>>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>> UNSUBSCRIBE:
>>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>>> http://www.freeswitch.org
>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Thanks & Regards,
>>>>> Godson Gera
>>>>> http://godson.in
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From jason at jasonjgw.net  Tue Aug 11 22:30:18 2009
From: jason at jasonjgw.net (Jason White)
Date: Wed, 12 Aug 2009 15:30:18 +1000
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
	<8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
	<86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>
	<86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
Message-ID: <20090812053018.GA3821@jdc.jasonjgw.net>

Diego Viola <diego.viola at gmail.com> wrote:
> Resolving files.freeswitch.org... failed: Temporary failure in name
> resolution.

It must be a problem at your end.

jason at jdc:~$ host files.freeswitch.org
files.freeswitch.org is an alias for filessync.freeswitch.netdna-cdn.com.
filessync.freeswitch.netdna-cdn.com has address 69.174.57.101

Note that I am running my own Bind 9 daemon on this host. The record wasn't
cached, as it has been a few weeks since I last upgraded FreeSWITCH.




From brian at freeswitch.org  Tue Aug 11 22:57:38 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 00:57:38 -0500
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
	<8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
	<86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>
	<86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
Message-ID: <2A5563F4-3197-42B0-851B-ABAB804A08F9@freeswitch.org>

I would have to say its YOUR system and not ours.

/b

On Aug 11, 2009, at 11:56 PM, Diego Viola wrote:

> Resolving files.freeswitch.org... failed: Temporary failure in name  
> resolution.
>
> Again...
>
> On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> Nope.
>

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From plite2012 at gmail.com  Tue Aug 11 22:58:41 2009
From: plite2012 at gmail.com (Paul Li)
Date: Wed, 12 Aug 2009 00:58:41 -0500
Subject: [Freeswitch-users] How to delay IVR answer during an outbound call
Message-ID: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com>

I have a dummy question. Say, you have an outbound call to the demo
IVR as below:

originate sofia/gateway/myvoip/19876543210 5000

How do I delay the IVR response until the recipient at 19876543210
picks up the call? I tried "ignore_early_media=true", which had no
effect.

Many thanks in advance.



From brian at freeswitch.org  Tue Aug 11 23:01:58 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 01:01:58 -0500
Subject: [Freeswitch-users] How to delay IVR answer during an outbound
	call
In-Reply-To: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com>
References: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com>
Message-ID: <11E6F5A9-D997-473E-8880-0E5AC996D3C4@freeswitch.org>

Is your provider answering the call before its connected?  If so then  
they should be shot.  I can't imagine other way the call would be  
answered unless you're using ignore_early_media wrong.... can you show  
me who you're doing this?

/b

On Aug 12, 2009, at 12:58 AM, Paul Li wrote:

> I have a dummy question. Say, you have an outbound call to the demo
> IVR as below:
>
> originate sofia/gateway/myvoip/19876543210 5000
>
> How do I delay the IVR response until the recipient at 19876543210
> picks up the call? I tried "ignore_early_media=true", which had no
> effect.
>
> Many thanks in advance.




From plite2012 at gmail.com  Tue Aug 11 23:23:33 2009
From: plite2012 at gmail.com (Paul Li)
Date: Wed, 12 Aug 2009 01:23:33 -0500
Subject: [Freeswitch-users] How to delay IVR answer during an outbound call
In-Reply-To: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com>
References: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com>
Message-ID: <c43084ae0908112323m1c8d5856kc05acc6664b635cd@mail.gmail.com>

I am actually doing a lua script for IVR as follows

-- answer the call
session:answer();

while session:ready() == true do
	-- sleep a second
	session:sleep(1000);

	-- play a file
	session:streamFile("/path/to/blah.wav");
	
	-- hangup
	session:hangup();
end

The problem lies in: when I picked up my phone, blah.wav was already
played for a while, instead of from the beginning.

I shall greatly appreciate any input.

On Wed, Aug 12, 2009 at 12:58 AM, Paul Li<plite2012 at gmail.com> wrote:
> I have a dummy question. Say, you have an outbound call to the demo
> IVR as below:
>
> originate sofia/gateway/myvoip/19876543210 5000
>
> How do I delay the IVR response until the recipient at 19876543210
> picks up the call? I tried "ignore_early_media=true", which had no
> effect.
>
> Many thanks in advance.
>



From tzury.by at reguluslabs.com  Tue Aug 11 23:44:49 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Wed, 12 Aug 2009 09:44:49 +0300
Subject: [Freeswitch-users] problem when adding more extension
Message-ID: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>

Hi,

I wanted to add more extension to freeswitch.
to add extension 1050 with password 1234 I did the following:

$ cd /usr/local/freeswitch/conf/directory/default

created 1050.xml having all '1000' strings replaced by '1050' by typing
$ sed s/1000/1050/g < 1000.xml > 1050.xml

rescan and reload the xml by typing into the CLI

freeswitch at internal> sofia profile internal rescan reloadxml

However, when I tried to login with these credentials I got the
following in the fs_cli:

2009-08-12 06:35:01 [DEBUG] sofia_reg.c:1469 sofia_reg_parse_auth()
SIP username 1050 does not match auth username
2009-08-12 06:35:01 [DEBUG] sofia_reg.c:869
sofia_reg_handle_register() Send challenge for
[1050 at server_address.net]

below are the content of 1000 and 1050 xml files

please advise.


$ cat 1050.xml
<include>
 <user id="1050" mailbox="1050">
   <params>
     <param name="password" value="1234"/>
     <param name="vm-password" value="1050"/>
   </params>
   <variables>
     <variable name="toll_allow" value="domestic,international,local"/>
     <variable name="accountcode" value="1050"/>
     <variable name="user_context" value="default"/>
     <variable name="effective_caller_id_name" value="Extension 1050"/>
     <variable name="effective_caller_id_number" value="1050"/>
     <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
     <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
     <variable name="callgroup" value="techsupport"/>
   </variables>
 </user>
</include>

$ cat 1000.xml
<include>
 <user id="1000" mailbox="1000">
   <params>
     <param name="password" value="1234"/>
     <param name="vm-password" value="1000"/>
   </params>
   <variables>
     <variable name="toll_allow" value="domestic,international,local"/>
     <variable name="accountcode" value="1000"/>
     <variable name="user_context" value="default"/>
     <variable name="effective_caller_id_name" value="Extension 1000"/>
     <variable name="effective_caller_id_number" value="1000"/>
     <variable name="outbound_caller_id_name"
value="$${outbound_caller_name}"/>
     <variable name="outbound_caller_id_number"
value="$${outbound_caller_id}"/>
     <variable name="callgroup" value="techsupport"/>
   </variables>
 </user>
</include>



From diego.viola at gmail.com  Wed Aug 12 00:19:04 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Wed, 12 Aug 2009 03:19:04 -0400
Subject: [Freeswitch-users] files.freeswitch.org resets connection.
In-Reply-To: <2A5563F4-3197-42B0-851B-ABAB804A08F9@freeswitch.org>
References: <f9822b400908092226q7cad77efke60a08d3042199bc@mail.gmail.com>
	<80659736-4C9E-4F85-A9F6-A9F9C72CD29C@freeswitch.org>
	<f9822b400908102220l7a232efam243514560ae0a349@mail.gmail.com>
	<86a32abc0908102354i5b022a86y8d83a9f89efab161@mail.gmail.com>
	<86a32abc0908111259p44301323nc7a534e5296ed201@mail.gmail.com>
	<8734A8EB-A63A-4EC3-AFEC-0A850837E950@freeswitch.org>
	<86a32abc0908111303w201a188ep451581f818b5eaca@mail.gmail.com>
	<86a32abc0908112156r6ad9b784p735a8f0957d17a83@mail.gmail.com>
	<2A5563F4-3197-42B0-851B-ABAB804A08F9@freeswitch.org>
Message-ID: <86a32abc0908120019k5a78eb8fi603d1bf4bd082a9@mail.gmail.com>

Aww, ok.

Bad luck to me :).

On Wed, Aug 12, 2009 at 1:57 AM, Brian West <brian at freeswitch.org> wrote:

> I would have to say its YOUR system and not ours.
> /b
>
> On Aug 11, 2009, at 11:56 PM, Diego Viola wrote:
>
> Resolving files.freeswitch.org... failed: Temporary failure in name
> resolution.
>
> Again...
>
> On Tue, Aug 11, 2009 at 4:03 PM, Diego Viola <diego.viola at gmail.com>
> wrote:
>
>> Nope.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From samu60 at gmail.com  Wed Aug 12 00:25:25 2009
From: samu60 at gmail.com (samuel)
Date: Wed, 12 Aug 2009 09:25:25 +0200
Subject: [Freeswitch-users] Spanish Prompts
In-Reply-To: <87f2f3b90908111254pf443a35qa306f7da93fb336a@mail.gmail.com>
References: <ECB52329035B4CBBAA75C4C52FC1132F@voztovoice>
	<f120f4c80908071157r3fe9d4b3y25d4dde31520e6@mail.gmail.com>
	<87f2f3b90908101604i724ec534vc8e933156e787155@mail.gmail.com>
	<d18bd3a10908110049w2af177d8w379f824d23858687@mail.gmail.com>
	<1b46b4e80908110727k1b6ed974r519971971734e6f@mail.gmail.com>
	<d18bd3a10908110748y2d929bcbu65a407fb69f258f@mail.gmail.com>
	<4A81C59A.2020705@chandlerfamily.org.uk>
	<06427DB1-CA10-49FC-ACC7-E481B2662E0D@jerris.com>
	<5a8712120908111249t3e23ceeese1c18d6fef52a227@mail.gmail.com>
	<87f2f3b90908111254pf443a35qa306f7da93fb336a@mail.gmail.com>
Message-ID: <d18bd3a10908120025y203a887axb0567d7e122d8207@mail.gmail.com>

I would say there are no changes in gender for dialects...but with so many
languages around I can't assure it 100% ;)

Samuel.

2009/8/11 Michael Collins <msc at freeswitch.org>

>
>
> 2009/8/11 Jo?o Mesquita <jmesquita at gmail.com>
>
>> Mike, the gender thing will eventually have to change code, I guess. I
>> have not yet looked at the say code, so I am just imagining here.
>>
>
> Are there gender differences between dialects of the same language?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Aug 11 20:43:09 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 11 Aug 2009 23:43:09 -0400
Subject: [Freeswitch-users] hangup_after_bridge=false only works if
	bypass_media=false
In-Reply-To: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
References: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
Message-ID: <C5AD471E-636F-427A-A0A9-9F894D799CEB@jerris.com>

Please post a bug for this on jira.freeswitch.org.

Mike

On Aug 11, 2009, at 2:29 PM, Jeremiah Johnson wrote:

> This is an integral part of my application.  I need to have  
> FreeSWITCH outside of the media path as well as be able to do  
> multiple bridges for the same "A" leg.
>
> /*WORKS*/
> <action application="set" data="hangup_after_bridge=false"/>
> <action application="bridge" data="sofia/gateway/${mygateway}/1$ 
> {client_one}"/>
> <action application="bridge" data="sofia/gateway/${mygateway}/1$ 
> {client_two}"/>
>
> /*DOES NOT WORK*/
> <action application="set" data="hangup_after_bridge=false"/>
> <action application="set" data="bypass_media=true"/>
> <action application="bridge" data="sofia/gateway/${mygateway}/1$ 
> {client_one}"/>
> <action application="bridge" data="sofia/gateway/${mygateway}/1$ 
> {client_two}"/>
>
> In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the  
> leg for client_one hangs up.




From charlieb at cot.net  Tue Aug 11 23:37:03 2009
From: charlieb at cot.net (Charles Boening)
Date: Tue, 11 Aug 2009 23:37:03 -0700
Subject: [Freeswitch-users] FS 1.0.4 official core dumping LUA streamFile
Message-ID: <4FB2938B89459C41860C4DB9B1821D6FB6657A2421@exchange.calore.local>

Greetings,

I have the following LUA script (at end of email) in a fresh FS 1.0.4 install.  I originally did an upgrade from one of the 1.0.4preX versions but when I came across this issue I went fresh just to make sure there wasn't an incompatibility with my previous config.

What I'm seeing is a seg fault and a core dump after playing a sound file.  I originally had a file I recorded but when I ran into this issue I figured I'd try an included sound file but that doesn't seem to make a bit of difference.

2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Starting test.lua
2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Caller [XXXXXXXXXX] connected
2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Pre streamFile
Segmentation fault (core dumped)


Any ideas?

Thanks,
Charlie



freeswitch.consoleLog("INFO", string.format("Starting test.lua\n"))
session:answer();
session:setHangupHook("session_hangup_hook")
calleridnumber = session:getVariable("caller_id_number")
calleridname = session:getVariable("caller_id_name")

if session:ready() then
   freeswitch.consoleLog("INFO", string.format("Caller [" .. calleridnumber .. "] connected\n"))
   freeswitch.consoleLog("INFO", string.format("Pre streamFile\n"))

   session:streamFile("conference/8000/conf-welcome.wav")

   freeswitch.consoleLog("INFO", string.format("Post streamFile.\n"))
end

function session_hangup_hook(status)
  freeswitch.consoleLog("INFO", "Session hangup: \n") --[[ .. status .. "\n") ]]--
  error()
end

session:hangup()



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From kirk.bateman at gmail.com  Wed Aug 12 01:34:16 2009
From: kirk.bateman at gmail.com (Kirk Bateman)
Date: Wed, 12 Aug 2009 09:34:16 +0100
Subject: [Freeswitch-users] VoiceMail transcription
In-Reply-To: <5a8712120908110930s74b84bf8x673ca1209fbae965@mail.gmail.com>
References: <20090810115949.2ad02225396a31c9de30536f2e338977.0a55f152ab.wbe@email04.secureserver.net>
	<1249998014.20224.82.camel@dk-d820>
	<2bee4fc40908110717p43649678ue2b94ecb571bd09@mail.gmail.com>
	<5a8712120908110930s74b84bf8x673ca1209fbae965@mail.gmail.com>
Message-ID: <2bee4fc40908120134v5781a339ha7a31cc677b7f205@mail.gmail.com>

Not sure, but they do certainly have a reasonably large server farm for
doing processing :)

I note that sphinx4 I believe has a java example for doing dictation
transcription from an audio file (saw something on a sphinx forum or mailing
list while trawling the net).

I'm still investigating modifications to use pocketsphinx.

Regards

Kirk


2009/8/11 Jo?o Mesquita <jmesquita at gmail.com>

> I am sorry for the ignorance on the matter, but how does google voice does?
> Do they also have humans?
>
> jmesquita
>
>
> On Tue, Aug 11, 2009 at 11:17 AM, Kirk Bateman <kirk.bateman at gmail.com>wrote:
>
>> I'm still interested in getting pocketsphinx to attempt speech recognition
>> on an audio file.
>>
>> To be honest, most of the problem is that at 8Khz (mobile phone call
>> rate), speech detection is NOT very accurate, at 16Khz it IS significantly
>> better.
>>
>> I'm planning to have a play with the speechtools module and
>> mod_pocketsphinx etc to try and get an audio file parsed, spare time
>> permitting.
>>
>> Will let the list know if I get anywhere.
>>
>> Regards
>>
>> Kirk Bateman
>>
>>
>> 2009/8/11 David Knell <dave at 3c.co.uk>
>>
>> Hi Pete,
>>>
>>> I'm afraid that the answer's still the same: use a human.  Here's an
>>> article describing the state of the art:
>>> http://www.theregister.co.uk/2009/08/05/spinvox_demo_day/
>>> - the links to previous stories at the bottom provide good background.
>>>
>>> --Dave
>>>
>>> > I apologize, I should have been more clear.  We will be using humans
>>> > to scan the translated results.  But we are looking for a system to
>>> > perform the "first pass" on the audio to hopefully help the human type
>>> > less.
>>> >
>>> > Although the question has been raised if it's faster to have a human
>>> > just transcribe the whole thing, or fix up what the computer spit out.
>>> > If you have any insights on this, that would be great.
>>> >
>>> > -pete
>>> >
>>> >         -------- Original Message --------
>>> >         Subject: Re: [Freeswitch-users] VoiceMail transcription
>>> >         From: David Knell <dave at 3c.co.uk>
>>> >         Date: Mon, August 10, 2009 11:51 am
>>> >         To: freeswitch-users at lists.freeswitch.org
>>> >
>>> >         Good evening Pete,
>>> >
>>> >         The only way to do this is, I'm afraid, to use a human. We use
>>> >         Amazon's
>>> >         Mechanical Turk to good effect.
>>> >
>>> >         Cheers --
>>> >
>>> >         Dave
>>> >
>>> >         > Good morning all,
>>> >         >
>>> >         > I realize this is slightly off the FS topic, but I am
>>> >         wondering if
>>> >         > anyone out there has experience with software packages
>>> >         designed for
>>> >         > the transcription of voicemails to text. I've used
>>> >         pocketsphinx with
>>> >         > FS to handle IVR menus, but now have the task of figuring
>>> >         out how to
>>> >         > convert recorded phone conversations (voicemails mostly) to
>>> >         text.
>>> >         >
>>> >         > This does not have to be a real-time process, I can store
>>> >         the audio
>>> >         > files and process them over time. This would need to be a
>>> >         software
>>> >         > (preferable open source) solution. ASPs like VoiceCloud
>>> >         would not
>>> >         > work for this application.
>>> >         >
>>> >         > Thanks for any help
>>> >         > -pete
>>> >         > _______________________________________________
>>> >         > FreeSWITCH-users mailing list
>>> >         > FreeSWITCH-users at lists.freeswitch.org
>>> >         >
>>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >         >
>>> >         UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >         > http://www.freeswitch.org
>>> >         --
>>> >         David Knell, Director, 3C Limited
>>> >         T: +44 20 3298 2000
>>> >         E: dave at 3c.co.uk
>>> >         W: http://www.3c.co.uk
>>> >
>>> >
>>> >         _______________________________________________
>>> >         FreeSWITCH-users mailing list
>>> >         FreeSWITCH-users at lists.freeswitch.org
>>> >         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >         UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >         http://www.freeswitch.org
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> --
>>> David Knell, Director, 3C Limited
>>> T: +44 20 3298 2000
>>> E: dave at 3c.co.uk
>>> W: http://www.3c.co.uk
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bruce.mcalister at blueface.ie  Wed Aug 12 01:43:36 2009
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Wed, 12 Aug 2009 09:43:36 +0100
Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 &
	v1.0.4)
Message-ID: <4A8280B8.6050308@blueface.ie>

Hi All,

I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and 
1.0.4.

I am running on Solaris 10 Update 5 on x86 hardware (32-bit).

The build fails with:

--- snip ---
make: Fatal error: Command failed for target `all-recursive'
Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4
*** Error code 1
make: Fatal error: Command failed for target `all'
---

Looking back through the build I can see the following error:

--- snip ---
creating libfreeswitch.la
(cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la 
libfreeswitch.la)
/usr/bin/cc 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
-KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
-DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch 
freeswitch-switch.o  ./.libs/libfreeswitch.so 
-L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib 
/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib/.libs/libexpat.a 
/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libapr-1.a 
-lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp 
-L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt 
-lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl 
-lcurses -lsocket  -R/opt/freeswitch/lib -R/usr/sfw/lib
Undefined                       first referenced
  symbol                             in file
herror                              ./.libs/libfreeswitch.so
ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch
*** Error code 1
The following command caused the error:
`if test -z "" ; then echo /bin/bash 
/export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else 
echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` 
--tag=CC --mode=link /usr/bin/cc 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
-KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
-DPATH_MAX=2048 -g -v -Xc -xc99=all  -lm -R/opt/freeswitch/lib  -o 
freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib 
freeswitch-switch.o   libfreeswitch.la libs/apr/libapr-1.la 
libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl 
-lcrypto -lsocket -lnsl -ldl   -lcurses -lsocket
--- snip ---

Then a little above this error, there is the following warning that is 
displayed (I'm not sure if it is related):

--- snip ---
*** Warning: Linking the shared library libfreeswitch.la against the
*** static library libs/libedit/src/.libs/libedit.a is not portable!
--- snip ---

My configure line is as follows:

---
./configure --prefix=/opt/freeswitch
---

I have the complete configure and make output if anyone needs them.

Any help/pointers would be greatly appreciated.

Thanks
Bruce



From kevin at kgolding.co.uk  Wed Aug 12 01:47:20 2009
From: kevin at kgolding.co.uk (Kevin Golding)
Date: Wed, 12 Aug 2009 09:47:20 +0100
Subject: [Freeswitch-users] problem when adding more extension
In-Reply-To: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>
References: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>
Message-ID: <4A828198.6020000@kgolding.co.uk>

Hello,

I've just had the same problem. Solved it by adding the new extension to 
the default group.

i.e. In the /usr/local/freeswitch/conf/directory/default.xml file you 
need to add <user id="1050" type="pointer"> with one of the "group" 
blocks (e.g. after the line <group name="default">

Kevin

Tzury Bar Yochay wrote:
> Hi,
> 
> I wanted to add more extension to freeswitch.
> to add extension 1050 with password 1234 I did the following:
> 
> $ cd /usr/local/freeswitch/conf/directory/default
> 
> created 1050.xml having all '1000' strings replaced by '1050' by typing
> $ sed s/1000/1050/g < 1000.xml > 1050.xml
> 
> rescan and reload the xml by typing into the CLI
> 
> freeswitch at internal> sofia profile internal rescan reloadxml
> 
> However, when I tried to login with these credentials I got the
> following in the fs_cli:
> 
> 2009-08-12 06:35:01 [DEBUG] sofia_reg.c:1469 sofia_reg_parse_auth()
> SIP username 1050 does not match auth username
> 2009-08-12 06:35:01 [DEBUG] sofia_reg.c:869
> sofia_reg_handle_register() Send challenge for
> [1050 at server_address.net]
> 
> below are the content of 1000 and 1050 xml files
> 
> please advise.
> 
> 
> $ cat 1050.xml
> <include>
>  <user id="1050" mailbox="1050">
>    <params>
>      <param name="password" value="1234"/>
>      <param name="vm-password" value="1050"/>
>    </params>
>    <variables>
>      <variable name="toll_allow" value="domestic,international,local"/>
>      <variable name="accountcode" value="1050"/>
>      <variable name="user_context" value="default"/>
>      <variable name="effective_caller_id_name" value="Extension 1050"/>
>      <variable name="effective_caller_id_number" value="1050"/>
>      <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>      <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>      <variable name="callgroup" value="techsupport"/>
>    </variables>
>  </user>
> </include>
> 
> $ cat 1000.xml
> <include>
>  <user id="1000" mailbox="1000">
>    <params>
>      <param name="password" value="1234"/>
>      <param name="vm-password" value="1000"/>
>    </params>
>    <variables>
>      <variable name="toll_allow" value="domestic,international,local"/>
>      <variable name="accountcode" value="1000"/>
>      <variable name="user_context" value="default"/>
>      <variable name="effective_caller_id_name" value="Extension 1000"/>
>      <variable name="effective_caller_id_number" value="1000"/>
>      <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>      <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>      <variable name="callgroup" value="techsupport"/>
>    </variables>
>  </user>
> </include>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From maxim.tsvetov at gmail.com  Wed Aug 12 01:54:38 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Wed, 12 Aug 2009 01:54:38 -0700 (PDT)
Subject: [Freeswitch-users] answer command
In-Reply-To: <86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
Message-ID: <24931876.post@talk.nabble.com>


I've tried to use "answer" command from outbound event socket and it's
working, but 
the problem is that FS answering the call, but SIP Client (we tried this
with EyeBeam and CISCO 7960)
doesn't know that call was answered. So, as long as FS doesn't know what to
do with this number it then disconnects the call.

2009-08-12 11:25:07.599250 [NOTICE] mod_dptools.c:649 Channel
[sofia/internal/sip:1000 at 10.107.181.160:42840] has been answered
2009-08-12 11:25:07.599250 [NOTICE] switch_ivr_originate.c:2015 Channel
[sofia/internal/1003 at 10.107.249.12] has been answered
2009-08-12 11:25:07.614875 [ERR] switch_core_io.c:118
sofia/internal/sip:1000 at 10.107.181.160:42840 has no read codec.
2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:503 Hangup
sofia/internal/sip:1000 at 10.107.181.160:42840 [CS_EXCHANGE_MEDIA]
[NORMAL_CLEARING]
2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:1016 Hangup
sofia/internal/1003 at 10.107.249.12 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-12 11:25:07.630500 [NOTICE] switch_core_session.c:1086 Session 133
(sofia/internal/sip:1000 at 10.107.181.160:42840) Ended


Maybe there is the way to acknowledge SIP client that call was answered?

Regards,
Maxim Tsvetov

Diego Viola wrote:
> 
> I suggest that you learn the differences between mod_commands commands and
> mod_dptools applications, and also the interfaces where you can access and
> use them.
> 
> As said before, mod_dptools is accessible from dialplan, event socket
> outbound, etc. and mod_commands is accessible from the CLI, event socket
> (inbound/outbound), XML RPC, etc.
> 
> That's all described in the wiki I think.
> 
> Let us know if you have any questions =D.
> 
> On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola <diego.viola at gmail.com>
> wrote:
> 
>> Michael, you're welcome :).
>>
>> Milena, answer is a mod_dptools command, you can use it from the XML
>> dialplan or from the event socket outbound. mod_commands API are APIs
>> that
>> you execute from the socket, event socket inbound, etc. But you can also
>> execute them from event socket outbound using the "api" command.
>>
>> I hope that makes sense, correct me if I'm wrong =D.
>>
>> On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins
>> <msc at freeswitch.org>wrote:
>>
>>>
>>>
>>> On Tue, Aug 11, 2009 at 9:05 AM, Milena <testeador01 at gmail.com> wrote:
>>>
>>>>
>>>> Hello Brian,
>>>>
>>>> I wanna fix the wiki, but to make sure i got it right, does it only
>>>> work
>>>> on outbound event socket? or is there any other scenario where it would
>>>> work.
>>>>
>>>
>>> FYI, Diego Viola fixed the wiki. (Thanks Diego!)
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/answer-command-tp24912812p24931876.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From mike at jerris.com  Wed Aug 12 02:11:40 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 12 Aug 2009 05:11:40 -0400
Subject: [Freeswitch-users] FS 1.0.4 official core dumping LUA streamFile
In-Reply-To: <4FB2938B89459C41860C4DB9B1821D6FB6657A2421@exchange.calore.local>
References: <4FB2938B89459C41860C4DB9B1821D6FB6657A2421@exchange.calore.local>
Message-ID: <A9CD3A3C-923A-47FE-B9BC-DEAE701F3248@jerris.com>

If your seeing a segfault, please report it to jira.freeswitch.org  
with a backtrace and details of how to reproduce.

Mike

On Aug 12, 2009, at 2:37 AM, Charles Boening wrote:

> Greetings,
>
> I have the following LUA script (at end of email) in a fresh FS  
> 1.0.4 install.  I originally did an upgrade from one of the  
> 1.0.4preX versions but when I came across this issue I went fresh  
> just to make sure there wasn?t an incompatibility with my previous  
> config.
>
> What I?m seeing is a seg fault and a core dump after playing a sound  
> file.  I originally had a file I recorded but when I ran into this  
> issue I figured I?d try an included sound file but that doesn?t seem  
> to make a bit of difference.
>
> 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Starting  
> test.lua
> 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Caller  
> [XXXXXXXXXX] connected
> 2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Pre streamFile
> Segmentation fault (core dumped)
>
>
> Any ideas?
>
> Thanks,
> Charlie
>
>
>
> freeswitch.consoleLog("INFO", string.format("Starting test.lua\n"))
> session:answer();
> session:setHangupHook("session_hangup_hook")
> calleridnumber = session:getVariable("caller_id_number")
> calleridname = session:getVariable("caller_id_name")
>
> if session:ready() then
>    freeswitch.consoleLog("INFO", string.format("Caller [" ..  
> calleridnumber .. "] connected\n"))
>    freeswitch.consoleLog("INFO", string.format("Pre streamFile\n"))
>
>    session:streamFile("conference/8000/conf-welcome.wav")
>
>    freeswitch.consoleLog("INFO", string.format("Post streamFile.\n"))
> end
>
> function session_hangup_hook(status)
>   freeswitch.consoleLog("INFO", "Session hangup: \n") --[[ ..  
> status .. "\n") ]]--
>   error()
> end
>
> session:hangup()
>
-------------- next part --------------
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From tzury.by at reguluslabs.com  Wed Aug 12 02:50:52 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Wed, 12 Aug 2009 12:50:52 +0300
Subject: [Freeswitch-users] problem when adding more extension
In-Reply-To: <4A828198.6020000@kgolding.co.uk>
References: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>
	<4A828198.6020000@kgolding.co.uk>
Message-ID: <10128ef10908120250y77917460g5c6b9cd1991fe074@mail.gmail.com>

still not working, I mean, I can initiate a call from 1060 to 1000 but
not from 1000 to 1060.
1060 is just an example. This applies to all new extension I have
added (beyond to the default 1000-1019).
as you can see below I added them all to <group name="support">

This is how the confs look like

/usr/local/freeswitch/conf/directory# cat default.xml
<include>
  <!--the domain or ip (the right hand side of the @ in the addr-->
  <domain name="$${domain}">
    <params>
      <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
    </params>

    <variables>
      <variable name="record_stereo" value="true"/>
      <variable name="default_gateway" value="$${default_provider}"/>
      <variable name="default_areacode" value="$${default_areacode}"/>
      <variable name="transfer_fallback_extension" value="operator"/>
    </variables>

    <groups>
      <group name="default">
	<users>
	  <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
	</users>
      </group>

      <group name="sales">
	<users>
	  <!--
	      type="pointer" is a pointer so you can have the
	      same user in multiple groups.  It basically means
	      to keep searching for the user in the directory.
	  -->
	  <user id="1000" type="pointer"/>
	  <user id="1001" type="pointer"/>
	  <user id="1002" type="pointer"/>
	  <user id="1003" type="pointer"/>
	  <user id="1004" type="pointer"/>
	</users>
      </group>

      <group name="billing">
	<users>
	  <user id="1005" type="pointer"/>
	  <user id="1006" type="pointer"/>
	  <user id="1007" type="pointer"/>
	  <user id="1008" type="pointer"/>
	  <user id="1009" type="pointer"/>
	</users>
      </group>

      <group name="support">
	<users>
	  <user id="1010" type="pointer"/>
	  <user id="1011" type="pointer"/>
	  <user id="1012" type="pointer"/>
	  <user id="1013" type="pointer"/>
          <user id="1014" type="pointer"/>
          <user id="1015" type="pointer"/>
          <user id="1016" type="pointer"/>
          <user id="1017" type="pointer"/>
          <user id="1018" type="pointer"/>
          <user id="1019" type="pointer"/>
          <user id="1020" type="pointer"/>
          <user id="1050" type="pointer"/>
          <user id="1051" type="pointer"/>
	  <user id="1052" type="pointer"/>
	  <user id="1053" type="pointer"/>
	  <user id="1054" type="pointer"/>
          <user id="1055" type="pointer"/>
          <user id="1056" type="pointer"/>
          <user id="1057" type="pointer"/>
          <user id="1058" type="pointer"/>
          <user id="1059" type="pointer"/>
          <user id="1060" type="pointer"/>
          <user id="1061" type="pointer"/>
          <user id="1062" type="pointer"/>
          <user id="1063" type="pointer"/>

	</users>
      </group>
    </groups>

  </domain>
</include>

and the xml files under directory/default

root at snoip-srv-001:/usr/local/freeswitch/conf/directory/default# ls -l
total 164
-rw-r--r-- 1 root root  750 2009-07-21 19:44 1000.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1001.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1002.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1003.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1004.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1005.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1006.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1007.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1008.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1009.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1010.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1011.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1012.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1013.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1014.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1015.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1016.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1017.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1018.xml
-rw-r--r-- 1 root root  750 2009-07-20 23:47 1019.xml
-rw-r--r-- 1 root root  750 2009-08-12 06:49 1020.xml
-rw-r--r-- 1 root root  750 2009-07-27 13:14 1050.xml
-rw-r--r-- 1 root root  750 2009-07-27 13:14 1051.xml
-rw-r--r-- 1 root root  750 2009-07-27 13:14 1052.xml
-rw-r--r-- 1 root root  750 2009-08-06 08:59 1053.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:56 1054.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:56 1055.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:56 1056.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:56 1057.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:57 1058.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:57 1059.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:57 1060.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:57 1061.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:57 1062.xml
-rw-r--r-- 1 root root  750 2009-08-11 09:58 1063.xml
-rw-r--r-- 1 root root  750 2009-08-11 10:10 1064.xml
-rw-r--r-- 1 root root  750 2009-08-11 10:12 1065.xml
-rw-r--r-- 1 root root 5029 2009-07-20 23:47 brian.xml
-rw-r--r-- 1 root root  526 2009-07-20 23:47 default.xml
-rw-r--r-- 1 root root  905 2009-07-20 23:47 example.com.xml



From kevin at kgolding.co.uk  Wed Aug 12 03:31:01 2009
From: kevin at kgolding.co.uk (Kevin Golding)
Date: Wed, 12 Aug 2009 11:31:01 +0100
Subject: [Freeswitch-users] problem when adding more extension
In-Reply-To: <10128ef10908120250y77917460g5c6b9cd1991fe074@mail.gmail.com>
References: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>	<4A828198.6020000@kgolding.co.uk>
	<10128ef10908120250y77917460g5c6b9cd1991fe074@mail.gmail.com>
Message-ID: <4A8299E5.9050207@kgolding.co.uk>

Edit the line below as shown (in the dialplan/default.xml file)

Original line (about line 206)
<condition field="desination_number" expression="^(10[01][0-9])$">

Replacement line
<condition field="desination_number" expression="^(10[0-9][0-9])$">

This will allow extensions number 1000 to 1099.

Kevin

Tzury Bar Yochay wrote:
> still not working, I mean, I can initiate a call from 1060 to 1000 but
> not from 1000 to 1060.
> 1060 is just an example. This applies to all new extension I have
> added (beyond to the default 1000-1019).
> as you can see below I added them all to <group name="support">
> 
> This is how the confs look like
> 
> /usr/local/freeswitch/conf/directory# cat default.xml
> <include>
>   <!--the domain or ip (the right hand side of the @ in the addr-->
>   <domain name="$${domain}">
>     <params>
>       <param name="dial-string"
> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>     </params>
> 
>     <variables>
>       <variable name="record_stereo" value="true"/>
>       <variable name="default_gateway" value="$${default_provider}"/>
>       <variable name="default_areacode" value="$${default_areacode}"/>
>       <variable name="transfer_fallback_extension" value="operator"/>
>     </variables>
> 
>     <groups>
>       <group name="default">
> 	<users>
> 	  <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
> 	</users>
>       </group>
> 
>       <group name="sales">
> 	<users>
> 	  <!--
> 	      type="pointer" is a pointer so you can have the
> 	      same user in multiple groups.  It basically means
> 	      to keep searching for the user in the directory.
> 	  -->
> 	  <user id="1000" type="pointer"/>
> 	  <user id="1001" type="pointer"/>
> 	  <user id="1002" type="pointer"/>
> 	  <user id="1003" type="pointer"/>
> 	  <user id="1004" type="pointer"/>
> 	</users>
>       </group>
> 
>       <group name="billing">
> 	<users>
> 	  <user id="1005" type="pointer"/>
> 	  <user id="1006" type="pointer"/>
> 	  <user id="1007" type="pointer"/>
> 	  <user id="1008" type="pointer"/>
> 	  <user id="1009" type="pointer"/>
> 	</users>
>       </group>
> 
>       <group name="support">
> 	<users>
> 	  <user id="1010" type="pointer"/>
> 	  <user id="1011" type="pointer"/>
> 	  <user id="1012" type="pointer"/>
> 	  <user id="1013" type="pointer"/>
>           <user id="1014" type="pointer"/>
>           <user id="1015" type="pointer"/>
>           <user id="1016" type="pointer"/>
>           <user id="1017" type="pointer"/>
>           <user id="1018" type="pointer"/>
>           <user id="1019" type="pointer"/>
>           <user id="1020" type="pointer"/>
>           <user id="1050" type="pointer"/>
>           <user id="1051" type="pointer"/>
> 	  <user id="1052" type="pointer"/>
> 	  <user id="1053" type="pointer"/>
> 	  <user id="1054" type="pointer"/>
>           <user id="1055" type="pointer"/>
>           <user id="1056" type="pointer"/>
>           <user id="1057" type="pointer"/>
>           <user id="1058" type="pointer"/>
>           <user id="1059" type="pointer"/>
>           <user id="1060" type="pointer"/>
>           <user id="1061" type="pointer"/>
>           <user id="1062" type="pointer"/>
>           <user id="1063" type="pointer"/>
> 
> 	</users>
>       </group>
>     </groups>
> 
>   </domain>
> </include>
> 
> and the xml files under directory/default
> 
> root at snoip-srv-001:/usr/local/freeswitch/conf/directory/default# ls -l
> total 164
> -rw-r--r-- 1 root root  750 2009-07-21 19:44 1000.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1001.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1002.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1003.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1004.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1005.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1006.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1007.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1008.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1009.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1010.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1011.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1012.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1013.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1014.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1015.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1016.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1017.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1018.xml
> -rw-r--r-- 1 root root  750 2009-07-20 23:47 1019.xml
> -rw-r--r-- 1 root root  750 2009-08-12 06:49 1020.xml
> -rw-r--r-- 1 root root  750 2009-07-27 13:14 1050.xml
> -rw-r--r-- 1 root root  750 2009-07-27 13:14 1051.xml
> -rw-r--r-- 1 root root  750 2009-07-27 13:14 1052.xml
> -rw-r--r-- 1 root root  750 2009-08-06 08:59 1053.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:56 1054.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:56 1055.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:56 1056.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:56 1057.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:57 1058.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:57 1059.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:57 1060.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:57 1061.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:57 1062.xml
> -rw-r--r-- 1 root root  750 2009-08-11 09:58 1063.xml
> -rw-r--r-- 1 root root  750 2009-08-11 10:10 1064.xml
> -rw-r--r-- 1 root root  750 2009-08-11 10:12 1065.xml
> -rw-r--r-- 1 root root 5029 2009-07-20 23:47 brian.xml
> -rw-r--r-- 1 root root  526 2009-07-20 23:47 default.xml
> -rw-r--r-- 1 root root  905 2009-07-20 23:47 example.com.xml
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From tzury.by at reguluslabs.com  Wed Aug 12 03:48:09 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Wed, 12 Aug 2009 13:48:09 +0300
Subject: [Freeswitch-users] problem when adding more extension
In-Reply-To: <4A8299E5.9050207@kgolding.co.uk>
References: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>
	<4A828198.6020000@kgolding.co.uk>
	<10128ef10908120250y77917460g5c6b9cd1991fe074@mail.gmail.com>
	<4A8299E5.9050207@kgolding.co.uk>
Message-ID: <10128ef10908120348m645994b6k5935f17010cbc79c@mail.gmail.com>

Thanks allot Kevin.
I felt it is about a missing configuration parameter


On Wed, Aug 12, 2009 at 1:31 PM, Kevin Golding<kevin at kgolding.co.uk> wrote:
> Edit the line below as shown (in the dialplan/default.xml file)
>
> Original line (about line 206)
> <condition field="desination_number" expression="^(10[01][0-9])$">
>
> Replacement line
> <condition field="desination_number" expression="^(10[0-9][0-9])$">
>
> This will allow extensions number 1000 to 1099.
>
> Kevin
>
> Tzury Bar Yochay wrote:
>> still not working, I mean, I can initiate a call from 1060 to 1000 but
>> not from 1000 to 1060.
>> 1060 is just an example. This applies to all new extension I have
>> added (beyond to the default 1000-1019).
>> as you can see below I added them all to <group name="support">
>>
>> This is how the confs look like
>>
>> /usr/local/freeswitch/conf/directory# cat default.xml
>> <include>
>> ? <!--the domain or ip (the right hand side of the @ in the addr-->
>> ? <domain name="$${domain}">
>> ? ? <params>
>> ? ? ? <param name="dial-string"
>> value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
>> ? ? </params>
>>
>> ? ? <variables>
>> ? ? ? <variable name="record_stereo" value="true"/>
>> ? ? ? <variable name="default_gateway" value="$${default_provider}"/>
>> ? ? ? <variable name="default_areacode" value="$${default_areacode}"/>
>> ? ? ? <variable name="transfer_fallback_extension" value="operator"/>
>> ? ? </variables>
>>
>> ? ? <groups>
>> ? ? ? <group name="default">
>> ? ? ? <users>
>> ? ? ? ? <X-PRE-PROCESS cmd="include" data="default/*.xml"/>
>> ? ? ? </users>
>> ? ? ? </group>
>>
>> ? ? ? <group name="sales">
>> ? ? ? <users>
>> ? ? ? ? <!--
>> ? ? ? ? ? ? type="pointer" is a pointer so you can have the
>> ? ? ? ? ? ? same user in multiple groups. ?It basically means
>> ? ? ? ? ? ? to keep searching for the user in the directory.
>> ? ? ? ? -->
>> ? ? ? ? <user id="1000" type="pointer"/>
>> ? ? ? ? <user id="1001" type="pointer"/>
>> ? ? ? ? <user id="1002" type="pointer"/>
>> ? ? ? ? <user id="1003" type="pointer"/>
>> ? ? ? ? <user id="1004" type="pointer"/>
>> ? ? ? </users>
>> ? ? ? </group>
>>
>> ? ? ? <group name="billing">
>> ? ? ? <users>
>> ? ? ? ? <user id="1005" type="pointer"/>
>> ? ? ? ? <user id="1006" type="pointer"/>
>> ? ? ? ? <user id="1007" type="pointer"/>
>> ? ? ? ? <user id="1008" type="pointer"/>
>> ? ? ? ? <user id="1009" type="pointer"/>
>> ? ? ? </users>
>> ? ? ? </group>
>>
>> ? ? ? <group name="support">
>> ? ? ? <users>
>> ? ? ? ? <user id="1010" type="pointer"/>
>> ? ? ? ? <user id="1011" type="pointer"/>
>> ? ? ? ? <user id="1012" type="pointer"/>
>> ? ? ? ? <user id="1013" type="pointer"/>
>> ? ? ? ? ? <user id="1014" type="pointer"/>
>> ? ? ? ? ? <user id="1015" type="pointer"/>
>> ? ? ? ? ? <user id="1016" type="pointer"/>
>> ? ? ? ? ? <user id="1017" type="pointer"/>
>> ? ? ? ? ? <user id="1018" type="pointer"/>
>> ? ? ? ? ? <user id="1019" type="pointer"/>
>> ? ? ? ? ? <user id="1020" type="pointer"/>
>> ? ? ? ? ? <user id="1050" type="pointer"/>
>> ? ? ? ? ? <user id="1051" type="pointer"/>
>> ? ? ? ? <user id="1052" type="pointer"/>
>> ? ? ? ? <user id="1053" type="pointer"/>
>> ? ? ? ? <user id="1054" type="pointer"/>
>> ? ? ? ? ? <user id="1055" type="pointer"/>
>> ? ? ? ? ? <user id="1056" type="pointer"/>
>> ? ? ? ? ? <user id="1057" type="pointer"/>
>> ? ? ? ? ? <user id="1058" type="pointer"/>
>> ? ? ? ? ? <user id="1059" type="pointer"/>
>> ? ? ? ? ? <user id="1060" type="pointer"/>
>> ? ? ? ? ? <user id="1061" type="pointer"/>
>> ? ? ? ? ? <user id="1062" type="pointer"/>
>> ? ? ? ? ? <user id="1063" type="pointer"/>
>>
>> ? ? ? </users>
>> ? ? ? </group>
>> ? ? </groups>
>>
>> ? </domain>
>> </include>
>>
>> and the xml files under directory/default
>>
>> root at snoip-srv-001:/usr/local/freeswitch/conf/directory/default# ls -l
>> total 164
>> -rw-r--r-- 1 root root ?750 2009-07-21 19:44 1000.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1001.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1002.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1003.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1004.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1005.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1006.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1007.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1008.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1009.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1010.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1011.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1012.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1013.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1014.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1015.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1016.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1017.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1018.xml
>> -rw-r--r-- 1 root root ?750 2009-07-20 23:47 1019.xml
>> -rw-r--r-- 1 root root ?750 2009-08-12 06:49 1020.xml
>> -rw-r--r-- 1 root root ?750 2009-07-27 13:14 1050.xml
>> -rw-r--r-- 1 root root ?750 2009-07-27 13:14 1051.xml
>> -rw-r--r-- 1 root root ?750 2009-07-27 13:14 1052.xml
>> -rw-r--r-- 1 root root ?750 2009-08-06 08:59 1053.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:56 1054.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:56 1055.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:56 1056.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:56 1057.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:57 1058.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:57 1059.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:57 1060.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:57 1061.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:57 1062.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 09:58 1063.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 10:10 1064.xml
>> -rw-r--r-- 1 root root ?750 2009-08-11 10:12 1065.xml
>> -rw-r--r-- 1 root root 5029 2009-07-20 23:47 brian.xml
>> -rw-r--r-- 1 root root ?526 2009-07-20 23:47 default.xml
>> -rw-r--r-- 1 root root ?905 2009-07-20 23:47 example.com.xml
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Tzury Bar Yochay

Regulus Labs ltd.
http://reguluslabs.com
+972 52 5133399



From dujinfang at gmail.com  Wed Aug 12 04:17:21 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 12 Aug 2009 19:17:21 +0800
Subject: [Freeswitch-users] answer command
In-Reply-To: <24931876.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
Message-ID: <B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>

It's not Eyebeam but FS hung up the call because it have nothing to do  
after answer.

You should either playback a sound, do the echo command, record, hold  
the call, bridge to another channel or transfer somewhere else.....

On Aug 12, 2009, at 4:54 PM, Maxim Tsvetov wrote:
>
> I've tried to use "answer" command from outbound event socket and it's
> working, but
> the problem is that FS answering the call, but SIP Client (we tried  
> this
> with EyeBeam and CISCO 7960)
> doesn't know that call was answered. So, as long as FS doesn't know  
> what to
> do with this number it then disconnects the call.
>
> 2009-08-12 11:25:07.599250 [NOTICE] mod_dptools.c:649 Channel
> [sofia/internal/sip:1000 at 10.107.181.160:42840] has been answered
> 2009-08-12 11:25:07.599250 [NOTICE] switch_ivr_originate.c:2015  
> Channel
> [sofia/internal/1003 at 10.107.249.12] has been answered
> 2009-08-12 11:25:07.614875 [ERR] switch_core_io.c:118
> sofia/internal/sip:1000 at 10.107.181.160:42840 has no read codec.
> 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:503 Hangup
> sofia/internal/sip:1000 at 10.107.181.160:42840 [CS_EXCHANGE_MEDIA]
> [NORMAL_CLEARING]
> 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:1016 Hangup
> sofia/internal/1003 at 10.107.249.12 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-08-12 11:25:07.630500 [NOTICE] switch_core_session.c:1086  
> Session 133
> (sofia/internal/sip:1000 at 10.107.181.160:42840) Ended
>
>
> Maybe there is the way to acknowledge SIP client that call was  
> answered?
>
> Regards,
> Maxim Tsvetov
>
> Diego Viola wrote:
>>
>> I suggest that you learn the differences between mod_commands  
>> commands and
>> mod_dptools applications, and also the interfaces where you can  
>> access and
>> use them.
>>
>> As said before, mod_dptools is accessible from dialplan, event socket
>> outbound, etc. and mod_commands is accessible from the CLI, event  
>> socket
>> (inbound/outbound), XML RPC, etc.
>>
>> That's all described in the wiki I think.
>>
>> Let us know if you have any questions =D.
>>
>> On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola <diego.viola at gmail.com>
>> wrote:
>>
>>> Michael, you're welcome :).
>>>
>>> Milena, answer is a mod_dptools command, you can use it from the XML
>>> dialplan or from the event socket outbound. mod_commands API are  
>>> APIs
>>> that
>>> you execute from the socket, event socket inbound, etc. But you  
>>> can also
>>> execute them from event socket outbound using the "api" command.
>>>
>>> I hope that makes sense, correct me if I'm wrong =D.
>>>
>>> On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins
>>> <msc at freeswitch.org>wrote:
>>>
>>>>
>>>>
>>>> On Tue, Aug 11, 2009 at 9:05 AM, Milena <testeador01 at gmail.com>  
>>>> wrote:
>>>>
>>>>>
>>>>> Hello Brian,
>>>>>
>>>>> I wanna fix the wiki, but to make sure i got it right, does it  
>>>>> only
>>>>> work
>>>>> on outbound event socket? or is there any other scenario where  
>>>>> it would
>>>>> work.
>>>>>
>>>>
>>>> FYI, Diego Viola fixed the wiki. (Thanks Diego!)
>>>> -MC
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://www.nabble.com/answer-command-tp24912812p24931876.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From dujinfang at gmail.com  Wed Aug 12 04:20:31 2009
From: dujinfang at gmail.com (Seven Du)
Date: Wed, 12 Aug 2009 19:20:31 +0800
Subject: [Freeswitch-users] problem when adding more extension
In-Reply-To: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>
References: <10128ef10908112344s519337d3ja35b7f69ba33f870@mail.gmail.com>
Message-ID: <93FED77F-903E-4B38-BD33-D43C7690F0C1@gmail.com>


On Aug 12, 2009, at 2:44 PM, Tzury Bar Yochay wrote:
> Hi,
>
> I wanted to add more extension to freeswitch.
> to add extension 1050 with password 1234 I did the following:
>
> $ cd /usr/local/freeswitch/conf/directory/default
>
> created 1050.xml having all '1000' strings replaced by '1050' by  
> typing
> $ sed s/1000/1050/g < 1000.xml > 1050.xml
>

just run reloadxml should be ok no need to rescan the profile

> rescan and reload the xml by typing into the CLI
>
> freeswitch at internal> sofia profile internal rescan reloadxml
>
> However, when I tried to login with these credentials I got the
> following in the fs_cli:
>
> 2009-08-12 06:35:01 [DEBUG] sofia_reg.c:1469 sofia_reg_parse_auth()
> SIP username 1050 does not match auth username
> 2009-08-12 06:35:01 [DEBUG] sofia_reg.c:869
> sofia_reg_handle_register() Send challenge for
> [1050 at server_address.net]
>
> below are the content of 1000 and 1050 xml files
>
> please advise.
>
>
> $ cat 1050.xml
> <include>
> <user id="1050" mailbox="1050">
>   <params>
>     <param name="password" value="1234"/>
>     <param name="vm-password" value="1050"/>
>   </params>
>   <variables>
>     <variable name="toll_allow" value="domestic,international,local"/>
>     <variable name="accountcode" value="1050"/>
>     <variable name="user_context" value="default"/>
>     <variable name="effective_caller_id_name" value="Extension 1050"/>
>     <variable name="effective_caller_id_number" value="1050"/>
>     <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>     <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>     <variable name="callgroup" value="techsupport"/>
>   </variables>
> </user>
> </include>
>
> $ cat 1000.xml
> <include>
> <user id="1000" mailbox="1000">
>   <params>
>     <param name="password" value="1234"/>
>     <param name="vm-password" value="1000"/>
>   </params>
>   <variables>
>     <variable name="toll_allow" value="domestic,international,local"/>
>     <variable name="accountcode" value="1000"/>
>     <variable name="user_context" value="default"/>
>     <variable name="effective_caller_id_name" value="Extension 1000"/>
>     <variable name="effective_caller_id_number" value="1000"/>
>     <variable name="outbound_caller_id_name"
> value="$${outbound_caller_name}"/>
>     <variable name="outbound_caller_id_number"
> value="$${outbound_caller_id}"/>
>     <variable name="callgroup" value="techsupport"/>
>   </variables>
> </user>
> </include>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From markmorreny at gmail.com  Wed Aug 12 04:26:38 2009
From: markmorreny at gmail.com (mark morreny)
Date: Wed, 12 Aug 2009 19:26:38 +0800
Subject: [Freeswitch-users] Fwd:  Fwd: Scheduler in module
In-Reply-To: <5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
Message-ID: <20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>

Hi,

In my LOAD_FUNCTION, I am trying to have freeswitch to flush out some data
every 10 s.  The following lines of code does not show any effect at all.

    switch_scheduler_task_thread_start();
    switch_scheduler_add_task(switch_epoch_time_now(NULL),
data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|SSHF_NO_DEL);


SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {

switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to flush
data  buffer...\n");


    task->runtime = switch_time_now() + 10;

}

Does anyone know how to get it to work?

Thanks,
Mark


---------- Forwarded message ----------
From: Brian West <brian at freeswitch.org>
Date: Mon, Aug 10, 2009 at 8:53 PM
Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
To: freeswitch-users at lists.freeswitch.org


switch_rtp.c has a simple one for the zrtp cache storing.

/b

On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:

> Re schedule is done in your callback, take a look at places that use
> these apis in the code for details.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From juanbackson at gmail.com  Wed Aug 12 05:01:59 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Wed, 12 Aug 2009 20:01:59 +0800
Subject: [Freeswitch-users] freeswitch time conversion
Message-ID: <27c25bc40908120501l19815773ne1494040ddcab111@mail.gmail.com>

Does anyone know how to take the epoch time in switch_event_t and convert it
into a format such as "Sat Jul 5 02:44:33 2009"?

Is there any existing facility that I can use for this purpose?

br,
JB
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From boonedox at gmail.com  Wed Aug 12 06:02:48 2009
From: boonedox at gmail.com (Jeremiah Johnson)
Date: Wed, 12 Aug 2009 07:02:48 -0600
Subject: [Freeswitch-users] hangup_after_bridge=false only works if
	bypass_media=false
In-Reply-To: <C5AD471E-636F-427A-A0A9-9F894D799CEB@jerris.com>
References: <c626f0a50908111129w626cf026u494c5846a7444f52@mail.gmail.com>
	<C5AD471E-636F-427A-A0A9-9F894D799CEB@jerris.com>
Message-ID: <c626f0a50908120602l155bd7f5w6357ff6d2303ae5@mail.gmail.com>

I posted it yesterday evening:  http://jira.freeswitch.org/browse/FSCORE-417

On Tue, Aug 11, 2009 at 9:43 PM, Michael Jerris <mike at jerris.com> wrote:

> Please post a bug for this on jira.freeswitch.org.
>
> Mike
>
> On Aug 11, 2009, at 2:29 PM, Jeremiah Johnson wrote:
>
> > This is an integral part of my application.  I need to have
> > FreeSWITCH outside of the media path as well as be able to do
> > multiple bridges for the same "A" leg.
> >
> > /*WORKS*/
> > <action application="set" data="hangup_after_bridge=false"/>
> > <action application="bridge" data="sofia/gateway/${mygateway}/1$
> > {client_one}"/>
> > <action application="bridge" data="sofia/gateway/${mygateway}/1$
> > {client_two}"/>
> >
> > /*DOES NOT WORK*/
> > <action application="set" data="hangup_after_bridge=false"/>
> > <action application="set" data="bypass_media=true"/>
> > <action application="bridge" data="sofia/gateway/${mygateway}/1$
> > {client_one}"/>
> > <action application="bridge" data="sofia/gateway/${mygateway}/1$
> > {client_two}"/>
> >
> > In the "DOES NOT WORK" example, the "A" leg hangs up as soon as the
> > leg for client_one hangs up.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From pjintheusa at gmail.com  Wed Aug 12 06:29:31 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 12 Aug 2009 09:29:31 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<6B83755F-C219-42D3-9C9E-4FA7FE5CA35F@freeswitch.org>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
Message-ID: <367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>

David / Michael - thanks for your your replies. The SoftIVR example is
particularly useful. Must admit though - I was hoping not to have to
do any custom stuff at this stage.

It does appear there is no method to do this by staking bridge lines
so I will put an issue in jira to try and get loopback working with
bypass_media.

In the meantime I will also start looking to build a custom bridging
app. As I said though - not a road I wanted to go down.

Thanks for your help!


Phillip Jones

On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavo<mgg at giagnocavo.net> wrote:
> It's also simple enough to write a plugin in one of the scripting languages to add an app to do exactly what you want...
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Knell
> Sent: Tuesday, August 11, 2009 1:55 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Loopback and bypass_media
>
> Just to add my $0.02-worth (if you're feeling generous..) - I don't
> think that the dialplan is expressive enough to do what's needed here,
> and that's where the trouble's coming from. ?It's not enormously tricky
> to build a generic "dial this set of numbers according to these rules"
> service using something hanging off the event socket - there's a writeup
> here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
> be done on SoftIVR.
>
> To roll something similar yourself using the event socket, you'd need to
> map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
> and have some way of passing messages around between the threads
> handling the different call legs, assuming that you're using one thread
> per leg.
>
> --Dave
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From moises.silva at gmail.com  Wed Aug 12 07:17:29 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 12 Aug 2009 10:17:29 -0400
Subject: [Freeswitch-users] FW: Sangoma/FS...
In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C52ECE61290@cooper>
References: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED944C@cooper>
	<549CFEF87AEDE841A38E9D15EAB4C04C52ECE61290@cooper>
Message-ID: <c4d05cbe0908120717t26a9b7c2m3840fb3ab641b002@mail.gmail.com>

Hello Peter,
I'd appreciate if you can keep the discussion going in the freeswitch-users
mailing list, there are other people there that will benefit of the
discussion or even can help. Read my comments below.

On Wed, Aug 12, 2009 at 5:59 AM, Peter Olsson <
peter.olsson at visionutveckling.se> wrote:

>  Sorry for spamming you :) But I have some more results now. I?ve tried
> using another lab PBX with Q.SIG enabled, and when using that one I?m able
> to connect calls as I should. At least incoming to FS, outgoing seem to have
> some problems still.. So the problem for the PBX I used yesterday seems to
> be both related to Q.SIG (maybe) and the PBX itself (it does connect to
> other providers though, so I know the trunk works).
>
>
>
> Should I take some dumps from the PRI card to try to find out why it didn?t
> work with the first one, or is this ?as expected?, since they have Q.SIG
> enabled?
>
> I have no experience with Q.SIG, so I won't be able to help much. One thing
though, is that if I were you, I'd be using openzap with libpri support, is
that what you are using, or are you using the ISDN openzap stack?

As of the dumps, they may help, or not, but pastebin them anyways so I can
make an un-educated guess.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From rupa at rupa.com  Wed Aug 12 07:22:12 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 12 Aug 2009 09:22:12 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110746v19f9981am86c9c00937f2cbf6@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
Message-ID: <b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>

perhaps we need to add some syntax + logic to originate:

application="originate"
data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"

This would acomplish the equiv of

loopback/bar,loopback/yum  where bar and yum are then further expanded
in the dialplan as

sofia/foo/bar|sofia/baz/bar and sofia/foo/yum|sofia/baz/yum

except that the threads of execution are handled directly by
originate.  I'm not sure that is really the "solution" since each ()
group would still have to be a separate thread to run independently.

To me, loopback is the way to accomplish this issue (how I've done it
with the same requirements that you have) since all the hard work is
layered and works.  The problem is that you require bypass_media which
doesn't play nice with loopback.

Perhaps have an on answer hook that tries to enable bypass media
(re-invite) after the call is setup?

On Wed, Aug 12, 2009 at 8:29 AM, Phillip Jones<pjintheusa at gmail.com> wrote:
> David / Michael - thanks for your your replies. The SoftIVR example is
> particularly useful. Must admit though - I was hoping not to have to
> do any custom stuff at this stage.
>
> It does appear there is no method to do this by staking bridge lines
> so I will put an issue in jira to try and get loopback working with
> bypass_media.
>
> In the meantime I will also start looking to build a custom bridging
> app. As I said though - not a road I wanted to go down.
>
> Thanks for your help!
>
>
> Phillip Jones
>
> On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavo<mgg at giagnocavo.net> wrote:
>> It's also simple enough to write a plugin in one of the scripting languages to add an app to do exactly what you want...
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Knell
>> Sent: Tuesday, August 11, 2009 1:55 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Loopback and bypass_media
>>
>> Just to add my $0.02-worth (if you're feeling generous..) - I don't
>> think that the dialplan is expressive enough to do what's needed here,
>> and that's where the trouble's coming from. ?It's not enormously tricky
>> to build a generic "dial this set of numbers according to these rules"
>> service using something hanging off the event socket - there's a writeup
>> here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
>> be done on SoftIVR.
>>
>> To roll something similar yourself using the event socket, you'd need to
>> map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
>> and have some way of passing messages around between the threads
>> handling the different call legs, assuming that you're using one thread
>> per leg.
>>
>> --Dave
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa



From email.list.subscriber at gmail.com  Wed Aug 12 07:28:08 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Wed, 12 Aug 2009 10:28:08 -0400
Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9
	&	v1.0.4)
In-Reply-To: <4A8280B8.6050308@blueface.ie>
References: <4A8280B8.6050308@blueface.ie>
Message-ID: <4a82d138.1f588c0a.2915.4e3d@mx.google.com>

Hi Bruce,

I am having similar issues trying build freeswitch 1.0.4 on Solaris
x86 as well.  I sent some information over the mailing list, and I
received a response from Michal Bielicki (attached), stating he'd test
this and direct me to the steps to successfully build freeswitch.

Just an FYI in case you see his response.

Vladimir

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Bruce McAlister
Sent: Wednesday, August 12, 2009 4:44 AM
To: freeswitch-users at lists.freeswitch.org
Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 &
v1.0.4)

Hi All,

I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and

1.0.4.

I am running on Solaris 10 Update 5 on x86 hardware (32-bit).

The build fails with:

--- snip ---
make: Fatal error: Command failed for target `all-recursive'
Current working directory
/export/home/user/packages/BUILD/freeswitch-1.0.4
*** Error code 1
make: Fatal error: Command failed for target `all'
---

Looking back through the build I can see the following error:

--- snip ---
creating libfreeswitch.la
(cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la 
libfreeswitch.la)
/usr/bin/cc 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/s
rc 
-KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
-DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch 
freeswitch-switch.o  ./.libs/libfreeswitch.so 
-L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/
expat/lib 
/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/ex
pat/lib/.libs/libexpat.a 
/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libap
r-1.a 
-lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp 
-L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt 
-lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl 
-lcurses -lsocket  -R/opt/freeswitch/lib -R/usr/sfw/lib
Undefined                       first referenced
  symbol                             in file
herror                              ./.libs/libfreeswitch.so
ld: fatal: Symbol referencing errors. No output written to
.libs/freeswitch
*** Error code 1
The following command caused the error:
`if test -z "" ; then echo /bin/bash 
/export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else 
echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` 
--tag=CC --mode=link /usr/bin/cc 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
-I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/s
rc 
-KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
-DPATH_MAX=2048 -g -v -Xc -xc99=all  -lm -R/opt/freeswitch/lib  -o 
freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib 
freeswitch-switch.o   libfreeswitch.la libs/apr/libapr-1.la 
libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl 
-lcrypto -lsocket -lnsl -ldl   -lcurses -lsocket
--- snip ---

Then a little above this error, there is the following warning that is

displayed (I'm not sure if it is related):

--- snip ---
*** Warning: Linking the shared library libfreeswitch.la against the
*** static library libs/libedit/src/.libs/libedit.a is not portable!
--- snip ---

My configure line is as follows:

---
./configure --prefix=/opt/freeswitch
---

I have the complete configure and make output if anyone needs them.

Any help/pointers would be greatly appreciated.

Thanks
Bruce

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
http://www.freeswitch.org
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An embedded message was scrubbed...
From: "Michal Bielicki" <michal.bielicki at halo2.pl>
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
Date: Tue, 11 Aug 2009 05:33:06 -0400
Size: 19568
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From peter.olsson at visionutveckling.se  Wed Aug 12 07:29:11 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Wed, 12 Aug 2009 16:29:11 +0200
Subject: [Freeswitch-users] Sangoma/FS...
In-Reply-To: <c4d05cbe0908120717t26a9b7c2m3840fb3ab641b002@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED944C@cooper>
	<549CFEF87AEDE841A38E9D15EAB4C04C52ECE61290@cooper>
	<c4d05cbe0908120717t26a9b7c2m3840fb3ab641b002@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C52ECE61309@cooper>

Of course ? no problem!

I?m not using libpri support now, I don?t think it?s ported for Windows (yet)?

I?ll try it out some more, and try to detect what?s going wrong...

/Peter

Fr?n: Moises Silva [mailto:moises.silva at gmail.com]
Skickat: den 12 augusti 2009 16:17
Till: freeswitch-users at lists.freeswitch.org
Kopia: Peter Olsson
?mne: Re: FW: Sangoma/FS...

Hello Peter,

I'd appreciate if you can keep the discussion going in the freeswitch-users mailing list, there are other people there that will benefit of the discussion or even can help. Read my comments below.

On Wed, Aug 12, 2009 at 5:59 AM, Peter Olsson <peter.olsson at visionutveckling.se<mailto:peter.olsson at visionutveckling.se>> wrote:

Sorry for spamming you :) But I have some more results now. I?ve tried using another lab PBX with Q.SIG enabled, and when using that one I?m able to connect calls as I should. At least incoming to FS, outgoing seem to have some problems still.. So the problem for the PBX I used yesterday seems to be both related to Q.SIG (maybe) and the PBX itself (it does connect to other providers though, so I know the trunk works).



Should I take some dumps from the PRI card to try to find out why it didn?t work with the first one, or is this ?as expected?, since they have Q.SIG enabled?
I have no experience with Q.SIG, so I won't be able to help much. One thing though, is that if I were you, I'd be using openzap with libpri support, is that what you are using, or are you using the ISDN openzap stack?

As of the dumps, they may help, or not, but pastebin them anyways so I can make an un-educated guess.

--
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com<mailto:moy at sangoma.com>
!DSPAM:4a82cefe32931477278362!
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From mrene_lists at avgs.ca  Wed Aug 12 08:04:00 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 12 Aug 2009 11:04:00 -0400
Subject: [Freeswitch-users] freeswitch time conversion
In-Reply-To: <27c25bc40908120501l19815773ne1494040ddcab111@mail.gmail.com>
References: <27c25bc40908120501l19815773ne1494040ddcab111@mail.gmail.com>
Message-ID: <EECDBB6A-49A4-491C-8952-14901E2E50EC@avgs.ca>

Hi,


The standard C function is strftime.

FreeSWITCH has some wrapped ones:

switch_apr.h:SWITCH_DECLARE(switch_status_t) switch_strftime(char *s,  
switch_size_t *retsize, switch_size_t max, const char *format,  
switch_time_exp_t *tm);
switch_apr.h:SWITCH_DECLARE(switch_status_t)  
switch_strftime_nocheck(char *s, switch_size_t *retsize, switch_size_t  
max, const char *format, switch_time_exp_t *tm);
switch_core.h:SWITCH_DECLARE(switch_status_t) switch_strftime_tz(const  
char *tz, const char *format, char *date, size_t len, switch_time_t  
thetime);


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 12-Aug-09, at 8:01 AM, Juan Backson wrote:

> Does anyone know how to take the epoch time in switch_event_t and  
> convert it into a format such as "Sat Jul 5 02:44:33 2009"?
>
> Is there any existing facility that I can use for this purpose?
>
> br,
> JB
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mrene_lists at avgs.ca  Wed Aug 12 08:09:21 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 12 Aug 2009 11:09:21 -0400
Subject: [Freeswitch-users] Fwd:  Fwd: Scheduler in module
In-Reply-To: <20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
Message-ID: <3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>

Hi,

I did the same thing on my side....

API CALL [load(mod_skel)] output:
+OK

2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task 2  
data_flush (core) to run at 1250089698
2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889  
Successfully Loaded [mod_skel]
2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270 Adding  
API Function 'skel'
freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]  
mod_skel.c:120 starting to flush data  buffer...

Note that you don't need to start the thread manually, the core  
already has threads running for the scheduler.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 12-Aug-09, at 7:26 AM, mark morreny wrote:

> Hi,
>
> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out  
> some data every 10 s.  The following lines of code does not show any  
> effect at all.
>
>     switch_scheduler_task_thread_start();
>     switch_scheduler_add_task(switch_epoch_time_now(NULL),  
> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE| 
> SSHF_NO_DEL);
>
>
> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to  
> flush data  buffer...\n");
>
>
>     task->runtime = switch_time_now() + 10;
>
> }
>
> Does anyone know how to get it to work?
>
> Thanks,
> Mark
>
>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> Date: Mon, Aug 10, 2009 at 8:53 PM
> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
> To: freeswitch-users at lists.freeswitch.org
>
>
> switch_rtp.c has a simple one for the zrtp cache storing.
>
> /b
>
> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>
> > Re schedule is done in your callback, take a look at places that use
> > these apis in the code for details.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From charlieb at cot.net  Wed Aug 12 08:14:17 2009
From: charlieb at cot.net (Charles Boening)
Date: Wed, 12 Aug 2009 08:14:17 -0700
Subject: [Freeswitch-users] FS 1.0.4 official core dumping LUA streamFile
In-Reply-To: <A9CD3A3C-923A-47FE-B9BC-DEAE701F3248@jerris.com>
References: <4FB2938B89459C41860C4DB9B1821D6FB6657A2421@exchange.calore.local>
	<A9CD3A3C-923A-47FE-B9BC-DEAE701F3248@jerris.com>
Message-ID: <4FB2938B89459C41860C4DB9B1821D6FB6657A2427@exchange.calore.local>

Done.

Thanks,
Charlie


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Wednesday, August 12, 2009 2:12 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS 1.0.4 official core dumping LUA streamFile

If your seeing a segfault, please report it to jira.freeswitch.org with a backtrace and details of how to reproduce.

Mike

On Aug 12, 2009, at 2:37 AM, Charles Boening wrote:


Greetings,

I have the following LUA script (at end of email) in a fresh FS 1.0.4 install.  I originally did an upgrade from one of the 1.0.4preX versions but when I came across this issue I went fresh just to make sure there wasn't an incompatibility with my previous config.

What I'm seeing is a seg fault and a core dump after playing a sound file.  I originally had a file I recorded but when I ran into this issue I figured I'd try an included sound file but that doesn't seem to make a bit of difference.

2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Starting test.lua
2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Caller [XXXXXXXXXX] connected
2009-08-11 22:52:20.327071 [INFO] switch_cpp.cpp:1130 Pre streamFile
Segmentation fault (core dumped)


Any ideas?

Thanks,
Charlie



freeswitch.consoleLog("INFO", string.format("Starting test.lua\n"))
session:answer();
session:setHangupHook("session_hangup_hook")
calleridnumber = session:getVariable("caller_id_number")
calleridname = session:getVariable("caller_id_name")

if session:ready() then
   freeswitch.consoleLog("INFO", string.format("Caller [" .. calleridnumber .. "] connected\n"))
   freeswitch.consoleLog("INFO", string.format("Pre streamFile\n"))

   session:streamFile("conference/8000/conf-welcome.wav")

   freeswitch.consoleLog("INFO", string.format("Post streamFile.\n"))
end

function session_hangup_hook(status)
  freeswitch.consoleLog("INFO", "Session hangup: \n") --[[ .. status .. "\n") ]]--
  error()
end

session:hangup()

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From pjintheusa at gmail.com  Wed Aug 12 08:22:08 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Wed, 12 Aug 2009 11:22:08 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<D45D8BA9-393B-4A74-B611-CC86ECCB3D2D@freeswitch.org>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
Message-ID: <367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>

Hi there,

>> application="originate" data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"

I agree. However, perhaps the ideal is not to specify the carriers at
this level, as carriers are added and removed fairly often as costings
change. So it would be nice to have some sort of proxy that resolves
to a list of carriers:

application="originate" data="sofia/MyCarriers/bar,sofia/MyCarriers/yum"

<MyCarriers>
<action application="carrier" data="sofia/foo"/>
<action application="carrier" data="sofia/baz"/>
<action application="carrier" data="sofia/etc"/>
</MyCarriers>


or something similar. This would achieve the same as loopback in this
use case but without dangers of looping? Complicated stuff though.

>>Perhaps have an on answer hook that tries to enable bypass media (re-invite) after the call is setup?

That's a good idea - I will look into that.


Thanks again.


Phillip

On Wed, Aug 12, 2009 at 10:22 AM, Rupa Schomaker<rupa at rupa.com> wrote:
> perhaps we need to add some syntax + logic to originate:
>
> application="originate"
> data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"
>
> This would acomplish the equiv of
>
> loopback/bar,loopback/yum ?where bar and yum are then further expanded
> in the dialplan as
>
> sofia/foo/bar|sofia/baz/bar and sofia/foo/yum|sofia/baz/yum
>
> except that the threads of execution are handled directly by
> originate. ?I'm not sure that is really the "solution" since each ()
> group would still have to be a separate thread to run independently.
>
> To me, loopback is the way to accomplish this issue (how I've done it
> with the same requirements that you have) since all the hard work is
> layered and works. ?The problem is that you require bypass_media which
> doesn't play nice with loopback.
>
> Perhaps have an on answer hook that tries to enable bypass media
> (re-invite) after the call is setup?
>
> On Wed, Aug 12, 2009 at 8:29 AM, Phillip Jones<pjintheusa at gmail.com> wrote:
>> David / Michael - thanks for your your replies. The SoftIVR example is
>> particularly useful. Must admit though - I was hoping not to have to
>> do any custom stuff at this stage.
>>
>> It does appear there is no method to do this by staking bridge lines
>> so I will put an issue in jira to try and get loopback working with
>> bypass_media.
>>
>> In the meantime I will also start looking to build a custom bridging
>> app. As I said though - not a road I wanted to go down.
>>
>> Thanks for your help!
>>
>>
>> Phillip Jones
>>
>> On Tue, Aug 11, 2009 at 6:13 PM, Michael Giagnocavo<mgg at giagnocavo.net> wrote:
>>> It's also simple enough to write a plugin in one of the scripting languages to add an app to do exactly what you want...
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Knell
>>> Sent: Tuesday, August 11, 2009 1:55 PM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Loopback and bypass_media
>>>
>>> Just to add my $0.02-worth (if you're feeling generous..) - I don't
>>> think that the dialplan is expressive enough to do what's needed here,
>>> and that's where the trouble's coming from. ?It's not enormously tricky
>>> to build a generic "dial this set of numbers according to these rules"
>>> service using something hanging off the event socket - there's a writeup
>>> here: http://www.softivr.com/wiki/index.php/Find_me showing how it could
>>> be done on SoftIVR.
>>>
>>> To roll something similar yourself using the event socket, you'd need to
>>> map the dial function to 'originate', bridge to (IIRC) 'uuid_bridge',
>>> and have some way of passing messages around between the threads
>>> handling the different call legs, assuming that you're using one thread
>>> per leg.
>>>
>>> --Dave
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From rupa at rupa.com  Wed Aug 12 09:21:58 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 12 Aug 2009 11:21:58 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<367751820908110908j27e6c170k4675f899f7806826@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
Message-ID: <b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>

On Wed, Aug 12, 2009 at 10:22 AM, Phillip Jones<pjintheusa at gmail.com> wrote:
> Hi there,
>
>>> application="originate" data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"
>
> I agree. However, perhaps the ideal is not to specify the carriers at
> this level, as carriers are added and removed fairly often as costings
> change. So it would be nice to have some sort of proxy that resolves
> to a list of carriers:
>
> application="originate" data="sofia/MyCarriers/bar,sofia/MyCarriers/yum"

> <MyCarriers>
> <action application="carrier" data="sofia/foo"/>
> <action application="carrier" data="sofia/baz"/>
> <action application="carrier" data="sofia/etc"/>
> </MyCarriers>
>
>
> or something similar. This would achieve the same as loopback in this
> use case but without dangers of looping? Complicated stuff though.

Well, that is all done by mod_lcr.  I was simplifying to narrow down
to just originate.

First we need to see if this is worth pursuing over fixing (modifying,
whatever) loopback to handle bypass media.  If it is, then I'll modify
mod_lcr to deal with the situation in question (comma or pipe sep list
of numbers to call.  mod_lcr would then group as appropriate).

Right now, my bridge is setup in a small javascript script that builds
the appropriate dialstring (using loopback for external calls, user/
for internal calls) and then when doing the loopback call to mod_lcr
to get the dialstring with all providers in the right order.

>>>Perhaps have an on answer hook that tries to enable bypass media (re-invite) after the call is setup?
>
> That's a good idea - I will look into that.
>
>
> Thanks again.
>
>
> Phillip

Let us know how it works for you...

-- 
-Rupa



From maxim.tsvetov at gmail.com  Wed Aug 12 09:49:20 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Wed, 12 Aug 2009 09:49:20 -0700 (PDT)
Subject: [Freeswitch-users] answer command
In-Reply-To: <B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
Message-ID: <24940548.post@talk.nabble.com>


I will try to paraphrase my question.
Is there any possibility to answer call  from CTI application and
synchronise answer with answer in SIP client?Maybe we can use SIP functions
in our CTI application instead of FS api commands?
I'm trying to find the way to make prototype of lineAnswer command in TAPI.


Seven Du wrote:
> 
> It's not Eyebeam but FS hung up the call because it have nothing to do  
> after answer.
> 
> You should either playback a sound, do the echo command, record, hold  
> the call, bridge to another channel or transfer somewhere else.....
> 
> On Aug 12, 2009, at 4:54 PM, Maxim Tsvetov wrote:
>>
>> I've tried to use "answer" command from outbound event socket and it's
>> working, but
>> the problem is that FS answering the call, but SIP Client (we tried  
>> this
>> with EyeBeam and CISCO 7960)
>> doesn't know that call was answered. So, as long as FS doesn't know  
>> what to
>> do with this number it then disconnects the call.
>>
>> 2009-08-12 11:25:07.599250 [NOTICE] mod_dptools.c:649 Channel
>> [sofia/internal/sip:1000 at 10.107.181.160:42840] has been answered
>> 2009-08-12 11:25:07.599250 [NOTICE] switch_ivr_originate.c:2015  
>> Channel
>> [sofia/internal/1003 at 10.107.249.12] has been answered
>> 2009-08-12 11:25:07.614875 [ERR] switch_core_io.c:118
>> sofia/internal/sip:1000 at 10.107.181.160:42840 has no read codec.
>> 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:503 Hangup
>> sofia/internal/sip:1000 at 10.107.181.160:42840 [CS_EXCHANGE_MEDIA]
>> [NORMAL_CLEARING]
>> 2009-08-12 11:25:07.614875 [NOTICE] switch_ivr_bridge.c:1016 Hangup
>> sofia/internal/1003 at 10.107.249.12 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-08-12 11:25:07.630500 [NOTICE] switch_core_session.c:1086  
>> Session 133
>> (sofia/internal/sip:1000 at 10.107.181.160:42840) Ended
>>
>>
>> Maybe there is the way to acknowledge SIP client that call was  
>> answered?
>>
>> Regards,
>> Maxim Tsvetov
>>
>> Diego Viola wrote:
>>>
>>> I suggest that you learn the differences between mod_commands  
>>> commands and
>>> mod_dptools applications, and also the interfaces where you can  
>>> access and
>>> use them.
>>>
>>> As said before, mod_dptools is accessible from dialplan, event socket
>>> outbound, etc. and mod_commands is accessible from the CLI, event  
>>> socket
>>> (inbound/outbound), XML RPC, etc.
>>>
>>> That's all described in the wiki I think.
>>>
>>> Let us know if you have any questions =D.
>>>
>>> On Tue, Aug 11, 2009 at 2:10 PM, Diego Viola <diego.viola at gmail.com>
>>> wrote:
>>>
>>>> Michael, you're welcome :).
>>>>
>>>> Milena, answer is a mod_dptools command, you can use it from the XML
>>>> dialplan or from the event socket outbound. mod_commands API are  
>>>> APIs
>>>> that
>>>> you execute from the socket, event socket inbound, etc. But you  
>>>> can also
>>>> execute them from event socket outbound using the "api" command.
>>>>
>>>> I hope that makes sense, correct me if I'm wrong =D.
>>>>
>>>> On Tue, Aug 11, 2009 at 1:09 PM, Michael Collins
>>>> <msc at freeswitch.org>wrote:
>>>>
>>>>>
>>>>>
>>>>> On Tue, Aug 11, 2009 at 9:05 AM, Milena <testeador01 at gmail.com>  
>>>>> wrote:
>>>>>
>>>>>>
>>>>>> Hello Brian,
>>>>>>
>>>>>> I wanna fix the wiki, but to make sure i got it right, does it  
>>>>>> only
>>>>>> work
>>>>>> on outbound event socket? or is there any other scenario where  
>>>>>> it would
>>>>>> work.
>>>>>>
>>>>>
>>>>> FYI, Diego Viola fixed the wiki. (Thanks Diego!)
>>>>> -MC
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> -- 
>> View this message in context:
>> http://www.nabble.com/answer-command-tp24912812p24931876.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/answer-command-tp24912812p24940548.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From brian at freeswitch.org  Wed Aug 12 09:57:01 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 11:57:01 -0500
Subject: [Freeswitch-users] answer command
In-Reply-To: <24940548.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
Message-ID: <B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>

Well you can only truly answer an inbound call to FS... you can't  
force answer an outbound call.

/b

On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:

>
> I will try to paraphrase my question.
> Is there any possibility to answer call  from CTI application and
> synchronise answer with answer in SIP client?Maybe we can use SIP  
> functions
> in our CTI application instead of FS api commands?
> I'm trying to find the way to make prototype of lineAnswer command  
> in TAPI.




From juanbackson at gmail.com  Wed Aug 12 10:21:07 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 13 Aug 2009 01:21:07 +0800
Subject: [Freeswitch-users] random route selection
Message-ID: <27c25bc40908121021r6b108b1fye42ca4e3970ee93a@mail.gmail.com>

Hi,

I would like to implement a random route selection based on some arbitrary
percentage.

Does anyone know if there is any good way of doing that within freeswitch?

If there isn't any api that I can use, does freeswitch has any random
generator that I can be used for this purpose?

br,
JB
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From maxim.tsvetov at gmail.com  Wed Aug 12 10:38:22 2009
From: maxim.tsvetov at gmail.com (Maxim Tsvetov)
Date: Wed, 12 Aug 2009 10:38:22 -0700 (PDT)
Subject: [Freeswitch-users] answer command
In-Reply-To: <B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
	<B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
Message-ID: <24941422.post@talk.nabble.com>


If I have two FS extensions A and B. I'm calling from A to B and want to
answer from B-side in my CTI application and to make SIP phone to be
synchronised to my CTI application. Is it possible to do it?


Brian West-3 wrote:
> 
> Well you can only truly answer an inbound call to FS... you can't  
> force answer an outbound call.
> 
> /b
> 
> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
> 
>>
>> I will try to paraphrase my question.
>> Is there any possibility to answer call  from CTI application and
>> synchronise answer with answer in SIP client?Maybe we can use SIP  
>> functions
>> in our CTI application instead of FS api commands?
>> I'm trying to find the way to make prototype of lineAnswer command  
>> in TAPI.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/answer-command-tp24912812p24941422.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From rupa at rupa.com  Wed Aug 12 10:42:56 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Wed, 12 Aug 2009 12:42:56 -0500
Subject: [Freeswitch-users] random route selection
In-Reply-To: <27c25bc40908121021r6b108b1fye42ca4e3970ee93a@mail.gmail.com>
References: <27c25bc40908121021r6b108b1fye42ca4e3970ee93a@mail.gmail.com>
Message-ID: <b39273530908121042v18f9d1c0ne8c551bd1b416dd2@mail.gmail.com>

mod_lcr will do random route selection if the rates are the same.  But
that gives an equal distribution.  There is no weighting/percentage
supported.

On Wed, Aug 12, 2009 at 12:21 PM, Juan Backson<juanbackson at gmail.com> wrote:
> Hi,
>
> I would like to implement a random route selection based on some arbitrary
> percentage.
>
> Does anyone know if there is any good way of doing that within freeswitch?
>
> If there isn't any api that I can use, does freeswitch has any random
> generator that I can be used for this purpose?
>
> br,
> JB
>



-- 
-Rupa



From mike at jerris.com  Wed Aug 12 10:50:13 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 12 Aug 2009 10:50:13 -0700
Subject: [Freeswitch-users] answer command
In-Reply-To: <24941422.post@talk.nabble.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<4A812279.6080404@gcdf.pl> <24916765.post@talk.nabble.com>
	<ecd05ce30908110619s5d4a4d04l1b94708419c7e313@mail.gmail.com>
	<24917706.post@talk.nabble.com>
	<59A8E004-B1AB-445E-8C30-ECF13828622A@freeswitch.org>
	<c29b930b0908110705s22446200veec98ca00d965337@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
	<B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
	<24941422.post@talk.nabble.com>
Message-ID: <6539128A-D7A1-4555-A3DB-60E94E461DB3@jerris.com>

Sip does not support this functionality.  The called device would have  
to support this via some other mechanism such as ctsa which I have  
seen recently someone was looking at for freeswitch.  So the first  
issue you must resolve is the called device needs to support some way  
to do this.

Mike

On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov <maxim.tsvetov at gmail.com>  
wrote:

>
> If I have two FS extensions A and B. I'm calling from A to B and  
> want to
> answer from B-side in my CTI application and to make SIP phone to be
> synchronised to my CTI application. Is it possible to do it?
>
>
> Brian West-3 wrote:
>>
>> Well you can only truly answer an inbound call to FS... you can't
>> force answer an outbound call.
>>
>> /b
>>
>> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
>>
>>>
>>> I will try to paraphrase my question.
>>> Is there any possibility to answer call  from CTI application and
>>> synchronise answer with answer in SIP client?Maybe we can use SIP
>>> functions
>>> in our CTI application instead of FS api commands?
>>> I'm trying to find the way to make prototype of lineAnswer command
>>> in TAPI.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> -- 
> View this message in context: http://www.nabble.com/answer-command-tp24912812p24941422.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From alan at chandlerfamily.org.uk  Wed Aug 12 10:56:19 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Wed, 12 Aug 2009 18:56:19 +0100
Subject: [Freeswitch-users] Confused about conferences
Message-ID: <4A830243.8080703@chandlerfamily.org.uk>

I have been reading all the docs about conferences I can find and am 
getting somewhat confused.  What I am trying to do is set up a dialplan 
where I have subscribers with extensions in the 1xx range, and then to 
set an ability to have a series of conference rooms for each subscriber 
in the 21xx range where if the user enters is "own" conference he is the 
moderator, but if not he is just a normal user.

I want to be able for the moderator to do things like mute or kick people.

So dialplan would probably have something like this in it



     <extension name="user_conference">
       <condition field="destination_number" 
expression="^(2${caller_id_number})$">
         <action application="set" data="i_am_moderator=true">
	<action application="answer"/>
	<action application="conference" data="$1 at default+flags{moderator}"/>
       </condition>
       <condition field="destination_number" expression="^(21\d{2})^">
         <action application="set" data="i_am_moderator=false">
	<action application="answer"/>
	<action application="conference" data="$1 at default"/>
       </condition>
     </extension>
     <extension name="kick">
       <condition field="i_am_moderator" expression="true">
         <action application="play_and_get_digits" data="3 3 3 7000 # 
/ask-for-extension.wav /invalid.wav conf-user-id \d+" >

WHAT GOES HERE???

       </condition>
     </extension>


in the conference.conf.xml file, I would change the caller controls to 
include

   <caller-controls>
     <group name="somekeys">
        <control action="transfer" digits="9"   data="kick XML default"/>
     </group>
   </caller-controls>


My question (at the moment) is

In the WHAT GOES HERE place how do it

Kick extension ${conf-user-id}  (DOES IT REQUIRE A SCRIPT TO CALL THE 
Conference API?)
Re-Enter the moderator back into the conference
Re-Enter the ordinary user who happened to press 9 back into the conference

I am assuming I can't stop the non moderator getting the control - since 
all users get the same controls.


-- 
Alan Chandler
http://www.chandlerfamily.org.uk




From testeador01 at gmail.com  Wed Aug 12 10:59:08 2009
From: testeador01 at gmail.com (Milena)
Date: Wed, 12 Aug 2009 12:59:08 -0500
Subject: [Freeswitch-users] fs installation is broken in version: FreeSWITCH
	Version 1.0.trunk (14500M)
Message-ID: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>

I just did a rebootstrap on a fs box, it turned out the new revision has
this at the end of mod_sofia.h:

> char *sofia_glue_get_extra_headers(switch_channel_t *channel, const char
> *prefix);

<<<<<<< .mine

void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const
> *sip, const char *prefix);


> =======

void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t const
> *sip, const char *prefix);

>>>>>>> .r14490


no wonder why it wouldn't compile :P
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From brian at freeswitch.org  Wed Aug 12 11:18:06 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 13:18:06 -0500
Subject: [Freeswitch-users] fs installation is broken in version:
	FreeSWITCH Version 1.0.trunk (14500M)
In-Reply-To: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
Message-ID: <B8137064-55DE-48C9-87C3-A9BFE6DF3B3E@freeswitch.org>

You have a merge conflict. svn revert sofia_glue.c

/b

On Aug 12, 2009, at 12:59 PM, Milena wrote:

> I just did a rebootstrap on a fs box, it turned out the new revision  
> has this at the end of mod_sofia.h:
>
> char *sofia_glue_get_extra_headers(switch_channel_t *channel, const  
> char *prefix);
> <<<<<<< .mine
> void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t  
> const *sip, const char *prefix);
>
> =======
> void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t  
> const *sip, const char *prefix);
> >>>>>>> .r14490
>
> no wonder why it wouldn't compile :P




From krice at suspicious.org  Wed Aug 12 11:20:42 2009
From: krice at suspicious.org (Ken Rice)
Date: Wed, 12 Aug 2009 13:20:42 -0500
Subject: [Freeswitch-users] random route selection
In-Reply-To: <27c25bc40908121021r6b108b1fye42ca4e3970ee93a@mail.gmail.com>
Message-ID: <C6A8722A.5AC41%krice@suspicious.org>

Write you a small C app to randomly return them based on the percentages...
Currently we do something similar to this but use a random round robin based
thing using a simple sql backend and doing a select order by random sort of
thing... 

Contact me off list if you need some profession help figuring this out

K



From: Juan Backson <juanbackson at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Thu, 13 Aug 2009 01:21:07 +0800
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] random route selection

Hi,

I would like to implement a random route selection based on some arbitrary
percentage.? 

Does anyone know if there is any good way of doing that within freeswitch?

If there isn't any api that I can use, does freeswitch has any random
generator that I can be used for this purpose?

br,
JB


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From mike at jerris.com  Wed Aug 12 11:22:41 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 12 Aug 2009 11:22:41 -0700
Subject: [Freeswitch-users] fs installation is broken in version:
	FreeSWITCH Version 1.0.trunk (14500M)
In-Reply-To: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
Message-ID: <D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com>

The M in the version number means modified.  You had local code  
changes that conflicted when you updated trunk.  Revert the changes to  
that file and it should be fine.

Mike

On Aug 12, 2009, at 10:59 AM, Milena <testeador01 at gmail.com> wrote:

> I just did a rebootstrap on a fs box, it turned out the new revision  
> has this at the end of mod_sofia.h:
>
> char *sofia_glue_get_extra_headers(switch_channel_t *channel, const  
> char *prefix);
> <<<<<<< .mine
> void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t  
> const *sip, const char *prefix);
>
> =======
> void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t  
> const *sip, const char *prefix);
> >>>>>>> .r14490
>
> no wonder why it wouldn't compile :P
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From alan at chandlerfamily.org.uk  Wed Aug 12 11:25:21 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Wed, 12 Aug 2009 19:25:21 +0100
Subject: [Freeswitch-users] The application "hash"
Message-ID: <4A830911.1060004@chandlerfamily.org.uk>

I was trying to explore the documentation for the application "hash" 
which is in the default dialplan.  Its vaguely obvious what its doing, 
but I wanted to be sure.

It appears to be listed as a application under dp_tools, but when I 
click on it I get taken to a page that talks about limit_hash rather 
than hash.

Is there any documentation for this?  Am I looking in the wring place?
-- 
Alan Chandler
http://www.chandlerfamily.org.uk




From testeador01 at gmail.com  Wed Aug 12 11:28:23 2009
From: testeador01 at gmail.com (Milena)
Date: Wed, 12 Aug 2009 13:28:23 -0500
Subject: [Freeswitch-users] fs installation is broken in version:
	FreeSWITCH Version 1.0.trunk (14500M)
In-Reply-To: <D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
	<D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com>
Message-ID: <c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>

It's modified because it wouldn't compile with those "<<<<<<" at the end of
the file

2009/8/12 Michael Jerris <mike at jerris.com>

> The M in the version number means modified.  You had local code
> changes that conflicted when you updated trunk.  Revert the changes to
> that file and it should be fine.
>
> Mike
>
> On Aug 12, 2009, at 10:59 AM, Milena <testeador01 at gmail.com> wrote:
>
> > I just did a rebootstrap on a fs box, it turned out the new revision
> > has this at the end of mod_sofia.h:
> >
> > char *sofia_glue_get_extra_headers(switch_channel_t *channel, const
> > char *prefix);
> > <<<<<<< .mine
> > void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t
> > const *sip, const char *prefix);
> >
> > =======
> > void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t
> > const *sip, const char *prefix);
> > >>>>>>> .r14490
> >
> > no wonder why it wouldn't compile :P
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Aug 12 11:29:23 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 13:29:23 -0500
Subject: [Freeswitch-users] The application "hash"
In-Reply-To: <4A830911.1060004@chandlerfamily.org.uk>
References: <4A830911.1060004@chandlerfamily.org.uk>
Message-ID: <87FBD426-C373-49CF-8E12-1539796F4146@freeswitch.org>

hash is just like db but its all in memory.. you can interchange db  
and hash.

/b

On Aug 12, 2009, at 1:25 PM, Alan Chandler wrote:

> I was trying to explore the documentation for the application "hash"
> which is in the default dialplan.  Its vaguely obvious what its doing,
> but I wanted to be sure.
>
> It appears to be listed as a application under dp_tools, but when I
> click on it I get taken to a page that talks about limit_hash rather
> than hash.
>
> Is there any documentation for this?  Am I looking in the wring place?
> -- 
> Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From testeador01 at gmail.com  Wed Aug 12 11:34:42 2009
From: testeador01 at gmail.com (Milena)
Date: Wed, 12 Aug 2009 13:34:42 -0500
Subject: [Freeswitch-users] fs installation is broken in version:
	FreeSWITCH Version 1.0.trunk (14500M)
In-Reply-To: <c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
	<D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com>
	<c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>
Message-ID: <c29b930b0908121134m50b1c8b2j3a91b57e71b70b10@mail.gmail.com>

Ok, done and fixed, thank you very much :)

2009/8/12 Milena <testeador01 at gmail.com>

> It's modified because it wouldn't compile with those "<<<<<<" at the end of
> the file
>
> 2009/8/12 Michael Jerris <mike at jerris.com>
>
> The M in the version number means modified.  You had local code
>> changes that conflicted when you updated trunk.  Revert the changes to
>> that file and it should be fine.
>>
>> Mike
>>
>> On Aug 12, 2009, at 10:59 AM, Milena <testeador01 at gmail.com> wrote:
>>
>> > I just did a rebootstrap on a fs box, it turned out the new revision
>> > has this at the end of mod_sofia.h:
>> >
>> > char *sofia_glue_get_extra_headers(switch_channel_t *channel, const
>> > char *prefix);
>> > <<<<<<< .mine
>> > void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t
>> > const *sip, const char *prefix);
>> >
>> > =======
>> > void sofia_glue_set_extra_headers(switch_channel_t *channel, sip_t
>> > const *sip, const char *prefix);
>> > >>>>>>> .r14490
>> >
>> > no wonder why it wouldn't compile :P
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From Christian.Jensen at Teligence.Net  Wed Aug 12 12:18:05 2009
From: Christian.Jensen at Teligence.Net (Christian Jensen)
Date: Wed, 12 Aug 2009 12:18:05 -0700
Subject: [Freeswitch-users] ClueCon Presentations - Where?
In-Reply-To: <c29b930b0908121134m50b1c8b2j3a91b57e71b70b10@mail.gmail.com>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com><D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com><c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>
	<c29b930b0908121134m50b1c8b2j3a91b57e71b70b10@mail.gmail.com>
Message-ID: <F571A97F57A2D946B229C8231805867D32AD4527@svanexcsvr01.alldir.net>

Hi there,

 

Does anyone have the URL for where I might find all the electronic
versions of the presentations made at ClueCon last week?

 

Thanks!

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From brian at freeswitch.org  Wed Aug 12 12:25:50 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 14:25:50 -0500
Subject: [Freeswitch-users] ClueCon Presentations - Where?
In-Reply-To: <F571A97F57A2D946B229C8231805867D32AD4527@svanexcsvr01.alldir.net>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com><D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com><c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>
	<c29b930b0908121134m50b1c8b2j3a91b57e71b70b10@mail.gmail.com>
	<F571A97F57A2D946B229C8231805867D32AD4527@svanexcsvr01.alldir.net>
Message-ID: <65BC9732-5836-48F4-83AC-B305F4D1028C@freeswitch.org>

They are all getting gathered up and put online...  
files.freeswitch.org/cluecon_2009 just keep an eye there some of the  
videos are up also.

/b

On Aug 12, 2009, at 2:18 PM, Christian Jensen wrote:

> Hi there,
>
> Does anyone have the URL for where I might find all the electronic  
> versions of the presentations made at ClueCon last week?
>
> Thanks!
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jaugenstine at gmail.com  Wed Aug 12 12:57:28 2009
From: jaugenstine at gmail.com (jonathan augenstine)
Date: Wed, 12 Aug 2009 12:57:28 -0700
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <1249685648.16901.34.camel@dk-d820>
References: <1249685648.16901.34.camel@dk-d820>
Message-ID: <207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>

I to would like to put my thanks on the table.  I have been going to
conferences for a very long time and often question the value of taking time
off to attend these venues.  When I was asked to attend by a client I was
very hesitant.  I am very pleased that I decided to attend.
Now the skeptical among you may say that I am only pleased because I won the
MacBook.  I cannot deny there is truth in that statement.  However, on
Thursday morning I was sitting in the conference room waiting for the first
presentation.  I was thinking that it had been a valuable three days.  I had
been able to connect with some clients at the conference, made some new
contacts, met people face to face that I had only met online, listened to
good presentations, learned some valuable new information, and lastly
received some insight from Anthony and Michael Jerris on fixing a bug that
had been plaguing me for sometime.
As I sat pondering, I clicked on the picture of the MacBook and thought the
only way the conference could end better was if I won the MacBook.  Never
did I actually think that would happen and the conference did end better.  I
will be much more motivated to consider attending next year, even knowing
lightening does not strike the same spot twice.

Jonathan Augenstine

On Fri, Aug 7, 2009 at 3:54 PM, David Knell <dave at 3c.co.uk> wrote:

> Just a quick note to say thanks to Cluecon's organisers for putting
> together such a useful, informative and packed three days.  I've come
> away with a head full of ideas, a bunch of new contacts and a collection
> of things to do; I'd thoroughly recommend that anyone interested in IP
> telephony blocks out the first week of August 2010, right now..!
>
> Cheers --
>
> Dave
>
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Aug 12 13:03:59 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 15:03:59 -0500
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
References: <1249685648.16901.34.camel@dk-d820>
	<207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
Message-ID: <74BD91EA-8D4B-44DD-9CA3-C0B3E3031E32@freeswitch.org>

Remember next year we'll have more Mac Book's to give away and iPod  
Touches with engraved sponsor logos on them.  :)

/b

On Aug 12, 2009, at 2:57 PM, jonathan augenstine wrote:

> I to would like to put my thanks on the table.  I have been going to  
> conferences for a very long time and often question the value of  
> taking time off to attend these venues.  When I was asked to attend  
> by a client I was very hesitant.  I am very pleased that I decided  
> to attend.
>
> Now the skeptical among you may say that I am only pleased because I  
> won the MacBook.  I cannot deny there is truth in that statement.   
> However, on Thursday morning I was sitting in the conference room  
> waiting for the first presentation.  I was thinking that it had been  
> a valuable three days.  I had been able to connect with some clients  
> at the conference, made some new contacts, met people face to face  
> that I had only met online, listened to good presentations, learned  
> some valuable new information, and lastly received some insight from  
> Anthony and Michael Jerris on fixing a bug that had been plaguing me  
> for sometime.
>
> As I sat pondering, I clicked on the picture of the MacBook and  
> thought the only way the conference could end better was if I won  
> the MacBook.  Never did I actually think that would happen and the  
> conference did end better.  I will be much more motivated to  
> consider attending next year, even knowing lightening does not  
> strike the same spot twice.
>
> Jonathan Augenstine




From Prometheus001 at gmx.net  Wed Aug 12 13:04:52 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Wed, 12 Aug 2009 22:04:52 +0200
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
Message-ID: <4A832064.8000908@gmx.net>

Hello,

anybody has a clue what this message means?
  [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from
PROGRESS_MEDIA to PROGRESS
What does VETO mean here?

Best regards
Peter



From brian at freeswitch.org  Wed Aug 12 13:12:22 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 15:12:22 -0500
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <1249685648.16901.34.camel@dk-d820>
References: <1249685648.16901.34.camel@dk-d820>
Message-ID: <378EC23A-D23A-4CFC-98DB-693946CF91E6@freeswitch.org>

Dave,
	Thanks,	Hope to see you there next year...

/b

On Aug 7, 2009, at 5:54 PM, David Knell wrote:

> Just a quick note to say thanks to Cluecon's organisers for putting
> together such a useful, informative and packed three days.  I've come
> away with a head full of ideas, a bunch of new contacts and a  
> collection
> of things to do; I'd thoroughly recommend that anyone interested in IP
> telephony blocks out the first week of August 2010, right now..!
>
> Cheers --
>
> Dave




From tina at a2unlimited.com  Wed Aug 12 12:55:35 2009
From: tina at a2unlimited.com (Tina Martinez)
Date: Wed, 12 Aug 2009 15:55:35 -0400
Subject: [Freeswitch-users] Question about sharing conference between servers
Message-ID: <55518.1250106935@a2unlimited.com>

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From terrymr at gmail.com  Wed Aug 12 13:25:56 2009
From: terrymr at gmail.com (Terry Moore-Read)
Date: Wed, 12 Aug 2009 13:25:56 -0700
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
References: <1249685648.16901.34.camel@dk-d820>
	<207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
Message-ID: <2d9dff7e0908121325g664dd580j9304d316ce4183ee@mail.gmail.com>

Macbook ... that's nothing,  I got $1500 worth of coffee :-)


On Wed, Aug 12, 2009 at 12:57 PM, jonathan
augenstine<jaugenstine at gmail.com> wrote:
> I to would like to put my thanks on the table. ?I have been going to
> conferences for a very long time and often question the value of taking time
> off to attend these venues. ?When I was asked to attend by a client I was
> very hesitant. ?I am very pleased that I decided to attend.
> Now the skeptical among you may say that I am only pleased because I won the
> MacBook. ?I cannot deny there is truth in that statement. ?However, on
> Thursday morning I was sitting in the conference room waiting for the first
> presentation. ?I was thinking that it had been a valuable three days. ?I had
> been able to connect with some clients at the conference, made some new
> contacts, met people face to face that I had only met online, listened to
> good presentations, learned some valuable new information, and lastly
> received some insight from Anthony and Michael Jerris on fixing a bug that
> had been plaguing me for sometime.
> As I sat pondering,?I clicked on the picture of the MacBook and thought the
> only way the conference could end better was if I won the MacBook. ?Never
> did I actually think that would happen and the conference did end better. ?I
> will be much more motivated to consider attending next year, even knowing
> lightening does not strike the same spot twice.
> Jonathan Augenstine
> On Fri, Aug 7, 2009 at 3:54 PM, David Knell <dave at 3c.co.uk> wrote:
>>
>> Just a quick note to say thanks to Cluecon's organisers for putting
>> together such a useful, informative and packed three days. ?I've come
>> away with a head full of ideas, a bunch of new contacts and a collection
>> of things to do; I'd thoroughly recommend that anyone interested in IP
>> telephony blocks out the first week of August 2010, right now..!
>>
>> Cheers --
>>
>> Dave
>>
>> --
>> David Knell, Director, 3C Limited
>> T: +44 20 3298 2000
>> E: dave at 3c.co.uk
>> W: http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Want to buy my photo's ? :
http://www.shutterstock.com/gallery.mhtml?id=309295&rid=309295



From moises.silva at gmail.com  Wed Aug 12 13:33:35 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 12 Aug 2009 16:33:35 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <4A832064.8000908@gmx.net>
References: <4A832064.8000908@gmx.net>
Message-ID: <c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>

On Wed, Aug 12, 2009 at 4:04 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> anybody has a clue what this message means?
>  [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from
> PROGRESS_MEDIA to PROGRESS
> What does VETO mean here?
>
> Best regards
> Peter
>

Means that state transition should not occur. The only thing that it would
cause that (I think) is a bug in the the openzap code.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From brian at freeswitch.org  Wed Aug 12 13:42:08 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 15:42:08 -0500
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <2d9dff7e0908121325g664dd580j9304d316ce4183ee@mail.gmail.com>
References: <1249685648.16901.34.camel@dk-d820>
	<207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
	<2d9dff7e0908121325g664dd580j9304d316ce4183ee@mail.gmail.com>
Message-ID: <FCF9FC37-0377-4947-BEC7-9264166574BF@freeswitch.org>

So you're the one that drank 16 gallons of coffee!  Good luck sleeping!

/b

On Aug 12, 2009, at 3:25 PM, Terry Moore-Read wrote:

> Macbook ... that's nothing,  I got $1500 worth of coffee :-)




From brian at freeswitch.org  Wed Aug 12 13:42:47 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 15:42:47 -0500
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
Message-ID: <15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>

Isn't progress_media already past progress in the state machine?  so  
the state machine can't move backwards in states right?

/b

On Aug 12, 2009, at 3:33 PM, Moises Silva wrote:

> On Wed, Aug 12, 2009 at 4:04 PM, Peter P GMX <Prometheus001 at gmx.net>  
> wrote:
> Hello,
>
> anybody has a clue what this message means?
>  [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from
> PROGRESS_MEDIA to PROGRESS
> What does VETO mean here?
>
> Best regards
> Peter
>
> Means that state transition should not occur. The only thing that it  
> would cause that (I think) is a bug in the the openzap code.
>

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From moises.silva at gmail.com  Wed Aug 12 13:57:37 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 12 Aug 2009 16:57:37 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
Message-ID: <c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>

Correct, so the question is why ozmod_libpri attempting to move from
progress_media to progress ... may be a delayed libpri event? or some crap
along those lines.

On Wed, Aug 12, 2009 at 4:42 PM, Brian West <brian at freeswitch.org> wrote:

> Isn't progress_media already past progress in the state machine?  so the
> state machine can't move backwards in states right?
> /b
>
> On Aug 12, 2009, at 3:33 PM, Moises Silva wrote:
>
> On Wed, Aug 12, 2009 at 4:04 PM, Peter P GMX <Prometheus001 at gmx.net>
> wrote:
>
>> Hello,
>>
>> anybody has a clue what this message means?
>>  [WARNING] ozmod_libpri.c:729 VETO Changing state on 1:1 from
>> PROGRESS_MEDIA to PROGRESS
>> What does VETO mean here?
>>
>> Best regards
>> Peter
>>
>
> Means that state transition should not occur. The only thing that it would
> cause that (I think) is a bug in the the openzap code.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From brian at freeswitch.org  Wed Aug 12 14:01:49 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 16:01:49 -0500
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
Message-ID: <90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>

Yes you can get a progress after you get a progress with media ... I  
have seen it.

/b

On Aug 12, 2009, at 3:57 PM, Moises Silva wrote:

> Correct, so the question is why ozmod_libpri attempting to move from  
> progress_media to progress ... may be a delayed libpri event? or  
> some crap along those lines.




From msc at freeswitch.org  Wed Aug 12 14:07:52 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Aug 2009 16:07:52 -0500
Subject: [Freeswitch-users] ClueCon Presentations - Where?
In-Reply-To: <65BC9732-5836-48F4-83AC-B305F4D1028C@freeswitch.org>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
	<D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com>
	<c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>
	<c29b930b0908121134m50b1c8b2j3a91b57e71b70b10@mail.gmail.com>
	<F571A97F57A2D946B229C8231805867D32AD4527@svanexcsvr01.alldir.net>
	<65BC9732-5836-48F4-83AC-B305F4D1028C@freeswitch.org>
Message-ID: <87f2f3b90908121407n77efcde2v374008b07af77e92@mail.gmail.com>

On Wed, Aug 12, 2009 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:

> They are all getting gathered up and put online...
> files.freeswitch.org/cluecon_2009 just keep an eye there some of the
> videos are up also.
> /b
>

FYI,

I've uploaded the first batch and they should get synched up on
files.freeswitch.org/cluecon_2009/presentations any time...
-MC
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From msc at freeswitch.org  Wed Aug 12 14:10:39 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Aug 2009 16:10:39 -0500
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
Message-ID: <87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>

On Wed, Aug 12, 2009 at 4:01 PM, Brian West <brian at freeswitch.org> wrote:

> Yes you can get a progress after you get a progress with media ... I
> have seen it.
>

Yes, you definitely can and I believe that some of the PRI specs suggest
that this is totally legal, even though it's kind of silly.
-MC
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From terrymr at gmail.com  Wed Aug 12 14:27:19 2009
From: terrymr at gmail.com (Terry Moore-Read)
Date: Wed, 12 Aug 2009 14:27:19 -0700
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <FCF9FC37-0377-4947-BEC7-9264166574BF@freeswitch.org>
References: <1249685648.16901.34.camel@dk-d820>
	<207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
	<2d9dff7e0908121325g664dd580j9304d316ce4183ee@mail.gmail.com>
	<FCF9FC37-0377-4947-BEC7-9264166574BF@freeswitch.org>
Message-ID: <2d9dff7e0908121427u33151adcj2bf9fb1968338b21@mail.gmail.com>

That's the trouble with a 8am conference in a town where the bars
close at 4am :-)

On Wed, Aug 12, 2009 at 1:42 PM, Brian West<brian at freeswitch.org> wrote:
> So you're the one that drank 16 gallons of coffee! ?Good luck sleeping!
>
> /b
>
> On Aug 12, 2009, at 3:25 PM, Terry Moore-Read wrote:
>
>> Macbook ... that's nothing, ?I got $1500 worth of coffee :-)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Want to buy my photo's ? :
http://www.shutterstock.com/gallery.mhtml?id=309295&rid=309295



From brian at freeswitch.org  Wed Aug 12 14:30:55 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 16:30:55 -0500
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <2d9dff7e0908121427u33151adcj2bf9fb1968338b21@mail.gmail.com>
References: <1249685648.16901.34.camel@dk-d820>
	<207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com>
	<2d9dff7e0908121325g664dd580j9304d316ce4183ee@mail.gmail.com>
	<FCF9FC37-0377-4947-BEC7-9264166574BF@freeswitch.org>
	<2d9dff7e0908121427u33151adcj2bf9fb1968338b21@mail.gmail.com>
Message-ID: <82A46A35-0CFF-4A5F-8D5A-CF013937ABB0@freeswitch.org>

And it didn't help we had an open bar two of the nights!

/b

On Aug 12, 2009, at 4:27 PM, Terry Moore-Read wrote:

> That's the trouble with a 8am conference in a town where the bars
> close at 4am :-)




From gmaruzz at celliax.org  Wed Aug 12 14:40:30 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 12 Aug 2009 23:40:30 +0200
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <82A46A35-0CFF-4A5F-8D5A-CF013937ABB0@freeswitch.org>
References: <1249685648.16901.34.camel@dk-d820>
	<207e7a5e0908121257t342d6fd5ked056ecfa8c213d4@mail.gmail.com> 
	<2d9dff7e0908121325g664dd580j9304d316ce4183ee@mail.gmail.com> 
	<FCF9FC37-0377-4947-BEC7-9264166574BF@freeswitch.org>
	<2d9dff7e0908121427u33151adcj2bf9fb1968338b21@mail.gmail.com> 
	<82A46A35-0CFF-4A5F-8D5A-CF013937ABB0@freeswitch.org>
Message-ID: <7b197bef0908121440o57fde283l7cf1f7566781f730@mail.gmail.com>

it helped me!
oh... well, I helped myself!

-giovanni


On Wed, Aug 12, 2009 at 11:30 PM, Brian West<brian at freeswitch.org> wrote:
> And it didn't help we had an open bar two of the nights!
>
> /b
>
> On Aug 12, 2009, at 4:27 PM, Terry Moore-Read wrote:
>
>> That's the trouble with a 8am conference in a town where the bars
>> close at 4am :-)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From bjbrashier at gmail.com  Wed Aug 12 14:40:02 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Wed, 12 Aug 2009 14:40:02 -0700
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <4A830243.8080703@chandlerfamily.org.uk>
References: <4A830243.8080703@chandlerfamily.org.uk>
Message-ID: <7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>

You've thought through some of the difficult points, which is good. You're
right that the moderator can't have different controls (unless you're
controlling the conference yourself from outside, using, say, the event
socket).

Before I go further, I want to make sure I understand what you're proposing.
What you're essentially saying is that when the command to kick someone is
pressed the person should be transferred out of the conference, checked for
moderator status, asked whom to kick (if so), and then let back in while the
system kicks that person. The other commands would work similarly. Does this
sound like a correct summary?

If so, I think what you've got is mostly right, but I'm not sure it will
work, having not tried to do it that way myself (I went the event socket
route). In theory, it should look something like this:

In the scripts entering the conference, you'll need to save the current
conference number. Something like (in both cases):

        <action application="set" data="conf-id=$1"/>

"Extension" doesn't actually check for anything, so you'll need to check
for "kick" with a condition. Try this:

    <extension name="kick">
      <condition field="destination_number" expression="^kick$"/>
      <condition field="$(i_am_moderator)" expression="true">
        <action application="play_and_get_digits" data="3 3 3 7000 #
/ask-for-extension.wav /invalid.wav conf-user-id \d+" >

Followed by (I picked up the "kick" syntax from the wiki):

        <action application="conference" data="$(conf-id) kick
$(conf-user-id)"/>
        <action application="conference"
data="$(conf-id)@default+flags{mod}"/>
        <anti-action application="conference" data="$(conf-id)@default"/>
      </condition>
    </extension>

In theory, then, you've transferred out with the "9" key, you check the
moderator flag you've made, and do stuff based on the conf-id and the data
you enter.  Obviously, if this works, the moderator won't be able to hear
what's going on in the conference while he's entering stuff.

Bigger problem: the conf-user-id he enters has to be the member ID that FS
chose for the user he's trying to kick. That number will make sense if
you're following the system closely, but for someone who doesn't know FS
internals, it will be impossible to know unless you broadcast it to him
somehow. To my knowledge, there's no way to use any other identifier (like
caller-id) to kick them with.

BB

On Wed, Aug 12, 2009 at 10:56 AM, Alan Chandler
<alan at chandlerfamily.org.uk>wrote:

> I have been reading all the docs about conferences I can find and am
> getting somewhat confused.  What I am trying to do is set up a dialplan
> where I have subscribers with extensions in the 1xx range, and then to
> set an ability to have a series of conference rooms for each subscriber
> in the 21xx range where if the user enters is "own" conference he is the
> moderator, but if not he is just a normal user.
>
> I want to be able for the moderator to do things like mute or kick people.
>
> So dialplan would probably have something like this in it
>
>
>
>     <extension name="user_conference">
>       <condition field="destination_number"
> expression="^(2${caller_id_number})$">
>         <action application="set" data="i_am_moderator=true">
>        <action application="answer"/>
>        <action application="conference" data="$1 at default
> +flags{moderator}"/>
>       </condition>
>       <condition field="destination_number" expression="^(21\d{2})^">
>         <action application="set" data="i_am_moderator=false">
>        <action application="answer"/>
>        <action application="conference" data="$1 at default"/>
>       </condition>
>     </extension>
>     <extension name="kick">
>       <condition field="i_am_moderator" expression="true">
>         <action application="play_and_get_digits" data="3 3 3 7000 #
> /ask-for-extension.wav /invalid.wav conf-user-id \d+" >
>
> WHAT GOES HERE???
>
>       </condition>
>     </extension>
>
>
> in the conference.conf.xml file, I would change the caller controls to
> include
>
>   <caller-controls>
>     <group name="somekeys">
>        <control action="transfer" digits="9"   data="kick XML default"/>
>     </group>
>   </caller-controls>
>
>
> My question (at the moment) is
>
> In the WHAT GOES HERE place how do it
>
> Kick extension ${conf-user-id}  (DOES IT REQUIRE A SCRIPT TO CALL THE
> Conference API?)
> Re-Enter the moderator back into the conference
> Re-Enter the ordinary user who happened to press 9 back into the conference
>
> I am assuming I can't stop the non moderator getting the control - since
> all users get the same controls.
>
>
> --
> Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Wed Aug 12 14:41:26 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Aug 2009 16:41:26 -0500
Subject: [Freeswitch-users] Question about sharing conference between
	servers
In-Reply-To: <55518.1250106935@a2unlimited.com>
References: <55518.1250106935@a2unlimited.com>
Message-ID: <87f2f3b90908121441p741e73cco4a947c0115722082@mail.gmail.com>

On Wed, Aug 12, 2009 at 2:55 PM, Tina Martinez <tina at a2unlimited.com> wrote:

> Hello,
>
> I have spent the past couple of weeks toying around with FS to evaluate the
> possibility of using it for a large scale conference server for our
> organization.  The plan is to have several FS servers initiate calls to
> participants and connect them together, but not have to transfer all of the
> calls to one server supporting the conference.  My problem is that I do not
> see a simple way to link the servers together.  Has anyone done anything
> like this, or know of a way to create a connection between servers without
> an actual caller on the line?  I started to go down a path of registering a
> soft-phone on each machine and place a call those extensions prior to
> initiating the conference call -- then connecting, but that feels like a
> kludge.
>
> Any thoughts, suggestions or guidance would be greatly appreciated.
> T


Tina,

Welcome to the FreeSWITCH community! I'm pretty sure that FS can do what you
need. My first question to you is this: do envision having some sort of call
control mechanism, like a script or something that knows which callers to
connect to and which ones to put into conferences, etc.? The reason I ask is
that controlling calls using the event-socket allows for great power and
flexibility.

As for "connecting" the FS servers you have several options. You can use
ACLs to allow the servers to call each other via SIP without needing
authentication. You could also use gateways to allow FS machines to do SIP
registrations with each other. (I suppose it depends on your exact needs,
but ACLs are pretty easy and clean.) Once you have FS machines able to dial
each other then it's pretty much a matter of configuring your dialplans to
route the calls. You can create conferences on the fly so that part is easy.
The tricky part will be controlling how to link the conferences together,
although FS has a handy API to add a conference member to an existing
conference by dialing out. (See
http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference)

Let's say you have two servers, FS_A and FS_B. Each server can dial the
other and each server has a conference running, named FS_A_Conf and
FS_B_Conf, respectively. You can have either conference dial the other
server. For example, from the FS_A command line:

conference FS_A_Conf dial
{originate_timeout=30}sofia/internal/3900 at FS_B.IP.Address 1111 FS_A_Conf

Where 3900 is an extension set up in FS_B's dialplan to route to FS_B_Conf
and "1111 FS_A_Conf" are the caller ID number and name that FS_B should
receive from FS_A.

You still need to figure out other things like if you want the ability to
"break apart" the two conferences after they've been connected together,
etc. This is definitely doable but you'll need to handle the call control
stuff some how.

HTH,
MC
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From bjbrashier at gmail.com  Wed Aug 12 15:18:48 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Wed, 12 Aug 2009 15:18:48 -0700
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>
References: <4A830243.8080703@chandlerfamily.org.uk>
	<7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>
Message-ID: <7bcfdd290908121518i42b8eebv6a96f20dea5c3078@mail.gmail.com>

Whoops.  All of my parens () should be curly braces {}. Wasn't paying
attention.

BB

On Wed, Aug 12, 2009 at 2:40 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> You've thought through some of the difficult points, which is good. You're
> right that the moderator can't have different controls (unless you're
> controlling the conference yourself from outside, using, say, the event
> socket).
>
> Before I go further, I want to make sure I understand what you're
> proposing. What you're essentially saying is that when the command to kick
> someone is pressed the person should be transferred out of the conference,
> checked for moderator status, asked whom to kick (if so), and then let back
> in while the system kicks that person. The other commands would work
> similarly. Does this sound like a correct summary?
>
> If so, I think what you've got is mostly right, but I'm not sure it will
> work, having not tried to do it that way myself (I went the event socket
> route). In theory, it should look something like this:
>
> In the scripts entering the conference, you'll need to save the current
> conference number. Something like (in both cases):
>
>         <action application="set" data="conf-id=$1"/>
>
> "Extension" doesn't actually check for anything, so you'll need to check
> for "kick" with a condition. Try this:
>
>     <extension name="kick">
>       <condition field="destination_number" expression="^kick$"/>
>       <condition field="$(i_am_moderator)" expression="true">
>         <action application="play_and_get_digits" data="3 3 3 7000 #
> /ask-for-extension.wav /invalid.wav conf-user-id \d+" >
>
> Followed by (I picked up the "kick" syntax from the wiki):
>
>         <action application="conference" data="$(conf-id) kick
> $(conf-user-id)"/>
>         <action application="conference"
> data="$(conf-id)@default+flags{mod}"/>
>         <anti-action application="conference" data="$(conf-id)@default"/>
>       </condition>
>     </extension>
>
> In theory, then, you've transferred out with the "9" key, you check the
> moderator flag you've made, and do stuff based on the conf-id and the data
> you enter.  Obviously, if this works, the moderator won't be able to hear
> what's going on in the conference while he's entering stuff.
>
> Bigger problem: the conf-user-id he enters has to be the member ID that FS
> chose for the user he's trying to kick. That number will make sense if
> you're following the system closely, but for someone who doesn't know FS
> internals, it will be impossible to know unless you broadcast it to him
> somehow. To my knowledge, there's no way to use any other identifier (like
> caller-id) to kick them with.
>
> BB
>
> On Wed, Aug 12, 2009 at 10:56 AM, Alan Chandler <
> alan at chandlerfamily.org.uk> wrote:
>
>> I have been reading all the docs about conferences I can find and am
>> getting somewhat confused.  What I am trying to do is set up a dialplan
>> where I have subscribers with extensions in the 1xx range, and then to
>> set an ability to have a series of conference rooms for each subscriber
>> in the 21xx range where if the user enters is "own" conference he is the
>> moderator, but if not he is just a normal user.
>>
>> I want to be able for the moderator to do things like mute or kick people.
>>
>> So dialplan would probably have something like this in it
>>
>>
>>
>>     <extension name="user_conference">
>>       <condition field="destination_number"
>> expression="^(2${caller_id_number})$">
>>         <action application="set" data="i_am_moderator=true">
>>        <action application="answer"/>
>>        <action application="conference" data="$1 at default
>> +flags{moderator}"/>
>>       </condition>
>>       <condition field="destination_number" expression="^(21\d{2})^">
>>         <action application="set" data="i_am_moderator=false">
>>        <action application="answer"/>
>>        <action application="conference" data="$1 at default"/>
>>       </condition>
>>     </extension>
>>     <extension name="kick">
>>       <condition field="i_am_moderator" expression="true">
>>         <action application="play_and_get_digits" data="3 3 3 7000 #
>> /ask-for-extension.wav /invalid.wav conf-user-id \d+" >
>>
>> WHAT GOES HERE???
>>
>>       </condition>
>>     </extension>
>>
>>
>> in the conference.conf.xml file, I would change the caller controls to
>> include
>>
>>   <caller-controls>
>>     <group name="somekeys">
>>        <control action="transfer" digits="9"   data="kick XML default"/>
>>     </group>
>>   </caller-controls>
>>
>>
>> My question (at the moment) is
>>
>> In the WHAT GOES HERE place how do it
>>
>> Kick extension ${conf-user-id}  (DOES IT REQUIRE A SCRIPT TO CALL THE
>> Conference API?)
>> Re-Enter the moderator back into the conference
>> Re-Enter the ordinary user who happened to press 9 back into the
>> conference
>>
>> I am assuming I can't stop the non moderator getting the control - since
>> all users get the same controls.
>>
>>
>> --
>> Alan Chandler
>> http://www.chandlerfamily.org.uk
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
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From moises.silva at gmail.com  Wed Aug 12 15:28:59 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Wed, 12 Aug 2009 18:28:59 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
	<87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
Message-ID: <c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>

then probably we should check the current state and ignore the libpri event
when already in progress with media.

On Wed, Aug 12, 2009 at 5:10 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 12, 2009 at 4:01 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Yes you can get a progress after you get a progress with media ... I
>> have seen it.
>>
>
> Yes, you definitely can and I believe that some of the PRI specs suggest
> that this is totally legal, even though it's kind of silly.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From alan at chandlerfamily.org.uk  Wed Aug 12 15:33:48 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Wed, 12 Aug 2009 23:33:48 +0100
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>
References: <4A830243.8080703@chandlerfamily.org.uk>
	<7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>
Message-ID: <4A83434C.3090505@chandlerfamily.org.uk>

Bradley Brashier wrote:
...
> Before I go further, I want to make sure I understand what you're 
> proposing. What you're essentially saying is that when the command to 
> kick someone is pressed the person should be transferred out of the 
> conference, checked for moderator status, asked whom to kick (if so), 
> and then let back in while the system kicks that person. The other 
> commands would work similarly. Does this sound like a correct summary?

Absolutely what I am trying to do.

...
>  
> Followed by (I picked up the "kick" syntax from the wiki):
>  
>         <action application="conference" data="$(conf-id) kick 
> $(conf-user-id)"/>

This is a significant new fact for me.  What you seem to be doing is 
calling the commands referenced in the conference api here

http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference

by using application="conference" and then the data string as the second 
part of the command.  Am I correct in the assumption that you can do this.


>  
> Bigger problem: the conf-user-id he enters has to be the member ID that 
> FS chose for the user he's trying to kick. That number will make sense 
> if you're following the system closely, but for someone who doesn't know 
> FS internals, it will be impossible to know unless you broadcast it to 
> him somehow. To my knowledge, there's no way to use any other identifier 
> (like caller-id) to kick them with.

If my assumption about the use of the API above is correct, then 
couldn't I do something like

<action application="set" data="member-list=${conference(${conf-id} 
list)}"/>
<action application="execute_extension" data="do-kick"/>
and the figure out a regular expression to get participant-id and 
caller-id out of the resultant string.

Or is this not how its done?




-- 
Alan Chandler
http://www.chandlerfamily.org.uk




From bjbrashier at gmail.com  Wed Aug 12 16:09:51 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Wed, 12 Aug 2009 16:09:51 -0700
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <4A83434C.3090505@chandlerfamily.org.uk>
References: <4A830243.8080703@chandlerfamily.org.uk>
	<7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com> 
	<4A83434C.3090505@chandlerfamily.org.uk>
Message-ID: <7bcfdd290908121609i5289fae6n40b1be48b0173ff0@mail.gmail.com>

>This is a significant new fact for me.  What you seem to be doing is
>calling the commands referenced in the conference api here
>
>http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
>
>by using application="conference" and then the data string as the second
>part of the command.  Am I correct in the assumption that you can do this.

I agree that that's what it looks like. What I don't know is if it works. I
got this example from the page
http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did
exactly what you're trying, and never tried using the API in this fashion.

><action application="set" data="member-list=${conference(${conf-id}
>list)}"/>
><action application="execute_extension" data="do-kick"/>
>and the figure out a regular expression to get participant-id and
>caller-id out of the resultant string.
>
>Or is this not how its done?

Even assuming that our understanding on the kick command is correct, and
there's an accessible API here, I don't believe the above would actually
populate your variable with the data that you want. I don't think it
actually issues the command with the "set" application. Instead, you'd need
to do something like

<action application="conference" data="${conf-id} list"/>

and then you'd have to capture the output somehow, which I also don't
believe is possible.

I'm hardly a master of XML, though, or of dialplans, so feel free to try it
if you want. Just make sure you're logging everything so you can watch it
all unfold.

BB
On Wed, Aug 12, 2009 at 3:33 PM, Alan Chandler
<alan at chandlerfamily.org.uk>wrote:

> Bradley Brashier wrote:
> ...
> > Before I go further, I want to make sure I understand what you're
> > proposing. What you're essentially saying is that when the command to
> > kick someone is pressed the person should be transferred out of the
> > conference, checked for moderator status, asked whom to kick (if so),
> > and then let back in while the system kicks that person. The other
> > commands would work similarly. Does this sound like a correct summary?
>
> Absolutely what I am trying to do.
>
> ...
> >
> > Followed by (I picked up the "kick" syntax from the wiki):
> >
> >         <action application="conference" data="$(conf-id) kick
> > $(conf-user-id)"/>
>
> This is a significant new fact for me.  What you seem to be doing is
> calling the commands referenced in the conference api here
>
> http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
>
> by using application="conference" and then the data string as the second
> part of the command.  Am I correct in the assumption that you can do this.
>
>
> >
> > Bigger problem: the conf-user-id he enters has to be the member ID that
> > FS chose for the user he's trying to kick. That number will make sense
> > if you're following the system closely, but for someone who doesn't know
> > FS internals, it will be impossible to know unless you broadcast it to
> > him somehow. To my knowledge, there's no way to use any other identifier
> > (like caller-id) to kick them with.
>
> If my assumption about the use of the API above is correct, then
> couldn't I do something like
>
> <action application="set" data="member-list=${conference(${conf-id}
> list)}"/>
> <action application="execute_extension" data="do-kick"/>
> and the figure out a regular expression to get participant-id and
> caller-id out of the resultant string.
>
> Or is this not how its done?
>
>
>
>
> --
>  Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tina at a2unlimited.com  Wed Aug 12 16:10:59 2009
From: tina at a2unlimited.com (Tina Martinez)
Date: Wed, 12 Aug 2009 19:10:59 -0400
Subject: [Freeswitch-users]  Question about sharing conference between
Message-ID: <41480.1250118659@a2unlimited.com>

Michael,

Thanks for the welcome, and for the response to my question.

The call control and dynamic setup of conferences I have working (pretty cool stuff).
The tricky part, as you said, is "linking" the servers together.

Basically, what I need to do is establish a connection that will not be dependent
on a live person being on the call.  And I would prefer to avoid having to
register actual phone extensions for every server -- and for every conference call.

I apologize if I'm slow, but I'm new to working an application like this.  

In your example, you stated, "3900 is an extension set up in FS_B's dialplan",
does this extension have to be a live person (or soft-phone connected using an
auto-answer mechanism)?  or can I setup something where there is not a phone
actually connected?

Right now I'm able to place a call from one machine to another, but I'm calling
an X-Lite phone on the second server.

Also, if it is possible to call a "virtual" extension, I have no problem
incorporating application logic that would clean-up the orphaned conferences on
all machines when a conference call is complete.  I'm more concerned with being
able to setup the "links" quickly and in an elegant fashion.

- T






From brian at freeswitch.org  Wed Aug 12 16:19:08 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 12 Aug 2009 18:19:08 -0500
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
	<87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
	<c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>
Message-ID: <A681A304-5B94-49B8-BA25-E3089ED3F4A9@freeswitch.org>

Well you really can't ignore it... it happens with our ISDN stack  
too.   Thats what the VETO handles.

/b

On Aug 12, 2009, at 5:28 PM, Moises Silva wrote:

> then probably we should check the current state and ignore the  
> libpri event when already in progress with media.




From msc at freeswitch.org  Wed Aug 12 17:34:38 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 12 Aug 2009 19:34:38 -0500
Subject: [Freeswitch-users] Question about sharing conference between
In-Reply-To: <41480.1250118659@a2unlimited.com>
References: <41480.1250118659@a2unlimited.com>
Message-ID: <87f2f3b90908121734r78b9830bw38a135c340a6a845@mail.gmail.com>

On Wed, Aug 12, 2009 at 6:10 PM, Tina Martinez <tina at a2unlimited.com> wrote:

> Michael,
>
> Thanks for the welcome, and for the response to my question.
>
> The call control and dynamic setup of conferences I have working (pretty
> cool stuff).
> The tricky part, as you said, is "linking" the servers together.
>
> Basically, what I need to do is establish a connection that will not be
> dependent
> on a live person being on the call.  And I would prefer to avoid having to
> register actual phone extensions for every server -- and for every
> conference call.
>
> I apologize if I'm slow, but I'm new to working an application like this.


No worries. :)


>
>
> In your example, you stated, "3900 is an extension set up in FS_B's
> dialplan",
> does this extension have to be a live person (or soft-phone connected using
> an
> auto-answer mechanism)?  or can I setup something where there is not a
> phone
> actually connected?
>

No actual telephone is needed. Here's an example dialplan snippet that you
could drop right into conf/dialplan/default/ in a new file. (I prefer to put
my own custom dialplan entries into a separate file instead of editing
default.xml)

<include>
  <extension name="Sample Conference at 3900">
    <condition field="destination_number" expression="^3900$">
      <action application="conference" data="Conf at FS_B"/>
    </condition>
  </extension>
</include>

Put the above into conf/dialplan/default/01_ConfB.xml on the "FS_B" server.
(You can make a similar file on the "FS_A" server or any other server if
you'd like.)

You will also need to create a "public" extension which will route the
inbound calls appropriately. (If that doesn't make any sense right now then
don't worry, just do it. :)

Put the following into a file named conf/dialplan/public/01_ConfB.xml:
<include>
  <extension name="Sample Conference at 3900">
    <condition field="destination_number" expression="^3900$">
      <action application="transfer" data="3900 XML default"/>
    </condition>
  </extension>
</include>

(Again, you can do the same on all of your servers - this will allow all
servers to receive calls and route them to x3900.)

Now that you've got 3900 set up you can test it. Press F6 (or type
"reloadxml") at the CLI for FS_B. Then, have a phone that is registered to
FS_B make a call to 3900. It should be alone in the conference.

Now you'll need to set up some sort of dialplan routing to call from FS_A to
FS_B, unless you have a SIP phone registered to FS_A that can dial a SIP
URI. The SIP URI is: sip:3900 at FS_B.IP.Address. For kicks, let's add a simple
dialplan extension on FS_A that allows you to dial "23900" to get to FS_B's
3900 extension. Put this into conf/dialplan/default/01_Dial_ConfB.xml on
FS_A:

<include>
  <extension name="Sample Conference at 3900">
    <condition field="destination_number" expression="^2(3900)$">
      <action application="bridge" data="sofia/internal/$1 at FS_B.IP.Address
"/>
    </condition>
  </extension>
</include>

Save, and do the reloadxml (or F6) thing on FS_A CLI. Now on FS_A you can
dial "23900" and it will ring right into 3900 on FS_B so that the phone
registered at FS_A is in the conference on FS_B. Got it?

Have fun tinkering and let us know how it all goes.
-MC



> Right now I'm able to place a call from one machine to another, but I'm
> calling
> an X-Lite phone on the second server.
>
> Also, if it is possible to call a "virtual" extension, I have no problem
> incorporating application logic that would clean-up the orphaned
> conferences on
> all machines when a conference call is complete.  I'm more concerned with
> being
> able to setup the "links" quickly and in an elegant fashion.
>
> - T
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From string01 at gmail.com  Wed Aug 12 17:42:02 2009
From: string01 at gmail.com (String Larson)
Date: Wed, 12 Aug 2009 19:42:02 -0500
Subject: [Freeswitch-users] Setting max inbound for UA
Message-ID: <34D23392-2847-41BA-81A1-EF173B2F5061@gmail.com>

Is there a way to limit the number of calls a UA can receive in the FS  
configs?

I'm doing some testing with XLite as the UA, and can not figure out  
how to keep line 2 from answering if line 1 is in use.

THanks.

-str



From krice at freeswitch.org  Wed Aug 12 19:22:49 2009
From: krice at freeswitch.org (Ken Rice)
Date: Wed, 12 Aug 2009 21:22:49 -0500
Subject: [Freeswitch-users] Setting max inbound for UA
In-Reply-To: <34D23392-2847-41BA-81A1-EF173B2F5061@gmail.com>
Message-ID: <C6A8E329.5ACC0%krice@freeswitch.org>

Check out mod_limit... Other wise you'll have to look specifically at the UA
you are trying to use, some like polycom and sipura offer a way to disable
"call waiting"

Remember with SIP there is no such thing as a line, its a SESSION and you
can have as many sessions as the software allows (and most software doesn't
put sane limits based on CPU/RAM/Bandwidth etc)


> From: String Larson <string01 at gmail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Wed, 12 Aug 2009 19:42:02 -0500
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Setting max inbound for UA
> 
> Is there a way to limit the number of calls a UA can receive in the FS
> configs?
> 
> I'm doing some testing with XLite as the UA, and can not figure out
> how to keep line 2 from answering if line 1 is in use.
> 
> THanks.
> 
> -str
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From jlenk at frontiernet.net  Wed Aug 12 21:11:33 2009
From: jlenk at frontiernet.net (Jeff Lenk)
Date: Wed, 12 Aug 2009 23:11:33 -0500 (CDT)
Subject: [Freeswitch-users] OpenZAP/Sangoma in Windows
In-Reply-To: <c4d05cbe0908110759j2c929796g1ce96f530b7ad681@mail.gmail.com>
References: <549CFEF87AEDE841A38E9D15EAB4C04C52ECED9435@cooper>
	<c4d05cbe0908110759j2c929796g1ce96f530b7ad681@mail.gmail.com>
Message-ID: <1250136693271-3435444.post@n2.nabble.com>


I have been testing analog support under Windows with good results so far. I
am waiting on another driver fix to solve some small problems with the api.

I have been running this code for 6 weeks or so now under a home test
environment(light call traffic) and the reliabilty has been fine - no errors
or other abnormalites.

The openzap for windows code is very simular to the others now(with the new
LibSangoma) so we should be in pretty good shape.

I would love to hear more from you regarding your testing of PRI support
under windows.

-Jeff


Moises Silva wrote:
> 
> On Tue, Aug 11, 2009 at 2:16 AM, Peter Olsson <
> peter.olsson at visionutveckling.se> wrote:
> 
>> Hi, I'm trying to evaluate the OpenZAP/Sangoma-support in Windows, using
>> PRI E1 connections.
>>
> 
> Thanks for testing this :-)
> 
> I have been meaning to install FreeSWITCH on Windows but just could not
> find
> the time.
> 
> 
>> 1. Has anyone tested it in Windows at all? I know the build-files for
>> Visual Studio has only been checked in for a couple of months, so that's
>> why
>> I'm asking.
>>
> 
> The drivers have been tested quite well but not using FreeSWITCH.
> 
> 
>> 2. Does anyone have any directions how to configure the driver within
>> Windows? Should I use BitStream or HDLC, and how should the channel
>> groups
>> be configured?
>>
> 
> You should use HDLC for the D-channel and Bitstream for the B-Channels.
> Typically you would create 2 groups, one with channels 1-23 and the other
> with just channel 24. The first group would work in bitstream and
> TDM_CHAN_VOICE_API operational mode and the second in HDLC/API mode.
> 
> You can find me in #openzap, #freeswitch or #freeswitch-dev as "moy" if
> you
> have more questions.
> 
> -- 
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3
> Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://n2.nabble.com/OpenZAP-Sangoma-in-Windows-tp3422060p3435444.html
Sent from the freeswitch-users mailing list archive at Nabble.com.



From diego.viola at gmail.com  Wed Aug 12 23:47:07 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Thu, 13 Aug 2009 02:47:07 -0400
Subject: [Freeswitch-users] ClueCon Presentations - Where?
In-Reply-To: <87f2f3b90908121407n77efcde2v374008b07af77e92@mail.gmail.com>
References: <c29b930b0908121059p3a540759yd48458e81cf6aa@mail.gmail.com>
	<D1A3A4B9-2093-4CD2-BA27-D32642F9D88A@jerris.com>
	<c29b930b0908121128g7485bb1fkff9a43233a3c6437@mail.gmail.com>
	<c29b930b0908121134m50b1c8b2j3a91b57e71b70b10@mail.gmail.com>
	<F571A97F57A2D946B229C8231805867D32AD4527@svanexcsvr01.alldir.net>
	<65BC9732-5836-48F4-83AC-B305F4D1028C@freeswitch.org>
	<87f2f3b90908121407n77efcde2v374008b07af77e92@mail.gmail.com>
Message-ID: <86a32abc0908122347y4f9a702dg7f11615597311ce9@mail.gmail.com>

Just seen Anthony presentation, very cool ;)

Everyone, watch it!

http://files.freeswitch.org/cluecon_2009/presentations/Day%2001%20Presentation%2002.Anthony%20Minessale.1500kbps.mp4

=D

On Wed, Aug 12, 2009 at 5:07 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 12, 2009 at 2:25 PM, Brian West <brian at freeswitch.org> wrote:
>
>> They are all getting gathered up and put online...
>> files.freeswitch.org/cluecon_2009 just keep an eye there some of the
>> videos are up also.
>> /b
>>
>
> FYI,
>
> I've uploaded the first batch and they should get synched up on
> files.freeswitch.org/cluecon_2009/presentations any time...
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From alan at chandlerfamily.org.uk  Thu Aug 13 00:11:48 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Thu, 13 Aug 2009 08:11:48 +0100
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <7bcfdd290908121609i5289fae6n40b1be48b0173ff0@mail.gmail.com>
References: <4A830243.8080703@chandlerfamily.org.uk>	<7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>
	<4A83434C.3090505@chandlerfamily.org.uk>
	<7bcfdd290908121609i5289fae6n40b1be48b0173ff0@mail.gmail.com>
Message-ID: <4A83BCB4.2050604@chandlerfamily.org.uk>

Bradley Brashier wrote:
 >  I wrote:
>  >This is a significant new fact for me.  What you seem to be doing is
>  >calling the commands referenced in the conference api here
>  >
>  >http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
>  >
>  >by using application="conference" and then the data string as the second
>  >part of the command.  Am I correct in the assumption that you can do this.
>  
> I agree that that's what it looks like. What I don't know is if it 
> works. I got this example from the page 
> http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did 
> exactly what you're trying, and never tried using the API in this fashion.

I just found this - which I think helps

http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan

An API can be called from the dialplan but it is not recommended. Example:

  <extension name="Make API call from Dialplan">
    <condition field="destination_number" expression="^(999)$">
      <!-- next line calls hupall, so be careful! -->
      <action application="set" 
data="api_result=${hupall(normal_clearing)}"/>
    </condition>
  </extension>

Anyway - thanks for you help - I am going away to rethink that 
particular interface again.  Its getting so complicated that it might be 
better to copy the Javascript approach in the examples.



-- 
Alan Chandler
http://www.chandlerfamily.org.uk




From diego.viola at gmail.com  Thu Aug 13 01:03:24 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Thu, 13 Aug 2009 04:03:24 -0400
Subject: [Freeswitch-users] answer command
In-Reply-To: <6539128A-D7A1-4555-A3DB-60E94E461DB3@jerris.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
	<B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
	<24941422.post@talk.nabble.com>
	<6539128A-D7A1-4555-A3DB-60E94E461DB3@jerris.com>
Message-ID: <86a32abc0908130103h6c0093d4hcef03ceed2f2aa3@mail.gmail.com>

Hey Michael,

Just wondering something, I have found that you added
"conference_set_auto_outcall" on the dptools wiki, but I could not find that
function in the mod_dptools.c, shouldn't that be part of the mod_conference
wiki article? =D.

Best regards,

Diego

On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris <mike at jerris.com> wrote:

> Sip does not support this functionality.  The called device would have
> to support this via some other mechanism such as ctsa which I have
> seen recently someone was looking at for freeswitch.  So the first
> issue you must resolve is the called device needs to support some way
> to do this.
>
> Mike
>
> On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov <maxim.tsvetov at gmail.com>
> wrote:
>
> >
> > If I have two FS extensions A and B. I'm calling from A to B and
> > want to
> > answer from B-side in my CTI application and to make SIP phone to be
> > synchronised to my CTI application. Is it possible to do it?
> >
> >
> > Brian West-3 wrote:
> >>
> >> Well you can only truly answer an inbound call to FS... you can't
> >> force answer an outbound call.
> >>
> >> /b
> >>
> >> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
> >>
> >>>
> >>> I will try to paraphrase my question.
> >>> Is there any possibility to answer call  from CTI application and
> >>> synchronise answer with answer in SIP client?Maybe we can use SIP
> >>> functions
> >>> in our CTI application instead of FS api commands?
> >>> I'm trying to find the way to make prototype of lineAnswer command
> >>> in TAPI.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > --
> > View this message in context:
> http://www.nabble.com/answer-command-tp24912812p24941422.html
> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From yivzhenko at mksat.net  Thu Aug 13 01:14:02 2009
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Thu, 13 Aug 2009 11:14:02 +0300
Subject: [Freeswitch-users] Grangstream Early Dial
Message-ID: <200908131114.02970.yivzhenko@mksat.net>

Hello all.

I want to use Grandstream Early Dial future.
How i can enable support 484 response?

I tried simply use
<action application="hangup" data="484"/>
and
<action application="respond" data="484"/>
on uncompleted extensions, 
but there is not work


Thanks.

Yuriy .




From tzury.by at reguluslabs.com  Thu Aug 13 01:40:55 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Thu, 13 Aug 2009 11:40:55 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
Message-ID: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>

Hi,

Googeling about this shows that FS aims to support this (in fact it supports
all 3: UDP/TCP/TLS).
Yet I could not find the way to configure FS in order to support that.
In fact, it does not work in my current install.
I have TLS configured and work, but could not make TCP works

thanks in advance

/tzury
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From mike at jerris.com  Thu Aug 13 06:10:57 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 13 Aug 2009 09:10:57 -0400
Subject: [Freeswitch-users] answer command
In-Reply-To: <86a32abc0908130103h6c0093d4hcef03ceed2f2aa3@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<87f2f3b90908111009g42caba70u7e108595a69d6000@mail.gmail.com>
	<86a32abc0908111110q26defd10ka3aa70c02235abfe@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
	<B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
	<24941422.post@talk.nabble.com>
	<6539128A-D7A1-4555-A3DB-60E94E461DB3@jerris.com>
	<86a32abc0908130103h6c0093d4hcef03ceed2f2aa3@mail.gmail.com>
Message-ID: <EA91B26D-1BB9-40BC-9D94-6614F9890456@jerris.com>

It probably belongs there.  It's a wiki, feel free to fix it.  What  
does this have to do with this thread?

On Aug 13, 2009, at 4:03 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Hey Michael,
>
> Just wondering something, I have found that you added  
> "conference_set_auto_outcall" on the dptools wiki, but I could not  
> find that function in the mod_dptools.c, shouldn't that be part of  
> the mod_conference wiki article? =D.
>
> Best regards,
>
> Diego
>
> On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> Sip does not support this functionality.  The called device would have
> to support this via some other mechanism such as ctsa which I have
> seen recently someone was looking at for freeswitch.  So the first
> issue you must resolve is the called device needs to support some way
> to do this.
>
> Mike
>
> On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov <maxim.tsvetov at gmail.com>
> wrote:
>
> >
> > If I have two FS extensions A and B. I'm calling from A to B and
> > want to
> > answer from B-side in my CTI application and to make SIP phone to be
> > synchronised to my CTI application. Is it possible to do it?
> >
> >
> > Brian West-3 wrote:
> >>
> >> Well you can only truly answer an inbound call to FS... you can't
> >> force answer an outbound call.
> >>
> >> /b
> >>
> >> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
> >>
> >>>
> >>> I will try to paraphrase my question.
> >>> Is there any possibility to answer call  from CTI application and
> >>> synchronise answer with answer in SIP client?Maybe we can use SIP
> >>> functions
> >>> in our CTI application instead of FS api commands?
> >>> I'm trying to find the way to make prototype of lineAnswer command
> >>> in TAPI.
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
> > --
> > View this message in context: http://www.nabble.com/answer-command-tp24912812p24941422.html
> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-------------- next part --------------
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From brian at freeswitch.org  Thu Aug 13 06:45:58 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 13 Aug 2009 08:45:58 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
Message-ID: <2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>

It just works... to force TCP you append ;transport=tcp

In reality you should be using SRV records.

/b

On Aug 13, 2009, at 3:40 AM, Tzury Bar Yochay wrote:

> Hi,
>
> Googeling about this shows that FS aims to support this (in fact it  
> supports all 3: UDP/TCP/TLS).
> Yet I could not find the way to configure FS in order to support that.
> In fact, it does not work in my current install.
> I have TLS configured and work, but could not make TCP works
>
> thanks in advance
>
> /tzury




From irmatov at gmail.com  Thu Aug 13 02:35:05 2009
From: irmatov at gmail.com (Timur Irmatov)
Date: Thu, 13 Aug 2009 14:35:05 +0500
Subject: [Freeswitch-users] calling through same gateway with multiple
	registrations
Message-ID: <241d382f0908130235w460339baxcc39527c03b0724@mail.gmail.com>

Hi,


I am new to FreeSWITCH and need an advice.

All calls to PSTN from our server will go through single gateway,
which is a soft switch supporting SIP protocol. FreeSWITCH will need
to register with soft switch, but soft switch permits only single
active call (in either direction) per registration. So we will need 10
SIP accounts to allow 10 simultaneous connections.

Question is - how should I configure FreeSWITCH for this scenario? I
see two options:

1) Create 10 gateways with different registrations, use mod_limit to
route only one outgoing call per gateway;
2) Create 10 gateways with different registrations, use event socket
to route calls manually and monitor used lines (incoming and outgoing
calls through soft switch).

Are there any other possibilities? Corrections/ suggestions are very welcome.


-- 
Timur Irmatov, xmpp:irmatov at jabber.ru



From brian at freeswitch.org  Thu Aug 13 06:47:18 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 13 Aug 2009 08:47:18 -0500
Subject: [Freeswitch-users] Grangstream Early Dial
In-Reply-To: <200908131114.02970.yivzhenko@mksat.net>
References: <200908131114.02970.yivzhenko@mksat.net>
Message-ID: <CF2F282D-7E28-4F19-ACD4-6692F54E5052@freeswitch.org>

I don't think we ever got this working correctly.  Can you do a trace  
of it working vs not working?

/b

On Aug 13, 2009, at 3:14 AM, Yuriy Ivzhenko wrote:

> Hello all.
>
> I want to use Grandstream Early Dial future.
> How i can enable support 484 response?
>
> I tried simply use
> <action application="hangup" data="484"/>
> and
> <action application="respond" data="484"/>
> on uncompleted extensions,
> but there is not work
>
>
> Thanks.
>
> Yuriy .




From tzury.by at reguluslabs.com  Thu Aug 13 07:04:18 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Thu, 13 Aug 2009 17:04:18 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
Message-ID: <10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>

> It just works... to force TCP you append ;transport=tcp
Hi Brian

In fact this is exactly what I did and I could not get it work.
I am using a console based application supplied by pjsip.org and when
trying to register i get some error messages saying 'invalid transport
"SIP/2.0/tcp"' and 'REGISTER (30669) has invalid Via'

using the very same client against iptel.org seems to work.

> In reality you should be using SRV records.
can you please elaborate a bit more about this?


I am dumping below the cli output.
thanks in advance for your time and attention


tport_wakeup(0x7fd82c2afaf0): events IN
tport_recv_event(0x7fd82c2afaf0)
tport_recv_iovec(0x7fd82c2afaf0) msg 0x7fd82c2a1830 from
(tcp/80.74.97.189:42634) has 472 bytes, veclen = 1
tport_deliver(0x7fd82c2afaf0): msg 0x7fd82c2a1830 (472 bytes) from
tcp/80.74.97.189:42634/sip next=(nil)
nta: received REGISTER sip:cheerfulsanity.net;transport=tcp SIP/2.0 (CSeq 30669)
nta: Via check: invalid transport "SIP/2.0/tcp" from 80.74.97.189:42634
nta: REGISTER (30669) has invalid Via
tport(0x7fd82c2afaf0): reset timer
tport(0x7fd82c2afaf0): set timer at 1800000 ms because idle since recv
tport_wakeup_pri(0x713dd0): events IN
tport_recv_event(0x713dd0)
tport_recv_iovec(0x713dd0) msg 0x7fd82c2a1830 from
(udp/67.23.5.142:5060) has 2 bytes, veclen = 1
tport_deliver(0x713dd0): bad msg 0x7fd82c2a1830 (2 bytes) from
udp/199.245.214.130:5060/sip next=(nil)
tport_wakeup_pri(0x713dd0): events IN
tport_recv_event(0x713dd0)
tport_recv_iovec(0x713dd0) msg 0x7fd82c2a1830 from
(udp/67.23.5.142:5060) has 2 bytes, veclen = 1
tport_deliver(0x713dd0): bad msg 0x7fd82c2a1830 (2 bytes) from
udp/199.245.214.130:5060/sip next=(nil)
tport_wakeup(0x7fd84027d7d0): events IN
tport_recv_event(0x7fd84027d7d0)



From brian at freeswitch.org  Thu Aug 13 07:11:03 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 13 Aug 2009 09:11:03 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
Message-ID: <43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>

I need to see the sip packet.  TCP should be uppercase I'm pretty sure.

/b

On Aug 13, 2009, at 9:04 AM, Tzury Bar Yochay wrote:

> nta: Via check: invalid transport "SIP/2.0/tcp" from  
> 80.74.97.189:42634
> nta: REGISTER (30669) has invalid Via




From csa at nowthor.com  Thu Aug 13 07:14:56 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 13 Aug 2009 10:14:56 -0400
Subject: [Freeswitch-users] bind_server_ip issue
Message-ID: <4A841FE0.90904@nowthor.com>

Hello!

First of all, I would like to express my thanks to all the developers of 
Freeswitch.

I am testing Freeswitch on a Debian machine with physical network 
interface with four virtual IP addresses. One of these IP addresses, 
aliased as eth0:3, has been created specifically for Freeswitch.  I then 
set bind_server_ip with the IP addresses associated with eth0:3. To my 
surprise, however, tow things happen more or less randomly: 1) in 
certain cases, Freeswitch binds to eth0:2 instead (with a different IP 
address); and in another, although Freeswitch binds initially to eth0:3, 
after a few hours it changes its mind and rebinds to eth0:2. Is this an 
issue with bind_server_ip or am I missing some configuration detail?

Thanks!

Carlos



From brian at freeswitch.org  Thu Aug 13 07:21:35 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 13 Aug 2009 09:21:35 -0500
Subject: [Freeswitch-users] bind_server_ip issue
In-Reply-To: <4A841FE0.90904@nowthor.com>
References: <4A841FE0.90904@nowthor.com>
Message-ID: <12E3506D-3F79-4987-9FA7-B40AE461B97A@freeswitch.org>

If you read the latest vars.xml I have clarified this:

   <!--
        THIS IS ONLY USED FOR DINGALING

        bind_server_ip

        Can be an ip address, a dns name, or "auto".
        This determines an ip address available on this host to bind.
        If you are separating RTP and SIP traffic, you will want to have
        use different addresses where this variable appears.
        Used by: dingaling.conf.xml
   -->
   <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>


So you'll need to open up the sip profile in sip_profiles and set the  
bind ip to exactly what you want.

Thanks,
Brian


On Aug 13, 2009, at 9:14 AM, Carlos S. Antunes wrote:

> Hello!
>
> First of all, I would like to express my thanks to all the  
> developers of
> Freeswitch.
>
> I am testing Freeswitch on a Debian machine with physical network
> interface with four virtual IP addresses. One of these IP addresses,
> aliased as eth0:3, has been created specifically for Freeswitch.  I  
> then
> set bind_server_ip with the IP addresses associated with eth0:3. To my
> surprise, however, tow things happen more or less randomly: 1) in
> certain cases, Freeswitch binds to eth0:2 instead (with a different IP
> address); and in another, although Freeswitch binds initially to  
> eth0:3,
> after a few hours it changes its mind and rebinds to eth0:2. Is this  
> an
> issue with bind_server_ip or am I missing some configuration detail?
>
> Thanks!
>
> Carlos
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From tzury.by at reguluslabs.com  Thu Aug 13 07:37:18 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Thu, 13 Aug 2009 17:37:18 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
Message-ID: <10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>

> I need to see the sip packet.
dumped below

> TCP should be uppercase I'm pretty sure.
you mean the via should be Via: SIP/2.0/TCP right?

If so, then that would a bug in the client then.

#wireshark dumps:
000  00 21 29 66 15 9f 00 23  ae 64 af a6 08 00 45 00   .!)f...# .d....E.
0010  02 3e 80 c2 40 00 40 06  64 e6 0a 00 00 6d 43 17   .>.. at .@. d....mC.
0020  05 8e db f4 13 c4 8a 0e  91 93 f6 a7 50 06 80 18   ........ ....P...
0030  00 5c 55 42 00 00 01 01  08 0a 0e 94 11 e4 01 ba   .\UB.... ........
0040  bf 9d 52 45 47 49 53 54  45 52 20 73 69 70 3a 63   ..REGIST ER sip:c
0050  68 65 65 72 66 75 6c 73  61 6e 69 74 79 2e 6e 65   heerfuls anity.ne
0060  74 3b 74 72 61 6e 73 70  6f 72 74 3d 74 63 70 20   t;transp ort=tcp
0070  53 49 50 2f 32 2e 30 0d  0a 56 69 61 3a 20 53 49   SIP/2.0. .Via: SI
0080  50 2f 32 2e 30 2f 74 63  70 20 31 30 2e 30 2e 30   P/2.0/tc p 10.0.0
0090  2e 31 30 39 3a 35 36 33  30 38 3b 72 70 6f 72 74   .109:563 08;rport
00a0  3b 62 72 61 6e 63 68 3d  7a 39 68 47 34 62 4b 50   ;branch= z9hG4bKP
00b0  6a 69 43 34 39 35 5a 50  52 6d 32 39 39 65 4f 6e   jiC495ZP Rm299eOn
00c0  6e 32 4f 4b 34 79 6a 6b  64 37 32 4c 2d 41 73 76   n2OK4yjk d72L-Asv
00d0  67 0d 0a 52 6f 75 74 65  3a 20 3c 73 69 70 3a 63   g..Route : <sip:c
00e0  68 65 65 72 66 75 6c 73  61 6e 69 74 79 2e 6e 65   heerfuls anity.ne
00f0  74 3b 74 72 61 6e 73 70  6f 72 74 3d 74 63 70 3b   t;transp ort=tcp;
0100  6c 72 3e 0d 0a 4d 61 78  2d 46 6f 72 77 61 72 64   lr>..Max -Forward
0110  73 3a 20 37 30 0d 0a 46  72 6f 6d 3a 20 3c 73 69   s: 70..F rom: <si
0120  70 3a 31 30 36 30 40 63  68 65 65 72 66 75 6c 73   p:1060 at c heerfuls
0130  61 6e 69 74 79 2e 6e 65  74 3e 3b 74 61 67 3d 52   anity.ne t>;tag=R
0140  45 2e 58 78 5a 6e 4b 6a  71 66 32 5a 48 2d 38 4f   E.XxZnKj qf2ZH-8O
0150  58 37 30 71 70 73 78 5a  44 43 69 51 4a 37 65 0d   X70qpsxZ DCiQJ7e.
0160  0a 54 6f 3a 20 3c 73 69  70 3a 31 30 36 30 40 63   .To: <si p:1060 at c
0170  68 65 65 72 66 75 6c 73  61 6e 69 74 79 2e 6e 65   heerfuls anity.ne
0180  74 3e 0d 0a 43 61 6c 6c  2d 49 44 3a 20 5a 6e 61   t>..Call -ID: Zna
0190  30 61 56 68 72 2d 51 43  4a 4a 58 34 62 70 79 31   0aVhr-QC JJX4bpy1
01a0  4c 78 6b 62 4a 46 78 6a  50 4f 2d 4f 6d 0d 0a 43   LxkbJFxj PO-Om..C
01b0  53 65 71 3a 20 34 37 37  38 30 20 52 45 47 49 53   Seq: 477 80 REGIS
01c0  54 45 52 0d 0a 55 73 65  72 2d 41 67 65 6e 74 3a   TER..Use r-Agent:
01d0  20 50 4a 53 55 41 20 76  31 2e 33 2d 74 72 75 6e    PJSUA v 1.3-trun
01e0  6b 2f 69 36 38 36 2d 70  63 2d 6c 69 6e 75 78 2d   k/i686-p c-linux-
01f0  67 6e 75 0d 0a 43 6f 6e  74 61 63 74 3a 20 3c 73   gnu..Con tact: <s
0200  69 70 3a 31 30 36 30 40  31 30 2e 30 2e 30 2e 31   ip:1060@ 10.0.0.1
0210  30 39 3a 35 30 36 30 3b  74 72 61 6e 73 70 6f 72   09:5060; transpor
0220  74 3d 54 43 50 3e 0d 0a  45 78 70 69 72 65 73 3a   t=TCP>.. Expires:
0230  20 33 30 30 0d 0a 43 6f  6e 74 65 6e 74 2d 4c 65    300..Co ntent-Le
0240  6e 67 74 68 3a 20 20 30  0d 0a 0d 0a               ngth:  0 ....



From brian at freeswitch.org  Thu Aug 13 07:42:22 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 13 Aug 2009 09:42:22 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
Message-ID: <45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>


On Aug 13, 2009, at 9:37 AM, Tzury Bar Yochay wrote:

>> I need to see the sip packet.
> dumped below
>
>> TCP should be uppercase I'm pretty sure.
> you mean the via should be Via: SIP/2.0/TCP right?
>

Yep

> If so, then that would a bug in the client then.

Some things might accept it but sofia is usually strict about some of  
this stuff.

/b





From string01 at gmail.com  Thu Aug 13 07:43:09 2009
From: string01 at gmail.com (String Larson)
Date: Thu, 13 Aug 2009 09:43:09 -0500
Subject: [Freeswitch-users] Setting max inbound for UA
In-Reply-To: <C6A8E329.5ACC0%krice@freeswitch.org>
References: <C6A8E329.5ACC0%krice@freeswitch.org>
Message-ID: <D3C49EE3-E1F0-47FA-AF55-545DF0B92075@gmail.com>

Thanks Ken.
I'll look at mod_limit

The XLite softphone doesn't seem to have a switch for controlling it.
-str

On Aug 12, 2009, at 9:22 PM, Ken Rice wrote:

> Check out mod_limit... Other wise you'll have to look specifically  
> at the UA
> you are trying to use, some like polycom and sipura offer a way to  
> disable
> "call waiting"
>
> Remember with SIP there is no such thing as a line, its a SESSION  
> and you
> can have as many sessions as the software allows (and most software  
> doesn't
> put sane limits based on CPU/RAM/Bandwidth etc)
>
>
>> From: String Larson <string01 at gmail.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Wed, 12 Aug 2009 19:42:02 -0500
>> To: <freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] Setting max inbound for UA
>>
>> Is there a way to limit the number of calls a UA can receive in the  
>> FS
>> configs?
>>
>> I'm doing some testing with XLite as the UA, and can not figure out
>> how to keep line 2 from answering if line 1 is in use.
>>
>> THanks.
>>
>> -str
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From tina at a2unlimited.com  Thu Aug 13 07:43:01 2009
From: tina at a2unlimited.com (Tina Martinez)
Date: Thu, 13 Aug 2009 10:43:01 -0400
Subject: [Freeswitch-users]  Question about sharing conference between
Message-ID: <39253.1250174581@a2unlimited.com>

Thank you Michael,

I will tinker around with it and definitely follow-up with the results.


- T



From moises.silva at gmail.com  Thu Aug 13 08:02:05 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 13 Aug 2009 11:02:05 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <A681A304-5B94-49B8-BA25-E3089ED3F4A9@freeswitch.org>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
	<87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
	<c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>
	<A681A304-5B94-49B8-BA25-E3089ED3F4A9@freeswitch.org>
Message-ID: <c4d05cbe0908130802w6528ca34ma9d6e34ab9ce28c2@mail.gmail.com>

On Wed, Aug 12, 2009 at 7:19 PM, Brian West <brian at freeswitch.org> wrote:

> Well you really can't ignore it... it happens with our ISDN stack
> too.   Thats what the VETO handles.
>
> /b
>

You lost me. What do you mean we can't ignore it? the way I see it, sure we
can and we should.

Currently that warning comes from the on_ringing() callback which blindly
attempts to move the state of the zap channel to ZAP_CHANNEL_STATE_PROGRESS,
even when the state may be already ZAP_CHANNEL_STATE_PROGRESS_MEDIA (which
means on_proceed() was called first).

As I see it, the VETO warning is more an aid to the programmer so you
quickly realize your doing a useless state change, which should be fixed. In
this case, the fix is simply checking the state of the channel before trying
to move it to progress, and don't even try to move it if already in progress
with media.

 --
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From yivzhenko at mksat.net  Thu Aug 13 08:11:43 2009
From: yivzhenko at mksat.net (Yuriy Ivzhenko)
Date: Thu, 13 Aug 2009 18:11:43 +0300
Subject: [Freeswitch-users] Grangstream Early Dial
In-Reply-To: <CF2F282D-7E28-4F19-ACD4-6692F54E5052@freeswitch.org>
References: <200908131114.02970.yivzhenko@mksat.net>
	<CF2F282D-7E28-4F19-ACD4-6692F54E5052@freeswitch.org>
Message-ID: <200908131811.43414.yivzhenko@mksat.net>

On Thursday 13 August 2009 16:47:18 Brian West wrote:
> I don't think we ever got this working correctly.  Can you do a trace
> of it working vs not working?

I can't do working trace, only not working

http://pastebin.freeswitch.org/9980

with dialplan action
 <action application="hangup" data="INVALID_NUMBER_FORMAT"/>



>
> /b
>
> On Aug 13, 2009, at 3:14 AM, Yuriy Ivzhenko wrote:
> > Hello all.
> >
> > I want to use Grandstream Early Dial future.
> > How i can enable support 484 response?
> >
> > I tried simply use
> > <action application="hangup" data="484"/>
> > and
> > <action application="respond" data="484"/>
> > on uncompleted extensions,
> > but there is not work
> >
> >
> > Thanks.
> >
> > Yuriy .
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From bjbrashier at gmail.com  Thu Aug 13 08:48:13 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 08:48:13 -0700
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <4A83BCB4.2050604@chandlerfamily.org.uk>
References: <4A830243.8080703@chandlerfamily.org.uk>
	<7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com> 
	<4A83434C.3090505@chandlerfamily.org.uk>
	<7bcfdd290908121609i5289fae6n40b1be48b0173ff0@mail.gmail.com> 
	<4A83BCB4.2050604@chandlerfamily.org.uk>
Message-ID: <7bcfdd290908130848t15592b50y68a6b0ee8808d095@mail.gmail.com>

So it sounds like "set" can work. But you'd still have to parse it. And even
then it's not recommended.

I have another couple of possible methods for you:
1) modification of mod_conference.
2) event socket.

If you modify mod_conference, you can probably do what you want, but it
obviously requires using C and modifying existing code.

If you use the event socket, you've got a bigger learning curve, perhaps,
but you can use a variety of languages, your code is separate (and therefore
easier to maintain), and you then know how the event socket works in case
you need to do something else later.
 Good luck with whatever you end up doing.

BB
On Thu, Aug 13, 2009 at 12:11 AM, Alan Chandler
<alan at chandlerfamily.org.uk>wrote:

> Bradley Brashier wrote:
>  >  I wrote:
> >  >This is a significant new fact for me.  What you seem to be doing is
> >  >calling the commands referenced in the conference api here
> >  >
> >  >http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
> >  >
> >  >by using application="conference" and then the data string as the
> second
> >  >part of the command.  Am I correct in the assumption that you can do
> this.
> >
> > I agree that that's what it looks like. What I don't know is if it
> > works. I got this example from the page
> > http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did
> > exactly what you're trying, and never tried using the API in this
> fashion.
>
> I just found this - which I think helps
>
> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
>
> An API can be called from the dialplan but it is not recommended. Example:
>
>  <extension name="Make API call from Dialplan">
>    <condition field="destination_number" expression="^(999)$">
>      <!-- next line calls hupall, so be careful! -->
>      <action application="set"
> data="api_result=${hupall(normal_clearing)}"/>
>    </condition>
>  </extension>
>
> Anyway - thanks for you help - I am going away to rethink that
> particular interface again.  Its getting so complicated that it might be
> better to copy the Javascript approach in the examples.
>
>
>
> --
>  Alan Chandler
> http://www.chandlerfamily.org.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Thu Aug 13 09:06:03 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 13 Aug 2009 09:06:03 -0700
Subject: [Freeswitch-users] calling through same gateway with multiple
	registrations
In-Reply-To: <241d382f0908130235w460339baxcc39527c03b0724@mail.gmail.com>
References: <241d382f0908130235w460339baxcc39527c03b0724@mail.gmail.com>
Message-ID: <87f2f3b90908130906h6f40889erc33a6ef18deace32@mail.gmail.com>

On Thu, Aug 13, 2009 at 2:35 AM, Timur Irmatov <irmatov at gmail.com> wrote:

> Hi,
>
>
> I am new to FreeSWITCH and need an advice.
>
> All calls to PSTN from our server will go through single gateway,
> which is a soft switch supporting SIP protocol. FreeSWITCH will need
> to register with soft switch, but soft switch permits only single
> active call (in either direction) per registration. So we will need 10
> SIP accounts to allow 10 simultaneous connections.


Are you going to have incoming calls as well? If so, how does the
soft-switch handle two concurrent calls to the same number?


>
>
> Question is - how should I configure FreeSWITCH for this scenario? I
> see two options:
>
> 1) Create 10 gateways with different registrations, use mod_limit to
> route only one outgoing call per gateway;
> 2) Create 10 gateways with different registrations, use event socket
> to route calls manually and monitor used lines (incoming and outgoing
> calls through soft switch).
>
> Are there any other possibilities? Corrections/ suggestions are very
> welcome.


This seems like a serious defect in the soft-switch. I can understand if it
allows you to specify only one call per SIP registration, but to hard-code
that limit seems pretty silly. Can you find out more about the soft-switch
in question and see if that limitation is flexible?
-MC
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From bjbrashier at gmail.com  Thu Aug 13 09:37:11 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 09:37:11 -0700
Subject: [Freeswitch-users] Conference silence timeouts
Message-ID: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>

Hi all.

The solution to this one should be short.

My conference hangs up when there's 2+ users and silence for 5 sec or so.
I'm trying to find a parameter that changes that (I'd rather it be, say, 60
seconds).

I didn't see a parameter like this specific to conferences, so I looked
abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300
(the default), so I'm pretty sure that's not the problem. I also searched
through the mod_conference.c code and didn't see it, though I was only
skimming.

I'm not 100% convinced that this is limited to conferences, but I don't
currently have a way to test in a non-conference environment.

Anybody know how to increase the conference silence-hangup timeout?

BB
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From msc at freeswitch.org  Thu Aug 13 09:50:55 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 13 Aug 2009 11:50:55 -0500
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>
Message-ID: <87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>

Check out the 'waste' member flag. I think if at least one member has that
set then RTP will get sent out even during silence. Let us know if that
helps...

-MC

On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> Hi all.
>
> The solution to this one should be short.
>
> My conference hangs up when there's 2+ users and silence for 5 sec or so.
> I'm trying to find a parameter that changes that (I'd rather it be, say, 60
> seconds).
>
> I didn't see a parameter like this specific to conferences, so I looked
> abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300
> (the default), so I'm pretty sure that's not the problem. I also searched
> through the mod_conference.c code and didn't see it, though I was only
> skimming.
>
> I'm not 100% convinced that this is limited to conferences, but I don't
> currently have a way to test in a non-conference environment.
>
> Anybody know how to increase the conference silence-hangup timeout?
>
> BB
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bjbrashier at gmail.com  Thu Aug 13 10:47:48 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 10:47:48 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>
Message-ID: <7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>

I'm sure that would work, but I'm worried about it sucking up bandwidth,
especially since you'd need it on every caller (since otherwise the one
person who had it could hang up and you'd be back to square 1).

Any other ideas, or should I hunt through the code to try to override the
behavior manually?

BB

On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins <msc at freeswitch.org> wrote:

> Check out the 'waste' member flag. I think if at least one member has that
> set then RTP will get sent out even during silence. Let us know if that
> helps...
>
> -MC
>
>   On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <bjbrashier at gmail.com
> > wrote:
>
>>   Hi all.
>>
>> The solution to this one should be short.
>>
>> My conference hangs up when there's 2+ users and silence for 5 sec or so.
>> I'm trying to find a parameter that changes that (I'd rather it be, say, 60
>> seconds).
>>
>> I didn't see a parameter like this specific to conferences, so I looked
>> abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300
>> (the default), so I'm pretty sure that's not the problem. I also searched
>> through the mod_conference.c code and didn't see it, though I was only
>> skimming.
>>
>> I'm not 100% convinced that this is limited to conferences, but I don't
>> currently have a way to test in a non-conference environment.
>>
>> Anybody know how to increase the conference silence-hangup timeout?
>>
>> BB
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From chris at cloudtel.com  Thu Aug 13 11:14:23 2009
From: chris at cloudtel.com (Chris Burns)
Date: Thu, 13 Aug 2009 11:14:23 -0700
Subject: [Freeswitch-users] Confused about conferences
In-Reply-To: <7bcfdd290908130848t15592b50y68a6b0ee8808d095@mail.gmail.com>
References: <4A830243.8080703@chandlerfamily.org.uk>
	<7bcfdd290908121440k64386676u2453fd68630238f5@mail.gmail.com>
	<4A83434C.3090505@chandlerfamily.org.uk>
	<7bcfdd290908121609i5289fae6n40b1be48b0173ff0@mail.gmail.com>
	<4A83BCB4.2050604@chandlerfamily.org.uk>
	<7bcfdd290908130848t15592b50y68a6b0ee8808d095@mail.gmail.com>
Message-ID: <e6ba88620908131114k7e48f529t8809bc9673639d0a@mail.gmail.com>

I couldn't imagine managing a conference without a GUI. I need to see who is
making noise so I can boot/mute em ;)

If I were you I would dive into ESL and make a simple web app to frontend
the conferences. There will surely be something in contrib to get you
started.

On Thu, Aug 13, 2009 at 8:48 AM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> So it sounds like "set" can work. But you'd still have to parse it. And
> even then it's not recommended.
>
> I have another couple of possible methods for you:
> 1) modification of mod_conference.
> 2) event socket.
>
> If you modify mod_conference, you can probably do what you want, but it
> obviously requires using C and modifying existing code.
>
> If you use the event socket, you've got a bigger learning curve, perhaps,
> but you can use a variety of languages, your code is separate (and therefore
> easier to maintain), and you then know how the event socket works in case
> you need to do something else later.
>  Good luck with whatever you end up doing.
>
> BB
> On Thu, Aug 13, 2009 at 12:11 AM, Alan Chandler <
> alan at chandlerfamily.org.uk> wrote:
>
>> Bradley Brashier wrote:
>>  >  I wrote:
>> >  >This is a significant new fact for me.  What you seem to be doing is
>> >  >calling the commands referenced in the conference api here
>> >  >
>> >  >http://wiki.freeswitch.org/wiki/Mod_conference#API_Reference
>> >  >
>> >  >by using application="conference" and then the data string as the
>> second
>> >  >part of the command.  Am I correct in the assumption that you can do
>> this.
>> >
>> > I agree that that's what it looks like. What I don't know is if it
>> > works. I got this example from the page
>> > http://wiki.freeswitch.org/wiki/Conferencing_and_Intercom. I never did
>> > exactly what you're trying, and never tried using the API in this
>> fashion.
>>
>> I just found this - which I think helps
>>
>> http://wiki.freeswitch.org/wiki/Mod_commands#From_the_Dialplan
>>
>> An API can be called from the dialplan but it is not recommended. Example:
>>
>>  <extension name="Make API call from Dialplan">
>>    <condition field="destination_number" expression="^(999)$">
>>      <!-- next line calls hupall, so be careful! -->
>>      <action application="set"
>> data="api_result=${hupall(normal_clearing)}"/>
>>    </condition>
>>  </extension>
>>
>> Anyway - thanks for you help - I am going away to rethink that
>> particular interface again.  Its getting so complicated that it might be
>> better to copy the Javascript approach in the examples.
>>
>>
>>
>> --
>>  Alan Chandler
>> http://www.chandlerfamily.org.uk
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Thu Aug 13 11:15:12 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Thu, 13 Aug 2009 14:15:12 -0400
Subject: [Freeswitch-users] answer command
In-Reply-To: <EA91B26D-1BB9-40BC-9D94-6614F9890456@jerris.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<86a32abc0908111124v3234f821k37c8d54d0a344203@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
	<B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
	<24941422.post@talk.nabble.com>
	<6539128A-D7A1-4555-A3DB-60E94E461DB3@jerris.com>
	<86a32abc0908130103h6c0093d4hcef03ceed2f2aa3@mail.gmail.com>
	<EA91B26D-1BB9-40BC-9D94-6614F9890456@jerris.com>
Message-ID: <86a32abc0908131115x4bbefcdfy3e3f84b8cb692a06@mail.gmail.com>

I was talking with Michael about fixing stuff in the wiki, so I just asked
to fix that also.

On Thu, Aug 13, 2009 at 9:10 AM, Michael Jerris <mike at jerris.com> wrote:

> It probably belongs there.  It's a wiki, feel free to fix it.  What does
> this have to do with this thread?
>
>
> On Aug 13, 2009, at 4:03 AM, Diego Viola <diego.viola at gmail.com> wrote:
>
> Hey Michael,
>
> Just wondering something, I have found that you added
> "conference_set_auto_outcall" on the dptools wiki, but I could not find that
> function in the mod_dptools.c, shouldn't that be part of the mod_conference
> wiki article? =D.
>
> Best regards,
>
> Diego
>
> On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris < <mike at jerris.com>
> mike at jerris.com> wrote:
>
>> Sip does not support this functionality.  The called device would have
>> to support this via some other mechanism such as ctsa which I have
>> seen recently someone was looking at for freeswitch.  So the first
>> issue you must resolve is the called device needs to support some way
>> to do this.
>>
>> Mike
>>
>> On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov < <maxim.tsvetov at gmail.com>
>> maxim.tsvetov at gmail.com>
>> wrote:
>>
>> >
>> > If I have two FS extensions A and B. I'm calling from A to B and
>> > want to
>> > answer from B-side in my CTI application and to make SIP phone to be
>> > synchronised to my CTI application. Is it possible to do it?
>> >
>> >
>> > Brian West-3 wrote:
>> >>
>> >> Well you can only truly answer an inbound call to FS... you can't
>> >> force answer an outbound call.
>> >>
>> >> /b
>> >>
>> >> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
>> >>
>> >>>
>> >>> I will try to paraphrase my question.
>> >>> Is there any possibility to answer call  from CTI application and
>> >>> synchronise answer with answer in SIP client?Maybe we can use SIP
>> >>> functions
>> >>> in our CTI application instead of FS api commands?
>> >>> I'm trying to find the way to make prototype of lineAnswer command
>> >>> in TAPI.
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> <http://www.freeswitch.org>http://www.freeswitch.org
>> >>
>> >>
>> >
>> > --
>> > View this message in context:
>> <http://www.nabble.com/answer-command-tp24912812p24941422.html>
>> http://www.nabble.com/answer-command-tp24912812p24941422.html
>> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Thu Aug 13 11:16:29 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Thu, 13 Aug 2009 14:16:29 -0400
Subject: [Freeswitch-users] answer command
In-Reply-To: <86a32abc0908131115x4bbefcdfy3e3f84b8cb692a06@mail.gmail.com>
References: <89c9bbf80908110010g7902e0f9g3d02c4a9c98b51ad@mail.gmail.com>
	<24931876.post@talk.nabble.com>
	<B0DCDC9F-D29D-498D-B97F-34D218A636E3@gmail.com>
	<24940548.post@talk.nabble.com>
	<B4523C60-18B8-4D2A-BB48-172AB46765A5@freeswitch.org>
	<24941422.post@talk.nabble.com>
	<6539128A-D7A1-4555-A3DB-60E94E461DB3@jerris.com>
	<86a32abc0908130103h6c0093d4hcef03ceed2f2aa3@mail.gmail.com>
	<EA91B26D-1BB9-40BC-9D94-6614F9890456@jerris.com>
	<86a32abc0908131115x4bbefcdfy3e3f84b8cb692a06@mail.gmail.com>
Message-ID: <86a32abc0908131116g1fa3218egc146f519b93c7cd0@mail.gmail.com>

Err, I asked if that was wrong to fix it.

On Thu, Aug 13, 2009 at 2:15 PM, Diego Viola <diego.viola at gmail.com> wrote:

> I was talking with Michael about fixing stuff in the wiki, so I just asked
> to fix that also.
>
>
> On Thu, Aug 13, 2009 at 9:10 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> It probably belongs there.  It's a wiki, feel free to fix it.  What does
>> this have to do with this thread?
>>
>>
>> On Aug 13, 2009, at 4:03 AM, Diego Viola <diego.viola at gmail.com> wrote:
>>
>> Hey Michael,
>>
>> Just wondering something, I have found that you added
>> "conference_set_auto_outcall" on the dptools wiki, but I could not find that
>> function in the mod_dptools.c, shouldn't that be part of the mod_conference
>> wiki article? =D.
>>
>> Best regards,
>>
>> Diego
>>
>> On Wed, Aug 12, 2009 at 1:50 PM, Michael Jerris < <mike at jerris.com>
>> mike at jerris.com> wrote:
>>
>>> Sip does not support this functionality.  The called device would have
>>> to support this via some other mechanism such as ctsa which I have
>>> seen recently someone was looking at for freeswitch.  So the first
>>> issue you must resolve is the called device needs to support some way
>>> to do this.
>>>
>>> Mike
>>>
>>> On Aug 12, 2009, at 10:38 AM, Maxim Tsvetov < <maxim.tsvetov at gmail.com>
>>> maxim.tsvetov at gmail.com>
>>> wrote:
>>>
>>> >
>>> > If I have two FS extensions A and B. I'm calling from A to B and
>>> > want to
>>> > answer from B-side in my CTI application and to make SIP phone to be
>>> > synchronised to my CTI application. Is it possible to do it?
>>> >
>>> >
>>> > Brian West-3 wrote:
>>> >>
>>> >> Well you can only truly answer an inbound call to FS... you can't
>>> >> force answer an outbound call.
>>> >>
>>> >> /b
>>> >>
>>> >> On Aug 12, 2009, at 11:49 AM, Maxim Tsvetov wrote:
>>> >>
>>> >>>
>>> >>> I will try to paraphrase my question.
>>> >>> Is there any possibility to answer call  from CTI application and
>>> >>> synchronise answer with answer in SIP client?Maybe we can use SIP
>>> >>> functions
>>> >>> in our CTI application instead of FS api commands?
>>> >>> I'm trying to find the way to make prototype of lineAnswer command
>>> >>> in TAPI.
>>> >>
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> >> <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> <http://www.freeswitch.org>http://www.freeswitch.org
>>> >>
>>> >>
>>> >
>>> > --
>>> > View this message in context:
>>> <http://www.nabble.com/answer-command-tp24912812p24941422.html>
>>> http://www.nabble.com/answer-command-tp24912812p24941422.html
>>> > Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>  <FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From irmatov at gmail.com  Thu Aug 13 11:21:48 2009
From: irmatov at gmail.com (Timur Irmatov)
Date: Thu, 13 Aug 2009 23:21:48 +0500
Subject: [Freeswitch-users] calling through same gateway with multiple
	registrations
In-Reply-To: <87f2f3b90908130906h6f40889erc33a6ef18deace32@mail.gmail.com>
References: <241d382f0908130235w460339baxcc39527c03b0724@mail.gmail.com>
	<87f2f3b90908130906h6f40889erc33a6ef18deace32@mail.gmail.com>
Message-ID: <241d382f0908131121s66298225xa5fe49036b6ec5e@mail.gmail.com>

Thanks for responding, Michael!

On Thu, Aug 13, 2009 at 9:06 PM, Michael Collins<msc at freeswitch.org> wrote:
> Are you going to have incoming calls as well? If so, how does the
> soft-switch handle two concurrent calls to the same number?

Yes, we'll have incoming calls as well. I did not performed any tests
myself yet, but a guy working with this soft-switch says it just
rejects second call. So soft-switch treats each sip registration more
or less like single phone line. Soft-switch is Huawei SoftX3000, if
somebody experienced in it can prove me wrong, I'll be glad.. :)

>> Question is - how should I configure FreeSWITCH for this scenario? I
>> see two options:
>>
>> 1) Create 10 gateways with different registrations, use mod_limit to
>> route only one outgoing call per gateway;
>> 2) Create 10 gateways with different registrations, use event socket
>> to route calls manually and monitor used lines (incoming and outgoing
>> calls through soft switch).
>>
>> Are there any other possibilities? Corrections/ suggestions are very
>> welcome.
>
> This seems like a serious defect in the soft-switch. I can understand if it
> allows you to specify only one call per SIP registration, but to hard-code
> that limit seems pretty silly. Can you find out more about the soft-switch
> in question and see if that limitation is flexible?

Yes, that was my initial thought too - that this limit should be
configurable. But a guy working with it says it is not configurable,
and local support engineers from Huawei also confirm this. But, who
knows, may be they are all wrong.

-- 
Timur Irmatov, xmpp:irmatov at jabber.ru



From pjintheusa at gmail.com  Thu Aug 13 11:33:30 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 13 Aug 2009 14:33:30 -0400
Subject: [Freeswitch-users] mod_managed users?
In-Reply-To: <6E8D2069C08AA84A83D336E996AE4C6702C4A8E0C8@mse17be1.mse17.exchange.ms>
References: <6E8D2069C08AA84A83D336E996AE4C6702C4A8E02E@mse17be1.mse17.exchange.ms>
	<501641.90132.qm@web33503.mail.mud.yahoo.com>
	<6E8D2069C08AA84A83D336E996AE4C6702C4A8E0C8@mse17be1.mse17.exchange.ms>
Message-ID: <367751820908131133q20a724cdw3fa4ab1026e6f908@mail.gmail.com>

Hey Michael,

I am a little late to the party I know - but just want to say thanks
for your latest efforts.

I updated my dev environment with the latest managed mod and swapped
my app to the latest plugin architecture last night and all is working
well.

Love the dynamic loading of my dll into freeswitch - no more starting
and stopping freeswitch!

Also appreciate the f# example.

Thanks again.


Phillip Jones

On Wed, Jul 29, 2009 at 8:04 PM, Michael Giagnocavo<mgg at giagnocavo.net> wrote:
> Which directory ends with ; ?
>
>
>
> I?m not following ? if you want email me off list and we can work together
> on it .
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego
> Toro
> Sent: Wednesday, July 29, 2009 5:06 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_managed users?
>
>
>
> Hi,
>
>
>
> My dll's are loaded correctly, I have a trouble becouse the directory where
> are loaded it finish with ";" (whitout ""), i mean the default path assembly
> finish with ";"
>
>
>
> Is possible remove?the character ";"? ?
>
>
>
> thanks
>
>
>
> Diego
>
> --- On Wed, 7/29/09, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
>
> From: Michael Giagnocavo <mgg at giagnocavo.net>
> Subject: Re: [Freeswitch-users] mod_managed users?
> To: "freeswitch-users at lists.freeswitch.org"
> <freeswitch-users at lists.freeswitch.org>
> Date: Wednesday, July 29, 2009, 2:32 PM
>
> The error is that it just doesn?t find that alias to call the plugin.
> Assuming everything is spelled correctly, this probably means the DLL did
> not load.
>
>
>
> I just checked in a fix for dlls that don?t have both Api and App interfaces
> ? it would not load them at all. Try with it now and see if that?s the
> problem. If it is, I apologize.
>
>
>
> If it still doesn?t, paste the full log of when it loads your file.
>
>
>
> Thanks,
>
> Michael
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego
> Toro
> Sent: Wednesday, July 29, 2009 12:44 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_managed users?
>
>
>
> Hi Michael,
>
>
>
> I am working with lastest managed module, I have assemblies wich dependen of
> others assemblies (dll's), the past version it works fine, but I have now
> next the error message:
>
>
>
> EXECUTE sofia/internal/10510 at 192.168.27.10 managed(CIV_BPFSProcess)
> 2009-07-29 13:31:27.718750 [DEBUG] switch_cpp.cpp:1130 FreeSWITCH.Managed:
> attempting to run application 'CIV_BPFSProcess'.
> 2009-07-29 13:31:27.718750 [ERR] switch_cpp.cpp:1130 App plugin
> CIV_BPFSProcess not found.
> 2009-07-29 13:31:27.718750 [ERR] mod_managed.cpp:405 Application run failed
> for CIV_BPFSProcess (unknown module or exception).
>
>
>
> My declaration class? is:
> public class CIV_BPFSProcess : FreeSWITCH.IAppPlugin
>
> It has public method:
> public void Run(AppContext context).
>
>
>
> I have VS2008 on Windows
>
> Thanks
>
>
>
> Diego
>
> --- On Wed, 7/29/09, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
>
> From: Michael Giagnocavo <mgg at giagnocavo.net>
> Subject: Re: [Freeswitch-users] mod_managed users?
> To: "freeswitch-users at lists.freeswitch.org"
> <freeswitch-users at lists.freeswitch.org>
> Date: Wednesday, July 29, 2009, 1:46 AM
>
> Hi ?ukasz,
>
> ? ? ? ? Would you please send me the DLL offlist and I'll figure it out?
>
> ? ? ? ? The new session you create is the b-leg. The parameter it takes in
> originate is the a-leg. So you'd do:
>
> var session = new ManagedSession();
> session.Originate(context.Session, "sofia/default/1000",10);
>
> ? ? ? ? As to non-blocking, I'm quite sure it's possible, but I don't recall
> offhand which functions. This should be the same as in any other language
> for FreeSWITCH -- these functions are just passthrough from the FS C++ API.
>
> -Michael
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lukasz
> Zwierko
> Sent: Wednesday, July 29, 2009 12:13 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] mod_managed users?
>
> Hi Michael ,
>
> thanks a lot for support on this.
>
>> As to the main problem of your DLL not working, can you send me the full
>> source code, or all the logging output from loading it? Try "managedreload
>> my.dll" to reload the DLL and see how it is registering them. It should
>> output something like "Registering API FullName with Aliases fullname,
>> shortname".
>>
>
> I'm just using the Demo.cs example, I compile it to dll undef VC#, not
> mono, maybe that is the difference?
> The output from the log is just as you stated "Registering API
> FullName with Aliases fullname, shortname". The difference between
> loading dll and csx is that, when loading csx all api and app classes
> are listed as registered, while with dll nothing is listed..
>
> Anyway, I have another question regarding usage of the CoreSession and
> ManagedSession object.
> Basically in my script I want to start new session and originate a call.
> So what I do is
>
> ManageSession session = new ManagedSession();
> session.Originate(session,"sofia/default/1000",10);
>
> And it works but I have some doubts. First thing is, why the first
> param of the Originate method is the CoreSession object? Can't it just
> use 'this'?? Or is there more to it?
>
> Second thing is the third parameter - timeout in seconds. Can't the
> call be started in non-blocking mode? I can start in a different
> thread of course, is that the intended behavior?
>
> Thanks for help,
>
> ?ukasz
>
>
> 2009/7/28 Michael Giagnocavo <mgg at giagnocavo.net>:
>> Hello Lukasz,
>>
>>? ? ? ? Thanks for testing mod_managed. I apologize for the problems you've
>> encountered, and I'll try to sort them out for you.
>>
>> A few things first:
>>
>>? ? ? ? - Scripting support: This is made to allow "true" scripts, as
>> invoked as an EXE - similar to the Lua and spidermonkey support. So, without
>> a Main(), it won't compile as an EXE. If you aren't using it as a script,
>> then an empty Main method will work fine.
>>
>>? ? ? ? - Entry points must be public for Mono. I'll update the demo code
>> to make sure that Main is public. This is a bug in Mono's lightweight code
>> generation -- it won't skip the JIT access checks.
>>
>> As to the main problem of your DLL not working, can you send me the full
>> source code, or all the logging output from loading it? Try "managedreload
>> my.dll" to reload the DLL and see how it is registering them. It should
>> output something like "Registering API FullName with Aliases fullname,
>> shortname".
>>
>> Thanks,
>> Michael
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lukasz
>> Zwierko
>> Sent: Tuesday, July 28, 2009 1:26 PM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] mod_managed users?
>>
>> -----BEGIN PGP SIGNED MESSAGE-----
>> Hash: SHA1
>>
>> Hi,
>>
>> I've just tried new mod_managed under Win32 and I get a weird behavior.
>> I try the example below:
>>
>> public class DemoScript : IApiPlugin
>> {
>>? ? public void Execute(ApiContext context)
>>? ? {
>>? ? ? ? context.Stream.Write(string.Format("DemoScripts executed with
>> args '{0}' and event type {1}.",
>>? ? ? ? ? ? context.Arguments, context.Event == null ? "<none>" :
>> context.Event.GetEventType()));
>>? ? }
>>
>>? ? public void ExecuteBackground(ApiBackgroundContext context)
>>? ? {
>>? ? ? ? Log.WriteLine(LogLevel.Notice, "DemoScripts on a background
>> thread #({0}), with args '{1}'.",
>>? ? ? ? ? ? System.Threading.Thread.CurrentThread.ManagedThreadId,
>>? ? ? ? ? ? context.Arguments);
>>? ? }
>> }
>>
>> It's just like the ApiDemo from Demo.cs
>>
>> So When I copy DemoScript.csx to managed dir the console log is:
>>
>> "
>> freeswitch at Zwierko-laptop> Loading
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx from
>> domain DemoScripts.csx_3
>> 2009-07-28 20:57:16.710000 [INFO] switch_cpp.cpp:1130 Compiling
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx
>> 2009-07-28 20:57:16.970000 [ERR] switch_cpp.cpp:1130 There were 1 errors
>> compiling
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\DemoScripts.csx.
>> 2009-07-28 20:57:16.970000 [ERR] switch_cpp.cpp:1130 CS5001: Program
>> 'c:\Users\Zwierko\AppData\Local\Temp\fc8jnlir.exe' does not contain a
>> static 'Main' method
>> suitable for an entry point
>> "
>>
>> Adding
>>
>> "
>>? ? public static void Main()
>>? ? {
>>? ? }
>> "
>>
>> solves the issue. Is this how it's supposed to work?
>>
>> Another strange thing is that when I compile this class to DLL (release)
>> it does not work at all...
>>
>> freeswitch at Zwierko-laptop> Loading
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\FSScripts.dll from
>> domain FSScripts.dll_5
>>
>>
>> freeswitch at Zwierko-laptop>
>>
>> freeswitch at Zwierko-laptop> managed DemoScript 111
>> API CALL [managed(DemoScript 111)] output:
>>
>> 2009-07-28 21:13:03.542000 [ERR] switch_cpp.cpp:1130 API plugin
>> DemoScript not found.
>> 2009-07-28 21:13:03.542000 [ERR] mod_managed.cpp:393 Execute failed for
>> DemoScript 111 (unknown module or exception).
>>
>> And another issue with scripts. I use script code as example:
>>
>> "
>> public class ScriptDemo
>> {
>>? ? static void Main()
>>? ? {
>>? ? ? ? switch (FreeSWITCH.Script.ContextType)
>>? ? ? ? {
>>? ? ? ? ? ? case ScriptContextType.Api:
>>? ? ? ? ? ? ? ? {
>>? ? ? ? ? ? ? ? ? ? var ctx = FreeSWITCH.Script.GetApiContext();
>>? ? ? ? ? ? ? ? ? ? ctx.Stream.Write("Script executing as API with args:
>> " + ctx.Arguments);
>>? ? ? ? ? ? ? ? ? ? break;
>>? ? ? ? ? ? ? ? }
>>? ? ? ? ? ? case ScriptContextType.ApiBackground:
>>? ? ? ? ? ? ? ? {
>>? ? ? ? ? ? ? ? ? ? var ctx = FreeSWITCH.Script.GetApiBackgroundContext();
>>? ? ? ? ? ? ? ? ? ? Log.WriteLine(LogLevel.Notice, "Executing as
>> APIBackground with args: " + ctx.Arguments);
>>? ? ? ? ? ? ? ? ? ? break;
>>? ? ? ? ? ? ? ? }
>>? ? ? ? ? ? case ScriptContextType.App:
>>? ? ? ? ? ? ? ? {
>>? ? ? ? ? ? ? ? ? ? var ctx = FreeSWITCH.Script.GetAppContext();
>>? ? ? ? ? ? ? ? ? ? Log.WriteLine(LogLevel.Notice, "Executing as App
>> with args: " + ctx.Arguments);
>>? ? ? ? ? ? ? ? ? ? break;
>>? ? ? ? ? ? ? ? }
>>? ? ? ? }
>>
>>? ? }
>> }
>> "
>>
>>
>> console log is as follows:
>>
>>
>> freeswitch at Zwierko-laptop> Loading
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx from
>> domain Script.csx_8
>> 2009-07-28 21:19:36.289000 [INFO] switch_cpp.cpp:1130 Compiling
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx
>> 2009-07-28 21:19:36.438000 [INFO] switch_cpp.cpp:1130 File
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx compiled
>> successfully.
>> 2009-07-28 21:19:36.451000 [ERR] switch_cpp.cpp:1130 Entry point:
>> ScriptDemo.Main is not public. This may cause errors with Mono.
>> 2009-07-28 21:19:36.458000 [NOTICE] switch_cpp.cpp:1130 Loaded App
>> Script.csx, aliases 'Script.csx', into domain Script.csx_8.
>> 2009-07-28 21:19:36.459000 [NOTICE] switch_cpp.cpp:1130 Loaded Api
>> Script.csx, aliases 'Script.csx', into domain Script.csx_8.
>> 2009-07-28 21:19:36.459000 [INFO] switch_cpp.cpp:1130 Finished loading
>> c:\Users\Zwierko\Projects\fs\svn\Debug\mod\managed\Script.csx into
>> domain Script.csx_8.
>> managed ScriptDemo 111
>> 2009-07-28 21:19:53.452000 [ERR] switch_cpp.cpp:1130 API plugin
>> ScriptDemo not found.
>> API CALL [managed(ScriptDemo 111)] output:
>>
>> 2009-07-28 21:19:53.452000 [ERR] mod_managed.cpp:393 Execute failed for
>> ScriptDemo 111 (unknown module or exception).
>>
>>
>> Again, am I doing something wrong in here?
>>
>> Thanks, for any feedback
>>
>> Lukasz Zwierko
>>
>>
>>
>>
>>
>> Michael Giagnocavo wrote:
>>> Ah, that's embarrassing. I added them and tried building
>>> FreeSWITCH.Managed from svn and it worked fine now. (I'll kick off a new
>>> complete build in a minute.)
>>>
>>> -Michael
>>>
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego
>>> Toro
>>> Sent: Sunday, July 26, 2009 8:47 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] mod_managed users?
>>>
>>> Hi Michael,
>>>
>>> Thank you for your job with mod_managed, I get lastest version with
>>> mod_managed but the files PluginInterfaces.cs, PluginManager.cs and
>>> ScriptPluginManager.cs were not downloaded.
>>>
>>> Diego
>>>
>>>
>>> --- On Sun, 7/26/09, Michael Giagnocavo <mgg at giagnocavo.net> wrote:
>>>
>>> From: Michael Giagnocavo <mgg at giagnocavo.net>
>>> Subject: Re: [Freeswitch-users] mod_managed users?
>>> To: "freeswitch-users at lists.freeswitch.org"
>>> <freeswitch-users at lists.freeswitch.org>
>>> Date: Sunday, July 26, 2009, 2:18 AM
>>> Hello,
>>>
>>>? ? ? ? ? ? ? ???I just checked in a new mod_managed. It breaks backwards
>>> compatibility, but adds scripting and reloading support.
>>>
>>>? ? ? ? ? ? ? ???I tested it on CentOS 5.3 x64 with Mono 2.4.2.2. Just
>>> make & make install seemed to take care of everything.
>>>
>>>? ? ? ? ? ? ? ???Let me know if you have better luck with this version.
>>>
>>> Thanks,
>>> Michael
>>>
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Muhammad
>>> Shahzad
>>> Sent: Saturday, July 18, 2009 6:58 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] mod_managed users?
>>>
>>> Sorry for replying late.
>>>
>>> I have tried mod_managed again on same machine (Lenovo 3000 N200), same
>>> OS (Ubuntu-9.04) and Mono version 2.0.1 but with latest FS revision 14249.
>>>
>>> It compiles correctly this time but gives following error upon "make
>>> install",
>>>
>>> =====================================================================
>>> making install mod_managed
>>> make[5]: *** No rule to make target
>>> `/usr/local/freeswitch/mod/mod_managed.so', needed by `local_install'.
>>> Stop.
>>> make[4]: *** [install] Error 1
>>> make[3]: *** [mod_managed-install] Error 1
>>> make[2]: *** [install-recursive] Error 1
>>>
>>> =====================================================================
>>>
>>> Here is compilation log when executing "make", if it could of any help.
>>>
>>> =====================================================================
>>> making all mod_managed
>>> Compiling freeswitch_managed.cpp...
>>> g++ -I/usr/src/svn-src/freeswitch/src/include
>>> -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>> -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0
>>> -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_managed.o
>>> freeswitch_managed.cpp
>>> Compiling freeswitch_wrap.cpp...
>>> g++ -I/usr/src/svn-src/freeswitch/src/include
>>> -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>> -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0
>>> -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o freeswitch_wrap.o
>>> freeswitch_wrap.cpp
>>> Demo.cs(58,14): warning CS0169: The private method
>>> `FreeSWITCH.Demo.AppDemo.hangupHook()' is never used
>>> Compilation succeeded - 1 warning(s)
>>> Compiling mod_managed.cpp...
>>> /usr/src/svn-src/freeswitch/libtool --mode=compile --tag=CXX g++
>>> -I/usr/src/svn-src/freeswitch/src/include
>>> -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>> -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0
>>> -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c -o mod_managed.lo
>>> mod_managed.cpp
>>> libtool: compile:? g++ -I/usr/src/svn-src/freeswitch/src/include
>>> -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>> -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0
>>> -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp? -fPIC -DPIC
>>> -o .libs/mod_managed.o
>>> mod_managed.cpp: In function 'void InitManagedSession(ManagedSession*,
>>> char* (*)(void*, switch_input_type_t), void (*)())':
>>> mod_managed.cpp:97: warning: deprecated conversion from string constant
>>> to 'char*'
>>> libtool: compile:? g++ -I/usr/src/svn-src/freeswitch/src/include
>>> -I/usr/src/svn-src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden
>>> -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE
>>> -D_REENTRANT -pthread -I/usr/include/mono-1.0 -I/usr/include/glib-2.0
>>> -I/usr/lib/glib-2.0/include -DHAVE_CONFIG_H -c mod_managed.cpp -o
>>> mod_managed.o >/dev/null 2>&1
>>> Creating mod_managed.la...
>>> cat: .libs/mod_managed.log: No such file or directory
>>>
>>> =====================================================================
>>>
>>> Thank you.
>>> On Fri, Jul 17, 2009 at 10:59 AM, Muhammad Shahzad
>>> <shaheryarkh at googlemail.com<http://us.mc335.mail.yahoo.com/mc/compose?to=shaheryarkh at googlemail.com>>
>>> wrote:
>>> I tried to install mod_managed on ubuntu-9.04, mono framework version
>>> 2.0. It gave me a lots of errors in Loader.cs, which seems to be SWIG
>>> related. Since i am not a expert in SWIG so i disabled this module. This
>>> happend long ago, i think FS svn revision 136xx.
>>>
>>> Let me try to compile it from latest FS revision and see if it works. I
>>> will let you know the results.
>>>
>>> Thank you.
>>>
>>> On Fri, Jul 17, 2009 at 3:54 AM, Diego Toro
>>> <dftoro at yahoo.com<http://us.mc335.mail.yahoo.com/mc/compose?to=dftoro at yahoo.com>>
>>> wrote:
>>> Hey, I am here? :)
>>>
>>> I am working with mod_managed on Windows 2003 and Windows Vista with
>>> sucessfull.? I noted on user list the issue with LoadFile on Loader.cs when
>>> a assembly had reference to others assemblies, I change LoadFile by LoadFrom
>>> and the load is made fine.
>>>
>>> I use c# application and sqlserver 2005, using FS and mod_managed.
>>>
>>> Diego
>>>
>>> --- On Thu, 7/16/09, Michael Giagnocavo
>>> <mgg at giagnocavo.net<http://us.mc335.mail.yahoo.com/mc/compose?to=mgg at giagnocavo.net>>
>>> wrote:
>>>
>>> From: Michael Giagnocavo
>>> <mgg at giagnocavo.net<http://us.mc335.mail.yahoo.com/mc/compose?to=mgg at giagnocavo.net>>
>>> Subject: [Freeswitch-users] mod_managed users?
>>> To:
>>> "freeswitch-users at lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users at lists.freeswitch.org>"
>>> <freeswitch-users at lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=freeswitch-users at lists.freeswitch.org>>
>>> Date: Thursday, July 16, 2009, 4:43 PM
>>>
>>> Hey, if there are any mod_managed users on this list, I'd love it if you
>>> were able to let me know. I'd like to get feedback, positive or negative, on
>>> what worked, what didn't, and how mod_managed can improve for you. Feel free
>>> to write on list or directly to me: mgg at
>>> giagnocavo.net<http://giagnocavo.net/>
>>>
>>> Thanks!
>>> -Michael
>>> -----Inline Attachment Follows-----
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org<http://www.freeswitch.org/>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org<http://www.freeswitch.org/>
>>>
>>>
>>> --
>>> Muhammad Shahzad
>>> -----------------------------------
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +92 334 422 40 88
>>> MSN:
>>> shari_786pk at hotmail.com<http://us.mc335.mail.yahoo.com/mc/compose?to=shari_786pk at hotmail.com>
>>> Email:
>>> shaheryarkh at googlemail.com<http://us.mc335.mail.yahoo.com/mc/compose?to=shaheryarkh at googlemail.com>
>>>
>>>
>>>
>>> --
>>> Muhammad Shahzad
>>> -----------------------------------
>>> CISCO Rich Media Communication Specialist (CRMCS)
>>> CISCO Certified Network Associate (CCNA)
>>> Cell: +92 334 422 40 88
>>> MSN:
>>> shari_786pk at hotmail.com<http://us.mc335.mail.yahoo.com/mc/compose?to=shari_786pk at hotmail.com>
>>> Email:
>>> shaheryarkh at googlemail.com<http://us.mc335.mail.yahoo.com/mc/compose?to=shaheryarkh at googlemail.com>
>>>
>>> -----Inline Attachment Follows-----
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>>
>>> FreeSWITCH-users at lists.freeswitch.org<http://us.mc335.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org<http://www.freeswitch.org/>
>>>
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> -----BEGIN PGP SIGNATURE-----
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>> Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org
>>
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>> =LgK4
>> -----END PGP SIGNATURE-----
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From email.list.subscriber at gmail.com  Thu Aug 13 12:10:13 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Thu, 13 Aug 2009 15:10:13 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>
	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>
Message-ID: <4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>

Hi Michal,

Just checking in to see if you've been able to take a stab at this.

Thanks,
Vladimir


-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Michal Bielicki
Sent: Tuesday, August 11, 2009 5:33 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

I'll retst it later today and give you a link with instructions

Am 10.08.2009 um 20:14 schrieb vmorales:

> By "./compile" I was referring to "./configure"
>
> Vladimir
>
> -----Original Message-----
> From: vmorales [mailto:email.list.subscriber at gmail.com]
> Sent: Monday, August 10, 2009 11:49 AM
> To: 'freeswitch-users at lists.freeswitch.org'
> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> Thanks for the response(s):
>
> I ran the "./compile" script with a set PREFIX.  This took a few
> attempts with errors before it was able to complete error-free, as I
> had to install libtool.
>
> Since then, I have tried running 'make', 'gmake', and
> '/opt/gnu/bin/make', but each results with an error.  This is the
> error when running 'make' or 'gmake':
>
> <snip>
> make: Fatal error: Command failed for target `all-recursive'
> Current working directory /home/vmorales/freeswitch-1.0.4
> *** Error code 1
> make: Fatal error: Command failed for target `all'
> </snip>
>
>
> This is the error when running '/opt/gnu/bin/make':
>
> <snip>
> make[5]: *** [mod_amr.so] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_amr-all] Error 1
> make[2]: *** [all-recursive] Error 1
> Making all in build
> +-------- FreeSWITCH Build Complete -----------+
> + FreeSWITCH has been successfully built.      +
> + Install by running:                          +
> +                                              +
> +               /opt/gnu/bin/make install                   +
> +----------------------------------------------+
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> </snip>
>
>
> I re-untar'd before each compile/make attempt.  Let me know if this
is
> something that I can resolve.
>
> Vladimir
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michael Jerris
> Sent: Saturday, August 08, 2009 12:37 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> This is not currently a supported platform, it only builds on 64 bit
> right now I think on solaris.
>
> Mike
>
> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>
>> Hello,
>>
>> Does anyone have, or know where to get, a pre-compiled copy of
>> FreeSwitch for Solaris 10/x86?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
> http://www.freeswitch.org

Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. info at halokwadrat.de
e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
Hauptgesch?ftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 0000153539





From rswagoner at gmail.com  Thu Aug 13 12:34:05 2009
From: rswagoner at gmail.com (Ryan Wagoner)
Date: Thu, 13 Aug 2009 15:34:05 -0400
Subject: [Freeswitch-users] Sangoma A102 Overrun Issue
Message-ID: <7d86ddb90908131234t7aad15afk2fa3c85e46db5499@mail.gmail.com>

I've been trying to bridge FreeSWITCH with a Toshiba CIX using qsig
over a PRI. I have a Sangoma A102 card installed in a Dell PowerEdge
with CentOS 5.3. The issue I am having is no packets are being
transmitted back to FreeSWITCH. ifconfig w1g1 shows every frame
received as an overrun. I've tried a different server with CentOS 5.3
with the same issue. I have a support ticket in with Sangoma, but was
wondering if anybody had seen this before.

The T1 shows connected so I think I have the Toshiba configured
properly. From what I've read the overrun has to deal with the driver
not reading the data in time so maybe this is a CentOS 5.3 specific
issue. Any recommendations on alternative Linux distros known to work
with the Sangoma A102 card?

Thanks,
Ryan

[root at voip ~]# ifconfig w1g1
w1g1      Link encap:Point-to-Point Protocol
          UP POINTOPOINT RUNNING NOARP  MTU:80  Metric:1
          RX packets:22298 errors:0 dropped:0 overruns:274 frame:274
          TX packets:22298 errors:0 dropped:0 overruns:0 carrier:0
          collisions:0 txqueuelen:100
          RX bytes:1783840 (1.7 MiB)  TX bytes:1783840 (1.7 MiB)
          Interrupt:169 Memory:f8e80000-f8e81fff


[root at voip ~]# wanrouter status

Devices currently active:
        wanpipe1


Wanpipe Config:

Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK | Baud rate |
wanpipe1    | N/A          | A101/1D/A102/2D/4/4D/8| 169 | 4       | 1
   | N/A | 0         |

Wanrouter Status:

Device name | Protocol | Station | Status        |
wanpipe1    | AFT TE1  | N/A     | Connected     |


[root at voip ~]# wanpipemon -i w1g1 -c Ta

***** w1g1: T1 Alarms (Framer) *****

ALOS:   OFF     | LOS:  OFF
RED:    OFF     | AIS:  OFF
RAI:    OFF     | OOF:  OFF

***** w1g1: T1 Alarms (LIU) *****

Short Circuit:  OFF
Open Circuit:   OFF
Loss of Signal: OFF


***** w1g1: T1 Performance Monitoring Counters *****

Line Code Violation     : 45
Bit Errors (CRC6/Ft/Fs) : 0
Out of Frame Errors     : 0


Rx Level        : > -2.5db



From msc at freeswitch.org  Thu Aug 13 12:44:35 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 13 Aug 2009 14:44:35 -0500
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>
Message-ID: <87f2f3b90908131244q4f157d6en2947ae5d7ed4e395@mail.gmail.com>

On Thu, Aug 13, 2009 at 12:47 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> I'm sure that would work, but I'm worried about it sucking up bandwidth,
> especially since you'd need it on every caller (since otherwise the one
> person who had it could hang up and you'd be back to square 1).
>
> Any other ideas, or should I hunt through the code to try to override the
> behavior manually?
>
> BB
>

Get a packet capture and debug trace of the symptom occurring. Put the PCAP
where we can download it and pastebin the debug log. We need to confirm who
the culprit is and what events precipitate the disconnect. I'd also be
curious to know if anyone else can reproduce these symptoms. BTW, which rev
of FS are you running?

Thanks,
MC
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From mike at jerris.com  Thu Aug 13 13:20:32 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 13 Aug 2009 16:20:32 -0400
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>
Message-ID: <BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>

My guess is that its the other end killing the call due to rtp  
timeouts, not us killing the call.  If you can confirm this the best  
method would be to get them not to do rtp timeouts.

On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:

> I'm sure that would work, but I'm worried about it sucking up  
> bandwidth, especially since you'd need it on every caller (since  
> otherwise the one person who had it could hang up and you'd be back  
> to square 1).
>
> Any other ideas, or should I hunt through the code to try to  
> override the behavior manually?
>
> BB
>
> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins  
> <msc at freeswitch.org> wrote:
> Check out the 'waste' member flag. I think if at least one member  
> has that set then RTP will get sent out even during silence. Let us  
> know if that helps...
>
> -MC
>
> On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <bjbrashier at gmail.com 
> > wrote:
> Hi all.
>
> The solution to this one should be short.
>
> My conference hangs up when there's 2+ users and silence for 5 sec  
> or so. I'm trying to find a parameter that changes that (I'd rather  
> it be, say, 60 seconds).
>
> I didn't see a parameter like this specific to conferences, so I  
> looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but  
> it's set to 300 (the default), so I'm pretty sure that's not the  
> problem. I also searched through the mod_conference.c code and  
> didn't see it, though I was only skimming.
>
> I'm not 100% convinced that this is limited to conferences, but I  
> don't currently have a way to test in a non-conference environment.
>
> Anybody know how to increase the conference silence-hangup timeout?
>
> BB
>
> _____
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From bjbrashier at gmail.com  Thu Aug 13 13:23:27 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 13:23:27 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <87f2f3b90908131244q4f157d6en2947ae5d7ed4e395@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com> 
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com> 
	<87f2f3b90908131244q4f157d6en2947ae5d7ed4e395@mail.gmail.com>
Message-ID: <7bcfdd290908131323r189f4585v72cf53e9a8b0321c@mail.gmail.com>

I'm currently running current trunk synched up Tues morning, but it was
happening in all of the versions I'd been using previous -- I first
downloaded around the end of May.

I'll look into getting you a PCap. I expected that this was a known thing
with a parameter somewhere, so I haven't looked into it too terribly far
myself, yet. I'm gonna try looking at the console outputs and logs myself,
first.

BB
On Thu, Aug 13, 2009 at 12:44 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
>  On Thu, Aug 13, 2009 at 12:47 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
>
>> I'm sure that would work, but I'm worried about it sucking up bandwidth,
>> especially since you'd need it on every caller (since otherwise the one
>> person who had it could hang up and you'd be back to square 1).
>>
>> Any other ideas, or should I hunt through the code to try to override the
>> behavior manually?
>>
>> BB
>>
>
> Get a packet capture and debug trace of the symptom occurring. Put the PCAP
> where we can download it and pastebin the debug log. We need to confirm who
> the culprit is and what events precipitate the disconnect. I'd also be
> curious to know if anyone else can reproduce these symptoms. BTW, which rev
> of FS are you running?
>
> Thanks,
> MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bjbrashier at gmail.com  Thu Aug 13 13:45:23 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 13:45:23 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com> 
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com> 
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
Message-ID: <7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com>

I had just thought of the exact same thing. I'm trying to test that now.
Thanks for your input.

BB

On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com> wrote:

> My guess is that its the other end killing the call due to rtp timeouts,
> not us killing the call.  If you can confirm this the best method would be
> to get them not to do rtp timeouts.
>  On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>
>  I'm sure that would work, but I'm worried about it sucking up bandwidth,
> especially since you'd need it on every caller (since otherwise the one
> person who had it could hang up and you'd be back to square 1).
>
> Any other ideas, or should I hunt through the code to try to override the
> behavior manually?
>
> BB
>
> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>> Check out the 'waste' member flag. I think if at least one member has that
>> set then RTP will get sent out even during silence. Let us know if that
>> helps...
>>
>> -MC
>>
>>   On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <
>> bjbrashier at gmail.com> wrote:
>>
>>>   Hi all.
>>>
>>> The solution to this one should be short.
>>>
>>> My conference hangs up when there's 2+ users and silence for 5 sec or so.
>>> I'm trying to find a parameter that changes that (I'd rather it be, say, 60
>>> seconds).
>>>
>>> I didn't see a parameter like this specific to conferences, so I looked
>>> abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300
>>> (the default), so I'm pretty sure that's not the problem. I also searched
>>> through the mod_conference.c code and didn't see it, though I was only
>>> skimming.
>>>
>>> I'm not 100% convinced that this is limited to conferences, but I don't
>>> currently have a way to test in a non-conference environment.
>>>
>>> Anybody know how to increase the conference silence-hangup timeout?
>>>
>>> BB
>>>
>>> _____
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bjbrashier at gmail.com  Thu Aug 13 14:48:55 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 14:48:55 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com> 
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com> 
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
	<7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com>
Message-ID: <7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com>

I took a closer look at the SIP messages on the console. From it, I
understand that it's not Freeswitch timing out, but rather FS is getting the
"BYE" msg from somewhere else. I've tested phones and tested calling without
going through the FS conference, though, and everything works fine. Then I
saw something else odd inside the BYE msg:

   Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"
So I Googled "RTP Broken Connection" and saw several sites talking about
AudioCodes gateways and Asterisk -- and our gateway is an AudioCodes. From
these sites it sounds like AudioCodes is rather aggressive in detecting RTP
breaks, and is interpreting the silence from FS as a break.

So now I'm looking into ways to maybe send "I'm still here" RTP packets from
FS or to tune the gateway to be less aggressive. I can't stop and get a
clean packet capture right now because I've got a bunch of testers working
on it today. I'll do that sometime when the system is less busy.

BB

On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> I had just thought of the exact same thing. I'm trying to test that now.
> Thanks for your input.
>
> BB
>
>   On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com> wrote:
>
>>   My guess is that its the other end killing the call due to rtp
>> timeouts, not us killing the call.  If you can confirm this the best method
>> would be to get them not to do rtp timeouts.
>>  On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>>
>>  I'm sure that would work, but I'm worried about it sucking up bandwidth,
>> especially since you'd need it on every caller (since otherwise the one
>> person who had it could hang up and you'd be back to square 1).
>>
>> Any other ideas, or should I hunt through the code to try to override the
>> behavior manually?
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins <msc at freeswitch.org>wrote:
>>
>>> Check out the 'waste' member flag. I think if at least one member has
>>> that set then RTP will get sent out even during silence. Let us know if that
>>> helps...
>>>
>>> -MC
>>>
>>>   On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <
>>> bjbrashier at gmail.com> wrote:
>>>
>>>>   Hi all.
>>>>
>>>> The solution to this one should be short.
>>>>
>>>> My conference hangs up when there's 2+ users and silence for 5 sec or
>>>> so. I'm trying to find a parameter that changes that (I'd rather it be,
>>>> say, 60 seconds).
>>>>
>>>> I didn't see a parameter like this specific to conferences, so I looked
>>>> abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300
>>>> (the default), so I'm pretty sure that's not the problem. I also searched
>>>> through the mod_conference.c code and didn't see it, though I was only
>>>> skimming.
>>>>
>>>> I'm not 100% convinced that this is limited to conferences, but I don't
>>>> currently have a way to test in a non-conference environment.
>>>>
>>>> Anybody know how to increase the conference silence-hangup timeout?
>>>>
>>>> BB
>>>>
>>>> _____
>>>
>>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From bjbrashier at gmail.com  Thu Aug 13 15:24:19 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Thu, 13 Aug 2009 15:24:19 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com> 
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com> 
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
	<7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com> 
	<7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com>
Message-ID: <7bcfdd290908131524s10e209dcla7bebada79ecfa51@mail.gmail.com>

OK, I finally got a moment to do a packet capture and take a look at the
streams.  It became very clear very quickly that what happens is that during
silence the gateway still sends RTP packets to Freeswitch, but Freeswitch
doesn't send any back to the gateway. After 10s of this, the gateway says
"Oh, the RPT must be broken" and it hangs up.

We found a way to turn off this behavior in the gateway, and the good news
is that it did indeed fix the problem. But we'd rather not rely on that as a
long-term solution because then we can't detect and drop RTP streams that
really are broken.

So now I'm back to looking at Freeswitch to figure out how to send just a
single packet every second or so during silence. If anyone knows of a way to
do this, let me know, otherwise I'll get back to you if and when I find one.

BB

On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> I took a closer look at the SIP messages on the console. From it, I
> understand that it's not Freeswitch timing out, but rather FS is getting the
> "BYE" msg from somewhere else. I've tested phones and tested calling without
> going through the FS conference, though, and everything works fine. Then I
> saw something else odd inside the BYE msg:
>
>    Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"
> So I Googled "RTP Broken Connection" and saw several sites talking about
> AudioCodes gateways and Asterisk -- and our gateway is an AudioCodes. From
> these sites it sounds like AudioCodes is rather aggressive in detecting RTP
> breaks, and is interpreting the silence from FS as a break.
>
> So now I'm looking into ways to maybe send "I'm still here" RTP packets
> from FS or to tune the gateway to be less aggressive. I can't stop and get a
> clean packet capture right now because I've got a bunch of testers working
> on it today. I'll do that sometime when the system is less busy.
>
> BB
>
> On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
>
>> I had just thought of the exact same thing. I'm trying to test that now.
>> Thanks for your input.
>>
>> BB
>>
>>   On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com>wrote:
>>
>>>   My guess is that its the other end killing the call due to rtp
>>> timeouts, not us killing the call.  If you can confirm this the best method
>>> would be to get them not to do rtp timeouts.
>>>  On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>>>
>>>  I'm sure that would work, but I'm worried about it sucking up
>>> bandwidth, especially since you'd need it on every caller (since otherwise
>>> the one person who had it could hang up and you'd be back to square 1).
>>>
>>> Any other ideas, or should I hunt through the code to try to override the
>>> behavior manually?
>>>
>>> BB
>>>
>>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>
>>>> Check out the 'waste' member flag. I think if at least one member has
>>>> that set then RTP will get sent out even during silence. Let us know if that
>>>> helps...
>>>>
>>>> -MC
>>>>
>>>>   On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <
>>>> bjbrashier at gmail.com> wrote:
>>>>
>>>>>   Hi all.
>>>>>
>>>>> The solution to this one should be short.
>>>>>
>>>>> My conference hangs up when there's 2+ users and silence for 5 sec or
>>>>> so. I'm trying to find a parameter that changes that (I'd rather it be,
>>>>> say, 60 seconds).
>>>>>
>>>>> I didn't see a parameter like this specific to conferences, so I looked
>>>>> abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set to 300
>>>>> (the default), so I'm pretty sure that's not the problem. I also searched
>>>>> through the mod_conference.c code and didn't see it, though I was only
>>>>> skimming.
>>>>>
>>>>> I'm not 100% convinced that this is limited to conferences, but I don't
>>>>> currently have a way to test in a non-conference environment.
>>>>>
>>>>> Anybody know how to increase the conference silence-hangup timeout?
>>>>>
>>>>> BB
>>>>>
>>>>> _____
>>>>
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>
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From pjintheusa at gmail.com  Thu Aug 13 15:59:19 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 13 Aug 2009 18:59:19 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
	<b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
Message-ID: <367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>

Rupa / all,

Just a quick follow up to this.

This is appears to a timing issue. If I try and do a "uuid_media off +
uuid"  in  "api_after_bridge"  it fails with "CHAN_NOT_IMPLEMENTED"
and  the call is dropped.

If appears to be trying to do a SIP reinvite on the loopback channel
which is of course just about to / has disappear/ed.

So I tried this, after the call is established, at the commend line, I
do "show calls" and using the uuid shown, type "uuid_media off  uuid".
The SIP REINVITE is issued and works.

I think the switch_ivr_nomedia function in switch_ivr_c is getting the
loopback uuid when it calls "other_uuid =
switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE)"

That's why the SIP REINVITE fails.

So... in api_after_bridge I issue:

"sched_api", "+3 none uuid_media off " + uuid. This calls the
switch_ivr_nomedia function 3 seconds after the calls bridge is
established.


And it works, Not nice - not scalable or production ready - but the
SIP-REINVITE is successful and at least now I understand what is going
on.

Make sense?

Thanks


Phil


On Wed, Aug 12, 2009 at 12:21 PM, Rupa Schomaker<rupa at rupa.com> wrote:
> On Wed, Aug 12, 2009 at 10:22 AM, Phillip Jones<pjintheusa at gmail.com> wrote:
>> Hi there,
>>
>>>> application="originate" data="(sofia/foo/bar|sofia/baz/bar),(sofia/foo/yum|sofia/baz/yum)"
>>
>> I agree. However, perhaps the ideal is not to specify the carriers at
>> this level, as carriers are added and removed fairly often as costings
>> change. So it would be nice to have some sort of proxy that resolves
>> to a list of carriers:
>>
>> application="originate" data="sofia/MyCarriers/bar,sofia/MyCarriers/yum"
>
>> <MyCarriers>
>> <action application="carrier" data="sofia/foo"/>
>> <action application="carrier" data="sofia/baz"/>
>> <action application="carrier" data="sofia/etc"/>
>> </MyCarriers>
>>
>>
>> or something similar. This would achieve the same as loopback in this
>> use case but without dangers of looping? Complicated stuff though.
>
> Well, that is all done by mod_lcr. ?I was simplifying to narrow down
> to just originate.
>
> First we need to see if this is worth pursuing over fixing (modifying,
> whatever) loopback to handle bypass media. ?If it is, then I'll modify
> mod_lcr to deal with the situation in question (comma or pipe sep list
> of numbers to call. ?mod_lcr would then group as appropriate).
>
> Right now, my bridge is setup in a small javascript script that builds
> the appropriate dialstring (using loopback for external calls, user/
> for internal calls) and then when doing the loopback call to mod_lcr
> to get the dialstring with all providers in the right order.
>
>>>>Perhaps have an on answer hook that tries to enable bypass media (re-invite) after the call is setup?
>>
>> That's a good idea - I will look into that.
>>
>>
>> Thanks again.
>>
>>
>> Phillip
>
> Let us know how it works for you...
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mrene_lists at avgs.ca  Thu Aug 13 16:54:08 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 13 Aug 2009 19:54:08 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<87f2f3b90908111107l3b90c721gf073c12e78e48f4c@mail.gmail.com>
	<367751820908111143k35dd95f7i84c02b3fbf0c9e34@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
	<b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
	<367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>
Message-ID: <FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>

Hi All,

The reason it works when you wait 3 seconds is that mod_loopback bails  
out of the equation as soon as it detects a bridge.
It'll call switch_ivr_uuid_bridge() and you'll end up with 2 sofia  
channels.

Now the reason why you can't do uuid_media over a loopback channel is  
because it doesn't pass on SWITCH_MESSAGE_INDICATE_MEDIA and   
SWITCH_MESSAGE_INDICATE_NOMEDIA onto the underlying channel.
The handler for those two events require accessing channel variables  
on the both channels to get the ip+port of where the audio should go  
through, so that mod_sofia can send a re-invite.
Since mod_loopback is a completely different channel, it has its own  
channel variables, independent from mod_sofia (provided you have sofia  
channels on both side).  That's why even sofia<>loopback won't do  
bypass media.

On another note, mod_sofia will behave differently when it detects its  
being bridge with another sofia channel, providing optimizations when  
both call legs are SIP.

My personal opinion is not to use mod_loopback unless absolutely  
necessary, FreeSWITCH's core is very flexible and there's often a  
(better) way than using mod_loopback.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 13-Aug-09, at 6:59 PM, Phillip Jones wrote:

> Rupa / all,
>
> Just a quick follow up to this.
>
> This is appears to a timing issue. If I try and do a "uuid_media off +
> uuid"  in  "api_after_bridge"  it fails with "CHAN_NOT_IMPLEMENTED"
> and  the call is dropped.
>
> If appears to be trying to do a SIP reinvite on the loopback channel
> which is of course just about to / has disappear/ed.
>
> So I tried this, after the call is established, at the commend line, I
> do "show calls" and using the uuid shown, type "uuid_media off  uuid".
> The SIP REINVITE is issued and works.
>
> I think the switch_ivr_nomedia function in switch_ivr_c is getting the
> loopback uuid when it calls "other_uuid =
> switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE)"
>
> That's why the SIP REINVITE fails.
>
> So... in api_after_bridge I issue:
>
> "sched_api", "+3 none uuid_media off " + uuid. This calls the
> switch_ivr_nomedia function 3 seconds after the calls bridge is
> established.
>
>
> And it works, Not nice - not scalable or production ready - but the
> SIP-REINVITE is successful and at least now I understand what is going
> on.
>
> Make sense?
>
> Thanks
>
>
> Phil
>
>
> On Wed, Aug 12, 2009 at 12:21 PM, Rupa Schomaker<rupa at rupa.com> wrote:
>> On Wed, Aug 12, 2009 at 10:22 AM, Phillip  
>> Jones<pjintheusa at gmail.com> wrote:
>>> Hi there,
>>>
>>>>> application="originate" data="(sofia/foo/bar|sofia/baz/bar), 
>>>>> (sofia/foo/yum|sofia/baz/yum)"
>>>
>>> I agree. However, perhaps the ideal is not to specify the carriers  
>>> at
>>> this level, as carriers are added and removed fairly often as  
>>> costings
>>> change. So it would be nice to have some sort of proxy that resolves
>>> to a list of carriers:
>>>
>>> application="originate" data="sofia/MyCarriers/bar,sofia/ 
>>> MyCarriers/yum"
>>
>>> <MyCarriers>
>>> <action application="carrier" data="sofia/foo"/>
>>> <action application="carrier" data="sofia/baz"/>
>>> <action application="carrier" data="sofia/etc"/>
>>> </MyCarriers>
>>>
>>>
>>> or something similar. This would achieve the same as loopback in  
>>> this
>>> use case but without dangers of looping? Complicated stuff though.
>>
>> Well, that is all done by mod_lcr.  I was simplifying to narrow down
>> to just originate.
>>
>> First we need to see if this is worth pursuing over fixing  
>> (modifying,
>> whatever) loopback to handle bypass media.  If it is, then I'll  
>> modify
>> mod_lcr to deal with the situation in question (comma or pipe sep  
>> list
>> of numbers to call.  mod_lcr would then group as appropriate).
>>
>> Right now, my bridge is setup in a small javascript script that  
>> builds
>> the appropriate dialstring (using loopback for external calls, user/
>> for internal calls) and then when doing the loopback call to mod_lcr
>> to get the dialstring with all providers in the right order.
>>
>>>>> Perhaps have an on answer hook that tries to enable bypass media  
>>>>> (re-invite) after the call is setup?
>>>
>>> That's a good idea - I will look into that.
>>>
>>>
>>> Thanks again.
>>>
>>>
>>> Phillip
>>
>> Let us know how it works for you...
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mrene_lists at avgs.ca  Thu Aug 13 16:57:28 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 13 Aug 2009 19:57:28 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <c4d05cbe0908130802w6528ca34ma9d6e34ab9ce28c2@mail.gmail.com>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
	<87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
	<c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>
	<A681A304-5B94-49B8-BA25-E3089ED3F4A9@freeswitch.org>
	<c4d05cbe0908130802w6528ca34ma9d6e34ab9ce28c2@mail.gmail.com>
Message-ID: <EEE57DE2-9176-49FB-A986-A53275E148E3@avgs.ca>

It probably just VETO it so it avoid sending  
SWITCH_MESSAGE_INDICATE_PROGRESS again since the call is already  
making progress from the core's point of view?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 13-Aug-09, at 11:02 AM, Moises Silva wrote:

> On Wed, Aug 12, 2009 at 7:19 PM, Brian West <brian at freeswitch.org>  
> wrote:
> Well you really can't ignore it... it happens with our ISDN stack
> too.   Thats what the VETO handles.
>
> /b
>
> You lost me. What do you mean we can't ignore it? the way I see it,  
> sure we can and we should.
>
> Currently that warning comes from the on_ringing() callback which  
> blindly attempts to move the state of the zap channel to  
> ZAP_CHANNEL_STATE_PROGRESS, even when the state may be already  
> ZAP_CHANNEL_STATE_PROGRESS_MEDIA (which means on_proceed() was  
> called first).
>
> As I see it, the VETO warning is more an aid to the programmer so  
> you quickly realize your doing a useless state change, which should  
> be fixed. In this case, the fix is simply checking the state of the  
> channel before trying to move it to progress, and don't even try to  
> move it if already in progress with media.
>
>  --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON  
> L3R 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From moises.silva at gmail.com  Thu Aug 13 18:04:13 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Thu, 13 Aug 2009 21:04:13 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <EEE57DE2-9176-49FB-A986-A53275E148E3@avgs.ca>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
	<87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
	<c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>
	<A681A304-5B94-49B8-BA25-E3089ED3F4A9@freeswitch.org>
	<c4d05cbe0908130802w6528ca34ma9d6e34ab9ce28c2@mail.gmail.com>
	<EEE57DE2-9176-49FB-A986-A53275E148E3@avgs.ca>
Message-ID: <c4d05cbe0908131804g18cd923ewdc629e0e99ccb50b@mail.gmail.com>

Yes, agreed, but there is no point in sending a WARNING since is a normal
condition, therefore should not even try to change the state of the channel.

On Thu, Aug 13, 2009 at 7:57 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> It probably just VETO it so it avoid sending
> SWITCH_MESSAGE_INDICATE_PROGRESS
> again since the call is already making progress from the core's point of view?
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 13-Aug-09, at 11:02 AM, Moises Silva wrote:
>
> On Wed, Aug 12, 2009 at 7:19 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Well you really can't ignore it... it happens with our ISDN stack
>> too.   Thats what the VETO handles.
>>
>> /b
>>
>
> You lost me. What do you mean we can't ignore it? the way I see it, sure we
> can and we should.
>
> Currently that warning comes from the on_ringing() callback which blindly
> attempts to move the state of the zap channel to ZAP_CHANNEL_STATE_PROGRESS,
> even when the state may be already ZAP_CHANNEL_STATE_PROGRESS_MEDIA (which
> means on_proceed() was called first).
>
> As I see it, the VETO warning is more an aid to the programmer so you
> quickly realize your doing a useless state change, which should be fixed. In
> this case, the fix is simply checking the state of the channel before trying
> to move it to progress, and don't even try to move it if already in progress
> with media.
>
>  --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From Nick.Lemberger at lkfd.net  Thu Aug 13 18:24:35 2009
From: Nick.Lemberger at lkfd.net (Nick Lemberger)
Date: Thu, 13 Aug 2009 20:24:35 -0500
Subject: [Freeswitch-users] OpenSolaris Compile Error [gcc]
Message-ID: <4A847695.2C9A.00FE.0@lkfd.net>

64bit OpenSolaris w/ gcc-4.3.2

After a bootstrap and configure I get the following error when running make:


---snip---

Compiling src/switch_caller.c ...
cc1: warnings being treated as errors
src/switch_caller.c: In function 'switch_caller_profile_event_set_data':
src/switch_caller.c:299: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
src/switch_caller.c:301: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
src/switch_caller.c:303: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
src/switch_caller.c:305: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
src/switch_caller.c:307: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
src/switch_caller.c:309: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
src/switch_caller.c:311: error: format '%lld' expects type 'long long int', but argument 5 has type 'switch_time_t'
make[2]: *** [libfreeswitch_la-switch_caller.lo] Error 1

---snip---

I get this error in both the source for 1.0.4 and last nights snapshot.  An suggestions or ideas?  There are no apparent errors during the bootstrap or configure processes.

Regards,
Nicholas Lemberger
Lakefield Communications





From pjintheusa at gmail.com  Thu Aug 13 20:48:17 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 13 Aug 2009 20:48:17 -0700
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
	<b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
	<367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>
	<FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>
Message-ID: <367751820908132048s2a39b0deu586aa3ecc01c1ac7@mail.gmail.com>

> The reason it works when you wait 3 seconds is that mod_loopback bails
> out of the equation as soon as it detects a bridge.
> It'll call switch_ivr_uuid_bridge() and you'll end up with 2 sofia
> channels.

Is there a hook that is fired when when that switch_ivr_uuid_bridge()
successfully executes? So the uuid_media off is called on the
appropriate sofia channels? Is "api_after_bridge" behaving correctly -
should that only be called on the sofia channels and not the loopback?
Is it being fired to early?



On Thu, Aug 13, 2009 at 4:54 PM, Mathieu Rene<mrene_lists at avgs.ca> wrote:
> Hi All,
>
> The reason it works when you wait 3 seconds is that mod_loopback bails
> out of the equation as soon as it detects a bridge.
> It'll call switch_ivr_uuid_bridge() and you'll end up with 2 sofia
> channels.
>
> Now the reason why you can't do uuid_media over a loopback channel is
> because it doesn't pass on SWITCH_MESSAGE_INDICATE_MEDIA and
> SWITCH_MESSAGE_INDICATE_NOMEDIA onto the underlying channel.
> The handler for those two events require accessing channel variables
> on the both channels to get the ip+port of where the audio should go
> through, so that mod_sofia can send a re-invite.
> Since mod_loopback is a completely different channel, it has its own
> channel variables, independent from mod_sofia (provided you have sofia
> channels on both side). ?That's why even sofia<>loopback won't do
> bypass media.
>
> On another note, mod_sofia will behave differently when it detects its
> being bridge with another sofia channel, providing optimizations when
> both call legs are SIP.
>
> My personal opinion is not to use mod_loopback unless absolutely
> necessary, FreeSWITCH's core is very flexible and there's often a
> (better) way than using mod_loopback.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 13-Aug-09, at 6:59 PM, Phillip Jones wrote:
>
>> Rupa / all,
>>
>> Just a quick follow up to this.
>>
>> This is appears to a timing issue. If I try and do a "uuid_media off +
>> uuid" ?in ?"api_after_bridge" ?it fails with "CHAN_NOT_IMPLEMENTED"
>> and ?the call is dropped.
>>
>> If appears to be trying to do a SIP reinvite on the loopback channel
>> which is of course just about to / has disappear/ed.
>>
>> So I tried this, after the call is established, at the commend line, I
>> do "show calls" and using the uuid shown, type "uuid_media off ?uuid".
>> The SIP REINVITE is issued and works.
>>
>> I think the switch_ivr_nomedia function in switch_ivr_c is getting the
>> loopback uuid when it calls "other_uuid =
>> switch_channel_get_variable(channel, SWITCH_BRIDGE_VARIABLE)"
>>
>> That's why the SIP REINVITE fails.
>>
>> So... in api_after_bridge I issue:
>>
>> "sched_api", "+3 none uuid_media off " + uuid. This calls the
>> switch_ivr_nomedia function 3 seconds after the calls bridge is
>> established.
>>
>>
>> And it works, Not nice - not scalable or production ready - but the
>> SIP-REINVITE is successful and at least now I understand what is going
>> on.
>>
>> Make sense?
>>
>> Thanks
>>
>>
>> Phil
>>
>>
>> On Wed, Aug 12, 2009 at 12:21 PM, Rupa Schomaker<rupa at rupa.com> wrote:
>>> On Wed, Aug 12, 2009 at 10:22 AM, Phillip
>>> Jones<pjintheusa at gmail.com> wrote:
>>>> Hi there,
>>>>
>>>>>> application="originate" data="(sofia/foo/bar|sofia/baz/bar),
>>>>>> (sofia/foo/yum|sofia/baz/yum)"
>>>>
>>>> I agree. However, perhaps the ideal is not to specify the carriers
>>>> at
>>>> this level, as carriers are added and removed fairly often as
>>>> costings
>>>> change. So it would be nice to have some sort of proxy that resolves
>>>> to a list of carriers:
>>>>
>>>> application="originate" data="sofia/MyCarriers/bar,sofia/
>>>> MyCarriers/yum"
>>>
>>>> <MyCarriers>
>>>> <action application="carrier" data="sofia/foo"/>
>>>> <action application="carrier" data="sofia/baz"/>
>>>> <action application="carrier" data="sofia/etc"/>
>>>> </MyCarriers>
>>>>
>>>>
>>>> or something similar. This would achieve the same as loopback in
>>>> this
>>>> use case but without dangers of looping? Complicated stuff though.
>>>
>>> Well, that is all done by mod_lcr. ?I was simplifying to narrow down
>>> to just originate.
>>>
>>> First we need to see if this is worth pursuing over fixing
>>> (modifying,
>>> whatever) loopback to handle bypass media. ?If it is, then I'll
>>> modify
>>> mod_lcr to deal with the situation in question (comma or pipe sep
>>> list
>>> of numbers to call. ?mod_lcr would then group as appropriate).
>>>
>>> Right now, my bridge is setup in a small javascript script that
>>> builds
>>> the appropriate dialstring (using loopback for external calls, user/
>>> for internal calls) and then when doing the loopback call to mod_lcr
>>> to get the dialstring with all providers in the right order.
>>>
>>>>>> Perhaps have an on answer hook that tries to enable bypass media
>>>>>> (re-invite) after the call is setup?
>>>>
>>>> That's a good idea - I will look into that.
>>>>
>>>>
>>>> Thanks again.
>>>>
>>>>
>>>> Phillip
>>>
>>> Let us know how it works for you...
>>>
>>> --
>>> -Rupa
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mike at jerris.com  Thu Aug 13 22:47:29 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 01:47:29 -0400
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908131524s10e209dcla7bebada79ecfa51@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
	<7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com>
	<7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com>
	<7bcfdd290908131524s10e209dcla7bebada79ecfa51@mail.gmail.com>
Message-ID: <DB70D4BE-CD40-4707-AD0C-333A6A0BA75E@jerris.com>

My suggestion is to use sip session timers not rtp timeouts as rtp is  
supposed to be discontinuous.  That being said, we have several  
settings to continuously send media, but then you are doing exactly  
what you said you didn't want to do.

Mike

On Aug 13, 2009, at 6:24 PM, Bradley Brashier wrote:

> OK, I finally got a moment to do a packet capture and take a look at  
> the streams.  It became very clear very quickly that what happens is  
> that during silence the gateway still sends RTP packets to  
> Freeswitch, but Freeswitch doesn't send any back to the gateway.  
> After 10s of this, the gateway says "Oh, the RPT must be broken" and  
> it hangs up.
>
> We found a way to turn off this behavior in the gateway, and the  
> good news is that it did indeed fix the problem. But we'd rather not  
> rely on that as a long-term solution because then we can't detect  
> and drop RTP streams that really are broken.
>
> So now I'm back to looking at Freeswitch to figure out how to send  
> just a single packet every second or so during silence. If anyone  
> knows of a way to do this, let me know, otherwise I'll get back to  
> you if and when I find one.
>
> BB
>
> On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier <bjbrashier at gmail.com 
> > wrote:
> I took a closer look at the SIP messages on the console. From it, I  
> understand that it's not Freeswitch timing out, but rather FS is  
> getting the "BYE" msg from somewhere else. I've tested phones and  
> tested calling without going through the FS conference, though, and  
> everything works fine. Then I saw something else odd inside the BYE  
> msg:
>
>    Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"
> So I Googled "RTP Broken Connection" and saw several sites talking  
> about AudioCodes gateways and Asterisk -- and our gateway is an  
> AudioCodes. From these sites it sounds like AudioCodes is rather  
> aggressive in detecting RTP breaks, and is interpreting the silence  
> from FS as a break.
>
> So now I'm looking into ways to maybe send "I'm still here" RTP  
> packets from FS or to tune the gateway to be less aggressive. I  
> can't stop and get a clean packet capture right now because I've got  
> a bunch of testers working on it today. I'll do that sometime when  
> the system is less busy.
>
> BB
>
> On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com 
> > wrote:
> I had just thought of the exact same thing. I'm trying to test that  
> now.  Thanks for your input.
>
> BB
>
> On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> My guess is that its the other end killing the call due to rtp  
> timeouts, not us killing the call.  If you can confirm this the best  
> method would be to get them not to do rtp timeouts.
>
> On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>
>> I'm sure that would work, but I'm worried about it sucking up  
>> bandwidth, especially since you'd need it on every caller (since  
>> otherwise the one person who had it could hang up and you'd be back  
>> to square 1).
>>
>> Any other ideas, or should I hunt through the code to try to  
>> override the behavior manually?
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins  
>> <msc at freeswitch.org> wrote:
>> Check out the 'waste' member flag. I think if at least one member  
>> has that set then RTP will get sent out even during silence. Let us  
>> know if that helps...
>>
>> -MC
>>
>> On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <bjbrashier at gmail.com 
>> > wrote:
>> Hi all.
>>
>> The solution to this one should be short.
>>
>> My conference hangs up when there's 2+ users and silence for 5 sec  
>> or so. I'm trying to find a parameter that changes that (I'd rather  
>> it be, say, 60 seconds).
>>
>> I didn't see a parameter like this specific to conferences, so I  
>> looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but  
>> it's set to 300 (the default), so I'm pretty sure that's not the  
>> problem. I also searched through the mod_conference.c code and  
>> didn't see it, though I was only skimming.
>>
>> I'm not 100% convinced that this is limited to conferences, but I  
>> don't currently have a way to test in a non-conference environment.
>>
>> Anybody know how to increase the conference silence-hangup timeout?
>>
>> BB
>>
>> _____
>

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From rupa at rupa.com  Thu Aug 13 23:33:55 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 14 Aug 2009 01:33:55 -0500
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<87f2f3b90908111219m76628ec6y284a15a28ae4699a@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
	<b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
	<367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>
	<FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>
Message-ID: <b39273530908132333y52279d64qe926eca47e15b5eb@mail.gmail.com>

On Thu, Aug 13, 2009 at 6:54 PM, Mathieu Rene<mrene_lists at avgs.ca> wrote:
> Hi All,
>
> The reason it works when you wait 3 seconds is that mod_loopback bails

[snip]

Thanks for that explanation.  It umm.. explains a lot. :)

> On another note, mod_sofia will behave differently when it detects its
> being bridge with another sofia channel, providing optimizations when
> both call legs are SIP.
>
> My personal opinion is not to use mod_loopback unless absolutely
> necessary, FreeSWITCH's core is very flexible and there's often a
> (better) way than using mod_loopback.

So, I think the temp solution is to use loopback+delayed no media.

but the real "solution" is to either drive the forked dialing logic
externally (event socket) or consider supporting groupings in the
bridge which.. umm...  is gonna be a pain and will need buy in from
from Tony and other core devs since that is a core (no pun intended)
piece of code that nearly everything uses.

I'm not sure I want to take a wack at it.

-- 
-Rupa



From bruce.mcalister at blueface.ie  Fri Aug 14 00:49:16 2009
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Fri, 14 Aug 2009 08:49:16 +0100
Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9	&
 v1.0.4)
In-Reply-To: <4a82d138.1f588c0a.2915.4e3d@mx.google.com>
References: <4A8280B8.6050308@blueface.ie>
	<4a82d138.1f588c0a.2915.4e3d@mx.google.com>
Message-ID: <4A8516FC.2030102@blueface.ie>

Hi All,

Anyone have any ideas about this?

Thanks
Bruce

vmorales wrote:
> Hi Bruce,
> 
> I am having similar issues trying build freeswitch 1.0.4 on Solaris
> x86 as well.  I sent some information over the mailing list, and I
> received a response from Michal Bielicki (attached), stating he'd test
> this and direct me to the steps to successfully build freeswitch.
> 
> Just an FYI in case you see his response.
> 
> Vladimir
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Bruce McAlister
> Sent: Wednesday, August 12, 2009 4:44 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 &
> v1.0.4)
> 
> Hi All,
> 
> I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and
> 
> 1.0.4.
> 
> I am running on Solaris 10 Update 5 on x86 hardware (32-bit).
> 
> The build fails with:
> 
> --- snip ---
> make: Fatal error: Command failed for target `all-recursive'
> Current working directory
> /export/home/user/packages/BUILD/freeswitch-1.0.4
> *** Error code 1
> make: Fatal error: Command failed for target `all'
> ---
> 
> Looking back through the build I can see the following error:
> 
> --- snip ---
> creating libfreeswitch.la
> (cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la 
> libfreeswitch.la)
> /usr/bin/cc 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/s
> rc 
> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
> -DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch 
> freeswitch-switch.o  ./.libs/libfreeswitch.so 
> -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/
> expat/lib 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/ex
> pat/lib/.libs/libexpat.a 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libap
> r-1.a 
> -lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp 
> -L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt 
> -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl 
> -lcurses -lsocket  -R/opt/freeswitch/lib -R/usr/sfw/lib
> Undefined                       first referenced
>   symbol                             in file
> herror                              ./.libs/libfreeswitch.so
> ld: fatal: Symbol referencing errors. No output written to
> .libs/freeswitch
> *** Error code 1
> The following command caused the error:
> `if test -z "" ; then echo /bin/bash 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else 
> echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` 
> --tag=CC --mode=link /usr/bin/cc 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/s
> rc 
> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
> -DPATH_MAX=2048 -g -v -Xc -xc99=all  -lm -R/opt/freeswitch/lib  -o 
> freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib 
> freeswitch-switch.o   libfreeswitch.la libs/apr/libapr-1.la 
> libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl 
> -lcrypto -lsocket -lnsl -ldl   -lcurses -lsocket
> --- snip ---
> 
> Then a little above this error, there is the following warning that is
> 
> displayed (I'm not sure if it is related):
> 
> --- snip ---
> *** Warning: Linking the shared library libfreeswitch.la against the
> *** static library libs/libedit/src/.libs/libedit.a is not portable!
> --- snip ---
> 
> My configure line is as follows:
> 
> ---
> ./configure --prefix=/opt/freeswitch
> ---
> 
> I have the complete configure and make output if anyone needs them.
> 
> Any help/pointers would be greatly appreciated.
> 
> Thanks
> Bruce
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From csa at nowthor.com  Fri Aug 14 05:14:29 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Fri, 14 Aug 2009 08:14:29 -0400
Subject: [Freeswitch-users] bind_server_ip issue
In-Reply-To: <12E3506D-3F79-4987-9FA7-B40AE461B97A@freeswitch.org>
References: <4A841FE0.90904@nowthor.com>
	<12E3506D-3F79-4987-9FA7-B40AE461B97A@freeswitch.org>
Message-ID: <4A855525.2080301@nowthor.com>

Hi Brian!

Thank you for your quick response. I ended up defining "local_ip_v4" at 
the top of vars.xml and set "bind_server_ip" to "local_ip_v4" as well. 
Is this the best way to go about selecting an IP address for FS to bind to?

In any case, even though it doesn't appear to affect operation, I am 
still getting this "IP change detected" somewhat periodically in my logs 
(running 1.0.4):

2009-08-14 01:30:21.215446 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 01:30:22.930551 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 02:00:21.226791 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 02:00:22.473798 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 02:10:21.255298 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 02:10:22.730414 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 02:20:21.279193 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 02:20:22.946537 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 02:30:21.297964 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 02:30:23.111418 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 02:40:21.333952 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 02:40:23.282789 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 02:50:21.362973 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 02:50:22.530569 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 03:20:21.409806 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 03:20:22.137788 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 03:30:21.414786 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 03:30:23.390800 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 03:50:21.422794 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 03:50:22.709792 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 04:00:21.422789 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 04:00:22.818539 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 05:00:21.2953 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 05:00:22.19405 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 05:10:21.29951 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 05:10:23.266294 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 05:50:21.98300 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 05:50:21.962418 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 06:00:21.125959 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 06:00:22.182413 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 06:50:21.238928 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 06:50:23.138401 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 07:00:21.246302 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.4]->[172.24.0.5] []->[]
2009-08-14 07:00:23.269782 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]
2009-08-14 08:00:21.373961 [INFO] mod_sofia.c:3232 IP change detected 
[172.24.0.5]->[172.24.0.4] []->[]
2009-08-14 08:00:22.503561 [NOTICE] sofia.c:1508 Adding Alias 
[172.24.0.5] for profile [internal]

Thanks!

Carlos

Brian West wrote:
> If you read the latest vars.xml I have clarified this:
>
>    <!--
>         THIS IS ONLY USED FOR DINGALING
>
>         bind_server_ip
>
>         Can be an ip address, a dns name, or "auto".
>         This determines an ip address available on this host to bind.
>         If you are separating RTP and SIP traffic, you will want to have
>         use different addresses where this variable appears.
>         Used by: dingaling.conf.xml
>    -->
>    <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
>
>
> So you'll need to open up the sip profile in sip_profiles and set the  
> bind ip to exactly what you want.
>
> Thanks,
> Brian
>
>
> On Aug 13, 2009, at 9:14 AM, Carlos S. Antunes wrote:
>
>   
>> Hello!
>>
>> First of all, I would like to express my thanks to all the  
>> developers of
>> Freeswitch.
>>
>> I am testing Freeswitch on a Debian machine with physical network
>> interface with four virtual IP addresses. One of these IP addresses,
>> aliased as eth0:3, has been created specifically for Freeswitch.  I  
>> then
>> set bind_server_ip with the IP addresses associated with eth0:3. To my
>> surprise, however, tow things happen more or less randomly: 1) in
>> certain cases, Freeswitch binds to eth0:2 instead (with a different IP
>> address); and in another, although Freeswitch binds initially to  
>> eth0:3,
>> after a few hours it changes its mind and rebinds to eth0:2. Is this  
>> an
>> issue with bind_server_ip or am I missing some configuration detail?
>>
>> Thanks!
>>
>> Carlos
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
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From markmorreny at gmail.com  Fri Aug 14 05:32:53 2009
From: markmorreny at gmail.com (mark morreny)
Date: Fri, 14 Aug 2009 20:32:53 +0800
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
Message-ID: <20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>

Hi,

Thank you for your help.

I get that too, but the callback does not execute the second time.

When I do task->runtime = switch_time_now() + 10;, what does +10 mean?  Does
it mean 10 s or 10 mins?

Thanks,
Mark

On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
> I did the same thing on my side....
>  API CALL [load(mod_skel)] output:
> +OK
>
> 2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task 2
> data_flush (core) to run at 1250089698
> 2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889
> Successfully Loaded [mod_skel]
> 2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270 Adding API
> Function 'skel'
> freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]
> mod_skel.c:120 starting to flush data  buffer...
>
> Note that you don't need to start the thread manually, the core already has
> threads running for the scheduler.
>
>   Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
>  On 12-Aug-09, at 7:26 AM, mark morreny wrote:
>
> Hi,
>
> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out some data
> every 10 s.  The following lines of code does not show any effect at all.
>
>     switch_scheduler_task_thread_start();
>     switch_scheduler_add_task(switch_epoch_time_now(NULL),
> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|SSHF_NO_DEL);
>
>
> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to flush
> data  buffer...\n");
>
>
>     task->runtime = switch_time_now() + 10;
>
> }
>
> Does anyone know how to get it to work?
>
> Thanks,
> Mark
>
>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> Date: Mon, Aug 10, 2009 at 8:53 PM
> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
> To: freeswitch-users at lists.freeswitch.org
>
>
> switch_rtp.c has a simple one for the zrtp cache storing.
>
> /b
>
> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>
> > Re schedule is done in your callback, take a look at places that use
> > these apis in the code for details.
>
>
>  _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dujinfang at gmail.com  Fri Aug 14 06:18:22 2009
From: dujinfang at gmail.com (Seven Du)
Date: Fri, 14 Aug 2009 21:18:22 +0800
Subject: [Freeswitch-users] 403 forbidden on files.freeswitch.org
Message-ID: <DF4D5954-19C7-457F-9958-7DF20A1A00CA@gmail.com>

Got this:

Forbidden
You don't have permission to access /cluecon_2009/presentations/ 
Dale_Building_FreeSWITCH_App_Lua.pptx on this server.

Apache/2.2.3 (CentOS) Server at files-sync.freeswitch.org Port 80

$ ping files.freeswitch.org
PING filessync.freeswitch.netdna-cdn.com (69.174.57.101): 56 data bytes
64 bytes from 69.174.57.101: icmp_seq=0 ttl=46 time=224.900 ms
64 bytes from 69.174.57.101: icmp_seq=1 ttl=46 time=221.597 ms
^C

$ ping files-sync.freeswitch.org
PING www-01.freeswitch.org (216.82.231.69): 56 data bytes
64 bytes from 216.82.231.69: icmp_seq=0 ttl=44 time=271.147 ms
64 bytes from 216.82.231.69: icmp_seq=1 ttl=44 time=273.161 ms
^C




From william.suffill at gmail.com  Fri Aug 14 06:37:27 2009
From: william.suffill at gmail.com (William Suffill)
Date: Fri, 14 Aug 2009 09:37:27 -0400
Subject: [Freeswitch-users] 403 forbidden on files.freeswitch.org
In-Reply-To: <DF4D5954-19C7-457F-9958-7DF20A1A00CA@gmail.com>
References: <DF4D5954-19C7-457F-9958-7DF20A1A00CA@gmail.com>
Message-ID: <6b65470d0908140637h2bf8bf7ch6f682b97d3be66c5@mail.gmail.com>

Given that people were downloading directly from the source that feeds
the CDN (files.freeswitch.org is hosted by a  CDN) it will be some
time before  the files appear available again. It became a problem
with how much bandwidth was being used without using the CDN. Best to
just wait I know it's being worked on but going to need time to get
resolved.



From brian at freeswitch.org  Fri Aug 14 07:09:31 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 14 Aug 2009 09:09:31 -0500
Subject: [Freeswitch-users] 403 forbidden on files.freeswitch.org
In-Reply-To: <6b65470d0908140637h2bf8bf7ch6f682b97d3be66c5@mail.gmail.com>
References: <DF4D5954-19C7-457F-9958-7DF20A1A00CA@gmail.com>
	<6b65470d0908140637h2bf8bf7ch6f682b97d3be66c5@mail.gmail.com>
Message-ID: <1BDE3BFC-4282-4C88-B71F-9919E903CE91@freeswitch.org>

No I'm going to have to put them on torrent.  Too many people were  
watching them off of files-sync on their media center PC's and chewing  
bandwidth galore.

/b

On Aug 14, 2009, at 8:37 AM, William Suffill wrote:

> Given that people were downloading directly from the source that feeds
> the CDN (files.freeswitch.org is hosted by a  CDN) it will be some
> time before  the files appear available again. It became a problem
> with how much bandwidth was being used without using the CDN. Best to
> just wait I know it's being worked on but going to need time to get
> resolved.




From brian at freeswitch.org  Fri Aug 14 07:10:25 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 14 Aug 2009 09:10:25 -0500
Subject: [Freeswitch-users] bind_server_ip issue
In-Reply-To: <4A855525.2080301@nowthor.com>
References: <4A841FE0.90904@nowthor.com>
	<12E3506D-3F79-4987-9FA7-B40AE461B97A@freeswitch.org>
	<4A855525.2080301@nowthor.com>
Message-ID: <CBF2E5B2-4707-4822-9E79-B81D1D86FA83@freeswitch.org>

Don't use local_ip_v4 then... please hard code the param in the sofia  
profile because the value of local_ip_v4 can change.

/b

On Aug 14, 2009, at 7:14 AM, Carlos S. Antunes wrote:

> Hi Brian!
>
> Thank you for your quick response. I ended up defining "local_ip_v4"  
> at the top of vars.xml and set "bind_server_ip" to "local_ip_v4" as  
> well. Is this the best way to go about selecting an IP address for  
> FS to bind to?
>
> In any case, even though it doesn't appear to affect operation, I am  
> still getting this "IP change detected" somewhat periodically in my  
> logs (running 1.0.4):




From juanbackson at gmail.com  Fri Aug 14 09:02:24 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Sat, 15 Aug 2009 00:02:24 +0800
Subject: [Freeswitch-users] sip contact header question
Message-ID: <27c25bc40908140902k78d5bbe1naa75a21df21caa6e@mail.gmail.com>

Hi,

I would like to set outbound INVITE with customer Contact: field.  I did
sip_contact_user="2033334545", but it still did not work.  I am using 1.0.4
release.

Is this a known issue?

br,
JB
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From tina at a2unlimited.com  Fri Aug 14 09:20:45 2009
From: tina at a2unlimited.com (Tina Martinez)
Date: Fri, 14 Aug 2009 12:20:45 -0400
Subject: [Freeswitch-users]  Question about sharing conference between
Message-ID: <37033.1250266845@a2unlimited.com>

Michael,

Thanks again for bearing with my novice perspective on this.

I was able to achieve the link between two FS servers as intended.  However, I
was not able to setup a "new" dialplan file as you described.  I had to place the
script into the default.xml dialplan to get it to work.  Is there something I'm
supposed to do to get FS to look in the conf/dialplan/default ?

Also, in using the code as we discussed, I was also able to establish the link
between the two servers without having an actual soft-phone registered/connected
to the FS server (which is ideal for my situation).  I can create the link
between the servers, and then dial out to external phone numbers from the
respective servers and connect the participants.  Then, regardless of who
hangs-up, the conference between the servers remains in tact.  So far it works
very well.

- T



From gmaruzz at celliax.org  Fri Aug 14 10:43:27 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 14 Aug 2009 19:43:27 +0200
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness patch
Message-ID: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>

Hi FreeSWITCHers,

all the users of mod_skypiax are kindly requested to test the svn trunk 14519.

It contains a lot of changes meant to add stability and robustness,
toward a production environment.

Let me know how your feelings, and please add to the Jira any possible
bug/issue/etc.

Thanks to you all,

-giovanni



Sincerely,

Giovanni Maruzzelli

Cell : +39-347-2665618



From chris at maxpowersoft.com  Fri Aug 14 10:54:42 2009
From: chris at maxpowersoft.com (Chris Danielson)
Date: Fri, 14 Aug 2009 10:54:42 -0700
Subject: [Freeswitch-users] FreeSWITCH Console Released For iPhone/iPod
	Touch Users
Message-ID: <4A85A4E2.2030702@maxpowersoft.com>

Announcing the release of FreeSWITCH Console in the Apple Application 
Store.  The application is FREE and allows you to connect to a 
FreeSWITCH event socket layer module that is bound to an external 
interface.  Great for development purposes and general remote debugging.

Blog announcement:
http://www.chrisdanielson.com/2009/08/14/release-iphoneipod-touch-freeswitch-console/

iTunes Store Link:
http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221&mt=8 
<http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221&mt=8>

Kind Regards,
Chris Danielson



From mike at jerris.com  Fri Aug 14 11:05:01 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 14:05:01 -0400
Subject: [Freeswitch-users] Sangoma A102 Overrun Issue
In-Reply-To: <7d86ddb90908131234t7aad15afk2fa3c85e46db5499@mail.gmail.com>
References: <7d86ddb90908131234t7aad15afk2fa3c85e46db5499@mail.gmail.com>
Message-ID: <BC34F7E2-5873-48D0-A8CE-CB450C1D3ED6@jerris.com>

I know sangoma is working on their new pri stack, maybe it has proper  
qsig support.  The openzap isdn stack does not right now and I was  
under the impression that the libpri support was pretty limited, but  
have no direct knowledge there.

Mike

On Aug 13, 2009, at 3:34 PM, Ryan Wagoner wrote:

> I've been trying to bridge FreeSWITCH with a Toshiba CIX using qsig
> over a PRI. I have a Sangoma A102 card installed in a Dell PowerEdge
> with CentOS 5.3. The issue I am having is no packets are being
> transmitted back to FreeSWITCH. ifconfig w1g1 shows every frame
> received as an overrun. I've tried a different server with CentOS 5.3
> with the same issue. I have a support ticket in with Sangoma, but was
> wondering if anybody had seen this before.
>
> The T1 shows connected so I think I have the Toshiba configured
> properly. From what I've read the overrun has to deal with the driver
> not reading the data in time so maybe this is a CentOS 5.3 specific
> issue. Any recommendations on alternative Linux distros known to work
> with the Sangoma A102 card?
>
> Thanks,
> Ryan
>
> [root at voip ~]# ifconfig w1g1
> w1g1      Link encap:Point-to-Point Protocol
>          UP POINTOPOINT RUNNING NOARP  MTU:80  Metric:1
>          RX packets:22298 errors:0 dropped:0 overruns:274 frame:274
>          TX packets:22298 errors:0 dropped:0 overruns:0 carrier:0
>          collisions:0 txqueuelen:100
>          RX bytes:1783840 (1.7 MiB)  TX bytes:1783840 (1.7 MiB)
>          Interrupt:169 Memory:f8e80000-f8e81fff
>
>
> [root at voip ~]# wanrouter status
>
> Devices currently active:
>        wanpipe1
>
>
> Wanpipe Config:
>
> Device name | Protocol Map | Adapter  | IRQ | Slot/IO | If's | CLK |  
> Baud rate |
> wanpipe1    | N/A          | A101/1D/A102/2D/4/4D/8| 169 | 4       | 1
>   | N/A | 0         |
>
> Wanrouter Status:
>
> Device name | Protocol | Station | Status        |
> wanpipe1    | AFT TE1  | N/A     | Connected     |
>
>
> [root at voip ~]# wanpipemon -i w1g1 -c Ta
>
> ***** w1g1: T1 Alarms (Framer) *****
>
> ALOS:   OFF     | LOS:  OFF
> RED:    OFF     | AIS:  OFF
> RAI:    OFF     | OOF:  OFF
>
> ***** w1g1: T1 Alarms (LIU) *****
>
> Short Circuit:  OFF
> Open Circuit:   OFF
> Loss of Signal: OFF
>
>
> ***** w1g1: T1 Performance Monitoring Counters *****
>
> Line Code Violation     : 45
> Bit Errors (CRC6/Ft/Fs) : 0
> Out of Frame Errors     : 0
>
>
> Rx Level        : > -2.5db




From mike at jerris.com  Fri Aug 14 11:08:05 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 14:08:05 -0400
Subject: [Freeswitch-users] Changing state on 1:1 from PROGRESS_MEDIA to
	PROGRESS
In-Reply-To: <c4d05cbe0908131804g18cd923ewdc629e0e99ccb50b@mail.gmail.com>
References: <4A832064.8000908@gmx.net>
	<c4d05cbe0908121333qbce7308p8eebdae0580b9e4a@mail.gmail.com>
	<15A3BE77-A077-4D1E-B95D-01861462C613@freeswitch.org>
	<c4d05cbe0908121357n398bb302vb07157490892af32@mail.gmail.com>
	<90B7DCE0-A6F7-4B4F-9D34-060979CF9550@freeswitch.org>
	<87f2f3b90908121410i621f2561r9dcbc3412e6a4e95@mail.gmail.com>
	<c4d05cbe0908121528h2db32adas15be1ecba5ff3749@mail.gmail.com>
	<A681A304-5B94-49B8-BA25-E3089ED3F4A9@freeswitch.org>
	<c4d05cbe0908130802w6528ca34ma9d6e34ab9ce28c2@mail.gmail.com>
	<EEE57DE2-9176-49FB-A986-A53275E148E3@avgs.ca>
	<c4d05cbe0908131804g18cd923ewdc629e0e99ccb50b@mail.gmail.com>
Message-ID: <3F9A015D-8583-4EB3-9211-C238628B6CE9@jerris.com>

Issue is we don't handle progress and progress media differently,  
maybe we should.  The message is however harmless, but annoying and  
should probably be dealt with a bit better.  Patches welcome.

Mike

On Aug 13, 2009, at 9:04 PM, Moises Silva wrote:

> Yes, agreed, but there is no point in sending a WARNING since is a  
> normal condition, therefore should not even try to change the state  
> of the channel.
>
> On Thu, Aug 13, 2009 at 7:57 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> It probably just VETO it so it avoid sending  
> SWITCH_MESSAGE_INDICATE_PROGRESS again since the call is already  
> making progress from the core's point of view?
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
> On 13-Aug-09, at 11:02 AM, Moises Silva wrote:
>
>> On Wed, Aug 12, 2009 at 7:19 PM, Brian West <brian at freeswitch.org>  
>> wrote:
>> Well you really can't ignore it... it happens with our ISDN stack
>> too.   Thats what the VETO handles.
>>
>> /b
>>
>> You lost me. What do you mean we can't ignore it? the way I see it,  
>> sure we can and we should.
>>
>> Currently that warning comes from the on_ringing() callback which  
>> blindly attempts to move the state of the zap channel to  
>> ZAP_CHANNEL_STATE_PROGRESS, even when the state may be already  
>> ZAP_CHANNEL_STATE_PROGRESS_MEDIA (which means on_proceed() was  
>> called first).
>>
>> As I see it, the VETO warning is more an aid to the programmer so  
>> you quickly realize your doing a useless state change, which should  
>> be fixed. In this case, the fix is simply checking the state of the  
>> channel before trying to move it to progress, and don't even try to  
>> move it if already in progress with media.
>>
>>  --
>> Moises Silva
>> Software Developer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham  
>> ON L3R 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>

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From mike at jerris.com  Fri Aug 14 11:11:25 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 14:11:25 -0400
Subject: [Freeswitch-users] OpenSolaris Compile Error [gcc]
In-Reply-To: <4A847695.2C9A.00FE.0@lkfd.net>
References: <4A847695.2C9A.00FE.0@lkfd.net>
Message-ID: <4E31892D-FC1B-4F8F-8D41-1D245D92699C@jerris.com>

It would appear that different versions of opensolaris / compiler /  
32/64 bit handle this totally differently.  I have tested this both 32  
and 64 on sun studio when I made the change and it works, seems gcc  
wants different format specifiers.  The patch that caused this is :

http://fisheye.freeswitch.org/changelog/FreeSWITCH?cs=14069

Anyone with access to 32 and 64 bit opensolaris with both sun cc and  
gcc build systems working I need a patch that works on all.

Mike



On Aug 13, 2009, at 9:24 PM, Nick Lemberger wrote:

> 64bit OpenSolaris w/ gcc-4.3.2
>
> After a bootstrap and configure I get the following error when  
> running make:
>
>
> ---snip---
>
> Compiling src/switch_caller.c ...
> cc1: warnings being treated as errors
> src/switch_caller.c: In function  
> 'switch_caller_profile_event_set_data':
> src/switch_caller.c:299: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> src/switch_caller.c:301: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> src/switch_caller.c:303: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> src/switch_caller.c:305: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> src/switch_caller.c:307: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> src/switch_caller.c:309: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> src/switch_caller.c:311: error: format '%lld' expects type 'long  
> long int', but argument 5 has type 'switch_time_t'
> make[2]: *** [libfreeswitch_la-switch_caller.lo] Error 1
>
> ---snip---
>
> I get this error in both the source for 1.0.4 and last nights  
> snapshot.  An suggestions or ideas?  There are no apparent errors  
> during the bootstrap or configure processes.
>
> Regards,
> Nicholas Lemberger
> Lakefield Communications
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mike at jerris.com  Fri Aug 14 11:13:47 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 14:13:47 -0400
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
	<20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
Message-ID: <EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>

thats in seconds.

Mike

On Aug 14, 2009, at 8:32 AM, mark morreny wrote:

> Hi,
>
> Thank you for your help.
>
> I get that too, but the callback does not execute the second time.
>
> When I do task->runtime = switch_time_now() + 10;, what does +10  
> mean?  Does it mean 10 s or 10 mins?
>
> Thanks,
> Mark
>
> On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene <mrene_lists at avgs.ca>  
> wrote:
> Hi,
>
> I did the same thing on my side....
>
> API CALL [load(mod_skel)] output:
> +OK
>
> 2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task  
> 2 data_flush (core) to run at 1250089698
> 2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889  
> Successfully Loaded [mod_skel]
> 2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270  
> Adding API Function 'skel'
> freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]  
> mod_skel.c:120 starting to flush data  buffer...
>
> Note that you don't need to start the thread manually, the core  
> already has threads running for the scheduler.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 12-Aug-09, at 7:26 AM, mark morreny wrote:
>
>> Hi,
>>
>> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out  
>> some data every 10 s.  The following lines of code does not show  
>> any effect at all.
>>
>>     switch_scheduler_task_thread_start();
>>     switch_scheduler_add_task(switch_epoch_time_now(NULL),  
>> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE| 
>> SSHF_NO_DEL);
>>
>>
>> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>>
>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting  
>> to flush data  buffer...\n");
>>
>>
>>     task->runtime = switch_time_now() + 10;
>>
>> }
>>
>> Does anyone know how to get it to work?
>>
>> Thanks,
>> Mark
>>
>>
>> ---------- Forwarded message ----------
>> From: Brian West <brian at freeswitch.org>
>> Date: Mon, Aug 10, 2009 at 8:53 PM
>> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
>> To: freeswitch-users at lists.freeswitch.org
>>
>>
>> switch_rtp.c has a simple one for the zrtp cache storing.
>>
>> /b
>>
>> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>>
>> > Re schedule is done in your callback, take a look at places that  
>> use
>> > these apis in the code for details.

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From mike at jerris.com  Fri Aug 14 11:20:31 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 14:20:31 -0400
Subject: [Freeswitch-users] sip contact header question
In-Reply-To: <27c25bc40908140902k78d5bbe1naa75a21df21caa6e@mail.gmail.com>
References: <27c25bc40908140902k78d5bbe1naa75a21df21caa6e@mail.gmail.com>
Message-ID: <543A09D4-B3A8-4BA0-8F5A-0C3C694EBB15@jerris.com>

How exactly are you setting the var?  It should be set on the b-leg,  
such as using [sip_contact_user=xxx] on the originate line.

Mike

On Aug 14, 2009, at 12:02 PM, Juan Backson wrote:

> Hi,
>
> I would like to set outbound INVITE with customer Contact: field.  I  
> did sip_contact_user="2033334545", but it still did not work.  I am  
> using 1.0.4 release.
>
> Is this a known issue?
>
> br,
> JB



From msc at freeswitch.org  Fri Aug 14 11:27:42 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 14 Aug 2009 13:27:42 -0500
Subject: [Freeswitch-users] Question about sharing conference between
In-Reply-To: <37033.1250266845@a2unlimited.com>
References: <37033.1250266845@a2unlimited.com>
Message-ID: <87f2f3b90908141127h5e216a06ta2ddd20c818fab9f@mail.gmail.com>

On Fri, Aug 14, 2009 at 11:20 AM, Tina Martinez <tina at a2unlimited.com>wrote:

> Michael,
>
> Thanks again for bearing with my novice perspective on this.
>
> I was able to achieve the link between two FS servers as intended.
>  However, I
> was not able to setup a "new" dialplan file as you described.  I had to
> place the
> script into the default.xml dialplan to get it to work.  Is there something
> I'm
> supposed to do to get FS to look in the conf/dialplan/default ?


Look at the sample files that are already in conf/dialplan/default/ and
conf/dialplan/public/ to see what they should look like. Just remember to
name the files with leading digits, so you have something like this:
conf/dialplan/default/01_My_Custom_Extensions.xml
conf/dialplan/public/01_More_Public_Extensions.xml

The files must be .xml files and preferable should have the <include> and
</include> tags at the beginning and end of the files. The other thing you
can do is just make a copy of one of the files that's already in
conf/dialplan/default/ or conf/dialplan/public/ and edit it.


>
>
> Also, in using the code as we discussed, I was also able to establish the
> link
> between the two servers without having an actual soft-phone
> registered/connected
> to the FS server (which is ideal for my situation).  I can create the link
> between the servers, and then dial out to external phone numbers from the
> respective servers and connect the participants.  Then, regardless of who
> hangs-up, the conference between the servers remains in tact.  So far it
> works
> very well.
>

Glad to hear it!
-MC


>
> - T
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bjbrashier at gmail.com  Fri Aug 14 11:41:41 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Fri, 14 Aug 2009 11:41:41 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <DB70D4BE-CD40-4707-AD0C-333A6A0BA75E@jerris.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com> 
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com> 
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
	<7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com> 
	<7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com> 
	<7bcfdd290908131524s10e209dcla7bebada79ecfa51@mail.gmail.com> 
	<DB70D4BE-CD40-4707-AD0C-333A6A0BA75E@jerris.com>
Message-ID: <7bcfdd290908141141w1372a10cw158f332e8bf13763@mail.gmail.com>

I didn't see any SIP session timers in the wiki. Since I'm already using the
event socket for control, my current plan is to use sched_api to play a file
with a short (20ms?) clip of silence, capture the play_file event and use it
to reschule another one for a couple of seconds later.

I'll let you know what happens.

BB

On Thu, Aug 13, 2009 at 10:47 PM, Michael Jerris <mike at jerris.com> wrote:

> My suggestion is to use sip session timers not rtp timeouts as rtp is
> supposed to be discontinuous.  That being said, we have several settings to
> continuously send media, but then you are doing exactly what you said you
> didn't want to do.
> Mike
>
>  On Aug 13, 2009, at 6:24 PM, Bradley Brashier wrote:
>
>  OK, I finally got a moment to do a packet capture and take a look at the
> streams.  It became very clear very quickly that what happens is that during
> silence the gateway still sends RTP packets to Freeswitch, but Freeswitch
> doesn't send any back to the gateway. After 10s of this, the gateway says
> "Oh, the RPT must be broken" and it hangs up.
>
> We found a way to turn off this behavior in the gateway, and the good news
> is that it did indeed fix the problem. But we'd rather not rely on that as a
> long-term solution because then we can't detect and drop RTP streams that
> really are broken.
>
> So now I'm back to looking at Freeswitch to figure out how to send just a
> single packet every second or so during silence. If anyone knows of a way to
> do this, let me know, otherwise I'll get back to you if and when I find one.
>
> BB
>
> On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
>
>> I took a closer look at the SIP messages on the console. From it, I
>> understand that it's not Freeswitch timing out, but rather FS is getting the
>> "BYE" msg from somewhere else. I've tested phones and tested calling without
>> going through the FS conference, though, and everything works fine. Then I
>> saw something else odd inside the BYE msg:
>>
>>    Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"
>> So I Googled "RTP Broken Connection" and saw several sites talking about
>> AudioCodes gateways and Asterisk -- and our gateway is an AudioCodes. From
>> these sites it sounds like AudioCodes is rather aggressive in detecting RTP
>> breaks, and is interpreting the silence from FS as a break.
>>
>> So now I'm looking into ways to maybe send "I'm still here" RTP packets
>> from FS or to tune the gateway to be less aggressive. I can't stop and get a
>> clean packet capture right now because I've got a bunch of testers working
>> on it today. I'll do that sometime when the system is less busy.
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
>>
>>> I had just thought of the exact same thing. I'm trying to test that now.
>>> Thanks for your input.
>>>
>>> BB
>>>
>>>   On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com>wrote:
>>>
>>>>   My guess is that its the other end killing the call due to rtp
>>>> timeouts, not us killing the call.  If you can confirm this the best method
>>>> would be to get them not to do rtp timeouts.
>>>>  On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>>>>
>>>>  I'm sure that would work, but I'm worried about it sucking up
>>>> bandwidth, especially since you'd need it on every caller (since otherwise
>>>> the one person who had it could hang up and you'd be back to square 1).
>>>>
>>>> Any other ideas, or should I hunt through the code to try to override
>>>> the behavior manually?
>>>>
>>>> BB
>>>>
>>>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>
>>>>> Check out the 'waste' member flag. I think if at least one member has
>>>>> that set then RTP will get sent out even during silence. Let us know if that
>>>>> helps...
>>>>>
>>>>> -MC
>>>>>
>>>>>   On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <
>>>>> bjbrashier at gmail.com> wrote:
>>>>>
>>>>>>   Hi all.
>>>>>>
>>>>>> The solution to this one should be short.
>>>>>>
>>>>>> My conference hangs up when there's 2+ users and silence for 5 sec or
>>>>>> so. I'm trying to find a parameter that changes that (I'd rather it be,
>>>>>> say, 60 seconds).
>>>>>>
>>>>>> I didn't see a parameter like this specific to conferences, so I
>>>>>> looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set
>>>>>> to 300 (the default), so I'm pretty sure that's not the problem. I also
>>>>>> searched through the mod_conference.c code and didn't see it, though I was
>>>>>> only skimming.
>>>>>>
>>>>>> I'm not 100% convinced that this is limited to conferences, but I
>>>>>> don't currently have a way to test in a non-conference environment.
>>>>>>
>>>>>> Anybody know how to increase the conference silence-hangup timeout?
>>>>>>
>>>>>> BB
>>>>>>
>>>>>> _____
>>>>>
>>>>>
>>>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From csa at nowthor.com  Fri Aug 14 11:52:25 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Fri, 14 Aug 2009 14:52:25 -0400
Subject: [Freeswitch-users] bind_server_ip issue
In-Reply-To: <CBF2E5B2-4707-4822-9E79-B81D1D86FA83@freeswitch.org>
References: <4A841FE0.90904@nowthor.com>	<12E3506D-3F79-4987-9FA7-B40AE461B97A@freeswitch.org>	<4A855525.2080301@nowthor.com>
	<CBF2E5B2-4707-4822-9E79-B81D1D86FA83@freeswitch.org>
Message-ID: <4A85B269.3070009@nowthor.com>

Thanks, Brian. If you don't mind my asking, what is the ultimate purpose 
of local_ip_v4, then?

Brian West wrote:
> Don't use local_ip_v4 then... please hard code the param in the sofia  
> profile because the value of local_ip_v4 can change.
>
> /b
>
> On Aug 14, 2009, at 7:14 AM, Carlos S. Antunes wrote:
>
>   
>> Hi Brian!
>>
>> Thank you for your quick response. I ended up defining "local_ip_v4"  
>> at the top of vars.xml and set "bind_server_ip" to "local_ip_v4" as  
>> well. Is this the best way to go about selecting an IP address for  
>> FS to bind to?
>>
>> In any case, even though it doesn't appear to affect operation, I am  
>> still getting this "IP change detected" somewhat periodically in my  
>> logs (running 1.0.4):
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
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From lon at kickasspixels.com  Fri Aug 14 12:00:37 2009
From: lon at kickasspixels.com (Lon Baker)
Date: Fri, 14 Aug 2009 12:00:37 -0700
Subject: [Freeswitch-users] Offline or background ASR
Message-ID: <5d3e0dc60908141200s70b7af5r76cbc75ddf3a615@mail.gmail.com>

Hi there,
Has anyone found a good solution for processing recordings/voicemail with
automatic speech recognition software?

I am researching numerous solutions, both open source and commercial, but
was wondering if I missed anything obvious.

Lon Baker
Kickass Pixels
-
(office) +1-415-287-0973
(mobile) +1-415-279-5019
(skype) lonbaker
-
http://kickasspixels.com
http://twitter.com/kickasspixels
http://www.linkedin.com/in/lonbaker
--
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From mike at jerris.com  Fri Aug 14 12:17:35 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 15:17:35 -0400
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <7bcfdd290908141141w1372a10cw158f332e8bf13763@mail.gmail.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com>
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com>
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com>
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
	<7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com>
	<7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com>
	<7bcfdd290908131524s10e209dcla7bebada79ecfa51@mail.gmail.com>
	<DB70D4BE-CD40-4707-AD0C-333A6A0BA75E@jerris.com>
	<7bcfdd290908141141w1372a10cw158f332e8bf13763@mail.gmail.com>
Message-ID: <4649090E-BBA8-4677-B8C5-6AF3D6F4B891@jerris.com>

That sounds horrible.  There are settings both in sip/rtp and in  
conference to do this already.

http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec
http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec
http://wiki.freeswitch.org/wiki/VAD_and_CNG

Mike

On Aug 14, 2009, at 2:41 PM, Bradley Brashier wrote:

> I didn't see any SIP session timers in the wiki. Since I'm already  
> using the event socket for control, my current plan is to use  
> sched_api to play a file with a short (20ms?) clip of silence,  
> capture the play_file event and use it to reschule another one for a  
> couple of seconds later.
>
> I'll let you know what happens.
>
> BB
>
> On Thu, Aug 13, 2009 at 10:47 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> My suggestion is to use sip session timers not rtp timeouts as rtp  
> is supposed to be discontinuous.  That being said, we have several  
> settings to continuously send media, but then you are doing exactly  
> what you said you didn't want to do.
>
> Mike
>
> On Aug 13, 2009, at 6:24 PM, Bradley Brashier wrote:
>
>> OK, I finally got a moment to do a packet capture and take a look  
>> at the streams.  It became very clear very quickly that what  
>> happens is that during silence the gateway still sends RTP packets  
>> to Freeswitch, but Freeswitch doesn't send any back to the gateway.  
>> After 10s of this, the gateway says "Oh, the RPT must be broken"  
>> and it hangs up.
>>
>> We found a way to turn off this behavior in the gateway, and the  
>> good news is that it did indeed fix the problem. But we'd rather  
>> not rely on that as a long-term solution because then we can't  
>> detect and drop RTP streams that really are broken.
>>
>> So now I'm back to looking at Freeswitch to figure out how to send  
>> just a single packet every second or so during silence. If anyone  
>> knows of a way to do this, let me know, otherwise I'll get back to  
>> you if and when I find one.
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier <bjbrashier at gmail.com 
>> > wrote:
>> I took a closer look at the SIP messages on the console. From it, I  
>> understand that it's not Freeswitch timing out, but rather FS is  
>> getting the "BYE" msg from somewhere else. I've tested phones and  
>> tested calling without going through the FS conference, though, and  
>> everything works fine. Then I saw something else odd inside the BYE  
>> msg:
>>
>>    Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"
>> So I Googled "RTP Broken Connection" and saw several sites talking  
>> about AudioCodes gateways and Asterisk -- and our gateway is an  
>> AudioCodes. From these sites it sounds like AudioCodes is rather  
>> aggressive in detecting RTP breaks, and is interpreting the silence  
>> from FS as a break.
>>
>> So now I'm looking into ways to maybe send "I'm still here" RTP  
>> packets from FS or to tune the gateway to be less aggressive. I  
>> can't stop and get a clean packet capture right now because I've  
>> got a bunch of testers working on it today. I'll do that sometime  
>> when the system is less busy.
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com 
>> > wrote:
>> I had just thought of the exact same thing. I'm trying to test that  
>> now.  Thanks for your input.
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com>  
>> wrote:
>> My guess is that its the other end killing the call due to rtp  
>> timeouts, not us killing the call.  If you can confirm this the  
>> best method would be to get them not to do rtp timeouts.
>>
>> On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>>
>>> I'm sure that would work, but I'm worried about it sucking up  
>>> bandwidth, especially since you'd need it on every caller (since  
>>> otherwise the one person who had it could hang up and you'd be  
>>> back to square 1).
>>>
>>> Any other ideas, or should I hunt through the code to try to  
>>> override the behavior manually?
>>>
>>> BB
>>>
>>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins  
>>> <msc at freeswitch.org> wrote:
>>> Check out the 'waste' member flag. I think if at least one member  
>>> has that set then RTP will get sent out even during silence. Let  
>>> us know if that helps...
>>>
>>> -MC
>>>
>>> On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <bjbrashier at gmail.com 
>>> > wrote:
>>> Hi all.
>>>
>>> The solution to this one should be short.
>>>
>>> My conference hangs up when there's 2+ users and silence for 5 sec  
>>> or so. I'm trying to find a parameter that changes that (I'd  
>>> rather it be, say, 60 seconds).
>>>
>>> I didn't see a parameter like this specific to conferences, so I  
>>> looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but  
>>> it's set to 300 (the default), so I'm pretty sure that's not the  
>>> problem. I also searched through the mod_conference.c code and  
>>> didn't see it, though I was only skimming.
>>>
>>> I'm not 100% convinced that this is limited to conferences, but I  
>>> don't currently have a way to test in a non-conference environment.
>>>
>>> Anybody know how to increase the conference silence-hangup timeout?
>>>
>>> BB
>>>
>>> _____
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From pjintheusa at gmail.com  Fri Aug 14 12:17:39 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 14 Aug 2009 15:17:39 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <b39273530908132333y52279d64qe926eca47e15b5eb@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<1250020512.4659.16.camel@dk-d820>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
	<b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
	<367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>
	<FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>
	<b39273530908132333y52279d64qe926eca47e15b5eb@mail.gmail.com>
Message-ID: <367751820908141217g2facd41av25f65d6321967d74@mail.gmail.com>

Hi Rupa,

What about my suggestion above introduce a "api_after_bridge" event
that fires when the switch_ivr_uuid_bridge() bridges to the two sofia
channels that Mathieu mentioned?

Is that suggestion just way off the mark? If possible that would allow
me to move forward - although I agree that supporting groupings of
carriers is would be the most elegant solution.

Let me know if I am just talking rubbish re the api_after_bridge" event.

Thanks!


Phillip Jones


On Fri, Aug 14, 2009 at 2:33 AM, Rupa Schomaker<rupa at rupa.com> wrote:
> On Thu, Aug 13, 2009 at 6:54 PM, Mathieu Rene<mrene_lists at avgs.ca> wrote:
>> Hi All,
>>
>> The reason it works when you wait 3 seconds is that mod_loopback bails
>
> [snip]
>
> Thanks for that explanation. ?It umm.. explains a lot. :)
>
>> On another note, mod_sofia will behave differently when it detects its
>> being bridge with another sofia channel, providing optimizations when
>> both call legs are SIP.
>>
>> My personal opinion is not to use mod_loopback unless absolutely
>> necessary, FreeSWITCH's core is very flexible and there's often a
>> (better) way than using mod_loopback.
>
> So, I think the temp solution is to use loopback+delayed no media.
>
> but the real "solution" is to either drive the forked dialing logic
> externally (event socket) or consider supporting groupings in the
> bridge which.. umm... ?is gonna be a pain and will need buy in from
> from Tony and other core devs since that is a core (no pun intended)
> piece of code that nearly everything uses.
>
> I'm not sure I want to take a wack at it.
>
> --
> -Rupa
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From bjbrashier at gmail.com  Fri Aug 14 12:44:54 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Fri, 14 Aug 2009 12:44:54 -0700
Subject: [Freeswitch-users] Conference silence timeouts
In-Reply-To: <4649090E-BBA8-4677-B8C5-6AF3D6F4B891@jerris.com>
References: <7bcfdd290908130937uef674f5hf04ab02f8ee5b3b@mail.gmail.com> 
	<87f2f3b90908130950i7f14c1betca768c2c568d439f@mail.gmail.com> 
	<7bcfdd290908131047t6c60d50eja1601a43f4ecb482@mail.gmail.com> 
	<BB20FF66-F3D6-40E4-A4B9-111432F4910C@jerris.com>
	<7bcfdd290908131345r573cbdf7k5e37c8fe7bf2e067@mail.gmail.com> 
	<7bcfdd290908131448p5442ee5dja4acc5537a82d8b9@mail.gmail.com> 
	<7bcfdd290908131524s10e209dcla7bebada79ecfa51@mail.gmail.com> 
	<DB70D4BE-CD40-4707-AD0C-333A6A0BA75E@jerris.com>
	<7bcfdd290908141141w1372a10cw158f332e8bf13763@mail.gmail.com> 
	<4649090E-BBA8-4677-B8C5-6AF3D6F4B891@jerris.com>
Message-ID: <7bcfdd290908141244k7e469966y2c69f665f15df1df@mail.gmail.com>

All right, I'm confused.

The RTP timeout parameters have no documentation, and from the names, I'd
have guessed that they hang up after a specified amount of time, not send
some other signal.

The session-timeout timer talks about calls expiring, and sending another
SIP invite, which I don't think is what I need. It also says it shouldn't be
set to less than 30 minutes.

I've been poking through RFC 4028, which the enable-timer parameter
mentions, but I'm not sure those will be helpful, either, because I've got
the SIP and RTP traffic going on different paths, and it's the RTP that's
having problems. Still, maybe I can use one of the timers to do something
different, but then I'm not sure how that's any better than using the event
socket the way I planned.

Clearly I'm missing something.

BB

On Fri, Aug 14, 2009 at 12:17 PM, Michael Jerris <mike at jerris.com> wrote:

> That sounds horrible.  There are settings both in sip/rtp and in conference
> to do this already.
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-timer
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#session-timeout
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-timeout-sec
>
> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#rtp-hold-timeout-sec
> http://wiki.freeswitch.org/wiki/VAD_and_CNG
>
> Mike
>
>  On Aug 14, 2009, at 2:41 PM, Bradley Brashier wrote:
>
>  I didn't see any SIP session timers in the wiki. Since I'm already using
> the event socket for control, my current plan is to use sched_api to play a
> file with a short (20ms?) clip of silence, capture the play_file event and
> use it to reschule another one for a couple of seconds later.
>
> I'll let you know what happens.
>
> BB
>
> On Thu, Aug 13, 2009 at 10:47 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> My suggestion is to use sip session timers not rtp timeouts as rtp is
>> supposed to be discontinuous.  That being said, we have several settings to
>> continuously send media, but then you are doing exactly what you said you
>> didn't want to do.
>> Mike
>>
>>  On Aug 13, 2009, at 6:24 PM, Bradley Brashier wrote:
>>
>>  OK, I finally got a moment to do a packet capture and take a look at the
>> streams.  It became very clear very quickly that what happens is that during
>> silence the gateway still sends RTP packets to Freeswitch, but Freeswitch
>> doesn't send any back to the gateway. After 10s of this, the gateway says
>> "Oh, the RPT must be broken" and it hangs up.
>>
>> We found a way to turn off this behavior in the gateway, and the good news
>> is that it did indeed fix the problem. But we'd rather not rely on that as a
>> long-term solution because then we can't detect and drop RTP streams that
>> really are broken.
>>
>> So now I'm back to looking at Freeswitch to figure out how to send just a
>> single packet every second or so during silence. If anyone knows of a way to
>> do this, let me know, otherwise I'll get back to you if and when I find one.
>>
>> BB
>>
>> On Thu, Aug 13, 2009 at 2:48 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
>>
>>> I took a closer look at the SIP messages on the console. From it, I
>>> understand that it's not Freeswitch timing out, but rather FS is getting the
>>> "BYE" msg from somewhere else. I've tested phones and tested calling without
>>> going through the FS conference, though, and everything works fine. Then I
>>> saw something else odd inside the BYE msg:
>>>
>>>    Reason: Q.850 ;cause=31 ;text="RTP Broken Connection"
>>> So I Googled "RTP Broken Connection" and saw several sites talking about
>>> AudioCodes gateways and Asterisk -- and our gateway is an AudioCodes. From
>>> these sites it sounds like AudioCodes is rather aggressive in detecting RTP
>>> breaks, and is interpreting the silence from FS as a break.
>>>
>>> So now I'm looking into ways to maybe send "I'm still here" RTP packets
>>> from FS or to tune the gateway to be less aggressive. I can't stop and get a
>>> clean packet capture right now because I've got a bunch of testers working
>>> on it today. I'll do that sometime when the system is less busy.
>>>
>>> BB
>>>
>>> On Thu, Aug 13, 2009 at 1:45 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:
>>>
>>>> I had just thought of the exact same thing. I'm trying to test that
>>>> now.  Thanks for your input.
>>>>
>>>> BB
>>>>
>>>>   On Thu, Aug 13, 2009 at 1:20 PM, Michael Jerris <mike at jerris.com>wrote:
>>>>
>>>>>   My guess is that its the other end killing the call due to rtp
>>>>> timeouts, not us killing the call.  If you can confirm this the best method
>>>>> would be to get them not to do rtp timeouts.
>>>>>  On Aug 13, 2009, at 1:47 PM, Bradley Brashier wrote:
>>>>>
>>>>>  I'm sure that would work, but I'm worried about it sucking up
>>>>> bandwidth, especially since you'd need it on every caller (since otherwise
>>>>> the one person who had it could hang up and you'd be back to square 1).
>>>>>
>>>>> Any other ideas, or should I hunt through the code to try to override
>>>>> the behavior manually?
>>>>>
>>>>> BB
>>>>>
>>>>> On Thu, Aug 13, 2009 at 9:50 AM, Michael Collins <msc at freeswitch.org>wrote:
>>>>>
>>>>>> Check out the 'waste' member flag. I think if at least one member has
>>>>>> that set then RTP will get sent out even during silence. Let us know if that
>>>>>> helps...
>>>>>>
>>>>>> -MC
>>>>>>
>>>>>>   On Thu, Aug 13, 2009 at 11:37 AM, Bradley Brashier <
>>>>>> bjbrashier at gmail.com> wrote:
>>>>>>
>>>>>>>   Hi all.
>>>>>>>
>>>>>>> The solution to this one should be short.
>>>>>>>
>>>>>>> My conference hangs up when there's 2+ users and silence for 5 sec or
>>>>>>> so. I'm trying to find a parameter that changes that (I'd rather it be,
>>>>>>> say, 60 seconds).
>>>>>>>
>>>>>>> I didn't see a parameter like this specific to conferences, so I
>>>>>>> looked abroad a bit. I found rtp-timeout-sec in sip_profiles, but it's set
>>>>>>> to 300 (the default), so I'm pretty sure that's not the problem. I also
>>>>>>> searched through the mod_conference.c code and didn't see it, though I was
>>>>>>> only skimming.
>>>>>>>
>>>>>>> I'm not 100% convinced that this is limited to conferences, but I
>>>>>>> don't currently have a way to test in a non-conference environment.
>>>>>>>
>>>>>>> Anybody know how to increase the conference silence-hangup timeout?
>>>>>>>
>>>>>>> BB
>>>>>>>
>>>>>>> _____
>>>>>>
>>>>>>
>>>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Fri Aug 14 14:02:06 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 14 Aug 2009 23:02:06 +0200
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness patch
In-Reply-To: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
Message-ID: <7b197bef0908141402k15d4144bt87a287e07c51b5c6@mail.gmail.com>

svn 14521: skypiax: compiles on windoz, not yet tested (on windoz)



On Fri, Aug 14, 2009 at 7:43 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org> wrote:
> Hi FreeSWITCHers,
>
> all the users of mod_skypiax are kindly requested to test the svn trunk 14519.
>
> It contains a lot of changes meant to add stability and robustness,
> toward a production environment.
>
> Let me know how your feelings, and please add to the Jira any possible
> bug/issue/etc.
>
> Thanks to you all,
>
> -giovanni
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
>
> Cell : +39-347-2665618
>



From marc at kasteris.com  Fri Aug 14 15:27:30 2009
From: marc at kasteris.com (Marc Orenberg)
Date: Fri, 14 Aug 2009 15:27:30 -0700 (PDT)
Subject: [Freeswitch-users] FreeSWITCH crashes when bridging two calls in
	python script
Message-ID: <492335.3738.qm@web50806.mail.re2.yahoo.com>

Hello, 

I'm trying to play a prompt to the B-leg of a bridged call in Python. 
I place the call to the B-leg, play the prompt, and then bridge it with the A-Leg, but then  FreeSWITCH crashes when the call is completed.

Here's the code I'm using:

def bridge_call_with_prompt(session,carrier,caller_id,phonenum,promptfile,recordfile):
  try:
    sessiondata = "{ignore_early_media=true,origination_caller_id_number=" + str(caller_id) + "}sofia/gateway/" + carrier + "/" + phonenum
    new_session =  Session(sessiondata)
    if (not new_session.ready()): return(False)
    if (recordfile <> ""): new_session.streamFile(promptfile)
    new_session.execute("set","call_timeout=60")
    new_session.execute("set","continue_on_fail=true")
    new_session.execute("set","hangup_after_bridge=false")
    new_session.execute("set", "ringback=%(2000,4000,440.0,480.0)")
    if (not new_session.ready()): return(False)
    if (not session.ready()): return(False)
    bridge(session,new_session)
    new_session.hangup()
  except:
    report_exception()
    return(False)
  return(True)

Does anybody know how I can stop it from crashing, and/or another way to go about this?   If I simply do:

session.execute("bridge","sofia/gateway/" + carrier + "/" + phonenum)


it won't cause FreeSWITCH to crash, but then I can't play my prompt.

Thanks,
Marc
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From markmorreny at gmail.com  Fri Aug 14 19:19:09 2009
From: markmorreny at gmail.com (mark morreny)
Date: Sat, 15 Aug 2009 10:19:09 +0800
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
	<20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
	<EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>
Message-ID: <20ad6b920908141919u1b2aa1c4jacc75cd4896d72b0@mail.gmail.com>

Hi Michael,

The following code was executed once, but not after the next 10 s.

SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to flush
data  buffer...\n");

    task->runtime = switch_time_now() + 10;
}

Any suggestion why?


Thanks,
Mark
On Sat, Aug 15, 2009 at 2:13 AM, Michael Jerris <mike at jerris.com> wrote:

> thats in seconds.
> Mike
>
>  On Aug 14, 2009, at 8:32 AM, mark morreny wrote:
>
>  Hi,
>
> Thank you for your help.
>
> I get that too, but the callback does not execute the second time.
>
> When I do task->runtime = switch_time_now() + 10;, what does +10 mean?
> Does it mean 10 s or 10 mins?
>
> Thanks,
> Mark
>
> On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene <mrene_lists at avgs.ca>wrote:
>
>> Hi,
>> I did the same thing on my side....
>>  API CALL [load(mod_skel)] output:
>> +OK
>>
>> 2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task 2
>> data_flush (core) to run at 1250089698
>> 2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889
>> Successfully Loaded [mod_skel]
>> 2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270 Adding API
>> Function 'skel'
>> freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]
>> mod_skel.c:120 starting to flush data  buffer...
>>
>> Note that you don't need to start the thread manually, the core already
>> has threads running for the scheduler.
>>
>>   Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>>  On 12-Aug-09, at 7:26 AM, mark morreny wrote:
>>
>> Hi,
>>
>> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out some data
>> every 10 s.  The following lines of code does not show any effect at all.
>>
>>     switch_scheduler_task_thread_start();
>>     switch_scheduler_add_task(switch_epoch_time_now(NULL),
>> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|SSHF_NO_DEL);
>>
>>
>> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>>
>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to flush
>> data  buffer...\n");
>>
>>
>>     task->runtime = switch_time_now() + 10;
>>
>> }
>>
>> Does anyone know how to get it to work?
>>
>> Thanks,
>> Mark
>>
>>
>> ---------- Forwarded message ----------
>> From: Brian West <brian at freeswitch.org>
>> Date: Mon, Aug 10, 2009 at 8:53 PM
>> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
>> To: freeswitch-users at lists.freeswitch.org
>>
>>
>> switch_rtp.c has a simple one for the zrtp cache storing.
>>
>> /b
>>
>> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>>
>> > Re schedule is done in your callback, take a look at places that use
>> > these apis in the code for details.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Fri Aug 14 20:16:47 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 23:16:47 -0400
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <20ad6b920908141919u1b2aa1c4jacc75cd4896d72b0@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
	<20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
	<EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>
	<20ad6b920908141919u1b2aa1c4jacc75cd4896d72b0@mail.gmail.com>
Message-ID: <EF4B4642-E586-42E0-8D40-D9FCB7A93DD2@jerris.com>

  task->runtime = switch_epoch_time_now(NULL) + 10;

On Aug 14, 2009, at 10:19 PM, mark morreny wrote:

> Hi Michael,
>
> The following code was executed once, but not after the next 10 s.
>
> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to  
> flush data  buffer...\n");
>
>     task->runtime = switch_time_now() + 10;
> }
>
> Any suggestion why?
>
>
> Thanks,
> Mark
> On Sat, Aug 15, 2009 at 2:13 AM, Michael Jerris <mike at jerris.com>  
> wrote:
> thats in seconds.
>
> Mike
>
> On Aug 14, 2009, at 8:32 AM, mark morreny wrote:
>
>> Hi,
>>
>> Thank you for your help.
>>
>> I get that too, but the callback does not execute the second time.
>>
>> When I do task->runtime = switch_time_now() + 10;, what does +10  
>> mean?  Does it mean 10 s or 10 mins?
>>
>> Thanks,
>> Mark
>>
>> On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene  
>> <mrene_lists at avgs.ca> wrote:
>> Hi,
>>
>> I did the same thing on my side....
>>
>> API CALL [load(mod_skel)] output:
>> +OK
>>
>> 2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task  
>> 2 data_flush (core) to run at 1250089698
>> 2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889  
>> Successfully Loaded [mod_skel]
>> 2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270  
>> Adding API Function 'skel'
>> freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]  
>> mod_skel.c:120 starting to flush data  buffer...
>>
>> Note that you don't need to start the thread manually, the core  
>> already has threads running for the scheduler.
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 12-Aug-09, at 7:26 AM, mark morreny wrote:
>>
>>> Hi,
>>>
>>> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out  
>>> some data every 10 s.  The following lines of code does not show  
>>> any effect at all.
>>>
>>>     switch_scheduler_task_thread_start();
>>>     switch_scheduler_add_task(switch_epoch_time_now(NULL),  
>>> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE| 
>>> SSHF_NO_DEL);
>>>
>>>
>>> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>>>
>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting  
>>> to flush data  buffer...\n");
>>>
>>>
>>>     task->runtime = switch_time_now() + 10;
>>>
>>> }
>>>
>>> Does anyone know how to get it to work?
>>>
>>> Thanks,
>>> Mark
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Brian West <brian at freeswitch.org>
>>> Date: Mon, Aug 10, 2009 at 8:53 PM
>>> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
>>> To: freeswitch-users at lists.freeswitch.org
>>>
>>>
>>> switch_rtp.c has a simple one for the zrtp cache storing.
>>>
>>> /b
>>>
>>> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>>>
>>> > Re schedule is done in your callback, take a look at places that  
>>> use
>>> > these apis in the code for details.
>
>

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From mike at jerris.com  Fri Aug 14 20:17:44 2009
From: mike at jerris.com (Michael Jerris)
Date: Fri, 14 Aug 2009 23:17:44 -0400
Subject: [Freeswitch-users] FreeSWITCH crashes when bridging two calls
	in python script
In-Reply-To: <492335.3738.qm@web50806.mail.re2.yahoo.com>
References: <492335.3738.qm@web50806.mail.re2.yahoo.com>
Message-ID: <D1D6EEEA-C269-4E6F-A055-373D37CB1B10@jerris.com>

Please open a bug on jira.freeswitch.org

mike

On Aug 14, 2009, at 6:27 PM, Marc Orenberg wrote:

> Hello,
>
> I'm trying to play a prompt to the B-leg of a bridged call in Python.
> I place the call to the B-leg, play the prompt, and then bridge it  
> with the A-Leg, but then  FreeSWITCH crashes when the call is  
> completed.
>
> Here's the code I'm using:
>
> def  
> bridge_call_with_prompt 
> (session,carrier,caller_id,phonenum,promptfile,recordfile):
>   try:
>     sessiondata =  
> "{ignore_early_media=true,origination_caller_id_number=" +  
> str(caller_id) + "}sofia/gateway/" + carrier + "/" + phonenum
>     new_session =  Session(sessiondata)
>     if (not new_session.ready()): return(False)
>     if (recordfile <> ""): new_session.streamFile(promptfile)
>     new_session.execute("set","call_timeout=60")
>     new_session.execute("set","continue_on_fail=true")
>     new_session.execute("set","hangup_after_bridge=false")
>     new_session.execute("set", "ringback=%(2000,4000,440.0,480.0)")
>     if (not new_session.ready()): return(False)
>     if (not session.ready()): return(False)
>     bridge(session,new_session)
>     new_session.hangup()
>   except:
>     report_exception()
>     return(False)
>   return(True)
>
> Does anybody know how I can stop it from crashing, and/or another  
> way to go about this?   If I simply do:
>
>  	session.execute("bridge","sofia/gateway/" + carrier + "/" +  
> phonenum)
>
> it won't cause FreeSWITCH to crash, but then I can't play my prompt.
>
> Thanks,
> Marc

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From mrene_lists at avgs.ca  Fri Aug 14 20:20:07 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 14 Aug 2009 23:20:07 -0400
Subject: [Freeswitch-users] FreeSWITCH crashes when bridging two calls
	in python script
In-Reply-To: <492335.3738.qm@web50806.mail.re2.yahoo.com>
References: <492335.3738.qm@web50806.mail.re2.yahoo.com>
Message-ID: <2932436E-C699-4090-A658-BE8F17D255F3@avgs.ca>

Hi,

Can you provide us with a backtrace of the crash? you can open a bug  
report on http://jira.freeswitch.org/

Also, if you want to play a prompt file and wait for a dtmf to accept  
the call, there are variables called group_confirm_file and  
group_confirm_key that will make the core take care of that for you.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 14-Aug-09, at 6:27 PM, Marc Orenberg wrote:

> Hello,
>
> I'm trying to play a prompt to the B-leg of a bridged call in Python.
> I place the call to the B-leg, play the prompt, and then bridge it  
> with the A-Leg, but then  FreeSWITCH crashes when the call is  
> completed.
>
> Here's the code I'm using:
>
> def  
> bridge_call_with_prompt 
> (session,carrier,caller_id,phonenum,promptfile,recordfile):
>   try:
>     sessiondata =  
> "{ignore_early_media=true,origination_caller_id_number=" +  
> str(caller_id) + "}sofia/gateway/" + carrier + "/" + phonenum
>     new_session =  Session(sessiondata)
>     if (not new_session.ready()): return(False)
>     if (recordfile <> ""): new_session.streamFile(promptfile)
>     new_session.execute("set","call_timeout=60")
>     new_session.execute("set","continue_on_fail=true")
>     new_session.execute("set","hangup_after_bridge=false")
>     new_session.execute("set", "ringback=%(2000,4000,440.0,480.0)")
>     if (not new_session.ready()): return(False)
>     if (not session.ready()): return(False)
>     bridge(session,new_session)
>     new_session.hangup()
>   except:
>     report_exception()
>     return(False)
>   return(True)
>
> Does anybody know how I can stop it from crashing, and/or another  
> way to go about this?   If I simply do:
>
>  	session.execute("bridge","sofia/gateway/" + carrier + "/" +  
> phonenum)
>
> it won't cause FreeSWITCH to crash, but then I can't play my prompt.
>
> Thanks,
> Marc
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mrene_lists at avgs.ca  Fri Aug 14 20:26:24 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 14 Aug 2009 23:26:24 -0400
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <EF4B4642-E586-42E0-8D40-D9FCB7A93DD2@jerris.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<c4d05cbe0908090905g16d15493i65db0b1383c4ae1f@mail.gmail.com>
	<20ad6b920908100358t782a5929jea5f86bef08b674d@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
	<20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
	<EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>
	<20ad6b920908141919u1b2aa1c4jacc75cd4896d72b0@mail.gmail.com>
	<EF4B4642-E586-42E0-8D40-D9FCB7A93DD2@jerris.com>
Message-ID: <FD54C936-D80E-4F96-8502-740ADDD1225C@avgs.ca>

Because switch_time_now() is in microseconds.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 14-Aug-09, at 11:16 PM, Michael Jerris wrote:

>  task->runtime = switch_epoch_time_now(NULL) + 10;
>
> On Aug 14, 2009, at 10:19 PM, mark morreny wrote:
>
>> Hi Michael,
>>
>> The following code was executed once, but not after the next 10 s.
>>
>> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting  
>> to flush data  buffer...\n");
>>
>>     task->runtime = switch_time_now() + 10;
>> }
>>
>> Any suggestion why?
>>
>>
>> Thanks,
>> Mark
>> On Sat, Aug 15, 2009 at 2:13 AM, Michael Jerris <mike at jerris.com>  
>> wrote:
>> thats in seconds.
>>
>> Mike
>>
>> On Aug 14, 2009, at 8:32 AM, mark morreny wrote:
>>
>>> Hi,
>>>
>>> Thank you for your help.
>>>
>>> I get that too, but the callback does not execute the second time.
>>>
>>> When I do task->runtime = switch_time_now() + 10;, what does +10  
>>> mean?  Does it mean 10 s or 10 mins?
>>>
>>> Thanks,
>>> Mark
>>>
>>> On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene  
>>> <mrene_lists at avgs.ca> wrote:
>>> Hi,
>>>
>>> I did the same thing on my side....
>>>
>>> API CALL [load(mod_skel)] output:
>>> +OK
>>>
>>> 2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added  
>>> task 2 data_flush (core) to run at 1250089698
>>> 2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889  
>>> Successfully Loaded [mod_skel]
>>> 2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270  
>>> Adding API Function 'skel'
>>> freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]  
>>> mod_skel.c:120 starting to flush data  buffer...
>>>
>>> Note that you don't need to start the thread manually, the core  
>>> already has threads running for the scheduler.
>>>
>>> Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> mrene at avgs.ca
>>>
>>>
>>>
>>>
>>> On 12-Aug-09, at 7:26 AM, mark morreny wrote:
>>>
>>>> Hi,
>>>>
>>>> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out  
>>>> some data every 10 s.  The following lines of code does not show  
>>>> any effect at all.
>>>>
>>>>     switch_scheduler_task_thread_start();
>>>>     switch_scheduler_add_task(switch_epoch_time_now(NULL),  
>>>> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE| 
>>>> SSHF_NO_DEL);
>>>>
>>>>
>>>> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>>>>
>>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting  
>>>> to flush data  buffer...\n");
>>>>
>>>>
>>>>     task->runtime = switch_time_now() + 10;
>>>>
>>>> }
>>>>
>>>> Does anyone know how to get it to work?
>>>>
>>>> Thanks,
>>>> Mark
>>>>
>>>>
>>>> ---------- Forwarded message ----------
>>>> From: Brian West <brian at freeswitch.org>
>>>> Date: Mon, Aug 10, 2009 at 8:53 PM
>>>> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
>>>> To: freeswitch-users at lists.freeswitch.org
>>>>
>>>>
>>>> switch_rtp.c has a simple one for the zrtp cache storing.
>>>>
>>>> /b
>>>>
>>>> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>>>>
>>>> > Re schedule is done in your callback, take a look at places  
>>>> that use
>>>> > these apis in the code for details.
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From hyppolite72 at yahoo.com  Sat Aug 15 07:17:23 2009
From: hyppolite72 at yahoo.com (Jean-Marc Hyppolite)
Date: Sat, 15 Aug 2009 07:17:23 -0700 (PDT)
Subject: [Freeswitch-users] Question about an ESL function
Message-ID: <788835.43200.qm@web35604.mail.mud.yahoo.com>

Hello,
?
I would like to know the purpose of the ESL function named recvEventTimed.
?
Thank you for your help.


      __________________________________________________________________
Be smarter than spam. See how smart SpamGuard is at giving junk email the boot with the All-new Yahoo! Mail.  Click on Options in Mail and switch to New Mail today or register for free at http://mail.yahoo.ca
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From dujinfang at gmail.com  Sat Aug 15 08:41:29 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sat, 15 Aug 2009 23:41:29 +0800
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness patch
In-Reply-To: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
Message-ID: <38D6F955-9003-4862-ABA9-A79778508A46@gmail.com>

Great works. I tested and reported results in jira.

And as I noticed you removed the sequential line hunting methods.  
Though I don't use that I think someone else may need that. Think  
about the guy want skypeout accounts in a round robin manner, others  
might use that in a priority manner, that means sequential search. So  
it maybe a good idea to add it back. At the same time I think "ANY" is  
not a good name for that method, so named it "SEQ" maybe better. Then  
"ANY" can be removed later by announcing here or keep there for  
backwards compatibility.

Thank you very much for merging in the sk list with statistic patch,  
and for the other two features, I found it's a little hard to split  
codes, so, can make two jira, and upload one patch file?

Two features are:

continue load on fail: make sure the module continue load even it  
failed to talk to a skype instance
auto skype user: get the user name by the returned CURRENTUSERHANDLE  
other than  from the config xml, for easier config.

Thanks.

-7-

On Aug 15, 2009, at 1:43 AM, Giovanni Maruzzelli wrote:
> Hi FreeSWITCHers,
>
> all the users of mod_skypiax are kindly requested to test the svn  
> trunk 14519.
>
> It contains a lot of changes meant to add stability and robustness,
> toward a production environment.
>
> Let me know how your feelings, and please add to the Jira any possible
> bug/issue/etc.
>
> Thanks to you all,
>
> -giovanni
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
>
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From gmaruzz at celliax.org  Sat Aug 15 09:02:47 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Sat, 15 Aug 2009 18:02:47 +0200
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness 	patch
In-Reply-To: <38D6F955-9003-4862-ABA9-A79778508A46@gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com> 
	<38D6F955-9003-4862-ABA9-A79778508A46@gmail.com>
Message-ID: <7b197bef0908150902x18d4a68am7fb47dc76f781565@mail.gmail.com>

On Sat, Aug 15, 2009 at 5:41 PM, Seven Du<dujinfang at gmail.com> wrote:
> And as I noticed you removed the sequential line hunting methods.

Because was broken. So, I aliased it to the RR.
If you think it can be useful, add a Jira for it

> Thank you very much for merging in the sk list with statistic patch,
Thanks to you for sending the patch! I've only added the callflow of
the skype client to it

>
> Two features are:
>
> continue load on fail: make sure the module continue load even it
> failed to talk to a skype instance

mmmmh, I'm too conservative for this one: I prefer that if you
configured a skype instance, you expect it to work, so the module must
fail if there is not such instance

> auto skype user: get the user name by the returned CURRENTUSERHANDLE
> other than ?from the config xml, for easier config.

the username returned by CURRENTUSERHANDLE is checked against the
config file because is the only way you can associate interface_name
with its related Skype client instance on Windoz (no multiple X
servers there).

Thanks a lot for all your efforts!!!

-giovanni




>
> On Aug 15, 2009, at 1:43 AM, Giovanni Maruzzelli wrote:
>> Hi FreeSWITCHers,
>>
>> all the users of mod_skypiax are kindly requested to test the svn
>> trunk 14519.
>>
>> It contains a lot of changes meant to add stability and robustness,
>> toward a production environment.
>>
>> Let me know how your feelings, and please add to the Jira any possible
>> bug/issue/etc.
>>
>> Thanks to you all,
>>
>> -giovanni
>>
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>>
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From dujinfang at gmail.com  Sat Aug 15 09:41:19 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 16 Aug 2009 00:41:19 +0800
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness 	patch
In-Reply-To: <7b197bef0908150902x18d4a68am7fb47dc76f781565@mail.gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
	<38D6F955-9003-4862-ABA9-A79778508A46@gmail.com>
	<7b197bef0908150902x18d4a68am7fb47dc76f781565@mail.gmail.com>
Message-ID: <EABE5B4A-4572-400E-BA60-0512CA329982@gmail.com>


On Aug 16, 2009, at 12:02 AM, Giovanni Maruzzelli wrote:
> On Sat, Aug 15, 2009 at 5:41 PM, Seven Du<dujinfang at gmail.com> wrote:
>> And as I noticed you removed the sequential line hunting methods.
>
> Because was broken. So, I aliased it to the RR.
> If you think it can be useful, add a Jira for it
>

Ok, as you think it's broken, better to leave it as is.
>> Thank you very much for merging in the sk list with statistic patch,
> Thanks to you for sending the patch! I've only added the callflow of
> the skype client to it
>
I think maybe you merged in an old version of the patch, the last  
should be skype_status_new1.diff, which I added one more statistic  
line and the unsigned long should be uint32_t I think. however, I can  
add another patch if you think it's useful. Oh, one more, it also set  
channel name to skypiax/RR/sk_1/other_skype_name for easy check in log. 
(As I also would like to see sofia/external/xxxx to be sofia/gateway/ 
gw/xxxx :) ).

>>
>> Two features are:
>>
>> continue load on fail: make sure the module continue load even it
>> failed to talk to a skype instance
>
> mmmmh, I'm too conservative for this one: I prefer that if you
> configured a skype instance, you expect it to work, so the module must
> fail if there is not such instance
>

One of the reason I think it's useful is one can configure to load  
everything on server boot time.  I run two skypiax servers, one can  
start 20 instances in batch without any problem but the other only  
starts 50%, then I need manually start them over and over until I  
confirmed all works with client or skypiax_auth. The two servers are  
not in the same datacenter but all have public ip. Once it started  
working we never never experienced a skype instance stoped working.  
But I experienced that kill a skype instance immediately caused  
skypiax core dump.

Sure it might cause other bugs even it is configured to false by  
default. I gona merge in my branch in case others using that.

>> auto skype user: get the user name by the returned CURRENTUSERHANDLE
>> other than  from the config xml, for easier config.
>
> the username returned by CURRENTUSERHANDLE is checked against the
> config file because is the only way you can associate interface_name
> with its related Skype client instance on Windoz (no multiple X
> servers there).
>

It by default disabled so I guess nothing will break.

> Thanks a lot for all your efforts!!!
>
> -giovanni
>
>
>
>
>>
>> On Aug 15, 2009, at 1:43 AM, Giovanni Maruzzelli wrote:
>>> Hi FreeSWITCHers,
>>>
>>> all the users of mod_skypiax are kindly requested to test the svn
>>> trunk 14519.
>>>
>>> It contains a lot of changes meant to add stability and robustness,
>>> toward a production environment.
>>>
>>> Let me know how your feelings, and please add to the Jira any  
>>> possible
>>> bug/issue/etc.
>>>
>>> Thanks to you all,
>>>
>>> -giovanni
>>>
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>>
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From peder at networkoblivion.com  Sat Aug 15 11:13:25 2009
From: peder at networkoblivion.com (Peder)
Date: Sat, 15 Aug 2009 13:13:25 -0500
Subject: [Freeswitch-users] ClueCon2009 Torrents
Message-ID: <018d01ca1dd4$14c97620$3e5c6260$@com>

I created torrents for all of the presentations from ClueCon 2009 and have
them seeded on 2-3 machines.  If you want the torrents, email me off list.

Peder





From gabe at gundy.org  Sat Aug 15 12:33:00 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Sat, 15 Aug 2009 13:33:00 -0600
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <018d01ca1dd4$14c97620$3e5c6260$@com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
Message-ID: <903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>

On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com> wrote:
> If you want the torrents, email me off list.

Why off list?  Isn't the point of torrents to have more people sharing
in the load?

Gabe



From scott.torr.fs at letterboxes.org  Sat Aug 15 10:31:32 2009
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Sun, 16 Aug 2009 03:31:32 +1000
Subject: [Freeswitch-users] Audio only in one direction when calling FS from
	Skype
Message-ID: <1250357492.23345.1330000179@webmail.messagingengine.com>

Freeswitch-users,

I'm very new to Freeswitch and have installed the following in VMware
Server 2.0.1


ubuntu-8.04.3-server-i386.iso (udate/upgrade)
skype-debian_2.0.0.72-1_i386.deb
FreeSWITCH Version 1.0.trunk (14492)
mod_skypiax


Using the FS wiki to install/setup and making small changes to default
XML configs as required.


(phone)--(SIP ATA)--(FS)--(MS-Skype client | Skype 'online number' via
PSTN)


The MS-Skype client is on the same local network as FS.


I can make calls from the phone to the Skype Client and this works OK.
Audio path OK: phone<-->FS<-->Skype


But ,if I call from the MS-Skype client to the phone I hear no audio
from the MS-Skype client.  

Audio path: phone<-->FS-->Skype

Likewise a call from a Skype 'online number' can hear for example the
default 5000 ivr but the DTMF tones from the PSTN phone are not detected
by FS when <action application="start_dtmf" /> is used.



Any suggestions or pointers in the right direction would be much
appreciated. 

Most likely I have overlook something very obvious to others.


Thanks in advance,
Scott Torr 








From manoj.joshi.13jan at gmail.com  Sat Aug 15 12:33:41 2009
From: manoj.joshi.13jan at gmail.com (Manu)
Date: Sun, 16 Aug 2009 01:03:41 +0530
Subject: [Freeswitch-users] mod nibblebill question
Message-ID: <26474d4e0908151233s1376ab15h74d2ac32b58593bf@mail.gmail.com>

Hello,

If we use heartbeat option on in nibblebill.conf.xml does that mean ODBC
database table will be updated every microsecond or any other interval we
set?

If this is so and there are many users (Lets say 500 users) are connected to
FS wouldn't it create locking issues in DB?

Regards,

Manoj
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From jmesquita at gmail.com  Sat Aug 15 13:34:05 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 15 Aug 2009 17:34:05 -0300
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
Message-ID: <5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>

I am interested and would also seed to the community

On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com> wrote:
>> If you want the torrents, email me off list.
>
> Why off list?  Isn't the point of torrents to have more people sharing
> in the load?
>
> Gabe
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device



From jmesquita at gmail.com  Sat Aug 15 13:49:35 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 15 Aug 2009 17:49:35 -0300
Subject: [Freeswitch-users] Question about an ESL function
In-Reply-To: <788835.43200.qm@web35604.mail.mud.yahoo.com>
References: <788835.43200.qm@web35604.mail.mud.yahoo.com>
Message-ID: <5a8712120908151349p7f584ab3wa07f679d04b7c06f@mail.gmail.com>

Recv will lock the calling thread until it gets an event or gets
disconnected while recvtimed will return controll when event received
or timer expires. Whatever comes fisrt.

On 8/15/09, Jean-Marc Hyppolite <hyppolite72 at yahoo.com> wrote:
> Hello,
>
> I would like to know the purpose of the ESL function named recvEventTimed.
>
> Thank you for your help.
>
>
>       __________________________________________________________________
> Be smarter than spam. See how smart SpamGuard is at giving junk email the
> boot with the All-new Yahoo! Mail.  Click on Options in Mail and switch to
> New Mail today or register for free at http://mail.yahoo.ca

-- 
Sent from my mobile device



From mike at jerris.com  Sat Aug 15 14:20:11 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 15 Aug 2009 17:20:11 -0400
Subject: [Freeswitch-users] mod nibblebill question
In-Reply-To: <26474d4e0908151233s1376ab15h74d2ac32b58593bf@mail.gmail.com>
References: <26474d4e0908151233s1376ab15h74d2ac32b58593bf@mail.gmail.com>
Message-ID: <4E77075F-4EF0-4CD0-A36A-D68DA8F438EC@jerris.com>

increments are in seconds, not microseconds.  In IMS for example I  
think it defaults to 20 or 30 second nibbles, depending on your  
tolerances and billing increments something much larger may even make  
sense.  Doing billing in sub second increments doesn't make a lot of  
sense to me.  Remember that this is just keeping track of available  
credit so if there are multiple calls at the same time you won't go  
over balance.  Everything is still reconciled at hang up, so if you  
have a bit too much reserved from your nibble the worst that could  
happen is it could cut off calls a little too early when multiple  
calls are in progress on the same account.

Mike

On Aug 15, 2009, at 3:33 PM, Manu wrote:

> Hello,
>
> If we use heartbeat option on in nibblebill.conf.xml does that mean  
> ODBC database table will be updated every microsecond or any other  
> interval we set?
>
> If this is so and there are many users (Lets say 500 users) are  
> connected to FS wouldn't it create locking issues in DB?
>
> Regards,
>
> Manoj



From mike at jerris.com  Sat Aug 15 14:20:59 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 15 Aug 2009 17:20:59 -0400
Subject: [Freeswitch-users] Audio only in one direction when calling FS
	from Skype
In-Reply-To: <1250357492.23345.1330000179@webmail.messagingengine.com>
References: <1250357492.23345.1330000179@webmail.messagingengine.com>
Message-ID: <0031C8A1-BCE9-445F-8DE0-177F24D576F5@jerris.com>

Make sure your external ip addresses and local networks are configured  
properly.

Mike

On Aug 15, 2009, at 1:31 PM, Scott Torr wrote:

> Freeswitch-users,
>
> I'm very new to Freeswitch and have installed the following in VMware
> Server 2.0.1
>
>
> ubuntu-8.04.3-server-i386.iso (udate/upgrade)
> skype-debian_2.0.0.72-1_i386.deb
> FreeSWITCH Version 1.0.trunk (14492)
> mod_skypiax
>
>
> Using the FS wiki to install/setup and making small changes to default
> XML configs as required.
>
>
> (phone)--(SIP ATA)--(FS)--(MS-Skype client | Skype 'online number' via
> PSTN)
>
>
> The MS-Skype client is on the same local network as FS.
>
>
> I can make calls from the phone to the Skype Client and this works OK.
> Audio path OK: phone<-->FS<-->Skype
>
>
> But ,if I call from the MS-Skype client to the phone I hear no audio
> from the MS-Skype client.
>
> Audio path: phone<-->FS-->Skype
>
> Likewise a call from a Skype 'online number' can hear for example the
> default 5000 ivr but the DTMF tones from the PSTN phone are not  
> detected
> by FS when <action application="start_dtmf" /> is used.
>
>
>
> Any suggestions or pointers in the right direction would be much
> appreciated.
>
> Most likely I have overlook something very obvious to others.
>
>
> Thanks in advance,
> Scott Torr
>




From hyppolite72 at yahoo.com  Sat Aug 15 14:39:32 2009
From: hyppolite72 at yahoo.com (Jean-Marc Hyppolite)
Date: Sat, 15 Aug 2009 14:39:32 -0700 (PDT)
Subject: [Freeswitch-users] Question about an ESL function
In-Reply-To: <5a8712120908151349p7f584ab3wa07f679d04b7c06f@mail.gmail.com>
Message-ID: <82725.8785.qm@web35601.mail.mud.yahoo.com>

Thanks a lot.

--- On Sat, 8/15/09, Jo?o Mesquita <jmesquita at gmail.com> wrote:


From: Jo?o Mesquita <jmesquita at gmail.com>
Subject: Re: [Freeswitch-users] Question about an ESL function
To: freeswitch-users at lists.freeswitch.org
Received: Saturday, August 15, 2009, 4:49 PM


Recv will lock the calling thread until it gets an event or gets
disconnected while recvtimed will return controll when event received
or timer expires. Whatever comes fisrt.

On 8/15/09, Jean-Marc Hyppolite <hyppolite72 at yahoo.com> wrote:
> Hello,
>
> I would like to know the purpose of the ESL function named recvEventTimed.
>
> Thank you for your help.
>
>
>? ? ???__________________________________________________________________
> Be smarter than spam. See how smart SpamGuard is at giving junk email the
> boot with the All-new Yahoo! Mail.? Click on Options in Mail and switch to
> New Mail today or register for free at http://mail.yahoo.ca

-- 
Sent from my mobile device

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



      __________________________________________________________________
Ask a question on any topic and get answers from real people. Go to Yahoo! Answers and share what you know at http://ca.answers.yahoo.com
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From jaybinks at gmail.com  Sat Aug 15 14:43:20 2009
From: jaybinks at gmail.com (Jay Binks)
Date: Sun, 16 Aug 2009 07:43:20 +1000
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
Message-ID: <D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>

I'd also seed such a torrent.

Please send the link :)



On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:

> I am interested and would also seed to the community
>
> On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
>> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>  
>> wrote:
>>> If you want the torrents, email me off list.
>>
>> Why off list?  Isn't the point of torrents to have more people  
>> sharing
>> in the load?
>>
>> Gabe
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
> -- 
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From diego.viola at gmail.com  Sat Aug 15 14:47:28 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 15 Aug 2009 17:47:28 -0400
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
Message-ID: <86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>

Upload the torrent files in http://files.freeswitch.org ;)

On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com> wrote:

> I'd also seed such a torrent.
>
> Please send the link :)
>
>
>
> On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:
>
> > I am interested and would also seed to the community
> >
> > On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
> >> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>
> >> wrote:
> >>> If you want the torrents, email me off list.
> >>
> >> Why off list?  Isn't the point of torrents to have more people
> >> sharing
> >> in the load?
> >>
> >> Gabe
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >
> > --
> > Sent from my mobile device
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peder at networkoblivion.com  Sat Aug 15 15:34:32 2009
From: peder at networkoblivion.com (Peder)
Date: Sat, 15 Aug 2009 17:34:32 -0500
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
	<86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
Message-ID: <023901ca1df8$8f404810$adc0d830$@com>

I don?t have access to do that or I would.  That?s why I offered to email
them to whoever wants them.    I did send them to Brian earlier, but he must
have some sort of life outside of FreeSWITCH  because he hasn?t put them
there yet.  ;-)

 

 

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Diego
Viola
Sent: Saturday, August 15, 2009 4:47 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] ClueCon2009 Torrents

 

Upload the torrent files in http://files.freeswitch.org ;)

On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com> wrote:

I'd also seed such a torrent.

Please send the link :)




On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:

> I am interested and would also seed to the community
>
> On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
>> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>
>> wrote:
>>> If you want the torrents, email me off list.
>>
>> Why off list?  Isn't the point of torrents to have more people
>> sharing
>> in the load?
>>
>> Gabe
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>>
>
> --
> Sent from my mobile device
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> users
> http://www.freeswitch.org

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From jmesquita at gmail.com  Sat Aug 15 15:37:37 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Sat, 15 Aug 2009 19:37:37 -0300
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <023901ca1df8$8f404810$adc0d830$@com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
	<86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
	<023901ca1df8$8f404810$adc0d830$@com>
Message-ID: <5a8712120908151537t50ddc42ej1b47718b6f4e2d79@mail.gmail.com>

I am already seeding from here.

jmesquita

On Sat, Aug 15, 2009 at 7:34 PM, Peder <peder at networkoblivion.com> wrote:

>  I don?t have access to do that or I would.  That?s why I offered to email
> them to whoever wants them.    I did send them to Brian earlier, but he must
> have some sort of life outside of FreeSWITCH  because he hasn?t put them
> there yet.  ;-)
>
>
>
>
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Diego Viola
> *Sent:* Saturday, August 15, 2009 4:47 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] ClueCon2009 Torrents
>
>
>
> Upload the torrent files in http://files.freeswitch.org ;)
>
> On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com> wrote:
>
> I'd also seed such a torrent.
>
> Please send the link :)
>
>
>
>
> On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:
>
> > I am interested and would also seed to the community
> >
> > On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
> >> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>
> >> wrote:
> >>> If you want the torrents, email me off list.
> >>
> >> Why off list?  Isn't the point of torrents to have more people
> >> sharing
> >> in the load?
> >>
> >> Gabe
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >
> > --
> > Sent from my mobile device
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From terrymr at gmail.com  Sat Aug 15 16:29:28 2009
From: terrymr at gmail.com (Terry Moore-Read)
Date: Sat, 15 Aug 2009 16:29:28 -0700
Subject: [Freeswitch-users] BLF and Openzap
Message-ID: <2d9dff7e0908151629o1a5a16fan8179048c1371fa67@mail.gmail.com>

Is it possible to have a sip phone show blf status for a phone which
is connected to an openzap port ?



From kjv at ken-ton.com  Sat Aug 15 17:30:54 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Sat, 15 Aug 2009 20:30:54 -0400
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <5a8712120908151537t50ddc42ej1b47718b6f4e2d79@mail.gmail.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
	<86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
	<023901ca1df8$8f404810$adc0d830$@com>
	<5a8712120908151537t50ddc42ej1b47718b6f4e2d79@mail.gmail.com>
Message-ID: <9136B992-B844-420C-B8C0-4224F09C8C36@ken-ton.com>


Torrent philez are 5mall...  I'd think it woud be a simple matter to  
attach them to the e-mail you're sending to the list eh?
I'd also be interested in seeding them...


Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Aug 15, 2009, at 6:37 PM, Jo?o Mesquita wrote:

> I am already seeding from here.
>
> jmesquita
>
> On Sat, Aug 15, 2009 at 7:34 PM, Peder <peder at networkoblivion.com>  
> wrote:
> I don?t have access to do that or I would.  That?s why I offered to  
> email them to whoever wants them.    I did send them to Brian  
> earlier, but he must have some sort of life outside of FreeSWITCH   
> because he hasn?t put them there yet.  ;-)
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Diego Viola
> Sent: Saturday, August 15, 2009 4:47 PM
>
>
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] ClueCon2009 Torrents
>
>
> Upload the torrent files in http://files.freeswitch.org ;)
>
> On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com> wrote:
>
> I'd also seed such a torrent.
>
> Please send the link :)
>
>
>
>
> On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:
>
> > I am interested and would also seed to the community
> >
> > On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
> >> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>
> >> wrote:
> >>> If you want the torrents, email me off list.
> >>
> >> Why off list?  Isn't the point of torrents to have more people
> >> sharing
> >> in the load?
> >>
> >> Gabe
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >
> > --
> > Sent from my mobile device
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From nicolas at medularis.com  Sat Aug 15 18:28:15 2009
From: nicolas at medularis.com (Nicolas Brenner)
Date: Sat, 15 Aug 2009 21:28:15 -0400
Subject: [Freeswitch-users] Not receiving DTMF
Message-ID: <1b46b4e80908151828l2fc3b893rf39b203aba23c745@mail.gmail.com>

Hi, I'm trying to get dtmf input, but I'm not getting anything.

What I discovered though, is that my provider is at fault, since when I
switched to another voip provider, everything started to work beautifully.

My question is: since my provider is not doing RC2833 dtmf (even though they
say they do), is there another way to get dtmf to work? I'm doing everything
in a javascript file, so I tried doing:

- session.setVariable("dtmf_type","info");

Also:

- session.setVariable("dtmf_type","rfc2833");

And:

- session.execute("start_dtmf");

But none worked. The voip provider that works would be ideal, but the calls
are twice as expensive, hence besides testing, I wouldn't use them for a
real case scenario. BTW, anyone know of a good quality VoIP provider with
low rates for termination to Santiago, Chile and Chilean cell phones?

Thanks!


Nicolas
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From jason at jasonjgw.net  Sat Aug 15 18:47:18 2009
From: jason at jasonjgw.net (Jason White)
Date: Sun, 16 Aug 2009 11:47:18 +1000
Subject: [Freeswitch-users] Not receiving DTMF
In-Reply-To: <1b46b4e80908151828l2fc3b893rf39b203aba23c745@mail.gmail.com>
References: <1b46b4e80908151828l2fc3b893rf39b203aba23c745@mail.gmail.com>
Message-ID: <20090816014718.GA25066@jdc.jasonjgw.net>

Nicolas Brenner <nicolas at medularis.com> wrote:
> My question is: since my provider is not doing RC2833 dtmf (even though they
> say they do), is there another way to get dtmf to work? 

You can try info and tone detection just in case one of those is being used,
but it appears from your message that you have attempted both of these
already.

I think it's time to find a more reliable provider, or try to persuade your
current provider to fix it (the latter is probably a waste of time and effort,
however).




From dujinfang at gmail.com  Sat Aug 15 18:51:13 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 16 Aug 2009 09:51:13 +0800
Subject: [Freeswitch-users] opal build error
Message-ID: <5BBB829D-BBFC-4A2D-A18F-A6D92F631EA2@gmail.com>

Hi,

According to wiki it still in development status, but should compile  
right? Any idea about this? thanks.

make

In file included from mod_opal.cpp:25:
mod_opal.h:151: error: conflicting return type specified for ?virtual  
OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)?
/usr/include/opal/opal/localep.h:267: error:   overriding ?virtual  
ptlib_virtual_function_changed_or_removed******  
OpalLocalEndPoint::CreateConnection(OpalCall&, void*)?
mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&,  
FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,  
switch_channel_t*)?:
mod_opal.cpp:564: error: no matching function for call to  
?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)?
/usr/include/opal/opal/localep.h:290: note: candidates are:  
OpalLocalConnection::OpalLocalConnection(OpalCall&,  
OpalLocalEndPoint&, void*, unsigned int,  
OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h: 
276: note:                  
OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
mod_opal.cpp: In member function ?switch_status_t  
FSConnection::receive_message(switch_core_session_message_t*)?:
mod_opal.cpp:1037: error: ?SWITCH_CHANNEL_SESSION_LOG? was not  
declared in this scope
make[1]: *** [mod_opal.lo] Error 1
make: *** [all] Error 1


From mitul at enterux.com  Sat Aug 15 20:28:38 2009
From: mitul at enterux.com (Mitul Limbani)
Date: Sun, 16 Aug 2009 08:58:38 +0530
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
	<86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
Message-ID: <87677B2E-6B68-43D0-A718-01CF5D08F1E2@enterux.com>

I would be glad to offer mirror service to Cluecon 2009 videos :)

Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com

On 16-Aug-2009, at 3:17 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Upload the torrent files in http://files.freeswitch.org ;)
>
> On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com> wrote:
> I'd also seed such a torrent.
>
> Please send the link :)
>
>
>
> On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:
>
> > I am interested and would also seed to the community
> >
> > On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
> >> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>
> >> wrote:
> >>> If you want the torrents, email me off list.
> >>
> >> Why off list?  Isn't the point of torrents to have more people
> >> sharing
> >> in the load?
> >>
> >> Gabe
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >
> > --
> > Sent from my mobile device
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From markmorreny at gmail.com  Sat Aug 15 20:44:08 2009
From: markmorreny at gmail.com (mark morreny)
Date: Sun, 16 Aug 2009 11:44:08 +0800
Subject: [Freeswitch-users] obtain dialplan content from switch_extension_t
Message-ID: <20ad6b920908152044r3b549875tc650e02262bf80d4@mail.gmail.com>

Hi,

After I do some operation on switch_extension_t, basically adding some
actions into it, I would like to display its content in the form of:

<extension name="xxx">
   <condition>
   <action>xxxx</action>
 </condition>

</extension>

Is that doable?  If so,  is there any existing functional call that I can
use?

Thanks,
Mark
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From mrene_lists at avgs.ca  Sat Aug 15 20:50:39 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sat, 15 Aug 2009 23:50:39 -0400
Subject: [Freeswitch-users] obtain dialplan content from
	switch_extension_t
In-Reply-To: <20ad6b920908152044r3b549875tc650e02262bf80d4@mail.gmail.com>
References: <20ad6b920908152044r3b549875tc650e02262bf80d4@mail.gmail.com>
Message-ID: <D17834AF-2569-488F-AFCE-9048678860C0@avgs.ca>

Hi,

caller_extension->applications is a linked list so you can loop  
through it and print the app name & params, but the structure doesn't  
contain any information about the conditions that were matched as this  
is a mod_xml_dialplan feature.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 15-Aug-09, at 11:44 PM, mark morreny wrote:

> Hi,
>
> After I do some operation on switch_extension_t, basically adding  
> some actions into it, I would like to display its content in the  
> form of:
>
> <extension name="xxx">
>    <condition>
>    <action>xxxx</action>
>  </condition>
>
> </extension>
>
> Is that doable?  If so,  is there any existing functional call that  
> I can use?
>
> Thanks,
> Mark
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From peter.olsson at visionutveckling.se  Sat Aug 15 23:20:07 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Sun, 16 Aug 2009 08:20:07 +0200
Subject: [Freeswitch-users] opal build error
In-Reply-To: <5BBB829D-BBFC-4A2D-A18F-A6D92F631EA2@gmail.com>
Message-ID: <C6AD71B7.16ED9%peter.olsson@visionutveckling.se>

Make sure to do a complete rebuild. And also read the comments in jira MODOPAL-10.

/Peter


On 09-08-16 03.51, "Seven Du" <dujinfang at gmail.com> wrote:

Hi,

According to wiki it still in development status, but should compile
right? Any idea about this? thanks.

make

In file included from mod_opal.cpp:25:
mod_opal.h:151: error: conflicting return type specified for 'virtual
OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)'
/usr/include/opal/opal/localep.h:267: error:   overriding 'virtual
ptlib_virtual_function_changed_or_removed******
OpalLocalEndPoint::CreateConnection(OpalCall&, void*)'
mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall&,
FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
switch_channel_t*)':
mod_opal.cpp:564: error: no matching function for call to
'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)'
/usr/include/opal/opal/localep.h:290: note: candidates are:
OpalLocalConnection::OpalLocalConnection(OpalCall&,
OpalLocalEndPoint&, void*, unsigned int,
OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h:
276: note:
OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
mod_opal.cpp: In member function 'switch_status_t
FSConnection::receive_message(switch_core_session_message_t*)':
mod_opal.cpp:1037: error: 'SWITCH_CHANNEL_SESSION_LOG' was not
declared in this scope
make[1]: *** [mod_opal.lo] Error 1
make: *** [all] Error 1
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

!DSPAM:4a8767f232931167913993!






From dujinfang at gmail.com  Sat Aug 15 23:47:04 2009
From: dujinfang at gmail.com (Seven Du)
Date: Sun, 16 Aug 2009 14:47:04 +0800
Subject: [Freeswitch-users] opal build error
In-Reply-To: <C6AD71B7.16ED9%peter.olsson@visionutveckling.se>
References: <C6AD71B7.16ED9%peter.olsson@visionutveckling.se>
Message-ID: <33340D04-0986-4F5B-B8FF-4DFB59E10F60@gmail.com>

Thanks, will try later.

On Aug 16, 2009, at 2:20 PM, Peter Olsson wrote:
> Make sure to do a complete rebuild. And also read the comments in  
> jira MODOPAL-10.
>
> /Peter
>
>
> On 09-08-16 03.51, "Seven Du" <dujinfang at gmail.com> wrote:
>
> Hi,
>
> According to wiki it still in development status, but should compile
> right? Any idea about this? thanks.
>
> make
>
> In file included from mod_opal.cpp:25:
> mod_opal.h:151: error: conflicting return type specified for 'virtual
> OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)'
> /usr/include/opal/opal/localep.h:267: error:   overriding 'virtual
> ptlib_virtual_function_changed_or_removed******
> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)'
> mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall&,
> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
> switch_channel_t*)':
> mod_opal.cpp:564: error: no matching function for call to
> 'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&,  
> NULL)'
> /usr/include/opal/opal/localep.h:290: note: candidates are:
> OpalLocalConnection::OpalLocalConnection(OpalCall&,
> OpalLocalEndPoint&, void*, unsigned int,
> OpalConnection::StringOptions*, char)/usr/include/opal/opal/localep.h:
> 276: note:
> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
> mod_opal.cpp: In member function 'switch_status_t
> FSConnection::receive_message(switch_core_session_message_t*)':
> mod_opal.cpp:1037: error: 'SWITCH_CHANNEL_SESSION_LOG' was not
> declared in this scope
> make[1]: *** [mod_opal.lo] Error 1
> make: *** [all] Error 1
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> !DSPAM:4a8767f232931167913993!
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From scott.torr.fs at letterboxes.org  Sun Aug 16 00:08:10 2009
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Sun, 16 Aug 2009 17:08:10 +1000
Subject: [Freeswitch-users] Audio only in one direction when calling FS
 from Skype
In-Reply-To: <0031C8A1-BCE9-445F-8DE0-177F24D576F5@jerris.com>
References: <1250357492.23345.1330000179@webmail.messagingengine.com>
	<0031C8A1-BCE9-445F-8DE0-177F24D576F5@jerris.com>
Message-ID: <1250406490.9719.1330052905@webmail.messagingengine.com>


I have used a MS-Skype client on the local area network to receive Skype
calls and do not have any audio continuity problems, and would conclude
that there is not an issue with the firewall if this is working
correctly.

To the best of my knowledge the external ip addresses and local networks
are configured properly in FS.


When I use the (linux Skpye client)<=>(mod_skypiax)<=>(FS) to recieve
Skype calls however I cannot hear the incoming audio.

This problem only occurs with an incoming Skype call (ie cannot hear
their audio, they can hear mine).

If I make an outgoing call from FS to Skype there are no audio problem.

So this is an asymmetrical problem.

I have used wireshark and can see UDP packets going in both directions
on the local area network.
So the incoming audio data is hitting the (linux Skype client).


How can I investigate where the data is being dropped?

Is it between:

(linux Skype client)-->(mod_skypiax)
or
(mod_skypiax)-->(FS)
 

I need some advice on what debug commands to use to see where the audio
is being dropped or some clues as to how to pinpoint where the problem
is occurring.


For example is (mod_skypiax) getting the audio data from (linux Skype
client)?

How do investigate that?
How do I see that?
Are there channel IO byte counters?

The machine is using snd_dummy, has no vmware sound card.

It is strange the problem is only in one direction?



Thanks,
Scott Torr



On Sat, 15 Aug 2009 17:20 -0400, "Michael Jerris" <mike at jerris.com>
wrote:
> Make sure your external ip addresses and local networks are configured  
> properly.
> 
> Mike
> 
> On Aug 15, 2009, at 1:31 PM, Scott Torr wrote:
> 
> > Freeswitch-users,
> >
> > I'm very new to Freeswitch and have installed the following in VMware
> > Server 2.0.1
> >
> >
> > ubuntu-8.04.3-server-i386.iso (udate/upgrade)
> > skype-debian_2.0.0.72-1_i386.deb
> > FreeSWITCH Version 1.0.trunk (14492)
> > mod_skypiax
> >
> >
> > Using the FS wiki to install/setup and making small changes to default
> > XML configs as required.
> >
> >
> > (phone)--(SIP ATA)--(FS)--(MS-Skype client | Skype 'online number' via
> > PSTN)
> >
> >
> > The MS-Skype client is on the same local network as FS.
> >
> >
> > I can make calls from the phone to the Skype Client and this works OK.
> > Audio path OK: phone<-->FS<-->Skype
> >
> >
> > But ,if I call from the MS-Skype client to the phone I hear no audio
> > from the MS-Skype client.
> >
> > Audio path: phone<-->FS-->Skype
> >
> > Likewise a call from a Skype 'online number' can hear for example the
> > default 5000 ivr but the DTMF tones from the PSTN phone are not  
> > detected
> > by FS when <action application="start_dtmf" /> is used.
> >
> >
> >
> > Any suggestions or pointers in the right direction would be much
> > appreciated.
> >
> > Most likely I have overlook something very obvious to others.
> >
> >
> > Thanks in advance,
> > Scott Torr
> >
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From manoj.joshi.13jan at gmail.com  Sun Aug 16 01:05:23 2009
From: manoj.joshi.13jan at gmail.com (Manu)
Date: Sun, 16 Aug 2009 13:35:23 +0530
Subject: [Freeswitch-users] mod nibblebill question
In-Reply-To: <4E77075F-4EF0-4CD0-A36A-D68DA8F438EC@jerris.com>
References: <26474d4e0908151233s1376ab15h74d2ac32b58593bf@mail.gmail.com>
	<4E77075F-4EF0-4CD0-A36A-D68DA8F438EC@jerris.com>
Message-ID: <26474d4e0908160105j3fc44d73s50e42f0940b420c7@mail.gmail.com>

Thank you for the reply Michael. Let me ask your opinion on another related
matter also.

I am using a MS SQL Database (which will be there on a remote server). I
wish to keep ...

1- User database in SQL.
2- Dialplans in SQL.
3- CDR logged in SQL.
4- I also require to cut the call in real time when credit is over.

I want to deply this to get good performance for 500 calls simultaneous. I
see in documents that using http responses i can fetch data from my web
server. I figure i can return number of gateways and other dial plan
parameters this way. In the same HTTP request i can also return "call Rates"
for the called destination (which i can use in nibble)

Is there any other efficiant apprroach you can suggest?

Regards,

Manoj





On Sun, Aug 16, 2009 at 2:50 AM, Michael Jerris <mike at jerris.com> wrote:

> increments are in seconds, not microseconds.  In IMS for example I
> think it defaults to 20 or 30 second nibbles, depending on your
> tolerances and billing increments something much larger may even make
> sense.  Doing billing in sub second increments doesn't make a lot of
> sense to me.  Remember that this is just keeping track of available
> credit so if there are multiple calls at the same time you won't go
> over balance.  Everything is still reconciled at hang up, so if you
> have a bit too much reserved from your nibble the worst that could
> happen is it could cut off calls a little too early when multiple
> calls are in progress on the same account.
>
> Mike
>
> On Aug 15, 2009, at 3:33 PM, Manu wrote:
>
> > Hello,
> >
> > If we use heartbeat option on in nibblebill.conf.xml does that mean
> > ODBC database table will be updated every microsecond or any other
> > interval we set?
> >
> > If this is so and there are many users (Lets say 500 users) are
> > connected to FS wouldn't it create locking issues in DB?
> >
> > Regards,
> >
> > Manoj
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mgg at giagnocavo.net  Sun Aug 16 01:54:59 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Sun, 16 Aug 2009 04:54:59 -0400
Subject: [Freeswitch-users] mod nibblebill question
In-Reply-To: <26474d4e0908160105j3fc44d73s50e42f0940b420c7@mail.gmail.com>
References: <26474d4e0908151233s1376ab15h74d2ac32b58593bf@mail.gmail.com>
	<4E77075F-4EF0-4CD0-A36A-D68DA8F438EC@jerris.com>
	<26474d4e0908160105j3fc44d73s50e42f0940b420c7@mail.gmail.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702D79D3B73@mse17be1.mse17.exchange.ms>

Just an in-general performance hint: Get your setup running, then use the SQL Profiler to generate a tuning trace, and run the "Database Engine Tuning Advisor" with it. That way it'll tell you which indexes you need (or need to drop) to optimize the database side of performance.

500 simultaneous calls isn't much though. Even with 30 second nibblebillings, that's only 16 updates/sec - and your calls/sec is probably lower than that. That's not too much work for any average server these days, even doing dialplan/cdr/billing against remote SQL.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Manu
Sent: Sunday, August 16, 2009 2:05 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod nibblebill question

Thank you for the reply Michael. Let me ask your opinion on another related matter also.

I am using a MS SQL Database (which will be there on a remote server). I wish to keep ...

1- User database in SQL.
2- Dialplans in SQL.
3- CDR logged in SQL.
4- I also require to cut the call in real time when credit is over.

I want to deply this to get good performance for 500 calls simultaneous. I see in documents that using http responses i can fetch data from my web server. I figure i can return number of gateways and other dial plan parameters this way. In the same HTTP request i can also return "call Rates" for the called destination (which i can use in nibble)

Is there any other efficiant apprroach you can suggest?

Regards,

Manoj





On Sun, Aug 16, 2009 at 2:50 AM, Michael Jerris <mike at jerris.com<mailto:mike at jerris.com>> wrote:
increments are in seconds, not microseconds.  In IMS for example I
think it defaults to 20 or 30 second nibbles, depending on your
tolerances and billing increments something much larger may even make
sense.  Doing billing in sub second increments doesn't make a lot of
sense to me.  Remember that this is just keeping track of available
credit so if there are multiple calls at the same time you won't go
over balance.  Everything is still reconciled at hang up, so if you
have a bit too much reserved from your nibble the worst that could
happen is it could cut off calls a little too early when multiple
calls are in progress on the same account.

Mike

On Aug 15, 2009, at 3:33 PM, Manu wrote:

> Hello,
>
> If we use heartbeat option on in nibblebill.conf.xml does that mean
> ODBC database table will be updated every microsecond or any other
> interval we set?
>
> If this is so and there are many users (Lets say 500 users) are
> connected to FS wouldn't it create locking issues in DB?
>
> Regards,
>
> Manoj
_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org<http://www.freeswitch.org/>

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From raffaele.p.guidi at gmail.com  Sun Aug 16 02:43:27 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Sun, 16 Aug 2009 11:43:27 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I
	have a little problem
Message-ID: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>

I had the sweet surprise to find the installer packaged with FreePBX...
really great! Why it has not been advertised as it deserves? It worked like
a breeze once launched, with the automatic configuration and all of that.,
Only thing: once stopped I cannot get it to load sofia profiles anymore -
issueing sofia status doesn't show anything. I had to copy internal.xml and
default.xml from a previous installation and everything got to work again -
but no FreePBX anymore :( I'm sure I'm missing something important.
Can you give me a hint? Should sofia profiles be served by curl or
something?

Thanks,
   Raffaele
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From woodydickson at gmail.com  Sun Aug 16 04:24:50 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Sun, 16 Aug 2009 19:24:50 +0800
Subject: [Freeswitch-users] how to set different action for different cause
	code
Message-ID: <a44688a0908160424n3569cef1vad8fcdc68cbfb96e@mail.gmail.com>

Hello,

I find hangup_hook, but I would like to define different actions for
different hangup codes.  Is there anyway to do that?

Woody
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From tzury.by at reguluslabs.com  Sun Aug 16 04:25:07 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Sun, 16 Aug 2009 14:25:07 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
Message-ID: <10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>

Hi Brian,

Just for your information here is a mail I got from a colleague of
mine which I consider as an experienced freeswitch integrator

> "... TCP used to work, and I had about 20-30 client phones connecting with it.
> About 3 months ago I did an upgrade and TCP no longer worked.? I was in panic mode, so I just ended up walking my
> clients through reconfiguring their phones to use UDP.
>
> It is definitely a bug in Freeswitch.? I can help you work up a jira bug report if you like.
> The bug happened somewhere between svn version 9800ish and 13223.
> 13223 is the version I'm running on my systems now, older, but stable for me..."

It would be just great if you can confirm the TCP functionality in the
latest release of FreeSwitch


/t



From asannucci at gmail.com  Sun Aug 16 05:07:18 2009
From: asannucci at gmail.com (bakko)
Date: Sun, 16 Aug 2009 14:07:18 +0200
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com><2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org><10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com><43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org><10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com><45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
Message-ID: <2278A0310B024C82B239E2FA9D6D2B77@voztovoice>

I'm just tryng TCP now and work (latest trunk)

I done this test with eyebeam phone registeres to TCP port and call other ip 
phone (registered over UDB port).

The audio work fine in both directions.

My configuration:

FS --> internet --> NAT --> Phones

In the internal profile:

Call-ID:        OTIyNzRlNTQxNTk5NzBhYzMzYzM1YmQyNTgzNTA3MTk.
User:           1004 at nydomain.com
Contact:        "User" 
<sip:1004 at 87.12.101.59:13035;transport=TCP;rinstance=f3fd252ea20dd34a>
Agent:          eyeBeam release 1102u stamp 52345
Status:         Registered(TCP)(unknown) EXP(2009-08-16 08:59:38)
Host:           mydomain.com
IP:             XXX:XXX.XXX.XXX
Port:           13035
Auth-User:      1004
Auth-Realm:     mydomain.com






From carlos.talbot at gmail.com  Sun Aug 16 07:04:30 2009
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Sun, 16 Aug 2009 09:04:30 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
Message-ID: <5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>

When you configure FreePBX for the first time it wipes out the sip_profiles
directory. If you follow the FreePBX shortcut on your desktop it'll mention
this on the last screen of the configuration. You might see something such
as the following below. If you plan to use FreePBX you'll need to define
trunk groups, trunks, etc in order to have the sip_profiles directory
populated.
regards,

Carlos


Incompatible ConfigurationWARNING: THE FOLLOWING FILES WILL BE DELETED!

   - D:/FreeSWITCH/conf/sip_profiles/external.xml
   - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
   - D:/FreeSWITCH/conf/sip_profiles/internal.xml


On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi <
raffaele.p.guidi at gmail.com> wrote:

> I had the sweet surprise to find the installer packaged with FreePBX...
> really great! Why it has not been advertised as it deserves? It worked like
> a breeze once launched, with the automatic configuration and all of that.,
> Only thing: once stopped I cannot get it to load sofia profiles anymore -
> issueing sofia status doesn't show anything. I had to copy internal.xml and
> default.xml from a previous installation and everything got to work again -
> but no FreePBX anymore :( I'm sure I'm missing something important.
> Can you give me a hint? Should sofia profiles be served by curl or
> something?
>
> Thanks,
>    Raffaele
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sun Aug 16 07:10:41 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Aug 2009 09:10:41 -0500
Subject: [Freeswitch-users] BLF and Openzap
In-Reply-To: <2d9dff7e0908151629o1a5a16fan8179048c1371fa67@mail.gmail.com>
References: <2d9dff7e0908151629o1a5a16fan8179048c1371fa67@mail.gmail.com>
Message-ID: <C9ABED9B-675E-4A30-AFD5-54A2E52C91CA@freeswitch.org>

Yes you sure can... setting the presence_id to exten at domain and have  
the phone sub to that you'll have the light come on when its in use.

/b

On Aug 15, 2009, at 6:29 PM, Terry Moore-Read wrote:

> Is it possible to have a sip phone show blf status for a phone which
> is connected to an openzap port ?




From brian at freeswitch.org  Sun Aug 16 07:11:20 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Aug 2009 09:11:20 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
Message-ID: <01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>

Works fine here.

/b

On Aug 16, 2009, at 6:25 AM, Tzury Bar Yochay wrote:

> Hi Brian,
>
> Just for your information here is a mail I got from a colleague of
> mine which I consider as an experienced freeswitch integrator
>
>> "... TCP used to work, and I had about 20-30 client phones  
>> connecting with it.
>> About 3 months ago I did an upgrade and TCP no longer worked.  I  
>> was in panic mode, so I just ended up walking my
>> clients through reconfiguring their phones to use UDP.
>>
>> It is definitely a bug in Freeswitch.  I can help you work up a  
>> jira bug report if you like.
>> The bug happened somewhere between svn version 9800ish and 13223.
>> 13223 is the version I'm running on my systems now, older, but  
>> stable for me..."
>
> It would be just great if you can confirm the TCP functionality in the
> latest release of FreeSwitch
>
>
> /t
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Sun Aug 16 07:11:56 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Aug 2009 09:11:56 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
Message-ID: <5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>

chances are mod_sofia isn't loaded.

/b

On Aug 16, 2009, at 4:43 AM, Raffaele P. Guidi wrote:

> Can you give me a hint? Should sofia profiles be served by curl or  
> something?
>
> Thanks,
>    Raffaele




From grevenx at me.com  Sun Aug 16 07:55:54 2009
From: grevenx at me.com (=?ISO-8859-1?Q?Even_Andr=E9_Fiskvik?=)
Date: Sun, 16 Aug 2009 16:55:54 +0200
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <87677B2E-6B68-43D0-A718-01CF5D08F1E2@enterux.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
	<86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
	<87677B2E-6B68-43D0-A718-01CF5D08F1E2@enterux.com>
Message-ID: <8FA1E6A3-1C07-4127-B25D-B91BA6FF2029@me.com>

Can we get the torrents added to a mail on this list soon please?

Best regards,
Even Andr?

On 16. aug.. 2009, at 05.28, Mitul Limbani wrote:

> I would be glad to offer mirror service to Cluecon 2009 videos :)
>
> Thanks & Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions Pvt. Ltd.,
> The Enterprise Linux Company (r),
> http://www.enterux.com
> http://www.entVoice.com
>
> On 16-Aug-2009, at 3:17 AM, Diego Viola <diego.viola at gmail.com> wrote:
>
>> Upload the torrent files in http://files.freeswitch.org ;)
>>
>> On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com>  
>> wrote:
>> I'd also seed such a torrent.
>>
>> Please send the link :)
>>
>>
>>
>> On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:
>>
>> > I am interested and would also seed to the community
>> >
>> > On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
>> >> On Sat, Aug 15, 2009 at 12:13 PM, Peder<peder at networkoblivion.com>
>> >> wrote:
>> >>> If you want the torrents, email me off list.
>> >>
>> >> Why off list?  Isn't the point of torrents to have more people
>> >> sharing
>> >> in the load?
>> >>
>> >> Gabe
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>> freeswitch-
>> >> users
>> >> http://www.freeswitch.org
>> >>
>> >
>> > --
>> > Sent from my mobile device
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> > users
>> > http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From kokoska.rokoska at post.cz  Sun Aug 16 09:41:08 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Sun, 16 Aug 2009 18:41:08 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
Message-ID: <4A8836A4.20402@post.cz>


Hi all,

I have just upgarded my (a little bit old :-) FreeSWITCH to current
trunk and now i have troubles with IP addresses in SIP & SDP, while
before upgrade everything work fine...

My FreeSWITCH is behind NAT and all relevant sip profiles have set:

<param name="ext-rtp-ip" value="1.2.3.4"/>
<param name="ext-sip-ip" value="1.2.3.4"/>

But when I send INVITE thru the profile to the gateway associated with
it i see local IP in "Via" header and in SDP "m" line too.
BTW: The "Contact" header is populated with "external" IP (1.2.3.4)

Could you, please, point me how could I modify IP in Via and in SDP?
Thank you.

Best regards,

kokoska.rokoska




From brian at freeswitch.org  Sun Aug 16 10:10:40 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Aug 2009 12:10:40 -0500
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A8836A4.20402@post.cz>
References: <4A8836A4.20402@post.cz>
Message-ID: <6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>

You have to set the local-network-acl on the profile so it knows when  
or not to fix up the VIA
/b

On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:

>
> Hi all,
>
> I have just upgarded my (a little bit old :-) FreeSWITCH to current
> trunk and now i have troubles with IP addresses in SIP & SDP, while
> before upgrade everything work fine...
>
> My FreeSWITCH is behind NAT and all relevant sip profiles have set:
>
> <param name="ext-rtp-ip" value="1.2.3.4"/>
> <param name="ext-sip-ip" value="1.2.3.4"/>
>
> But when I send INVITE thru the profile to the gateway associated with
> it i see local IP in "Via" header and in SDP "m" line too.
> BTW: The "Contact" header is populated with "external" IP (1.2.3.4)




From tayeb.meftah at gmail.com  Sun Aug 16 11:25:11 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 16 Aug 2009 18:25:11 +0000
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
 but I have a little problem
In-Reply-To: <5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>
Message-ID: <4A884F07.3040200@gmail.com>

hello,
could you give me the installer Link to test it?
thanks
meftah tayeb
DelphiWorld at irc.freenode.net

    * Call Me (SIP) <sip:voipworld at sip.voipuser.org>
    * Call me (skype) <skype:tayeb.meftah>
    * Call Me (PSTN) <callto:0021348526972>

Brian West wrote:
> chances are mod_sofia isn't loaded.
>
> /b
>
> On Aug 16, 2009, at 4:43 AM, Raffaele P. Guidi wrote:
>
>   
>> Can you give me a hint? Should sofia profiles be served by curl or  
>> something?
>>
>> Thanks,
>>    Raffaele
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4339 (20090816) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4339 (20090816) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From carlos.talbot at gmail.com  Sun Aug 16 11:35:58 2009
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Sun, 16 Aug 2009 13:35:58 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <4A884F07.3040200@gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>
	<4A884F07.3040200@gmail.com>
Message-ID: <5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>

http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe

On Sun, Aug 16, 2009 at 1:25 PM, Meftah Tayeb <tayeb.meftah at gmail.com>wrote:

>  hello,
> could you give me the installer Link to test it?
> thanks
> meftah tayeb
> DelphiWorld at irc.freenode.net
>
>    - Call Me (SIP)
>    - Call me (skype)
>    - Call Me (PSTN) <callto:0021348526972>
>
> Brian West wrote:
>
> chances are mod_sofia isn't loaded.
>
> /b
>
> On Aug 16, 2009, at 4:43 AM, Raffaele P. Guidi wrote:
>
>
>
>  Can you give me a hint? Should sofia profiles be served by curl or
> something?
>
> Thanks,
>    Raffaele
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4339 (20090816) __________
>
> The message was checked by ESET NOD32 Antivirus.
> http://www.eset.com
>
>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus
> signature database 4339 (20090816) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From terrymr at gmail.com  Sun Aug 16 11:47:00 2009
From: terrymr at gmail.com (Terry Moore-Read)
Date: Sun, 16 Aug 2009 11:47:00 -0700
Subject: [Freeswitch-users] BLF and Openzap
In-Reply-To: <C9ABED9B-675E-4A30-AFD5-54A2E52C91CA@freeswitch.org>
References: <2d9dff7e0908151629o1a5a16fan8179048c1371fa67@mail.gmail.com>
	<C9ABED9B-675E-4A30-AFD5-54A2E52C91CA@freeswitch.org>
Message-ID: <2d9dff7e0908161147g1a4f08feu30bc33d20f1a9811@mail.gmail.com>

On Sun, Aug 16, 2009 at 7:10 AM, Brian West<brian at freeswitch.org> wrote:
> Yes you sure can... setting the presence_id to exten at domain and have

Thats what I thought ... but where should I be setting it ?

> the phone sub to that you'll have the light come on when its in use.
>
> /b
>
> On Aug 15, 2009, at 6:29 PM, Terry Moore-Read wrote:
>
>> Is it possible to have a sip phone show blf status for a phone which
>> is connected to an openzap port ?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Want to buy my photo's ? :
http://www.shutterstock.com/gallery.mhtml?id=309295&rid=309295



From krice at freeswitch.org  Sun Aug 16 11:49:22 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 16 Aug 2009 13:49:22 -0500
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <8FA1E6A3-1C07-4127-B25D-B91BA6FF2029@me.com>
Message-ID: <C6ADBEE3.5AFB7%krice@freeswitch.org>

Lets see if this works... There are a few seeders out there...



From: Even Andr? Fiskvik <grevenx at me.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Sun, 16 Aug 2009 16:55:54 +0200
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] ClueCon2009 Torrents

Can we get the torrents added to a mail on this list soon please?

Best regards,
Even Andr?

On 16. aug.. 2009, at 05.28, Mitul Limbani wrote:

> I would be glad to offer mirror service to Cluecon 2009 videos :)
> 
> Thanks & Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions Pvt. Ltd.,
> The Enterprise Linux Company (r),
> http://www.enterux.com
> http://www.entVoice.com
> 
> On 16-Aug-2009, at 3:17 AM, Diego Viola <diego.viola at gmail.com> wrote:
> 
>> Upload the torrent files in  <http://files.freeswitch.org>
>> http://files.freeswitch.org ;)
>> 
>> On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks < <mailto:jaybinks at gmail.com>
>> jaybinks at gmail.com> wrote:
>>  
>>> I'd also seed such a torrent.
>>>  
>>>  Please send the link :)
>>>  
>>> 
>>>  
>>>  
>>>  On 16/08/2009, at 6:34, Jo?o Mesquita < <mailto:jmesquita at gmail.com>
>>> jmesquita at gmail.com> wrote:
>>>  
>>>>  > I am interested and would also seed to the community
>>>>  >
>>>>  > On 8/15/09, Gabriel Gunderson < <mailto:gabe at gundy.org> gabe at gundy.org>
>>>> wrote:
>>>>>  >> On Sat, Aug 15, 2009 at 12:13 PM, Peder<
>>>>> <mailto:peder at networkoblivion.com> peder at networkoblivion.com>
>>>>>  >> wrote:
>>>>>>  >>> If you want the torrents, email me off list.
>>>>>  >>
>>>>>  >> Why off list?  Isn't the point of torrents to have more people
>>>>>  >> sharing
>>>>>  >> in the load?
>>>>>  >>
>>>>>  >> Gabe
>>>>>  >>
>>>>>  >> _______________________________________________
>>>>>  >> FreeSWITCH-users mailing list
>>>>>  >>  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>>  >>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>>  >> UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch->
>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>>  >> users
>>>>>  >>  <http://www.freeswitch.org> http://www.freeswitch.org
>>>>>  >>
>>>>  >
>>>>  > --
>>>>  > Sent from my mobile device
>>>>  >
>>>>  > _______________________________________________
>>>>  > FreeSWITCH-users mailing list
>>>>  >  <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>  >  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>  > UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch->
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-
>>>>  > users
>>>>  >  <http://www.freeswitch.org> http://www.freeswitch.org
>>>  
>>>  _______________________________________________
>>>  FreeSWITCH-users mailing list
>>>   <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>> FreeSWITCH-users at lists.freeswitch.org
>>>   <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>  UNSUBSCRIBE: <http://lists.freeswitch.org/mailman/options/freeswitch-users>
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>   <http://www.freeswitch.org> http://www.freeswitch.org
>>>  
>>> 
>>>  
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org

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From kokoska.rokoska at post.cz  Sun Aug 16 12:40:06 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Sun, 16 Aug 2009 21:40:06 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>
References: <4A8836A4.20402@post.cz>
	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>
Message-ID: <4A886096.3050900@post.cz>


Thank you very much, Brian, for your help!

Now are all IPs correct :-) (BTW: In my previous post I mean "c" line in
SDP, not "m" line). Thanks once more!

Well, signaling works fine, but I still have troubles with RTPs:
I can see with ngrep RTPs going to FreeSWITCH from both legs (a, b) but
there are not outgoing RTPs from FreeSWITCH to any of the call legs.
And, of course, caller and callee hear nothing :-)

Could you, please, help me to solve this? Thanks.

Best regards,

kokoska.rokoska




Brian West napsal(a):
> You have to set the local-network-acl on the profile so it knows when  
> or not to fix up the VIA
> /b
> 
> On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:
> 
>> Hi all,
>>
>> I have just upgarded my (a little bit old :-) FreeSWITCH to current
>> trunk and now i have troubles with IP addresses in SIP & SDP, while
>> before upgrade everything work fine...
>>
>> My FreeSWITCH is behind NAT and all relevant sip profiles have set:
>>
>> <param name="ext-rtp-ip" value="1.2.3.4"/>
>> <param name="ext-sip-ip" value="1.2.3.4"/>
>>
>> But when I send INVITE thru the profile to the gateway associated with
>> it i see local IP in "Via" header and in SDP "m" line too.
>> BTW: The "Contact" header is populated with "external" IP (1.2.3.4)
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From shaheryarkh at googlemail.com  Sun Aug 16 13:33:29 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Mon, 17 Aug 2009 02:33:29 +0600
Subject: [Freeswitch-users] Is there any freeswitch show version command
Message-ID: <b16156850908161333m3b1f0bb0nd71836cf505ef3ec@mail.gmail.com>

Hi,

How can we check the version / svn revision of a running FS instance?

I know, this is kind of a stupid question, but i sometimes run into
situation where i don't know or don't have access to FS source, nor i can
restart it to get its version string.

Thank you.


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From mrene_lists at avgs.ca  Sun Aug 16 13:35:06 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 16 Aug 2009 16:35:06 -0400
Subject: [Freeswitch-users] Is there any freeswitch show version command
In-Reply-To: <b16156850908161333m3b1f0bb0nd71836cf505ef3ec@mail.gmail.com>
References: <b16156850908161333m3b1f0bb0nd71836cf505ef3ec@mail.gmail.com>
Message-ID: <496E941D-748E-4166-A78D-036193EFBADF@avgs.ca>

Hi,

type:
version

at the CLI, it'll tell you.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 16-Aug-09, at 4:33 PM, Muhammad Shahzad wrote:

> Hi,
>
> How can we check the version / svn revision of a running FS instance?
>
> I know, this is kind of a stupid question, but i sometimes run into  
> situation where i don't know or don't have access to FS source, nor  
> i can restart it to get its version string.
>
> Thank you.
>
>
> -- 
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From shaheryarkh at googlemail.com  Sun Aug 16 13:40:49 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Mon, 17 Aug 2009 02:40:49 +0600
Subject: [Freeswitch-users] Is there any freeswitch show version command
In-Reply-To: <496E941D-748E-4166-A78D-036193EFBADF@avgs.ca>
References: <b16156850908161333m3b1f0bb0nd71836cf505ef3ec@mail.gmail.com>
	<496E941D-748E-4166-A78D-036193EFBADF@avgs.ca>
Message-ID: <b16156850908161340i3c382dcfv1f483ea9a2327db4@mail.gmail.com>

thanks.

On Mon, Aug 17, 2009 at 2:35 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> type: version
> at the CLI, it'll tell you.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 16-Aug-09, at 4:33 PM, Muhammad Shahzad wrote:
>
> Hi,
>
> How can we check the version / svn revision of a running FS instance?
>
> I know, this is kind of a stupid question, but i sometimes run into
> situation where i don't know or don't have access to FS source, nor i can
> restart it to get its version string.
>
> Thank you.
>
>
> --
> Muhammad Shahzad
> -----------------------------------
> CISCO Rich Media Communication Specialist (CRMCS)
> CISCO Certified Network Associate (CCNA)
> Cell: +92 334 422 40 88
> MSN: shari_786pk at hotmail.com
> Email: shaheryarkh at googlemail.com
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From brian at freeswitch.org  Sun Aug 16 14:20:58 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Aug 2009 16:20:58 -0500
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A886096.3050900@post.cz>
References: <4A8836A4.20402@post.cz>
	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>
	<4A886096.3050900@post.cz>
Message-ID: <81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>

Ok update in a few I now recall why we had that && ! 
sofia_test_pflag(tech_pvt->profile, PFLAG_AUTO_NAT) in there.

/b

On Aug 16, 2009, at 2:40 PM, kokoska rokoska wrote:

>
> Thank you very much, Brian, for your help!
>
> Now are all IPs correct :-) (BTW: In my previous post I mean "c"  
> line in
> SDP, not "m" line). Thanks once more!
>
> Well, signaling works fine, but I still have troubles with RTPs:
> I can see with ngrep RTPs going to FreeSWITCH from both legs (a, b)  
> but
> there are not outgoing RTPs from FreeSWITCH to any of the call legs.
> And, of course, caller and callee hear nothing :-)
>
> Could you, please, help me to solve this? Thanks.
>
> Best regards,
>
> kokoska.rokoska
>
>
>
>
> Brian West napsal(a):
>> You have to set the local-network-acl on the profile so it knows when
>> or not to fix up the VIA
>> /b
>>
>> On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:
>>
>>> Hi all,
>>>
>>> I have just upgarded my (a little bit old :-) FreeSWITCH to current
>>> trunk and now i have troubles with IP addresses in SIP & SDP, while
>>> before upgrade everything work fine...
>>>
>>> My FreeSWITCH is behind NAT and all relevant sip profiles have set:
>>>
>>> <param name="ext-rtp-ip" value="1.2.3.4"/>
>>> <param name="ext-sip-ip" value="1.2.3.4"/>
>>>
>>> But when I send INVITE thru the profile to the gateway associated  
>>> with
>>> it i see local IP in "Via" header and in SDP "m" line too.
>>> BTW: The "Contact" header is populated with "external" IP (1.2.3.4)
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From tayeb.meftah at gmail.com  Sun Aug 16 15:02:15 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sun, 16 Aug 2009 22:02:15 +0000
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
 but I have a little problem
In-Reply-To: <5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>	<4A884F07.3040200@gmail.com>
	<5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>
Message-ID: <4A8881E7.8080509@gmail.com>

hello
i'm rewriting this executable file in MSI format
i can use:

    * MakeMSI
    * WIX (Windows installer XML)

what you like?
WIX is open source, but MakeMSI i'm not sur
for web server, i'm replacing WAMP with a great web server called 
uniform server
this server have a conventional Unix Path configuration, for example, 
web files can by stored in /www/ and apache is in /usr/local/apache2
this facilitate the deploiment of applications
any suggestion?
thanks
Meftah Tayeb
Carlos Talbot wrote:
> http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe
>
> On Sun, Aug 16, 2009 at 1:25 PM, Meftah Tayeb <tayeb.meftah at gmail.com 
> <mailto:tayeb.meftah at gmail.com>> wrote:
>
>     hello,
>     could you give me the installer Link to test it?
>     thanks
>     meftah tayeb
>     DelphiWorld at irc.freenode.net <mailto:DelphiWorld at irc.freenode.net>
>
>         * Call Me (SIP)
>         * Call me (skype)
>         * Call Me (PSTN) <callto:0021348526972>
>
>     Brian West wrote:
>>     chances are mod_sofia isn't loaded.
>>
>>     /b
>>
>>     On Aug 16, 2009, at 4:43 AM, Raffaele P. Guidi wrote:
>>
>>       
>>>     Can you give me a hint? Should sofia profiles be served by curl or  
>>>     something?
>>>
>>>     Thanks,
>>>        Raffaele
>>>         
>>     _______________________________________________ FreeSWITCH-users
>>     mailing list FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org
>>
>>
>>
>>     __________ Information from ESET NOD32 Antivirus, version of virus signature database 4339 (20090816) __________
>>
>>     The message was checked by ESET NOD32 Antivirus.
>>
>>     http://www.eset.com
>>
>>
>>       
>
>
>
>     __________ Information from ESET NOD32 Antivirus, version of virus
>     signature database 4339 (20090816) __________
>
>     The message was checked by ESET NOD32 Antivirus.
>
>     http://www.eset.com
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4340 (20090816) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4340 (20090816) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From brian at freeswitch.org  Sun Aug 16 15:04:36 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 16 Aug 2009 17:04:36 -0500
Subject: [Freeswitch-users] BLF and Openzap
In-Reply-To: <2d9dff7e0908161147g1a4f08feu30bc33d20f1a9811@mail.gmail.com>
References: <2d9dff7e0908151629o1a5a16fan8179048c1371fa67@mail.gmail.com>
	<C9ABED9B-675E-4A30-AFD5-54A2E52C91CA@freeswitch.org>
	<2d9dff7e0908161147g1a4f08feu30bc33d20f1a9811@mail.gmail.com>
Message-ID: <B92E348F-8320-4CA8-94F0-7FC541CB1F13@freeswitch.org>

Just as a variable on the session.

/b

On Aug 16, 2009, at 1:47 PM, Terry Moore-Read wrote:

> Thats what I thought ... but where should I be setting it ?




From kjv at ken-ton.com  Sun Aug 16 10:43:58 2009
From: kjv at ken-ton.com (Karl Vesterling)
Date: Sun, 16 Aug 2009 13:43:58 -0400
Subject: [Freeswitch-users] ClueCon2009 Torrents
In-Reply-To: <8FA1E6A3-1C07-4127-B25D-B91BA6FF2029@me.com>
References: <018d01ca1dd4$14c97620$3e5c6260$@com>
	<903da5680908151233o5ec00ae5qd9875fe9f72f1b02@mail.gmail.com>
	<5a8712120908151334o5a1a824bhe3d46f175b5174e3@mail.gmail.com>
	<D5B2B848-E4FD-4EB5-913D-70DD80E5F42B@gmail.com>
	<86a32abc0908151447p54240e07o82bd0ce2ff388341@mail.gmail.com>
	<87677B2E-6B68-43D0-A718-01CF5D08F1E2@enterux.com>
	<8FA1E6A3-1C07-4127-B25D-B91BA6FF2029@me.com>
Message-ID: <D5CF3ACC-A7E0-47EF-871D-D5013C250398@ken-ton.com>


Let's see if this flies onto the list...




Best Regards,
Karl J. Vesterling
kjv at ken-ton.com
202-461-3231 x0

On Aug 16, 2009, at 10:55 AM, Even Andr? Fiskvik wrote:

> Can we get the torrents added to a mail on this list soon please?
>
> Best regards,
> Even Andr?
>
> On 16. aug.. 2009, at 05.28, Mitul Limbani wrote:
>
>> I would be glad to offer mirror service to Cluecon 2009 videos :)
>>
>> Thanks & Regards,
>> Mitul Limbani,
>> Founder & CEO,
>> Enterux Solutions Pvt. Ltd.,
>> The Enterprise Linux Company (r),
>> http://www.enterux.com
>> http://www.entVoice.com
>>
>> On 16-Aug-2009, at 3:17 AM, Diego Viola <diego.viola at gmail.com>  
>> wrote:
>>
>>> Upload the torrent files in http://files.freeswitch.org ;)
>>>
>>> On Sat, Aug 15, 2009 at 5:43 PM, Jay Binks <jaybinks at gmail.com>  
>>> wrote:
>>> I'd also seed such a torrent.
>>>
>>> Please send the link :)
>>>
>>>
>>>
>>> On 16/08/2009, at 6:34, Jo?o Mesquita <jmesquita at gmail.com> wrote:
>>>
>>> > I am interested and would also seed to the community
>>> >
>>> > On 8/15/09, Gabriel Gunderson <gabe at gundy.org> wrote:
>>> >> On Sat, Aug 15, 2009 at 12:13 PM,  
>>> Peder<peder at networkoblivion.com>
>>> >> wrote:
>>> >>> If you want the torrents, email me off list.
>>> >>
>>> >> Why off list?  Isn't the point of torrents to have more people
>>> >> sharing
>>> >> in the load?
>>> >>
>>> >> Gabe
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> >> users
>>> >> http://www.freeswitch.org
>>> >>
>>> >
>>> > --
>>> > Sent from my mobile device
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>> freeswitch-
>>> > users
>>> > http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Sun Aug 16 16:04:33 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 16 Aug 2009 19:04:33 -0400
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <4A8881E7.8080509@gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>	<4A884F07.3040200@gmail.com>
	<5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>
	<4A8881E7.8080509@gmail.com>
Message-ID: <D3325007-BA3D-4B65-B533-AEC2BF7AB447@jerris.com>

I think sticking with standard WAMP is preferable.  What is the  
advantage to creating yet another installer over the one that we have  
already done and maintained?

Mike

On Aug 16, 2009, at 6:02 PM, Meftah Tayeb wrote:

> hello
> i'm rewriting this executable file in MSI format
> i can use:
> MakeMSI
> WIX (Windows installer XML)
> what you like?
> WIX is open source, but MakeMSI i'm not sur
> for web server, i'm replacing WAMP with a great web server called  
> uniform server
> this server have a conventional Unix Path configuration, for  
> example, web files can by stored in /www/ and apache is in /usr/ 
> local/apache2
> this facilitate the deploiment of applications
> any suggestion?
> thanks
> Meftah Tayeb
> Carlos Talbot wrote:
>>
>> http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe
>>

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From mattdfong at gmail.com  Sun Aug 16 16:35:23 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Sun, 16 Aug 2009 16:35:23 -0700
Subject: [Freeswitch-users] Maxmium Number of Concurrent Sessions on EIDE
	Drive
Message-ID: <4256bf830908161635l4b748a85v8ea5032842a06de4@mail.gmail.com>

I'm interested in doing some testing on the accuracy of mod_vmd (and
mod_amd) but wanted to see if anyone could provide some guidelines on the
maximum number of concurrent sessions I can record audio files to disk with
a typical EIDE drive under 64-bit linux without overloading my system.
Also, since I only need to record the incoming audio, would it be suggested
I use the api command uuid_record or session:record? Is there a way to only
record inbound audio with session:record? Thanks.

--matt
hello hunter corp.
http://www.hellohunter.com
hosted dialer & voice broadcasting
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From mattdfong at gmail.com  Sun Aug 16 21:26:14 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Sun, 16 Aug 2009 21:26:14 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
Message-ID: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>

I tried emailed Eric, seeking advice on this, but his email (the one in the
source code) is bouncing email (invalid user), so thought I would ask here
instead. If anyone has eric's new email address, I'd be interesting in it.

I did some tests with mod_vmd this afternoon, but I'm only finding about 33%
of the voice mail beeps and did have 1 false-positive in my test of 7 voice
mail machines. I've recorded the audio of the session in .wav files that
were both successful and not, as a comparison. I can upload the .wav files
if they would be useful.

mod_vmd works great for voicemails of Skype Users, and kall8.com, but has
issues dealing with mobile phone carriers.

sprint - not successful
tmobile - not successful
verizon - not successful
panasonic home answering machine system - not successful

kall8 - SUCCESS
skype - SUCCESS

I'm wondering if you can recommend a simple fix, like changing some of the
constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c source
file, or if better success requires more complex analysis.  Do you have any
recommendations on how this might be done? Listening to the .wav's
its apparent the beeps are not as loud for the mobile phone carriers as they
are with skype and kall8. Any guidance would be greatly appreciated.

--matt
hello hunter
http://www.hellohunter.com
voice broadcasting & hosted dialer
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From kokoska.rokoska at post.cz  Sun Aug 16 23:08:51 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Mon, 17 Aug 2009 08:08:51 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>
	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>
Message-ID: <4A88F3F3.3080007@post.cz>


Thank you very much, Brian, for your help!

I have just upgraded to curent trunk, but all remains the same - RTPs
from both endpoints arrives to the FreeSWITCH but no RTPs are send back
from FS (using ngrep I can't see even single RTP packet comming from FS).

I think it is due to my misconfiguration, but I have no idea where.

I have trid to blindly modify followed params to (I hope) all possible
values combination (including manual ones :-), but with no luck:

<param name='apply-nat-acl' value='rfc1918.auto'/>
<param name='local-network-acl' value='localnet.auto'/>
<param name='disable-rtp-auto-adjust' value='false'/>
<param name='rtp-autofix-timing' value='false'/>


Best regards,

kokoska.rokoska


Brian West napsal(a):
> Ok update in a few I now recall why we had that && ! 
> sofia_test_pflag(tech_pvt->profile, PFLAG_AUTO_NAT) in there.
> 
> /b
> 
> On Aug 16, 2009, at 2:40 PM, kokoska rokoska wrote:
> 
>> Thank you very much, Brian, for your help!
>>
>> Now are all IPs correct :-) (BTW: In my previous post I mean "c"  
>> line in
>> SDP, not "m" line). Thanks once more!
>>
>> Well, signaling works fine, but I still have troubles with RTPs:
>> I can see with ngrep RTPs going to FreeSWITCH from both legs (a, b)  
>> but
>> there are not outgoing RTPs from FreeSWITCH to any of the call legs.
>> And, of course, caller and callee hear nothing :-)
>>
>> Could you, please, help me to solve this? Thanks.
>>
>> Best regards,
>>
>> kokoska.rokoska
>>
>>
>>
>>
>> Brian West napsal(a):
>>> You have to set the local-network-acl on the profile so it knows when
>>> or not to fix up the VIA
>>> /b
>>>
>>> On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:
>>>
>>>> Hi all,
>>>>
>>>> I have just upgarded my (a little bit old :-) FreeSWITCH to current
>>>> trunk and now i have troubles with IP addresses in SIP & SDP, while
>>>> before upgrade everything work fine...
>>>>
>>>> My FreeSWITCH is behind NAT and all relevant sip profiles have set:
>>>>
>>>> <param name="ext-rtp-ip" value="1.2.3.4"/>
>>>> <param name="ext-sip-ip" value="1.2.3.4"/>
>>>>
>>>> But when I send INVITE thru the profile to the gateway associated  
>>>> with
>>>> it i see local IP in "Via" header and in SDP "m" line too.
>>>> BTW: The "Contact" header is populated with "external" IP (1.2.3.4)
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From kokoska.rokoska at post.cz  Sun Aug 16 23:29:12 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Mon, 17 Aug 2009 08:29:12 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A88F3F3.3080007@post.cz>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>
	<4A88F3F3.3080007@post.cz>
Message-ID: <4A88F8B8.8070003@post.cz>


After some testing I found following:

1. When I try to bridge call between endpoints, there are always no RTPs
from FreeSWITCH (like I wrote before).
2. When I just answer call on FS and play something localy (i.e. music)
everything works fine - FS sends RTPs like expected.

All tested with various endpoints (Linksys, Snom, Nokia, Asterisk :-)
and from public IPs, private IPs at foreign LAN, private IPs at the same
LAN as FreeSWITCH is (yes, I have separate profile for phones in the
same LAN). And I found no difference.

Any hint is really appreciated :-)

Best regards,

kokoska.rokoska


kokoska rokoska napsal(a):
> Thank you very much, Brian, for your help!
> 
> I have just upgraded to curent trunk, but all remains the same - RTPs
> from both endpoints arrives to the FreeSWITCH but no RTPs are send back
> from FS (using ngrep I can't see even single RTP packet comming from FS).
> 
> I think it is due to my misconfiguration, but I have no idea where.
> 
> I have trid to blindly modify followed params to (I hope) all possible
> values combination (including manual ones :-), but with no luck:
> 
> <param name='apply-nat-acl' value='rfc1918.auto'/>
> <param name='local-network-acl' value='localnet.auto'/>
> <param name='disable-rtp-auto-adjust' value='false'/>
> <param name='rtp-autofix-timing' value='false'/>
> 
> 
> Best regards,
> 
> kokoska.rokoska
> 
> 
> Brian West napsal(a):
>> Ok update in a few I now recall why we had that && ! 
>> sofia_test_pflag(tech_pvt->profile, PFLAG_AUTO_NAT) in there.
>>
>> /b
>>
>> On Aug 16, 2009, at 2:40 PM, kokoska rokoska wrote:
>>
>>> Thank you very much, Brian, for your help!
>>>
>>> Now are all IPs correct :-) (BTW: In my previous post I mean "c"  
>>> line in
>>> SDP, not "m" line). Thanks once more!
>>>
>>> Well, signaling works fine, but I still have troubles with RTPs:
>>> I can see with ngrep RTPs going to FreeSWITCH from both legs (a, b)  
>>> but
>>> there are not outgoing RTPs from FreeSWITCH to any of the call legs.
>>> And, of course, caller and callee hear nothing :-)
>>>
>>> Could you, please, help me to solve this? Thanks.
>>>
>>> Best regards,
>>>
>>> kokoska.rokoska
>>>
>>>
>>>
>>>
>>> Brian West napsal(a):
>>>> You have to set the local-network-acl on the profile so it knows when
>>>> or not to fix up the VIA
>>>> /b
>>>>
>>>> On Aug 16, 2009, at 11:41 AM, kokoska rokoska wrote:
>>>>
>>>>> Hi all,
>>>>>
>>>>> I have just upgarded my (a little bit old :-) FreeSWITCH to current
>>>>> trunk and now i have troubles with IP addresses in SIP & SDP, while
>>>>> before upgrade everything work fine...
>>>>>
>>>>> My FreeSWITCH is behind NAT and all relevant sip profiles have set:
>>>>>
>>>>> <param name="ext-rtp-ip" value="1.2.3.4"/>
>>>>> <param name="ext-sip-ip" value="1.2.3.4"/>
>>>>>
>>>>> But when I send INVITE thru the profile to the gateway associated  
>>>>> with
>>>>> it i see local IP in "Via" header and in SDP "m" line too.
>>>>> BTW: The "Contact" header is populated with "external" IP (1.2.3.4)
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From tzury.by at reguluslabs.com  Mon Aug 17 00:14:55 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 17 Aug 2009 10:14:55 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
Message-ID: <10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>

Brian/Bakko,

Would you please tell me which softphone are you using?
As you know, my own one is not working and when I tried tcp with xlite
(providing transport=tls) I see in wireshark that it is still
transporting it over udp(!)

thanks allot in advance,
Tzury

On Sun, Aug 16, 2009 at 5:11 PM, Brian West <brian at freeswitch.org> wrote:
>
> Works fine here.
>
> /b
>
> On Aug 16, 2009, at 6:25 AM, Tzury Bar Yochay wrote:
>
> > Hi Brian,
> >
> > Just for your information here is a mail I got from a colleague of
> > mine which I consider as an experienced freeswitch integrator
> >
> >> "... TCP used to work, and I had about 20-30 client phones
> >> connecting with it.
> >> About 3 months ago I did an upgrade and TCP no longer worked. ?I
> >> was in panic mode, so I just ended up walking my
> >> clients through reconfiguring their phones to use UDP.
> >>
> >> It is definitely a bug in Freeswitch. ?I can help you work up a
> >> jira bug report if you like.
> >> The bug happened somewhere between svn version 9800ish and 13223.
> >> 13223 is the version I'm running on my systems now, older, but
> >> stable for me..."
> >
> > It would be just great if you can confirm the TCP functionality in the
> > latest release of FreeSwitch
> >
> >
> > /t
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



--
Tzury Bar Yochay

Regulus Labs ltd.
http://reguluslabs.com
+972 52 5133399



From jason at jasonjgw.net  Mon Aug 17 00:34:18 2009
From: jason at jasonjgw.net (Jason White)
Date: Mon, 17 Aug 2009 17:34:18 +1000
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
Message-ID: <20090817073418.GA5443@jdc.jasonjgw.net>

Tzury Bar Yochay <tzury.by at reguluslabs.com> wrote:
> Brian/Bakko,
> 
> Would you please tell me which softphone are you using?
> As you know, my own one is not working and when I tried tcp with xlite
> (providing transport=tls) I see in wireshark that it is still
> transporting it over udp(!)

I've successfully used TLS with FreeSWITCH at both ends (yes, that's with
FreeSWITCH itself as the softphone).

Snom 320 phones are known to work as well if you set up SRV records for the
TLS.

I haven't tried other softphones because, basically, FreeSWITCH is a better
phone than anything else I can find.

To debug this, try wireshark or tshark to find out whether your softphone is
trying to connect over the TLS port at all.




From tzury.by at reguluslabs.com  Mon Aug 17 00:38:47 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 17 Aug 2009 10:38:47 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <20090817073418.GA5443@jdc.jasonjgw.net>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<20090817073418.GA5443@jdc.jasonjgw.net>
Message-ID: <10128ef10908170038l2574727p45285d78347803f5@mail.gmail.com>

> I've successfully used TLS with FreeSWITCH at both ends (yes, that's with
> FreeSWITCH itself as the softphone).

Well, as I said at the beginning of this thread, TLS works fine for me.
The problem is when using TCP (transport=tcp and not transport=tls)



From jason at jasonjgw.net  Mon Aug 17 01:04:49 2009
From: jason at jasonjgw.net (Jason White)
Date: Mon, 17 Aug 2009 18:04:49 +1000
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170038l2574727p45285d78347803f5@mail.gmail.com>
References: <2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<20090817073418.GA5443@jdc.jasonjgw.net>
	<10128ef10908170038l2574727p45285d78347803f5@mail.gmail.com>
Message-ID: <20090817080449.GA9172@jdc.jasonjgw.net>

Tzury Bar Yochay <tzury.by at reguluslabs.com> wrote:
 
> Well, as I said at the beginning of this thread, TLS works fine for me.
> The problem is when using TCP (transport=tcp and not transport=tls)

I'm not sure whether that's supposed to use TLS. I suspect not.




From tzury.by at reguluslabs.com  Mon Aug 17 01:23:11 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 17 Aug 2009 11:23:11 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <20090817080449.GA9172@jdc.jasonjgw.net>
References: <2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<20090817073418.GA5443@jdc.jasonjgw.net>
	<10128ef10908170038l2574727p45285d78347803f5@mail.gmail.com>
	<20090817080449.GA9172@jdc.jasonjgw.net>
Message-ID: <10128ef10908170123o52aa2a06p80094006948e66df@mail.gmail.com>

> I'm not sure whether that's supposed to use TLS. I suspect not.


Jason,

I think I confused you with this TLS/TCP thing.
For the sake of clarification, I am talking about TCP and _not_ about TLS.
That is simply transporting the signaling packets over TCP instead of UDP.
No TLS should be involved at this stage. It is simply communication layer
matter.
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From bruce.mcalister at blueface.ie  Mon Aug 17 01:27:30 2009
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Mon, 17 Aug 2009 09:27:30 +0100
Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 &
 v1.0.4)
In-Reply-To: <4A8280B8.6050308@blueface.ie>
References: <4A8280B8.6050308@blueface.ie>
Message-ID: <4A891472.5060302@blueface.ie>

Hi All,

Shall I log a JIRA for this issue?

Thanks
Bruce

Bruce McAlister wrote:
> Hi All,
> 
> I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and 
> 1.0.4.
> 
> I am running on Solaris 10 Update 5 on x86 hardware (32-bit).
> 
> The build fails with:
> 
> --- snip ---
> make: Fatal error: Command failed for target `all-recursive'
> Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4
> *** Error code 1
> make: Fatal error: Command failed for target `all'
> ---
> 
> Looking back through the build I can see the following error:
> 
> --- snip ---
> creating libfreeswitch.la
> (cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la 
> libfreeswitch.la)
> /usr/bin/cc 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
> -DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch 
> freeswitch-switch.o  ./.libs/libfreeswitch.so 
> -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib/.libs/libexpat.a 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libapr-1.a 
> -lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp 
> -L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt 
> -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl 
> -lcurses -lsocket  -R/opt/freeswitch/lib -R/usr/sfw/lib
> Undefined                       first referenced
>   symbol                             in file
> herror                              ./.libs/libfreeswitch.so
> ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch
> *** Error code 1
> The following command caused the error:
> `if test -z "" ; then echo /bin/bash 
> /export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else 
> echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` 
> --tag=CC --mode=link /usr/bin/cc 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
> -DPATH_MAX=2048 -g -v -Xc -xc99=all  -lm -R/opt/freeswitch/lib  -o 
> freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib 
> freeswitch-switch.o   libfreeswitch.la libs/apr/libapr-1.la 
> libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl 
> -lcrypto -lsocket -lnsl -ldl   -lcurses -lsocket
> --- snip ---
> 
> Then a little above this error, there is the following warning that is 
> displayed (I'm not sure if it is related):
> 
> --- snip ---
> *** Warning: Linking the shared library libfreeswitch.la against the
> *** static library libs/libedit/src/.libs/libedit.a is not portable!
> --- snip ---
> 
> My configure line is as follows:
> 
> ---
> ./configure --prefix=/opt/freeswitch
> ---
> 
> I have the complete configure and make output if anyone needs them.
> 
> Any help/pointers would be greatly appreciated.
> 
> Thanks
> Bruce
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 



From jason at jasonjgw.net  Mon Aug 17 01:49:23 2009
From: jason at jasonjgw.net (Jason White)
Date: Mon, 17 Aug 2009 18:49:23 +1000
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170123o52aa2a06p80094006948e66df@mail.gmail.com>
References: <43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<20090817073418.GA5443@jdc.jasonjgw.net>
	<10128ef10908170038l2574727p45285d78347803f5@mail.gmail.com>
	<20090817080449.GA9172@jdc.jasonjgw.net>
	<10128ef10908170123o52aa2a06p80094006948e66df@mail.gmail.com>
Message-ID: <20090817084923.GA15420@jdc.jasonjgw.net>

Tzury Bar Yochay <tzury.by at reguluslabs.com> wrote:
 
> I think I confused you with this TLS/TCP thing.
> For the sake of clarification, I am talking about TCP and _not_ about TLS.
> That is simply transporting the signaling packets over TCP instead of UDP.
> No TLS should be involved at this stage. It is simply communication layer
> matter.

I can confirm that transport=tcp works fine here from one FreeSWITCH to
another. (I'm doing it over IPv6, but that shouldn't make any difference).




From tayeb.meftah at gmail.com  Mon Aug 17 02:29:03 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Mon, 17 Aug 2009 09:29:03 +0000
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
 but I have a little problem
In-Reply-To: <D3325007-BA3D-4B65-B533-AEC2BF7AB447@jerris.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>	<4A884F07.3040200@gmail.com>	<5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>	<4A8881E7.8080509@gmail.com>
	<D3325007-BA3D-4B65-B533-AEC2BF7AB447@jerris.com>
Message-ID: <4A8922DF.1080008@gmail.com>

hi MikeJ,
i prefer creating MSI file that is easy to mintin$
unstid of using inno setup or advanced installer (not free), we can use 
WIX (Windows installer XML) that is a open source one
we can create a customised MSI that fully install mor features, 
including Sounds / MOH/...
and we can edit XML files easyly to let users chouse each module to 
install during setup, for example conferencing, voice mail, Event Socket 
and ...
also the WAMP server is not fully stable
i prefer Uniform Server, that mintin Full Compatibility with Unix like 
platform including for example, perl files that need this path: /usr/...
apache2 is installed in /usr/local/apache2, mySQL: /usr/local/mysql and 
php is in /usr/local/php
and perl is in /usr/bin
php my admin is in /etc/phpmyadmin
and mor featurs, including multiple instance, virtual hosting and full 
management using control pannel
this will help enterprise users that need that in windows unstid of WAMP
     ---------------    
if you don't want to mintin it, jive me a chance to mintin it
      ---------------     
thanks


    Contact Me


    * SIP <sip:voipworld at sip.voipuser.org>
    * INUM <callto:+883510009102769>
    * PSTN <callto:0021348526972>
    * Email <mailto:tayeb.meftah at gmail.com>


Michael Jerris wrote:
> I think sticking with standard WAMP is preferable.  What is the 
> advantage to creating yet another installer over the one that we have 
> already done and maintained?
>
> Mike
>
> On Aug 16, 2009, at 6:02 PM, Meftah Tayeb wrote:
>
>> hello
>> i'm rewriting this executable file in MSI format
>> i can use:
>>
>>     * MakeMSI
>>     * WIX (Windows installer XML)
>>
>> what you like?
>> WIX is open source, but MakeMSI i'm not sur
>> for web server, i'm replacing WAMP with a great web server called 
>> uniform server
>> this server have a conventional Unix Path configuration, for example, 
>> web files can by stored in /www/ and apache is in /usr/local/apache2
>> this facilitate the deploiment of applications
>> any suggestion?
>> thanks
>> Meftah Tayeb
>> Carlos Talbot wrote:
>>> http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe
>>>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus 
> signature database 4340 (20090816) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4340 (20090816) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   




__________ Information from ESET NOD32 Antivirus, version of virus signature database 4340 (20090816) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From enno.egbert at googlemail.com  Mon Aug 17 05:10:33 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Mon, 17 Aug 2009 05:10:33 -0700 (PDT)
Subject: [Freeswitch-users]  SIPGATE Problem
Message-ID: <25005636.post@talk.nabble.com>


Hi,

i have a problem using my freeswitch with a sipgate account. The gateway
entry is o.k.. Freeswitch try a call, sent an invite and sipgate answer with
"403" (forbidden). I guess the string is incorrect.

I modify the "effective_caller_id_number" to accountnumber at sipgate.de but
the trace show the strting from: sip:accountnumber at sipgate.de@my-ip-address

How can i modify the string to sip:accountnumber at sipgate.de without my
ip-address.


-- 
View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25005636.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From asannucci at gmail.com  Mon Aug 17 05:18:24 2009
From: asannucci at gmail.com (bakko)
Date: Mon, 17 Aug 2009 14:18:24 +0200
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com><2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org><10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com><43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org><10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com><45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org><10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com><01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
Message-ID: <C62C40FE5A6547719969BCD0653D1A4E@voztovoice>

Y make my tests with eyebeam.

I thing X-lite dont't support TCP transport.

BR



From chris.chen2004 at gmail.com  Mon Aug 17 05:37:47 2009
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Mon, 17 Aug 2009 08:37:47 -0400
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <25005636.post@talk.nabble.com>
References: <25005636.post@talk.nabble.com>
Message-ID: <507898380908170537r4c68871akc13b2b8bd7129af4@mail.gmail.com>

Hi, could you please check the destination number in your dial string? If it
is the right format, one of the reasons could be that number is not in
service
when you get "403" response from the SIP gateway.

Thanks,

Chris

On Mon, Aug 17, 2009 at 8:10 AM, NOx-WHV <enno.egbert at googlemail.com> wrote:

>
> Hi,
>
> i have a problem using my freeswitch with a sipgate account. The gateway
> entry is o.k.. Freeswitch try a call, sent an invite and sipgate answer
> with
> "403" (forbidden). I guess the string is incorrect.
>
> I modify the "effective_caller_id_number" to accountnumber at sipgate.de but
> the trace show the strting from: sip:accountnumber at sipgate.de<sip%3Aaccountnumber at sipgate.de>
> @my-ip-address
>
> How can i modify the string to sip:accountnumber at sipgate.de<sip%3Aaccountnumber at sipgate.de>without my
> ip-address.
>
>
> --
> View this message in context:
> http://www.nabble.com/SIPGATE-Problem-tp25005636p25005636.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Mon Aug 17 06:09:28 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 08:09:28 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com><2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org><10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com><43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org><10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com><45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org><10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com><01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
Message-ID: <3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>

They make you pay for TCP which is weird since the RFC says TCP is  
required.

/b

On Aug 17, 2009, at 7:18 AM, bakko wrote:

> Y make my tests with eyebeam.
>
> I thing X-lite dont't support TCP transport.
>
> BR




From brian at freeswitch.org  Mon Aug 17 06:11:05 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 08:11:05 -0500
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <25005636.post@talk.nabble.com>
References: <25005636.post@talk.nabble.com>
Message-ID: <672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>

You need to set from-domain on the gateway.  And set the  
effective_caller_id_number to just the  number not the number at host.

/b

On Aug 17, 2009, at 7:10 AM, NOx-WHV wrote:

>
> Hi,
>
> i have a problem using my freeswitch with a sipgate account. The  
> gateway
> entry is o.k.. Freeswitch try a call, sent an invite and sipgate  
> answer with
> "403" (forbidden). I guess the string is incorrect.
>
> I modify the "effective_caller_id_number" to  
> accountnumber at sipgate.de but
> the trace show the strting from: sip:accountnumber at sipgate.de@my-ip-address
>
> How can i modify the string to sip:accountnumber at sipgate.de without my
> ip-address.

-------------- next part --------------
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From brian at freeswitch.org  Mon Aug 17 06:11:43 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 08:11:43 -0500
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A88F8B8.8070003@post.cz>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>
	<4A88F3F3.3080007@post.cz> <4A88F8B8.8070003@post.cz>
Message-ID: <797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>

Update, Math refactored the functions that caused some of the problems.

/b

On Aug 17, 2009, at 1:29 AM, kokoska rokoska wrote:

>
> After some testing I found following:
>
> 1. When I try to bridge call between endpoints, there are always no  
> RTPs
> from FreeSWITCH (like I wrote before).
> 2. When I just answer call on FS and play something localy (i.e.  
> music)
> everything works fine - FS sends RTPs like expected.
>
> All tested with various endpoints (Linksys, Snom, Nokia, Asterisk :-)
> and from public IPs, private IPs at foreign LAN, private IPs at the  
> same
> LAN as FreeSWITCH is (yes, I have separate profile for phones in the
> same LAN). And I found no difference.
>
> Any hint is really appreciated :-)
>
> Best regards,
>
> kokoska.rokoska




From enno.egbert at googlemail.com  Mon Aug 17 06:36:17 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Mon, 17 Aug 2009 06:36:17 -0700 (PDT)
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
References: <25005636.post@talk.nabble.com>
	<672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
Message-ID: <25006858.post@talk.nabble.com>


Hi,

i have just taken some pictures. 

http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG 
http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG 
http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG 

the third picture is taken by a softphone that works. 

To Brian: 
If i set the effective_caller_id_name to 2395805 without @sipgate.de it?s
the same problem, because the freeswitch set in the from fielt:
2395805 at 139.13.37.160. And i can?t find the right parameter to change
"139.13.37.160" to "sipgate.de"

Which is the right parameter in the dialplan?

<action application=.....?????>

Thanks for your help





Brian West-3 wrote:
> 
> You need to set from-domain on the gateway.  And set the  
> effective_caller_id_number to just the  number not the number at host.
> 
> /b
> 
> On Aug 17, 2009, at 7:10 AM, NOx-WHV wrote:
> 
>>
>> Hi,
>>
>> i have a problem using my freeswitch with a sipgate account. The  
>> gateway
>> entry is o.k.. Freeswitch try a call, sent an invite and sipgate  
>> answer with
>> "403" (forbidden). I guess the string is incorrect.
>>
>> I modify the "effective_caller_id_number" to  
>> accountnumber at sipgate.de but
>> the trace show the strting from:
>> sip:accountnumber at sipgate.de@my-ip-address
>>
>> How can i modify the string to sip:accountnumber at sipgate.de without my
>> ip-address.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25006858.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From brian at freeswitch.org  Mon Aug 17 06:52:46 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 08:52:46 -0500
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <25006858.post@talk.nabble.com>
References: <25005636.post@talk.nabble.com>
	<672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
	<25006858.post@talk.nabble.com>
Message-ID: <11D5AA56-1609-4D2C-B5D1-8AAF4A575309@freeswitch.org>

Well if you pay attention I told you in the last email... set the  
param from-domain on the gateway to sipgate.de

/b

On Aug 17, 2009, at 8:36 AM, NOx-WHV wrote:

>
> Hi,
>
> i have just taken some pictures.
>
> http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
> http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG
> http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG
>
> the third picture is taken by a softphone that works.
>
> To Brian:
> If i set the effective_caller_id_name to 2395805 without @sipgate.de  
> it?s
> the same problem, because the freeswitch set in the from fielt:
> 2395805 at 139.13.37.160. And i can?t find the right parameter to change
> "139.13.37.160" to "sipgate.de"
>
> Which is the right parameter in the dialplan?
>
> <action application=.....?????>
>
> Thanks for your help
>

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From tzury.by at reguluslabs.com  Mon Aug 17 06:53:30 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 17 Aug 2009 16:53:30 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
Message-ID: <10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>

> They make you pay for TCP which is weird since the RFC says TCP is
> required.
>
> /b

my conclusion is then:
we are living in a sad world where there is no a production ready,
cross platform, fully compliant, decent and open source sip client



From brian at freeswitch.org  Mon Aug 17 06:57:48 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 08:57:48 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
Message-ID: <CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>

FreeSWITCH works very well as a client :P

/b

On Aug 17, 2009, at 8:53 AM, Tzury Bar Yochay wrote:

> my conclusion is then:
> we are living in a sad world where there is no a production ready,
> cross platform, fully compliant, decent and open source sip client




From brian at freeswitch.org  Mon Aug 17 06:57:48 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 08:57:48 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
Message-ID: <CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>

FreeSWITCH works very well as a client :P

/b

On Aug 17, 2009, at 8:53 AM, Tzury Bar Yochay wrote:

> my conclusion is then:
> we are living in a sad world where there is no a production ready,
> cross platform, fully compliant, decent and open source sip client




From tzury.by at reguluslabs.com  Mon Aug 17 07:06:49 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Mon, 17 Aug 2009 17:06:49 +0300
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
	<CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>
Message-ID: <10128ef10908170706p414098faq5d9d8024ba0cb3ef@mail.gmail.com>

> FreeSWITCH works very well as a client :P
I am currently porting it into iPhone and Symbian, I am almost done ;-)

anyway, seriously now, can one point to a wiki page about this?
How do I do that?
I would need 3 server instances to place a call, right?



From kokoska.rokoska at post.cz  Mon Aug 17 07:09:25 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Mon, 17 Aug 2009 16:09:25 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>	<4A88F3F3.3080007@post.cz>
	<4A88F8B8.8070003@post.cz>
	<797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>
Message-ID: <4A896495.5080009@post.cz>




Brian West napsal(a):
> Update, Math refactored the functions that caused some of the problems.
> 
> /b
> 

Still no luck - no RTPs from FreeSWITCH.
For sure I'll make fresh svn checkout in a minute and let you know if it
helps...

Thank you very much for your help!

Best regards,

kokoska.rokoska



> On Aug 17, 2009, at 1:29 AM, kokoska rokoska wrote:
> 
>> After some testing I found following:
>>
>> 1. When I try to bridge call between endpoints, there are always no  
>> RTPs
>> from FreeSWITCH (like I wrote before).
>> 2. When I just answer call on FS and play something localy (i.e.  
>> music)
>> everything works fine - FS sends RTPs like expected.
>>
>> All tested with various endpoints (Linksys, Snom, Nokia, Asterisk :-)
>> and from public IPs, private IPs at foreign LAN, private IPs at the  
>> same
>> LAN as FreeSWITCH is (yes, I have separate profile for phones in the
>> same LAN). And I found no difference.
>>
>> Any hint is really appreciated :-)
>>
>> Best regards,
>>
>> kokoska.rokoska
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From brian at freeswitch.org  Mon Aug 17 07:10:39 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 09:10:39 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <10128ef10908170706p414098faq5d9d8024ba0cb3ef@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
	<CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>
	<10128ef10908170706p414098faq5d9d8024ba0cb3ef@mail.gmail.com>
Message-ID: <9B65BFD4-C93F-4E26-82F1-AF8D687DCF52@freeswitch.org>

Well if you append ;transport=tcp on the bridge lines it will use TCP .

/b

On Aug 17, 2009, at 9:06 AM, Tzury Bar Yochay wrote:

>> FreeSWITCH works very well as a client :P
> I am currently porting it into iPhone and Symbian, I am almost  
> done ;-)
>
> anyway, seriously now, can one point to a wiki page about this?
> How do I do that?
> I would need 3 server instances to place a call, right?




From brian at freeswitch.org  Mon Aug 17 07:11:01 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 09:11:01 -0500
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A896495.5080009@post.cz>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>	<4A88F3F3.3080007@post.cz>
	<4A88F8B8.8070003@post.cz>
	<797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>
	<4A896495.5080009@post.cz>
Message-ID: <22F6C640-1662-4F13-B25C-F816DC9F40E7@freeswitch.org>

if all else fails get me access to the machine please.

/b

On Aug 17, 2009, at 9:09 AM, kokoska rokoska wrote:

> Still no luck - no RTPs from FreeSWITCH.
> For sure I'll make fresh svn checkout in a minute and let you know  
> if it
> helps...
>
> Thank you very much for your help!
>
> Best regards,
>
> kokoska.rokoska




From mike at jerris.com  Mon Aug 17 08:24:05 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 17 Aug 2009 11:24:05 -0400
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <4A8922DF.1080008@gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>	<4A884F07.3040200@gmail.com>	<5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>	<4A8881E7.8080509@gmail.com>
	<D3325007-BA3D-4B65-B533-AEC2BF7AB447@jerris.com>
	<4A8922DF.1080008@gmail.com>
Message-ID: <8181A0DB-67EE-464C-9A69-30BA0719C24C@jerris.com>


On Aug 17, 2009, at 5:29 AM, Meftah Tayeb wrote:

> hi MikeJ,
> i prefer creating MSI file that is easy to mintin$

How is this useful vs. something that is already maintained?

> unstid of using inno setup or advanced installer (not free), we can  
> use WIX (Windows installer XML) that is a open source one

All the tools being used are free, however all are not open source.

> we can create a customised MSI that fully install mor features,  
> including Sounds / MOH/...

This installer already does all that and more.

> and we can edit XML files easyly to let users chouse each module to  
> install during setup, for example conferencing, voice mail, Event  
> Socket and ...
> also the WAMP server is not fully stable
> i prefer Uniform Server, that mintin Full Compatibility with Unix  
> like platform including for example, perl files that need this  
> path: /usr/...
> apache2 is installed in /usr/local/apache2, mySQL: /usr/local/mysql  
> and php is in /usr/local/php
> and perl is in /usr/bin
> php my admin is in /etc/phpmyadmin
> and mor featurs, including multiple instance, virtual hosting and  
> full management using control pannel
> this will help enterprise users that need that in windows unstid of  
> WAMP
>      ---------------
> if you don't want to mintin it, jive me a chance to mintin it
>       ---------------
> thanks

I just don't want to split efforts here, every different way we do  
this is yet another way we have to support.  I see no compelling  
argument here other than personal preference.  If I am missing  
anything someone please chime in and correct me.

Mike

> Contact Me
>
>
> SIP
> INUM
> PSTN
> Email
>
> Michael Jerris wrote:
>>
>> I think sticking with standard WAMP is preferable.  What is the  
>> advantage to creating yet another installer over the one that we  
>> have already done and maintained?
>>
>> Mike
>>
>> On Aug 16, 2009, at 6:02 PM, Meftah Tayeb wrote:
>>
>>> hello
>>> i'm rewriting this executable file in MSI format
>>> i can use:
>>> MakeMSI
>>> WIX (Windows installer XML)
>>> what you like?
>>> WIX is open source, but MakeMSI i'm not sur
>>> for web server, i'm replacing WAMP with a great web server called  
>>> uniform server
>>> this server have a conventional Unix Path configuration, for  
>>> example, web files can by stored in /www/ and apache is in /usr/ 
>>> local/apache2
>>> this facilitate the deploiment of applications
>>> any suggestion?
>>> thanks
>>> Meftah Tayeb
>>> Carlos Talbot wrote:
>>>>
>>>> http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe
>>>>
>>
>>

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From msc at freeswitch.org  Mon Aug 17 08:59:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Aug 2009 08:59:31 -0700
Subject: [Freeswitch-users] how to set different action for different
	cause code
In-Reply-To: <a44688a0908160424n3569cef1vad8fcdc68cbfb96e@mail.gmail.com>
References: <a44688a0908160424n3569cef1vad8fcdc68cbfb96e@mail.gmail.com>
Message-ID: <87f2f3b90908170859p52259449q913b2ddab224709f@mail.gmail.com>

On Sun, Aug 16, 2009 at 4:24 AM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hello,
>
> I find hangup_hook, but I would like to define different actions for
> different hangup codes.  Is there anyway to do that?
>
>

I can think of at least two ways you could do this: one that uses only the
dialplan and one that uses a script. If you don't mind using a scripting
language then you can make it very clean:

<action application="set" data="api_hangup_hook=luarun chancleanup.lua
${hangup_cause}"/>

Then have your Lua script handle all the if-then-else or case stuff.

Question: are you trying to transfer the a-leg to some other destination if
the b-leg hangup is a specific cause, or are you just doing some external
cleanup stuff? Just curious...

-MC


> Woody
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From kokoska.rokoska at post.cz  Mon Aug 17 09:46:55 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Mon, 17 Aug 2009 18:46:55 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <22F6C640-1662-4F13-B25C-F816DC9F40E7@freeswitch.org>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>	<4A88F3F3.3080007@post.cz>	<4A88F8B8.8070003@post.cz>	<797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>	<4A896495.5080009@post.cz>
	<22F6C640-1662-4F13-B25C-F816DC9F40E7@freeswitch.org>
Message-ID: <4A89897F.9060607@post.cz>




Brian West napsal(a):
> if all else fails get me access to the machine please.
> 
> /b
> 

I'm sorry, but even after fresh svn checkout all goes wrong :-)

If you be so kind to look at that machine, I'll be very glad.
I still think it will be some trivial, stupid, misconfiguration, but
can't find what I did wrong. Especially if all worked fine before FS
update...
What public ssh key should I install (shinzon, freeswitch) or should I
get another one?

Thanks once more for your help!

Best regards,

kokoska.rokoska


> On Aug 17, 2009, at 9:09 AM, kokoska rokoska wrote:
> 
>> Still no luck - no RTPs from FreeSWITCH.
>> For sure I'll make fresh svn checkout in a minute and let you know  
>> if it
>> helps...
>>
>> Thank you very much for your help!
>>
>> Best regards,
>>
>> kokoska.rokoska
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From brian at freeswitch.org  Mon Aug 17 09:54:41 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 17 Aug 2009 11:54:41 -0500
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A89897F.9060607@post.cz>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>	<4A88F3F3.3080007@post.cz>	<4A88F8B8.8070003@post.cz>	<797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>	<4A896495.5080009@post.cz>
	<22F6C640-1662-4F13-B25C-F816DC9F40E7@freeswitch.org>
	<4A89897F.9060607@post.cz>
Message-ID: <45DDEB13-3FB7-4AAC-A3A5-48272360C0F5@freeswitch.org>

shinzon.pub and get on IRC and msg bkw_ and i'll take a look.

/b

On Aug 17, 2009, at 11:46 AM, kokoska rokoska wrote:

>
> I'm sorry, but even after fresh svn checkout all goes wrong :-)
>
> If you be so kind to look at that machine, I'll be very glad.
> I still think it will be some trivial, stupid, misconfiguration, but
> can't find what I did wrong. Especially if all worked fine before FS
> update...
> What public ssh key should I install (shinzon, freeswitch) or should I
> get another one?
>
> Thanks once more for your help!
>
> Best regards,
>
> kokoska.rokoska




From kokoska.rokoska at post.cz  Mon Aug 17 10:11:03 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Mon, 17 Aug 2009 19:11:03 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <45DDEB13-3FB7-4AAC-A3A5-48272360C0F5@freeswitch.org>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>	<4A88F3F3.3080007@post.cz>	<4A88F8B8.8070003@post.cz>	<797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>	<4A896495.5080009@post.cz>	<22F6C640-1662-4F13-B25C-F816DC9F40E7@freeswitch.org>	<4A89897F.9060607@post.cz>
	<45DDEB13-3FB7-4AAC-A3A5-48272360C0F5@freeswitch.org>
Message-ID: <4A898F27.40806@post.cz>




Brian West napsal(a):
> shinzon.pub and get on IRC and msg bkw_ and i'll take a look.
> 
> /b
> 

OK, I do it.
But wait few minutes, please:

1. I should asleep my chidren :-)
2. Before a while I discovered that if I disable IPv6 networking in my
CentOS, the RTPs from FreeSWITCH works great (FS stays runnig untouched)
=> it looks like it may be network layer/config (iptables, ip6tables,
roueting etc.) error not related to FS.

I'll be back in an hour or so and let you know what is current status.

Many thanks, Brian, for your valuable help.

Best regards,

kokoska.rokoska


> On Aug 17, 2009, at 11:46 AM, kokoska rokoska wrote:
> 
>> I'm sorry, but even after fresh svn checkout all goes wrong :-)
>>
>> If you be so kind to look at that machine, I'll be very glad.
>> I still think it will be some trivial, stupid, misconfiguration, but
>> can't find what I did wrong. Especially if all worked fine before FS
>> update...
>> What public ssh key should I install (shinzon, freeswitch) or should I
>> get another one?
>>
>> Thanks once more for your help!
>>
>> Best regards,
>>
>> kokoska.rokoska
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From kokoska.rokoska at post.cz  Mon Aug 17 12:15:43 2009
From: kokoska.rokoska at post.cz (kokoska rokoska)
Date: Mon, 17 Aug 2009 21:15:43 +0200
Subject: [Freeswitch-users] ext sip & rtp IP trouble
In-Reply-To: <4A898F27.40806@post.cz>
References: <4A8836A4.20402@post.cz>	<6E56D16C-5EF1-4489-A07A-3C1C7E1643D9@freeswitch.org>	<4A886096.3050900@post.cz>	<81A3155C-E61D-41DD-9F55-BC09445BCF98@freeswitch.org>	<4A88F3F3.3080007@post.cz>	<4A88F8B8.8070003@post.cz>	<797C95BE-7170-4AD3-B4DE-B10067B641C9@freeswitch.org>	<4A896495.5080009@post.cz>	<22F6C640-1662-4F13-B25C-F816DC9F40E7@freeswitch.org>	<4A89897F.9060607@post.cz>	<45DDEB13-3FB7-4AAC-A3A5-48272360C0F5@freeswitch.org>
	<4A898F27.40806@post.cz>
Message-ID: <4A89AC5F.3020900@post.cz>


Let me apologize to waste your time, Brian (and others too).

I'm still not 100% sure where problem is, but I'm nearly sure it is not
related to FreeSWITCH.

I found similar problems with SMB - with IPv6 active some clients (even
with only IPv4 stack) can't connect to server. Disabling IPv6 solves the
issue => it has to be network problem...

Thank you very very much for your help!

Best regards,

kokoska.rokoska



kokoska rokoska napsal(a):
> 
> 
> Brian West napsal(a):
>> shinzon.pub and get on IRC and msg bkw_ and i'll take a look.
>>
>> /b
>>
> 
> OK, I do it.
> But wait few minutes, please:
> 
> 1. I should asleep my chidren :-)
> 2. Before a while I discovered that if I disable IPv6 networking in my
> CentOS, the RTPs from FreeSWITCH works great (FS stays runnig untouched)
> => it looks like it may be network layer/config (iptables, ip6tables,
> roueting etc.) error not related to FS.
> 
> I'll be back in an hour or so and let you know what is current status.
> 
> Many thanks, Brian, for your valuable help.
> 
> Best regards,
> 
> kokoska.rokoska
> 
> 
>> On Aug 17, 2009, at 11:46 AM, kokoska rokoska wrote:
>>
>>> I'm sorry, but even after fresh svn checkout all goes wrong :-)
>>>
>>> If you be so kind to look at that machine, I'll be very glad.
>>> I still think it will be some trivial, stupid, misconfiguration, but
>>> can't find what I did wrong. Especially if all worked fine before FS
>>> update...
>>> What public ssh key should I install (shinzon, freeswitch) or should I
>>> get another one?
>>>
>>> Thanks once more for your help!
>>>
>>> Best regards,
>>>
>>> kokoska.rokoska
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 




From testa at voicetechnology.com.br  Mon Aug 17 13:04:07 2009
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Mon, 17 Aug 2009 17:04:07 -0300
Subject: [Freeswitch-users] JAVA ESL
In-Reply-To: <881036.91387.qm@web35603.mail.mud.yahoo.com>
References: <191c3a030907301800n221cab6cmfbc27b97e292bd85@mail.gmail.com>
	<881036.91387.qm@web35603.mail.mud.yahoo.com>
Message-ID: <9cb0e15e0908171304q1a47e358s9cbaac00fc284cfb@mail.gmail.com>

I found this same issue on my machine.
If you could compile a esl_wrap.o then you have to generate a libesl.so with
a cmd like this:
g++ -shared esl_wrap.o -o libesl.so
Then in your code, do something like this:
/* Test.java */
import org.freeswitch.esl.*;

class Test
{
  public static void main(String[] args)
  {
    System.loadLibrary("esl");
    System.out.println("hello");
  }
}



On Thu, Jul 30, 2009 at 10:31 PM, Jean-Marc Hyppolite <hyppolite72 at yahoo.com
> wrote:

> Thank you Anthony.
>
> --- On *Thu, 7/30/09, Anthony Minessale <anthony.minessale at gmail.com>*wrote:
>
>
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] JAVA ESL
> To: freeswitch-users at lists.freeswitch.org
> Received: Thursday, July 30, 2009, 9:00 PM
>
>
> it might be a build issue, I was not exactly sure how to build it etc.
> so it may need some help from a java expert
>
> I wrote all of that with swig and never was able to test it.
>
>
> On Thu, Jul 30, 2009 at 7:40 PM, Jean-Marc Hyppolite <
> hyppolite72 at yahoo.com<http://ca.mc356.mail.yahoo.com/mc/compose?to=hyppolite72 at yahoo.com>
> > wrote:
>
>>   Hello,
>>
>> I built libesl and JAVA mod. (make and make javamod). But when I try to
>> run a JAVA script with the following code
>>
>> ESLconnection connection = new ESLconnection("127.0.0.1", "9000", "");
>> ESLevent events = connection.getInfo();
>> System.out.println(events.toString());
>>
>> I get the following error message:
>>
>> Exception in thread "main" java.lang.UnsatisfiedLinkError:
>> /usr/lib/libesl.so: /usr/lib/libesl.so: undefined symbol:
>> __gxx_personality_v0
>>         at java.lang.ClassLoader$NativeLibrary.load(Native Method)
>>         at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778)
>>         at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703)
>>         at java.lang.Runtime.loadLibrary0(Runtime.java:823)
>>         at java.lang.System.loadLibrary(System.java:1030)
>>         at ivr.IVRServer.<clinit>(IVRServer.java:18)
>>
>> Any help would be appreciated.
>>
>> Thanks.
>>
>>
>> ------------------------------
>> The new Internet Explorer? 8 - Faster, safer, easier. Optimized for Yahoo!
>> *Get it Now for Free!* <http://downloads.yahoo.com/ca/internetexplorer/>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com<http://ca.mc356.mail.yahoo.com/mc/compose?to=MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<http://ca.mc356.mail.yahoo.com/mc/compose?to=PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> -----Inline Attachment Follows-----
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>    ------------------------------
>
> *Yahoo! Canada Toolbar :* Search from anywhere on the web and bookmark
> your favourite sites. Download it now! <http://ca.toolbar.yahoo.com/>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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From mike at jerris.com  Mon Aug 17 13:18:20 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 17 Aug 2009 16:18:20 -0400
Subject: [Freeswitch-users] JAVA ESL
In-Reply-To: <9cb0e15e0908171304q1a47e358s9cbaac00fc284cfb@mail.gmail.com>
References: <191c3a030907301800n221cab6cmfbc27b97e292bd85@mail.gmail.com>
	<881036.91387.qm@web35603.mail.mud.yahoo.com>
	<9cb0e15e0908171304q1a47e358s9cbaac00fc284cfb@mail.gmail.com>
Message-ID: <F0B83E91-DE86-486B-82AB-C3BF9DF3783D@jerris.com>

can someone post a patch to that makefile to jira.freeswitch.org please.

Mike

On Aug 17, 2009, at 4:04 PM, Fernando Testa wrote:

> I found this same issue on my machine.
> If you could compile a esl_wrap.o then you have to generate a  
> libesl.so with a cmd like this:
> g++ -shared esl_wrap.o -o libesl.so
> Then in your code, do something like this:
> /* Test.java */
> import org.freeswitch.esl.*;
>
> class Test
> {
>   public static void main(String[] args)
>   {
>     System.loadLibrary("esl");
>     System.out.println("hello");
>   }
> }
>
>
>
> On Thu, Jul 30, 2009 at 10:31 PM, Jean-Marc Hyppolite <hyppolite72 at yahoo.com 
> > wrote:
> Thank you Anthony.
>
> --- On Thu, 7/30/09, Anthony Minessale <anthony.minessale at gmail.com>  
> wrote:
>
> From: Anthony Minessale <anthony.minessale at gmail.com>
> Subject: Re: [Freeswitch-users] JAVA ESL
> To: freeswitch-users at lists.freeswitch.org
> Received: Thursday, July 30, 2009, 9:00 PM
>
>
> it might be a build issue, I was not exactly sure how to build it etc.
> so it may need some help from a java expert
>
> I wrote all of that with swig and never was able to test it.
>
>
> On Thu, Jul 30, 2009 at 7:40 PM, Jean-Marc Hyppolite <hyppolite72 at yahoo.com 
> > wrote:
> Hello,
>
> I built libesl and JAVA mod. (make and make javamod). But when I try  
> to run a JAVA script with the following code
>
> ESLconnection connection = new ESLconnection("127.0.0.1", "9000", "");
> ESLevent events = connection.getInfo();
> System.out.println(events.toString());
>
> I get the following error message:
>
> Exception in thread "main" java.lang.UnsatisfiedLinkError: /usr/lib/ 
> libesl.so: /usr/lib/libesl.so: undefined symbol: __gxx_personality_v0
>         at java.lang.ClassLoader$NativeLibrary.load(Native Method)
>         at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778)
>         at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703)
>         at java.lang.Runtime.loadLibrary0(Runtime.java:823)
>         at java.lang.System.loadLibrary(System.java:1030)
>         at ivr.IVRServer.<clinit>(IVRServer.java:18)
>
> Any help would be appreciated.
>
> Thanks.
>
>
> The new Internet Explorer? 8 - Faster, safer, easier. Optimized for  
> Yahoo! Get it Now for Free!
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> -----Inline Attachment Follows-----
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>  Yahoo! Canada Toolbar : Search from anywhere on the web and  
> bookmark your favourite sites. Download it now!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Fernando Gregianin Testa
> Voice Technology Ltda
> +55 11 35882166
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From larclap at yahoo.com  Mon Aug 17 13:18:44 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Mon, 17 Aug 2009 13:18:44 -0700
Subject: [Freeswitch-users] Eavesdrop getting killed after being answered
Message-ID: <005c01ca1f77$eb8e6ef0$c2ab4cd0$@com>

I used to be able to dial 88+extension to eavesdrop, but now it is killed
right after the call is answered by the extension. Can anyone tell me what I
have done wrong? 

 

I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7
10:39:21 EDT 2009 i686 i686 i386 GNU/Linux.

 

http://pastebin.freeswitch.org/10025

 

Thanks Lars

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From fax at virgintechnologies.com  Mon Aug 17 13:30:51 2009
From: fax at virgintechnologies.com (Justin Miller)
Date: Mon, 17 Aug 2009 20:30:51 +0000
Subject: [Freeswitch-users] G729 transcoding workaround
Message-ID: <W4098424883208801250541051@webmail30>

To overcome the G729 transcoding issue with voicemail, I'm using an Audiocodes Mediant 1000 for transcoding. Our SIP trunk provider and all of our phones use G729 exclusively. 
When a call needs to go to voicemail, the call is bridged to the M1000, which transcodes to G711, and returns the call to Freeswitch on another port (5090). This seems to be working well. 

I'm now working on getting the IVR working so that I can start using the PBX functionality of Freeswitch for our office. When a call needs to hit the IVR, it's bridged to the M1000 the same as voicemail. I hear the custom menu I've created, and I can choose options successfully, except that the call is now operating in G711. When I bridge to a phone, I get the transcoding error. One option would be to transcode through the M1000 yet again, but this would take up 2 more DSP sessions in the M1000, and I would be running a total of 6 call legs in Freeswitch for this one call. 

Is there a way to end the transcoded call legs, and bridge to the phone from the original call leg? This would free up the M1000, and just seems like a better way to do things. 

Thank you 
Justin

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From eric.des.courtis at gmail.com  Mon Aug 17 10:29:30 2009
From: eric.des.courtis at gmail.com (Eric des Courtis)
Date: Mon, 17 Aug 2009 13:29:30 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
Message-ID: <539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>

Matt,

You must first capture the audio beeps and verify that they are sine
waves. If not, simply tweaking the algorithm will not give you better
results.

It might be possible to use FFT and I would be happy to help you
implement such a solution but keep in mind FFT is very very demanding
on the hardware. Ideally what you want to find out is what functions
was use to generate the beep in the first place so that it can be
detected. Is it two sines waves like in DTMF? Or something more
complex?

Anyway my email is eric.des.courtis at benbria.ca.

Cheers.

Eric des Courtis

On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong at gmail.com> wrote:
> I tried emailed Eric, seeking advice on this, but his email (the one in the
> source code) is bouncing email (invalid user), so thought I would ask here
> instead. If anyone has eric's new email address, I'd be interesting in it.
>
> I did some tests with mod_vmd this afternoon, but I'm?only finding about 33%
> of the voice mail beeps and did have 1 false-positive in my test of 7?voice
> mail?machines. I've recorded the audio of the session in .wav files that
> were both successful and not, as a comparison. I can upload the .wav files
> if they would be useful.
> mod_vmd works great for voicemails of Skype Users, and?kall8.com, but has
> issues dealing with mobile phone carriers.
> sprint - not successful
> tmobile - not successful
> verizon - not successful
> panasonic home answering machine system - not successful
> kall8 - SUCCESS
> skype - SUCCESS
> I'm wondering if you can recommend a simple fix, like changing some of the
> constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c source
> file, or if better success requires more complex analysis. ?Do you have any
> recommendations on how this might be done??Listening to the .wav's
> its?apparent?the beeps are not as loud for the mobile phone carriers as they
> are with skype and kall8. Any guidance would be greatly appreciated.
> --matt
> hello hunter
> http://www.hellohunter.com
> voice broadcasting & hosted dialer
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From eric.des.courtis at gmail.com  Mon Aug 17 14:15:57 2009
From: eric.des.courtis at gmail.com (Eric des Courtis)
Date: Mon, 17 Aug 2009 17:15:57 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
Message-ID: <539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>

Matt,

Okay the good news is vmd should be able to handle these cases. The
bad news is for whatever reason they are not getting detected at the
moment.

vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
MAX_FREQ to 1450 and play with MIN_AMPL if that still doesn't help.

The following seem to use the same beep:

vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise

You can try to play with these values:

POINTS 32
VALID 22
MAX_CHIRP 22

If that doesn't work let me know I will try to improve the algorithm
to detect the providers.

Cheers!

Eric des Courtis



On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong at gmail.com> wrote:
> Hi Eric,
> Thanks for the response. I had tried emailing you @brenbria.com and the
> email had bounced, thanks for responding to my mail.
> If you'd be interested I .zipped up my sample voicemail beeps
> at?http://bandcon.hellohunter.com/vmd_wav.zip
> I'm relatively new to telephony, but can you point me in the right direction
> for figuring out if the beeps are sinewaves. About as far as I've come with
> audio is being able to open the .wav files in audacity. Any website
> ?recommendations I can read? Thanks so much.
> --matt
>
> On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
> <eric.des.courtis at gmail.com> wrote:
>>
>> Matt,
>>
>> You must first capture the audio beeps and verify that they are sine
>> waves. If not, simply tweaking the algorithm will not give you better
>> results.
>>
>> It might be possible to use FFT and I would be happy to help you
>> implement such a solution but keep in mind FFT is very very demanding
>> on the hardware. Ideally what you want to find out is what functions
>> was use to generate the beep in the first place so that it can be
>> detected. Is it two sines waves like in DTMF? Or something more
>> complex?
>>
>> Anyway my email is eric.des.courtis at benbria.ca.
>>
>> Cheers.
>>
>> Eric des Courtis
>>
>> On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong at gmail.com> wrote:
>> > I tried emailed Eric, seeking advice on this, but his email (the one in
>> > the
>> > source code) is bouncing email (invalid user), so thought I would ask
>> > here
>> > instead. If anyone has eric's new email address, I'd be interesting in
>> > it.
>> >
>> > I did some tests with mod_vmd this afternoon, but I'm?only finding about
>> > 33%
>> > of the voice mail beeps and did have 1 false-positive in my test of
>> > 7?voice
>> > mail?machines. I've recorded the audio of the session in .wav files that
>> > were both successful and not, as a comparison. I can upload the .wav
>> > files
>> > if they would be useful.
>> > mod_vmd works great for voicemails of Skype Users, and?kall8.com, but
>> > has
>> > issues dealing with mobile phone carriers.
>> > sprint - not successful
>> > tmobile - not successful
>> > verizon - not successful
>> > panasonic home answering machine system - not successful
>> > kall8 - SUCCESS
>> > skype - SUCCESS
>> > I'm wondering if you can recommend a simple fix, like changing some of
>> > the
>> > constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c source
>> > file, or if better success requires more complex analysis. ?Do you have
>> > any
>> > recommendations on how this might be done??Listening to the .wav's
>> > its?apparent?the beeps are not as loud for the mobile phone carriers as
>> > they
>> > are with skype and kall8. Any guidance would be greatly appreciated.
>> > --matt
>> > hello hunter
>> > http://www.hellohunter.com
>> > voice broadcasting & hosted dialer
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>
>



From msc at freeswitch.org  Mon Aug 17 14:22:07 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 17 Aug 2009 14:22:07 -0700
Subject: [Freeswitch-users] Eavesdrop getting killed after being answered
In-Reply-To: <005c01ca1f77$eb8e6ef0$c2ab4cd0$@com>
References: <005c01ca1f77$eb8e6ef0$c2ab4cd0$@com>
Message-ID: <87f2f3b90908171422x1990be2fm9c3d01a7f36d58e5@mail.gmail.com>

On Mon, Aug 17, 2009 at 1:18 PM, Lars Zeb <larclap at yahoo.com> wrote:

>  I used to be able to dial 88+extension to eavesdrop, but now it is killed
> right after the call is answered by the extension. Can anyone tell me what I
> have done wrong?
>
>
>
> I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7
> 10:39:21 EDT 2009 i686 i686 i386 GNU/Linux.
>
>
>
> http://pastebin.freeswitch.org/10025
>
>
>
Hmm, something isn't right. Here's a snippet from a successful eavesdrop:
http://pastebin.freeswitch.org/10027

As you can see, your log shows no media bug operation like mine does.

Can you do another "make current" just to make 100% certain that your build
environment isn't corrupted?
-MC


> Thanks Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From testa at voicetechnology.com.br  Mon Aug 17 14:35:41 2009
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Mon, 17 Aug 2009 18:35:41 -0300
Subject: [Freeswitch-users] JAVA ESL
In-Reply-To: <F0B83E91-DE86-486B-82AB-C3BF9DF3783D@jerris.com>
References: <191c3a030907301800n221cab6cmfbc27b97e292bd85@mail.gmail.com>
	<881036.91387.qm@web35603.mail.mud.yahoo.com>
	<9cb0e15e0908171304q1a47e358s9cbaac00fc284cfb@mail.gmail.com>
	<F0B83E91-DE86-486B-82AB-C3BF9DF3783D@jerris.com>
Message-ID: <9cb0e15e0908171435o54c7379fye4a3016d9b7c59b6@mail.gmail.com>

Done!
Check out patch at http://jira.freeswitch.org/browse/FSBUILD-185

Testa

On Mon, Aug 17, 2009 at 5:18 PM, Michael Jerris <mike at jerris.com> wrote:

> can someone post a patch to that makefile to jira.freeswitch.org please.
> Mike
>
> On Aug 17, 2009, at 4:04 PM, Fernando Testa wrote:
>
> I found this same issue on my machine.
> If you could compile a esl_wrap.o then you have to generate a libesl.so
> with a cmd like this:
> g++ -shared esl_wrap.o -o libesl.so
> Then in your code, do something like this:
> /* Test.java */
> import org.freeswitch.esl.*;
>
> class Test
> {
>   public static void main(String[] args)
>   {
>     System.loadLibrary("esl");
>     System.out.println("hello");
>   }
> }
>
>
>
> On Thu, Jul 30, 2009 at 10:31 PM, Jean-Marc Hyppolite <
> hyppolite72 at yahoo.com> wrote:
>
>>  Thank you Anthony.
>>
>> --- On *Thu, 7/30/09, Anthony Minessale <anthony.minessale at gmail.com>*wrote:
>>
>>
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Subject: Re: [Freeswitch-users] JAVA ESL
>> To: freeswitch-users at lists.freeswitch.org
>> Received: Thursday, July 30, 2009, 9:00 PM
>>
>>
>> it might be a build issue, I was not exactly sure how to build it etc.
>> so it may need some help from a java expert
>>
>> I wrote all of that with swig and never was able to test it.
>>
>>
>> On Thu, Jul 30, 2009 at 7:40 PM, Jean-Marc Hyppolite <
>> hyppolite72 at yahoo.com<http://ca.mc356.mail.yahoo.com/mc/compose?to=hyppolite72 at yahoo.com>
>> > wrote:
>>
>>>   Hello,
>>>
>>> I built libesl and JAVA mod. (make and make javamod). But when I try to
>>> run a JAVA script with the following code
>>>
>>> ESLconnection connection = new ESLconnection("127.0.0.1", "9000", "");
>>> ESLevent events = connection.getInfo();
>>> System.out.println(events.toString());
>>>
>>> I get the following error message:
>>>
>>> Exception in thread "main" java.lang.UnsatisfiedLinkError:
>>> /usr/lib/libesl.so: /usr/lib/libesl.so: undefined symbol:
>>> __gxx_personality_v0
>>>         at java.lang.ClassLoader$NativeLibrary.load(Native Method)
>>>         at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778)
>>>         at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703)
>>>         at java.lang.Runtime.loadLibrary0(Runtime.java:823)
>>>         at java.lang.System.loadLibrary(System.java:1030)
>>>         at ivr.IVRServer.<clinit>(IVRServer.java:18)
>>>
>>> Any help would be appreciated.
>>>
>>> Thanks.
>>>
>>>
>>> ------------------------------
>>> The new Internet Explorer? 8 - Faster, safer, easier. Optimized for
>>> Yahoo! *Get it Now for Free!*<http://downloads.yahoo.com/ca/internetexplorer/>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com<http://ca.mc356.mail.yahoo.com/mc/compose?to=MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<http://ca.mc356.mail.yahoo.com/mc/compose?to=PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> -----Inline Attachment Follows-----
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<http://ca.mc356.mail.yahoo.com/mc/compose?to=FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>   ------------------------------
>>
>> *Yahoo! Canada Toolbar :* Search from anywhere on the web and bookmark
>> your favourite sites. Download it now! <http://ca.toolbar.yahoo.com/>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Fernando Gregianin Testa
> Voice Technology Ltda
> +55 11 35882166
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Fernando Gregianin Testa
Voice Technology Ltda
+55 11 35882166
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From telles-listas at devel-it.com.br  Mon Aug 17 14:36:03 2009
From: telles-listas at devel-it.com.br (Rodrigo P. Telles)
Date: Mon, 17 Aug 2009 18:36:03 -0300
Subject: [Freeswitch-users] Cluecon 2009
In-Reply-To: <378EC23A-D23A-4CFC-98DB-693946CF91E6@freeswitch.org>
References: <1249685648.16901.34.camel@dk-d820>
	<378EC23A-D23A-4CFC-98DB-693946CF91E6@freeswitch.org>
Message-ID: <4A89CD43.8030306@devel-it.com.br>

Hi FS Team,

Thanks for your great work at Cluecon, we came back to Brazil with some 
good ideas (+ Snom 360 + Sangoma B600) ;-)

See you again next year!

Rodrigo Telles
Devel-IT



From mike at jerris.com  Mon Aug 17 15:05:13 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 17 Aug 2009 18:05:13 -0400
Subject: [Freeswitch-users] JAVA ESL
In-Reply-To: <9cb0e15e0908171435o54c7379fye4a3016d9b7c59b6@mail.gmail.com>
References: <191c3a030907301800n221cab6cmfbc27b97e292bd85@mail.gmail.com>
	<881036.91387.qm@web35603.mail.mud.yahoo.com>
	<9cb0e15e0908171304q1a47e358s9cbaac00fc284cfb@mail.gmail.com>
	<F0B83E91-DE86-486B-82AB-C3BF9DF3783D@jerris.com>
	<9cb0e15e0908171435o54c7379fye4a3016d9b7c59b6@mail.gmail.com>
Message-ID: <06FE817A-222B-45FA-9A5F-B5BB891436F0@jerris.com>

Thanks, I'll get that merged in and clear from this we need to start  
using proper autoconf checks for java.

mike

On Aug 17, 2009, at 5:35 PM, Fernando Testa wrote:

> Done!
> Check out patch at http://jira.freeswitch.org/browse/FSBUILD-185
>
> Testa
>
> On Mon, Aug 17, 2009 at 5:18 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> can someone post a patch to that makefile to jira.freeswitch.org  
> please.
>
> Mike
>
> On Aug 17, 2009, at 4:04 PM, Fernando Testa wrote:
>
>> I found this same issue on my machine.
>> If you could compile a esl_wrap.o then you have to generate a  
>> libesl.so with a cmd like this:
>> g++ -shared esl_wrap.o -o libesl.so
>> Then in your code, do something like this:
>> /* Test.java */
>> import org.freeswitch.esl.*;
>>
>> class Test
>> {
>>   public static void main(String[] args)
>>   {
>>     System.loadLibrary("esl");
>>     System.out.println("hello");
>>   }
>> }
>>
>>
>>
>> On Thu, Jul 30, 2009 at 10:31 PM, Jean-Marc Hyppolite <hyppolite72 at yahoo.com 
>> > wrote:
>> Thank you Anthony.
>>
>> --- On Thu, 7/30/09, Anthony Minessale  
>> <anthony.minessale at gmail.com> wrote:
>>
>> From: Anthony Minessale <anthony.minessale at gmail.com>
>> Subject: Re: [Freeswitch-users] JAVA ESL
>> To: freeswitch-users at lists.freeswitch.org
>> Received: Thursday, July 30, 2009, 9:00 PM
>>
>>
>> it might be a build issue, I was not exactly sure how to build it  
>> etc.
>> so it may need some help from a java expert
>>
>> I wrote all of that with swig and never was able to test it.
>>
>>
>> On Thu, Jul 30, 2009 at 7:40 PM, Jean-Marc Hyppolite <hyppolite72 at yahoo.com 
>> > wrote:
>> Hello,
>>
>> I built libesl and JAVA mod. (make and make javamod). But when I  
>> try to run a JAVA script with the following code
>>
>> ESLconnection connection = new ESLconnection("127.0.0.1", "9000",  
>> "");
>> ESLevent events = connection.getInfo();
>> System.out.println(events.toString());
>>
>> I get the following error message:
>>
>> Exception in thread "main" java.lang.UnsatisfiedLinkError: /usr/lib/ 
>> libesl.so: /usr/lib/libesl.so: undefined symbol: __gxx_personality_v0
>>         at java.lang.ClassLoader$NativeLibrary.load(Native Method)
>>         at java.lang.ClassLoader.loadLibrary0(ClassLoader.java:1778)
>>         at java.lang.ClassLoader.loadLibrary(ClassLoader.java:1703)
>>         at java.lang.Runtime.loadLibrary0(Runtime.java:823)
>>         at java.lang.System.loadLibrary(System.java:1030)
>>         at ivr.IVRServer.<clinit>(IVRServer.java:18)
>>
>> Any help would be appreciated.
>>
>> Thanks.
>>
>>
>> The new Internet Explorer? 8 - Faster, safer, easier. Optimized for  
>> Yahoo! Get it Now for Free!
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>>
>> -----Inline Attachment Follows-----
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>  Yahoo! Canada Toolbar : Search from anywhere on the web and  
>> bookmark your favourite sites. Download it now!
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Fernando Gregianin Testa
>> Voice Technology Ltda
>> +55 11 35882166
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Fernando Gregianin Testa
> Voice Technology Ltda
> +55 11 35882166
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From larclap at yahoo.com  Mon Aug 17 17:45:34 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Mon, 17 Aug 2009 17:45:34 -0700
Subject: [Freeswitch-users] Eavesdrop getting killed after being answered
In-Reply-To: <87f2f3b90908171422x1990be2fm9c3d01a7f36d58e5@mail.gmail.com>
References: <005c01ca1f77$eb8e6ef0$c2ab4cd0$@com>
	<87f2f3b90908171422x1990be2fm9c3d01a7f36d58e5@mail.gmail.com>
Message-ID: <00ac01ca1f9d$32428ff0$96c7afd0$@com>

Thanks, Michael, it's working at 14548.

 

From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Monday, August 17, 2009 2:22 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Eavesdrop getting killed after being
answered

 

 

On Mon, Aug 17, 2009 at 1:18 PM, Lars Zeb <larclap at yahoo.com> wrote:

I used to be able to dial 88+extension to eavesdrop, but now it is killed
right after the call is answered by the extension. Can anyone tell me what I
have done wrong? 

 

I am running version 14534 on Linux fs 2.6.18-128.1.10.el5 #1 SMP Thu May 7
10:39:21 EDT 2009 i686 i686 i386 GNU/Linux.

 

http://pastebin.freeswitch.org/10025

 

Hmm, something isn't right. Here's a snippet from a successful eavesdrop:
http://pastebin.freeswitch.org/10027

As you can see, your log shows no media bug operation like mine does.

Can you do another "make current" just to make 100% certain that your build
environment isn't corrupted?
-MC
 

Thanks Lars


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From intralanman at freeswitch.org  Mon Aug 17 18:27:12 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Mon, 17 Aug 2009 21:27:12 -0400
Subject: [Freeswitch-users] G729 transcoding workaround
In-Reply-To: <W4098424883208801250541051@webmail30>
References: <W4098424883208801250541051@webmail30>
Message-ID: <66B5BC8D-496B-4B7A-9C6F-DD430F97336D@freeswitch.org>

On Aug 17, 2009, at 4:30 PM, Justin Miller wrote:
> Is there a way to end the transcoded call legs, and bridge to the  
> phone from the original call leg?  This would free up the M1000, and  
> just seems like a better way to do things.


You might consider using a REFER if your endpoints support it.

Check out the "deflect" app


Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     

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From max.bridgewater at gmail.com  Mon Aug 17 21:59:22 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Tue, 18 Aug 2009 00:59:22 -0400
Subject: [Freeswitch-users] Setting WAV File as rinback
Message-ID: <c55d46950908172159w40232399l385ee9902869aa18@mail.gmail.com>

Hi,

I'm trying to have an audio as ringback (WAV, 8khz, mono) when originating a
call. Unfortunately, it doesn't seem to work; the RINGING teletone is being
used instead of my audio. I have debug level enabled on Freeswitch but i
don't find anything suspicious.

is there anything specific i should be taking into consideration?


Thanks,
max.
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From wangdq.no1 at gmail.com  Tue Aug 18 02:24:35 2009
From: wangdq.no1 at gmail.com (daqiang wang)
Date: Tue, 18 Aug 2009 17:24:35 +0800
Subject: [Freeswitch-users] Setting WAV File as rinback
In-Reply-To: <c55d46950908172159w40232399l385ee9902869aa18@mail.gmail.com>
References: <c55d46950908172159w40232399l385ee9902869aa18@mail.gmail.com>
Message-ID: <ecd05ce30908180224k30472c44mebb7f54fba21b384@mail.gmail.com>

set ring_back = " your ring file "


2009/8/18 Max Bridgewater <max.bridgewater at gmail.com>

> Hi,
>
> I'm trying to have an audio as ringback (WAV, 8khz, mono) when originating
> a call. Unfortunately, it doesn't seem to work; the RINGING teletone is
> being used instead of my audio. I have debug level enabled on Freeswitch but
> i don't find anything suspicious.
>
> is there anything specific i should be taking into consideration?
>
>
> Thanks,
> max.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From enno.egbert at googlemail.com  Tue Aug 18 04:34:53 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Tue, 18 Aug 2009 04:34:53 -0700 (PDT)
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <11D5AA56-1609-4D2C-B5D1-8AAF4A575309@freeswitch.org>
References: <25005636.post@talk.nabble.com>
	<672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
	<25006858.post@talk.nabble.com>
	<11D5AA56-1609-4D2C-B5D1-8AAF4A575309@freeswitch.org>
Message-ID: <25023035.post@talk.nabble.com>


Hi,

sorry - i have to ask again. This is my entry in
..../conf/sip_profiles/external/ :

<include>
<gateway name="sipgate.de">


 

</gateway>
</include>

and it?s still the same... from: 2395805 at 139.13.37.160

Do you have another tip?

Thanks 

NOx






Brian West-3 wrote:
> 
> Well if you pay attention I told you in the last email... set the  
> param from-domain on the gateway to sipgate.de
> 
> /b
> 
> On Aug 17, 2009, at 8:36 AM, NOx-WHV wrote:
> 
>>
>> Hi,
>>
>> i have just taken some pictures.
>>
>> http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
>> http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG
>> http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG
>>
>> the third picture is taken by a softphone that works.
>>
>> To Brian:
>> If i set the effective_caller_id_name to 2395805 without @sipgate.de  
>> it?s
>> the same problem, because the freeswitch set in the from fielt:
>> 2395805 at 139.13.37.160. And i can?t find the right parameter to change
>> "139.13.37.160" to "sipgate.de"
>>
>> Which is the right parameter in the dialplan?
>>
>> <action application=.....?????>
>>
>> Thanks for your help
>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25023035.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From enno.egbert at googlemail.com  Tue Aug 18 04:36:04 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Tue, 18 Aug 2009 04:36:04 -0700 (PDT)
Subject: [Freeswitch-users] SIPGATE Problem
Message-ID: <25023035.post@talk.nabble.com>


Hi,

sorry - i have to ask again. This is my entry in
..../conf/sip_profiles/external/ :

include
gateway name="sipgate.de"
param name="username" value="2395805"/
param name="from-domain" value="sipgate.de"/
param name="password" value="abcde"/ 
param name="proxy" value="sipgate.de"/
/gateway
/include

and it?s still the same... from: 2395805 at 139.13.37.160

Do you have another tip?

Thanks 

NOx






Brian West-3 wrote:
> 
> Well if you pay attention I told you in the last email... set the  
> param from-domain on the gateway to sipgate.de
> 
> /b
> 
> On Aug 17, 2009, at 8:36 AM, NOx-WHV wrote:
> 
>>
>> Hi,
>>
>> i have just taken some pictures.
>>
>> http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
>> http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG
>> http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG
>>
>> the third picture is taken by a softphone that works.
>>
>> To Brian:
>> If i set the effective_caller_id_name to 2395805 without @sipgate.de  
>> it?s
>> the same problem, because the freeswitch set in the from fielt:
>> 2395805 at 139.13.37.160. And i can?t find the right parameter to change
>> "139.13.37.160" to "sipgate.de"
>>
>> Which is the right parameter in the dialplan?
>>
>> <action application=.....?????>
>>
>> Thanks for your help
>>
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25023035.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From mayamatakeshi at gmail.com  Tue Aug 18 04:53:24 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 18 Aug 2009 20:53:24 +0900
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <25023035.post@talk.nabble.com>
References: <25005636.post@talk.nabble.com>
	<672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
	<25006858.post@talk.nabble.com>
	<11D5AA56-1609-4D2C-B5D1-8AAF4A575309@freeswitch.org>
	<25023035.post@talk.nabble.com>
Message-ID: <15b9404e0908180453r59cb95bbn55f3ab9721a298d2@mail.gmail.com>

On Tue, Aug 18, 2009 at 8:34 PM, NOx-WHV<enno.egbert at googlemail.com> wrote:
>
> Hi,
>
> sorry - i have to ask again. This is my entry in
> ..../conf/sip_profiles/external/ :
>
> <include>
> <gateway name="sipgate.de">
>
>
>
>
> </gateway>
> </include>
>
> and it?s still the same... from: 2395805 at 139.13.37.160
>
> Do you have another tip?

I don't see where you are setting the param from-domain to sipgate.de
as Brian told you to do. I can only see you are setting the name of
the gateway to sipgate.de.
I never used this, but in the samples that come with FS we can see this:

<param name="from-domain" value="asterlink.com"/>

>
> Brian West-3 wrote:
>>
>> Well if you pay attention I told you in the last email... set the
>> param from-domain on the gateway to sipgate.de
>>
>> /b
>>
>> On Aug 17, 2009, at 8:36 AM, NOx-WHV wrote:
>>
>>>
>>> Hi,
>>>
>>> i have just taken some pictures.
>>>
>>> http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
>>> http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG
>>> http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG
>>>
>>> the third picture is taken by a softphone that works.
>>>
>>> To Brian:
>>> If i set the effective_caller_id_name to 2395805 without @sipgate.de
>>> it?s
>>> the same problem, because the freeswitch set in the from fielt:
>>> 2395805 at 139.13.37.160. And i can?t find the right parameter to change
>>> "139.13.37.160" to "sipgate.de"
>>>
>>> Which is the right parameter in the dialplan?
>>>
>>> <action application=.....?????>
>>>
>>> Thanks for your help
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> --
> View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25023035.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From shiyanov at gmail.com  Tue Aug 18 05:54:04 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Tue, 18 Aug 2009 16:54:04 +0400
Subject: [Freeswitch-users] "mute" channel programmatically with
	mod_event_socket
Message-ID: <d211f54a0908180554s4b5324cfx39e641d44307b173@mail.gmail.com>

Hello all!

I'm trying to implement "mute" feature with mod_event_socket: I want
programmatically mute/unmute some channel in a call.. And I don't see any
other ways except to use conference room with special rule "mute".
Anybody knows the better way?


Thanks,
Artem
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From enno.egbert at googlemail.com  Tue Aug 18 06:10:09 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Tue, 18 Aug 2009 06:10:09 -0700 (PDT)
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <15b9404e0908180453r59cb95bbn55f3ab9721a298d2@mail.gmail.com>
References: <25005636.post@talk.nabble.com>
	<672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
	<25006858.post@talk.nabble.com>
	<11D5AA56-1609-4D2C-B5D1-8AAF4A575309@freeswitch.org>
	<25023035.post@talk.nabble.com>
	<15b9404e0908180453r59cb95bbn55f3ab9721a298d2@mail.gmail.com>
Message-ID: <25024536.post@talk.nabble.com>


Hi,

this is the text without brackets:

include 
gateway name="sipgate.de" 
param name="username" value="2395805"/ 
param name="from-domain" value="sipgate.de"/ 
param name="password" value="abcde"/ 
param name="proxy" value="sipgate.de"/ 
/gateway 
/include 

I don?t know, why you can?t see the lines between gateway and /gateway.

NOx

mayamatakeshi wrote:
> 
> On Tue, Aug 18, 2009 at 8:34 PM, NOx-WHV<enno.egbert at googlemail.com>
> wrote:
>>
>> Hi,
>>
>> sorry - i have to ask again. This is my entry in
>> ..../conf/sip_profiles/external/ :
>>
>> <include>
>> <gateway name="sipgate.de">
>>
>>
>>
>>
>> </gateway>
>> </include>
>>
>> and it?s still the same... from: 2395805 at 139.13.37.160
>>
>> Do you have another tip?
> 
> I don't see where you are setting the param from-domain to sipgate.de
> as Brian told you to do. I can only see you are setting the name of
> the gateway to sipgate.de.
> I never used this, but in the samples that come with FS we can see this:
> 
> 
> 
>>
>> Brian West-3 wrote:
>>>
>>> Well if you pay attention I told you in the last email... set the
>>> param from-domain on the gateway to sipgate.de
>>>
>>> /b
>>>
>>> On Aug 17, 2009, at 8:36 AM, NOx-WHV wrote:
>>>
>>>>
>>>> Hi,
>>>>
>>>> i have just taken some pictures.
>>>>
>>>> http://www.nabble.com/file/p25006858/pic1.JPG pic1.JPG
>>>> http://www.nabble.com/file/p25006858/pic2.JPG pic2.JPG
>>>> http://www.nabble.com/file/p25006858/pic3.JPG pic3.JPG
>>>>
>>>> the third picture is taken by a softphone that works.
>>>>
>>>> To Brian:
>>>> If i set the effective_caller_id_name to 2395805 without @sipgate.de
>>>> it?s
>>>> the same problem, because the freeswitch set in the from fielt:
>>>> 2395805 at 139.13.37.160. And i can?t find the right parameter to change
>>>> "139.13.37.160" to "sipgate.de"
>>>>
>>>> Which is the right parameter in the dialplan?
>>>>
>>>> <action application=.....?????>
>>>>
>>>> Thanks for your help
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> --
>> View this message in context:
>> http://www.nabble.com/SIPGATE-Problem-tp25005636p25023035.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/SIPGATE-Problem-tp25005636p25024536.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From mayamatakeshi at gmail.com  Tue Aug 18 07:03:43 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 18 Aug 2009 23:03:43 +0900
Subject: [Freeswitch-users] SIPGATE Problem
In-Reply-To: <25024536.post@talk.nabble.com>
References: <25005636.post@talk.nabble.com>
	<672D0744-FD6A-448E-9019-4F961621368E@freeswitch.org>
	<25006858.post@talk.nabble.com>
	<11D5AA56-1609-4D2C-B5D1-8AAF4A575309@freeswitch.org>
	<25023035.post@talk.nabble.com>
	<15b9404e0908180453r59cb95bbn55f3ab9721a298d2@mail.gmail.com>
	<25024536.post@talk.nabble.com>
Message-ID: <15b9404e0908180703s77a4877ek2cafd976ffc8a2f4@mail.gmail.com>

On Tue, Aug 18, 2009 at 10:10 PM, NOx-WHV<enno.egbert at googlemail.com> wrote:
>
> Hi,
>
> this is the text without brackets:
>
> include
> gateway name="sipgate.de"
> param name="username" value="2395805"/
> param name="from-domain" value="sipgate.de"/
> param name="password" value="abcde"/
> param name="proxy" value="sipgate.de"/
> /gateway
> /include
>
> I don?t know, why you can?t see the lines between gateway and /gateway.

I think my email client (gmail) doesn't like the brackets.

I've tried the above settings in my machine and it works.
I tried originating a call from the CLI:
originate sofia/gateway/sipgate.de/1234 &park

And I got this:
From: "FreeSWITCH" <sip:2395805 at sipgate.de;transport=udp>;



From tayeb.meftah at gmail.com  Tue Aug 18 07:49:11 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Tue, 18 Aug 2009 14:49:11 +0000
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
 but I have a little problem
In-Reply-To: <8181A0DB-67EE-464C-9A69-30BA0719C24C@jerris.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>	<5A1F9427-8BBC-4F51-9712-56C46FCCC282@freeswitch.org>	<4A884F07.3040200@gmail.com>	<5800526b0908161135o39e64592vd34d9ac280e2b054@mail.gmail.com>	<4A8881E7.8080509@gmail.com>	<D3325007-BA3D-4B65-B533-AEC2BF7AB447@jerris.com>	<4A8922DF.1080008@gmail.com>
	<8181A0DB-67EE-464C-9A69-30BA0719C24C@jerris.com>
Message-ID: <4A8ABF67.8040608@gmail.com>

hi MikeJ,
yes, is allready maintained with the latest APACHE/PHP/MySQL/Perl 
software in uniform server / sourceforge
i can provide you with any informations, if you want to help you about 
using Windows Installer XML
please just add a customisation to the installer and ignore Advanced 
installer
some users want to install FreePBX3 in IIS
we can provide Virtual directory installation optionaly, unstid of 
uniform server
in a custom dialog, we chouse if we use IIS or Uniform Server
http://www.wixedit.sourceforge.net/
thanks,
meftah tayeb
Michael Jerris wrote:
>
> On Aug 17, 2009, at 5:29 AM, Meftah Tayeb wrote:
>
>> hi MikeJ,
>> i prefer creating MSI file that is easy to mintin$
>
> How is this useful vs. something that is already maintained?
>
>> unstid of using inno setup or advanced installer (not free), we can 
>> use WIX (Windows installer XML) that is a open source one
>
> All the tools being used are free, however all are not open source.
>
>> we can create a customised MSI that fully install mor features, 
>> including Sounds / MOH/...
>
> This installer already does all that and more.
>
>> and we can edit XML files easyly to let users chouse each module to 
>> install during setup, for example conferencing, voice mail, Event 
>> Socket and ...
>> also the WAMP server is not fully stable
>> i prefer Uniform Server, that mintin Full Compatibility with Unix 
>> like platform including for example, perl files that need this path: 
>> /usr/...
>> apache2 is installed in /usr/local/apache2, mySQL: /usr/local/mysql 
>> and php is in /usr/local/php
>> and perl is in /usr/bin
>> php my admin is in /etc/phpmyadmin
>> and mor featurs, including multiple instance, virtual hosting and 
>> full management using control pannel
>> this will help enterprise users that need that in windows unstid of WAMP
>>      ---------------    
>> if you don't want to mintin it, jive me a chance to mintin it
>>       ---------------     
>> thanks
>
> I just don't want to split efforts here, every different way we do 
> this is yet another way we have to support.  I see no compelling 
> argument here other than personal preference.  If I am missing 
> anything someone please chime in and correct me.
>
> Mike
>
>>
>>     Contact Me
>>
>>
>>     * SIP <sip:voipworld at sip.voipuser.org>
>>     * INUM <callto:+883510009102769>
>>     * PSTN <callto:0021348526972>
>>     * Email <mailto:tayeb.meftah at gmail.com>
>>
>>
>> Michael Jerris wrote:
>>> I think sticking with standard WAMP is preferable.  What is the 
>>> advantage to creating yet another installer over the one that we 
>>> have already done and maintained?
>>>
>>> Mike
>>>
>>> On Aug 16, 2009, at 6:02 PM, Meftah Tayeb wrote:
>>>
>>>> hello
>>>> i'm rewriting this executable file in MSI format
>>>> i can use:
>>>>
>>>>     * MakeMSI
>>>>     * WIX (Windows installer XML)
>>>>
>>>> what you like?
>>>> WIX is open source, but MakeMSI i'm not sur
>>>> for web server, i'm replacing WAMP with a great web server called 
>>>> uniform server
>>>> this server have a conventional Unix Path configuration, for 
>>>> example, web files can by stored in /www/ and apache is in 
>>>> /usr/local/apache2
>>>> this facilitate the deploiment of applications
>>>> any suggestion?
>>>> thanks
>>>> Meftah Tayeb
>>>> Carlos Talbot wrote:
>>>>> http://files.freeswitch.org/windows_installer/freeswitch-1.0.4.exe
>>>>>
>>>
>>>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus 
> signature database 4345 (20090818) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4345 (20090818) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   



__________ Information from ESET NOD32 Antivirus, version of virus signature database 4345 (20090818) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com





From juanbackson at gmail.com  Tue Aug 18 08:04:31 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Tue, 18 Aug 2009 23:04:31 +0800
Subject: [Freeswitch-users] memory leak with switch_core_hash_insert
Message-ID: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>

Hi,

I am getting some strange vg malloc error message in switch_core_hash_insert.
Does anyone know what is wrong with these few lines?  Am I missing
something?


switch_core_hash_init(&hash,pool);
param_name =switch_core_sprintf(pool,"%s", key);
param_value =switch_core_sprintf(pool,"%s", value);
switch_core_hash_insert(hash,param_name,param_value);


==5459==
==5459== 8,973 (2,344 direct, 6,629 indirect) bytes in 10 blocks are
definitely lost in loss record 62 of 89
==5459==    at 0x4A05809: malloc (vg_replace_malloc.c:149)
==5459==    by 0x4CEBB68: sqlite3MallocX (sqliteInt.h:278)
==5459==    by 0x4CE1C7E: rehash (hash.c:227)
==5459==    by 0x4CE206F: sqlite3HashInsert (hash.c:386)
==5459==    by 0x4C70EA4: switch_core_hash_insert (switch_core_hash.c:67)
==5459==    by 0x7CEAD07: ??? (mod_specialivr.c:343)
==5459==    by 0x7CEBF87: ??? (mod_specialivr.c:784)
==5459==    by 0x7CEDBF0: ??? (mod_specialivr.c:1778)
==5459==    by 0x4C790AC: switch_core_session_run
(switch_core_state_machine.c:109)
==5459==    by 0x4C751FF: switch_core_session_thread
(switch_core_session.c:1066)
==5459==    by 0x4FBC306: start_thread (in /lib64/libpthread-2.5.so)
==5459==    by 0x35018D1DEC: clone (in /lib64/libc-2.5.so)
==5459==
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From mrene_lists at avgs.ca  Tue Aug 18 08:09:02 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 18 Aug 2009 11:09:02 -0400
Subject: [Freeswitch-users] memory leak with switch_core_hash_insert
In-Reply-To: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>
References: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>
Message-ID: <5BB3EE22-66F9-4D9B-9CB9-23A7680757AB@avgs.ca>

Hi,

You're missing:
switch_core_hash_destroy(&hash)

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 18-Aug-09, at 11:04 AM, Juan Backson wrote:

> Hi,
>
> I am getting some strange vg malloc error message in  
> switch_core_hash_insert.  Does anyone know what is wrong with these  
> few lines?  Am I missing something?
>
>
> switch_core_hash_init(&hash,pool);
> param_name =switch_core_sprintf(pool,"%s", key);
> param_value =switch_core_sprintf(pool,"%s", value);
> switch_core_hash_insert(hash,param_name,param_value);
>
>
> ==5459==
> ==5459== 8,973 (2,344 direct, 6,629 indirect) bytes in 10 blocks are  
> definitely lost in loss record 62 of 89
> ==5459==    at 0x4A05809: malloc (vg_replace_malloc.c:149)
> ==5459==    by 0x4CEBB68: sqlite3MallocX (sqliteInt.h:278)
> ==5459==    by 0x4CE1C7E: rehash (hash.c:227)
> ==5459==    by 0x4CE206F: sqlite3HashInsert (hash.c:386)
> ==5459==    by 0x4C70EA4: switch_core_hash_insert  
> (switch_core_hash.c:67)
> ==5459==    by 0x7CEAD07: ??? (mod_specialivr.c:343)
> ==5459==    by 0x7CEBF87: ??? (mod_specialivr.c:784)
> ==5459==    by 0x7CEDBF0: ??? (mod_specialivr.c:1778)
> ==5459==    by 0x4C790AC: switch_core_session_run  
> (switch_core_state_machine.c:109)
> ==5459==    by 0x4C751FF: switch_core_session_thread  
> (switch_core_session.c:1066)
> ==5459==    by 0x4FBC306: start_thread (in /lib64/libpthread-2.5.so)
> ==5459==    by 0x35018D1DEC: clone (in /lib64/libc-2.5.so)
> ==5459==
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From markmorreny at gmail.com  Tue Aug 18 08:18:33 2009
From: markmorreny at gmail.com (mark morreny)
Date: Tue, 18 Aug 2009 23:18:33 +0800
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <FD54C936-D80E-4F96-8502-740ADDD1225C@avgs.ca>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
	<20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
	<EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>
	<20ad6b920908141919u1b2aa1c4jacc75cd4896d72b0@mail.gmail.com>
	<EF4B4642-E586-42E0-8D40-D9FCB7A93DD2@jerris.com>
	<FD54C936-D80E-4F96-8502-740ADDD1225C@avgs.ca>
Message-ID: <20ad6b920908180818q7c5a7214vc20aa8ed864953e9@mail.gmail.com>

Hi,

Thank you very much.  I am able to get my sched api to work now.

I still have one problem.  I am getting core dump if freeswitch is shut down
by typing in "shutdown" in the CLI.  How can I control the shutdown process
so that it will wait until all the existing sched jobs are done before
starting to free the memory?

Mark

On Sat, Aug 15, 2009 at 11:26 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Because switch_time_now() is in microseconds.
>    Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
>   On 14-Aug-09, at 11:16 PM, Michael Jerris wrote:
>
>    task->runtime = switch_epoch_time_now(NULL) + 10;
>  On Aug 14, 2009, at 10:19 PM, mark morreny wrote:
>
>  Hi Michael,
>
> The following code was executed once, but not after the next 10 s.
>
> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to flush
> data  buffer...\n");
>
>     task->runtime = switch_time_now() + 10;
> }
>
> Any suggestion why?
>
>
> Thanks,
> Mark
> On Sat, Aug 15, 2009 at 2:13 AM, Michael Jerris <mike at jerris.com> wrote:
>
>> thats in seconds.
>> Mike
>>
>>  On Aug 14, 2009, at 8:32 AM, mark morreny wrote:
>>
>>  Hi,
>>
>> Thank you for your help.
>>
>> I get that too, but the callback does not execute the second time.
>>
>> When I do task->runtime = switch_time_now() + 10;, what does +10 mean?
>> Does it mean 10 s or 10 mins?
>>
>> Thanks,
>> Mark
>>
>> On Wed, Aug 12, 2009 at 11:09 PM, Mathieu Rene <mrene_lists at avgs.ca>wrote:
>>
>>> Hi,
>>> I did the same thing on my side....
>>>  API CALL [load(mod_skel)] output:
>>> +OK
>>>
>>> 2009-08-12 11:08:18.37891 [DEBUG] switch_scheduler.c:214 Added task 2
>>> data_flush (core) to run at 1250089698
>>> 2009-08-12 11:08:18.37891 [CONSOLE] switch_loadable_module.c:889
>>> Successfully Loaded [mod_skel]
>>> 2009-08-12 11:08:18.37891 [NOTICE] switch_loadable_module.c:270 Adding
>>> API Function 'skel'
>>> freeswitch at Maths-Mac.local> 2009-08-12 11:08:18.207113 [ERR]
>>> mod_skel.c:120 starting to flush data  buffer...
>>>
>>> Note that you don't need to start the thread manually, the core already
>>> has threads running for the scheduler.
>>>
>>>   Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> mrene at avgs.ca
>>>
>>>
>>>
>>>
>>>  On 12-Aug-09, at 7:26 AM, mark morreny wrote:
>>>
>>> Hi,
>>>
>>> In my LOAD_FUNCTION, I am trying to have freeswitch to flush out some
>>> data every 10 s.  The following lines of code does not show any effect at
>>> all.
>>>
>>>     switch_scheduler_task_thread_start();
>>>     switch_scheduler_add_task(switch_epoch_time_now(NULL),
>>> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|SSHF_NO_DEL);
>>>
>>>
>>> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>>>
>>> switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_ERROR, "starting to
>>> flush data  buffer...\n");
>>>
>>>
>>>     task->runtime = switch_time_now() + 10;
>>>
>>> }
>>>
>>> Does anyone know how to get it to work?
>>>
>>> Thanks,
>>> Mark
>>>
>>>
>>> ---------- Forwarded message ----------
>>> From: Brian West <brian at freeswitch.org>
>>> Date: Mon, Aug 10, 2009 at 8:53 PM
>>> Subject: Re: [Freeswitch-users] Fwd: Scheduler in module
>>> To: freeswitch-users at lists.freeswitch.org
>>>
>>>
>>> switch_rtp.c has a simple one for the zrtp cache storing.
>>>
>>> /b
>>>
>>> On Aug 10, 2009, at 7:13 AM, Michael Jerris wrote:
>>>
>>> > Re schedule is done in your callback, take a look at places that use
>>> > these apis in the code for details.
>>>
>>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Aug 18 08:21:21 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 18 Aug 2009 11:21:21 -0400
Subject: [Freeswitch-users] memory leak with switch_core_hash_insert
In-Reply-To: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>
References: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>
Message-ID: <BB05B7CA-F29D-48E0-A102-7518866AFAB9@jerris.com>

try running freeswitch with -vg command line arg.

Mike

On Aug 18, 2009, at 11:04 AM, Juan Backson wrote:

> Hi,
>
> I am getting some strange vg malloc error message in  
> switch_core_hash_insert.  Does anyone know what is wrong with these  
> few lines?  Am I missing something?
>
>
> switch_core_hash_init(&hash,pool);
> param_name =switch_core_sprintf(pool,"%s", key);
> param_value =switch_core_sprintf(pool,"%s", value);
> switch_core_hash_insert(hash,param_name,param_value);
>
>
> ==5459==
> ==5459== 8,973 (2,344 direct, 6,629 indirect) bytes in 10 blocks are  
> definitely lost in loss record 62 of 89
> ==5459==    at 0x4A05809: malloc (vg_replace_malloc.c:149)
> ==5459==    by 0x4CEBB68: sqlite3MallocX (sqliteInt.h:278)
> ==5459==    by 0x4CE1C7E: rehash (hash.c:227)
> ==5459==    by 0x4CE206F: sqlite3HashInsert (hash.c:386)
> ==5459==    by 0x4C70EA4: switch_core_hash_insert  
> (switch_core_hash.c:67)
> ==5459==    by 0x7CEAD07: ??? (mod_specialivr.c:343)
> ==5459==    by 0x7CEBF87: ??? (mod_specialivr.c:784)
> ==5459==    by 0x7CEDBF0: ??? (mod_specialivr.c:1778)
> ==5459==    by 0x4C790AC: switch_core_session_run  
> (switch_core_state_machine.c:109)
> ==5459==    by 0x4C751FF: switch_core_session_thread  
> (switch_core_session.c:1066)
> ==5459==    by 0x4FBC306: start_thread (in /lib64/libpthread-2.5.so)
> ==5459==    by 0x35018D1DEC: clone (in /lib64/libc-2.5.so)
> ==5459==
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tzury.by at reguluslabs.com  Tue Aug 18 08:22:19 2009
From: tzury.by at reguluslabs.com (Tzury Bar Yochay)
Date: Tue, 18 Aug 2009 18:22:19 +0300
Subject: [Freeswitch-users] how to use freeswitch as a sip client
Message-ID: <10128ef10908180822p373b2415rb558bbaa8df5020a@mail.gmail.com>

Hi,

Can someone instruct me (or point to a page where I can found this
information) how can freeswitch be used as a sip client.
I would like to test a server transporting SIP packets over TCP and was
reported on this mailing list that it is possible to do so with FS.

thanks,
Tzury
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From dujinfang at gmail.com  Tue Aug 18 08:26:37 2009
From: dujinfang at gmail.com (Seven Du)
Date: Tue, 18 Aug 2009 23:26:37 +0800
Subject: [Freeswitch-users] how to use freeswitch as a sip client
In-Reply-To: <10128ef10908180822p373b2415rb558bbaa8df5020a@mail.gmail.com>
References: <10128ef10908180822p373b2415rb558bbaa8df5020a@mail.gmail.com>
Message-ID: <7BC38AD3-79BB-4533-A4B7-501539DE678C@gmail.com>

http://wiki.freeswitch.org/wiki/Mod_portaudio
http://wiki.freeswitch.org/wiki/FsAir


On Aug 18, 2009, at 11:22 PM, Tzury Bar Yochay wrote:
> Hi,
>
> Can someone instruct me (or point to a page where I can found this  
> information) how can freeswitch be used as a sip client.
> I would like to test a server transporting SIP packets over TCP and  
> was reported on this mailing list that it is possible to do so with  
> FS.
>
> thanks,
> Tzury
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From dist.lists at gmail.com  Tue Aug 18 08:24:48 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Tue, 18 Aug 2009 18:24:48 +0300
Subject: [Freeswitch-users] failover based on initial INVITE timeout
Message-ID: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>

I am trying to implement failover dialing plan as described in:
http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout

I figured out that originate_timeout must be passed as
{originate_timeout=<timeout>} in front of the dial string to have any
effect (setting it as channel variable as described in the example
above has no effect).

I have set the timeout to 1 second, so expected behavior is to try the
second gateway if no response is received from the first one in 1
second. The problem is that FS cancels the first request with
[NO_ANSWER] and tries to route the call via the second gateway even
though it receives response from the first during that 1 second.

The response received is "100 Trying" provisional response (checked
with sofia siptrace). I'm guessing that either 100 provisional
responses don't cancel the originate_timeout timer (bug?) or I am
doing it the wrong way.

I was also thinking of using the timer-T1 or timer-T1X64 parameter in
the sip profile, but I need this to be set per dial string, not per
profile, besides, it seems that these timers (T1, T1X64) affect both
invite and non-invite requests, so this is not really an option.
Also, I tried leg_timeout, but it doesn't really do what I need it to.

Anyone has any idea how to implement this?



From brian at freeswitch.org  Tue Aug 18 08:26:57 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Aug 2009 10:26:57 -0500
Subject: [Freeswitch-users] Fwd: Fwd: Scheduler in module
In-Reply-To: <20ad6b920908180818q7c5a7214vc20aa8ed864953e9@mail.gmail.com>
References: <20ad6b920908090857t1f60562bv4fd8beb176ae63b6@mail.gmail.com>
	<985283E0-8DEA-411B-82EC-9753004FDE15@jerris.com>
	<5B94AA53-D374-4037-8C74-7BC10C522AF6@freeswitch.org>
	<20ad6b920908120426p147a7a65r4e8937dc1727378a@mail.gmail.com>
	<3F14D790-027D-4285-8D6D-B714980F6D2C@avgs.ca>
	<20ad6b920908140532n2dea8b5ev5a016a2805f26122@mail.gmail.com>
	<EE0915F9-32F0-4119-A75E-C65799416693@jerris.com>
	<20ad6b920908141919u1b2aa1c4jacc75cd4896d72b0@mail.gmail.com>
	<EF4B4642-E586-42E0-8D40-D9FCB7A93DD2@jerris.com>
	<FD54C936-D80E-4F96-8502-740ADDD1225C@avgs.ca>
	<20ad6b920908180818q7c5a7214vc20aa8ed864953e9@mail.gmail.com>
Message-ID: <28DCCC86-C909-4AD7-9AD5-3D5AC9D39EB1@freeswitch.org>

Sounds like you need to put a mutex on your scheduler routine.

/b

On Aug 18, 2009, at 10:18 AM, mark morreny wrote:

> Hi,
>
> Thank you very much.  I am able to get my sched api to work now.
>
> I still have one problem.  I am getting core dump if freeswitch is  
> shut down by typing in "shutdown" in the CLI.  How can I control the  
> shutdown process so that it will wait until all the existing sched  
> jobs are done before starting to free the memory?
>
> Mark




From brian at freeswitch.org  Tue Aug 18 08:33:01 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Aug 2009 10:33:01 -0500
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
Message-ID: <45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>

I think you wanna use progress_timeout

http://wiki.freeswitch.org/wiki/Channel_Variables#progress_timeout

/b

On Aug 18, 2009, at 10:24 AM, Hristo Trendev wrote:

> I am trying to implement failover dialing plan as described in:
> http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout
>
> I figured out that originate_timeout must be passed as
> {originate_timeout=<timeout>} in front of the dial string to have any
> effect (setting it as channel variable as described in the example
> above has no effect).
>
> I have set the timeout to 1 second, so expected behavior is to try the
> second gateway if no response is received from the first one in 1
> second. The problem is that FS cancels the first request with
> [NO_ANSWER] and tries to route the call via the second gateway even
> though it receives response from the first during that 1 second.
>
> The response received is "100 Trying" provisional response (checked
> with sofia siptrace). I'm guessing that either 100 provisional
> responses don't cancel the originate_timeout timer (bug?) or I am
> doing it the wrong way.
>
> I was also thinking of using the timer-T1 or timer-T1X64 parameter in
> the sip profile, but I need this to be set per dial string, not per
> profile, besides, it seems that these timers (T1, T1X64) affect both
> invite and non-invite requests, so this is not really an option.
> Also, I tried leg_timeout, but it doesn't really do what I need it to.
>
> Anyone has any idea how to implement this?

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From msc at freeswitch.org  Tue Aug 18 09:02:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Aug 2009 09:02:33 -0700
Subject: [Freeswitch-users] memory leak with switch_core_hash_insert
In-Reply-To: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>
References: <27c25bc40908180804u58b6d488h8e5acb4787095fd5@mail.gmail.com>
Message-ID: <87f2f3b90908180902i4ddb6c75mbc4c6b353ce39821@mail.gmail.com>

Just a heads up: this is a rather technical subject. I highly recommend that
you use freeswitch-dev list for this kind of thing.
-MC

On Tue, Aug 18, 2009 at 8:04 AM, Juan Backson <juanbackson at gmail.com> wrote:

> Hi,
>
> I am getting some strange vg malloc error message in switch_core_hash_insert.
> Does anyone know what is wrong with these few lines?  Am I missing
> something?
>
>
> switch_core_hash_init(&hash,pool);
> param_name =switch_core_sprintf(pool,"%s", key);
> param_value =switch_core_sprintf(pool,"%s", value);
> switch_core_hash_insert(hash,param_name,param_value);
>
>
> ==5459==
> ==5459== 8,973 (2,344 direct, 6,629 indirect) bytes in 10 blocks are
> definitely lost in loss record 62 of 89
> ==5459==    at 0x4A05809: malloc (vg_replace_malloc.c:149)
> ==5459==    by 0x4CEBB68: sqlite3MallocX (sqliteInt.h:278)
> ==5459==    by 0x4CE1C7E: rehash (hash.c:227)
> ==5459==    by 0x4CE206F: sqlite3HashInsert (hash.c:386)
> ==5459==    by 0x4C70EA4: switch_core_hash_insert (switch_core_hash.c:67)
> ==5459==    by 0x7CEAD07: ??? (mod_specialivr.c:343)
> ==5459==    by 0x7CEBF87: ??? (mod_specialivr.c:784)
> ==5459==    by 0x7CEDBF0: ??? (mod_specialivr.c:1778)
> ==5459==    by 0x4C790AC: switch_core_session_run
> (switch_core_state_machine.c:109)
> ==5459==    by 0x4C751FF: switch_core_session_thread
> (switch_core_session.c:1066)
> ==5459==    by 0x4FBC306: start_thread (in /lib64/libpthread-2.5.so)
> ==5459==    by 0x35018D1DEC: clone (in /lib64/libc-2.5.so)
> ==5459==
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From woodydickson at gmail.com  Tue Aug 18 09:23:02 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Wed, 19 Aug 2009 00:23:02 +0800
Subject: [Freeswitch-users] how to set different action for different
	cause code
In-Reply-To: <87f2f3b90908170859p52259449q913b2ddab224709f@mail.gmail.com>
References: <a44688a0908160424n3569cef1vad8fcdc68cbfb96e@mail.gmail.com>
	<87f2f3b90908170859p52259449q913b2ddab224709f@mail.gmail.com>
Message-ID: <a44688a0908180923q4b4b8e48od86d676872c97c3a@mail.gmail.com>

Hi,

I have my dialplan to do some simple routing. What I need to do is when
certain hangup code is received, route advance to the next or the route
after next based on the hangup code received.

So, I have:

<condition>
   <action application="set" data="hangup_after_bridge=true"/>
   <action application="set" data="continue_on_fail="false"/>
   <action application="bridge" data="sofia/internal/aaa at 123.3.3./>
   <action application="bridge" data="sofia/internal/bbb at 123.3.3./>
   <action application="bridge" data="sofia/internal/ccc at 123.3.3./>
   <action application="bridge" data="sofia/internal/ddd at 123.3.3./>

</condition>

What I want to do is that based on the error code received from ( bridge to
aaa), it can route to either bbb or ccc depends on the hangup code received
from aaa.

So there anyway to do it in the dialplan without any scripting?

thannks,
woody




On Mon, Aug 17, 2009 at 11:59 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Sun, Aug 16, 2009 at 4:24 AM, Woody Dickson <woodydickson at gmail.com>wrote:
>
>> Hello,
>>
>> I find hangup_hook, but I would like to define different actions for
>> different hangup codes.  Is there anyway to do that?
>>
>>
>
> I can think of at least two ways you could do this: one that uses only the
> dialplan and one that uses a script. If you don't mind using a scripting
> language then you can make it very clean:
>
> <action application="set" data="api_hangup_hook=luarun chancleanup.lua
> ${hangup_cause}"/>
>
> Then have your Lua script handle all the if-then-else or case stuff.
>
> Question: are you trying to transfer the a-leg to some other destination if
> the b-leg hangup is a specific cause, or are you just doing some external
> cleanup stuff? Just curious...
>
> -MC
>
>
>> Woody
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dist.lists at gmail.com  Tue Aug 18 10:48:19 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Tue, 18 Aug 2009 20:48:19 +0300
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
Message-ID: <2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>

progress_timeout will wait for media (tried it, but I don't need
that). I want to detect the case when the destination gateway is down
and there is no response (not even 100, 407, etc) to the initial
INVITE sent by FS.

According to the wiki, this is exactly what originate_timeout is used
for. Actually the wiki gives as example for originate_timeout exactly
what I'm trying to accomplish.

It seems to me that FS ignores 100 and alike messages, which are sent
as response to initial INVITE and doesn't cancel originate_timeout
timer if such message is received.

The more I look into this, the more I start to think that it's a bug.

On Tue, Aug 18, 2009 at 6:33 PM, Brian West<brian at freeswitch.org> wrote:
> I think you wanna use progress_timeout
> http://wiki.freeswitch.org/wiki/Channel_Variables#progress_timeout
> /b
> On Aug 18, 2009, at 10:24 AM, Hristo Trendev wrote:
>
> I am trying to implement failover dialing plan as described in:
> http://wiki.freeswitch.org/wiki/Channel_Variables#originate_timeout
>
> I figured out that originate_timeout must be passed as
> {originate_timeout=<timeout>} in front of the dial string to have any
> effect (setting it as channel variable as described in the example
> above has no effect).
>
> I have set the timeout to 1 second, so expected behavior is to try the
> second gateway if no response is received from the first one in 1
> second. The problem is that FS cancels the first request with
> [NO_ANSWER] and tries to route the call via the second gateway even
> though it receives response from the first during that 1 second.
>
> The response received is "100 Trying" provisional response (checked
> with sofia siptrace). I'm guessing that either 100 provisional
> responses don't cancel the originate_timeout timer (bug?) or I am
> doing it the wrong way.
>
> I was also thinking of using the timer-T1 or timer-T1X64 parameter in
> the sip profile, but I need this to be set per dial string, not per
> profile, besides, it seems that these timers (T1, T1X64) affect both
> invite and non-invite requests, so this is not really an option.
> Also, I tried leg_timeout, but it doesn't really do what I need it to.
>
> Anyone has any idea how to implement this?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Tue Aug 18 11:00:39 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Aug 2009 13:00:39 -0500
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
Message-ID: <F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>

use TCP then if you get an ICMP unreachable it'll move on instantly.

/b

On Aug 18, 2009, at 12:48 PM, Hristo Trendev wrote:

> progress_timeout will wait for media (tried it, but I don't need
> that). I want to detect the case when the destination gateway is down
> and there is no response (not even 100, 407, etc) to the initial
> INVITE sent by FS.
>
> According to the wiki, this is exactly what originate_timeout is used
> for. Actually the wiki gives as example for originate_timeout exactly
> what I'm trying to accomplish.
>
> It seems to me that FS ignores 100 and alike messages, which are sent
> as response to initial INVITE and doesn't cancel originate_timeout
> timer if such message is received.
>
> The more I look into this, the more I start to think that it's a bug.




From dist.lists at gmail.com  Tue Aug 18 11:24:57 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Tue, 18 Aug 2009 21:24:57 +0300
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
Message-ID: <2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>

TCP is not really an option for me. I have tried using different sofia
profile for that and set:

<param name="timer-T1X64" value="1000" />

This works the way I want and INVITEs to dead gateways are
disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
1. It affects all calls sent via this "custom timer" profile
2. I need to use one more SIP port to bind to (I need keep the
"default timers" profile as well)
3. The timer is used by transactions other than the initial invite
message and that may cause unexpected problems.

Obviously originate_timeout doesn't work the way it's supposed to
(according to the wiki) so I will report it as bug.

The profile trick above actually solves my problem and may happen to
be a better solution after all, but I will need to test it for some
time before I know.

On Tue, Aug 18, 2009 at 9:00 PM, Brian West<brian at freeswitch.org> wrote:
> use TCP then if you get an ICMP unreachable it'll move on instantly.
>
> /b
>
> On Aug 18, 2009, at 12:48 PM, Hristo Trendev wrote:
>
>> progress_timeout will wait for media (tried it, but I don't need
>> that). I want to detect the case when the destination gateway is down
>> and there is no response (not even 100, 407, etc) to the initial
>> INVITE sent by FS.
>>
>> According to the wiki, this is exactly what originate_timeout is used
>> for. Actually the wiki gives as example for originate_timeout exactly
>> what I'm trying to accomplish.
>>
>> It seems to me that FS ignores 100 and alike messages, which are sent
>> as response to initial INVITE and doesn't cancel originate_timeout
>> timer if such message is received.
>>
>> The more I look into this, the more I start to think that it's a bug.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Tue Aug 18 11:30:56 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Aug 2009 13:30:56 -0500
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
Message-ID: <A7B2812C-DA99-4284-AC44-9BCD5C78E967@freeswitch.org>

setup gateways and turn on the ping option.
/b

On Aug 18, 2009, at 1:24 PM, Hristo Trendev wrote:

> TCP is not really an option for me. I have tried using different sofia
> profile for that and set:
>
> <param name="timer-T1X64" value="1000" />
>
> This works the way I want and INVITEs to dead gateways are
> disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
> 1. It affects all calls sent via this "custom timer" profile
> 2. I need to use one more SIP port to bind to (I need keep the
> "default timers" profile as well)
> 3. The timer is used by transactions other than the initial invite
> message and that may cause unexpected problems.
>
> Obviously originate_timeout doesn't work the way it's supposed to
> (according to the wiki) so I will report it as bug.
>
> The profile trick above actually solves my problem and may happen to
> be a better solution after all, but I will need to test it for some
> time before I know.




From mrene_lists at avgs.ca  Tue Aug 18 11:32:59 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 18 Aug 2009 14:32:59 -0400
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
Message-ID: <C01FD4EC-8F34-4C64-8EE7-C4E18276B2C3@avgs.ca>

Hi,

originate_timeout does exactly what it is supposed to do, cancel the  
new call if originate doesnt return within X seconds.
In my opinion, I would keep state wether the gateway is alive or not,  
and have a reasonable t1x64 timeout value (5000 should be enough).
This way it'll only timeout the first time the call is attempted.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 18-Aug-09, at 2:24 PM, Hristo Trendev wrote:

> TCP is not really an option for me. I have tried using different sofia
> profile for that and set:
>
> <param name="timer-T1X64" value="1000" />
>
> This works the way I want and INVITEs to dead gateways are
> disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
> 1. It affects all calls sent via this "custom timer" profile
> 2. I need to use one more SIP port to bind to (I need keep the
> "default timers" profile as well)
> 3. The timer is used by transactions other than the initial invite
> message and that may cause unexpected problems.
>
> Obviously originate_timeout doesn't work the way it's supposed to
> (according to the wiki) so I will report it as bug.
>
> The profile trick above actually solves my problem and may happen to
> be a better solution after all, but I will need to test it for some
> time before I know.
>
> On Tue, Aug 18, 2009 at 9:00 PM, Brian West<brian at freeswitch.org>  
> wrote:
>> use TCP then if you get an ICMP unreachable it'll move on instantly.
>>
>> /b
>>
>> On Aug 18, 2009, at 12:48 PM, Hristo Trendev wrote:
>>
>>> progress_timeout will wait for media (tried it, but I don't need
>>> that). I want to detect the case when the destination gateway is  
>>> down
>>> and there is no response (not even 100, 407, etc) to the initial
>>> INVITE sent by FS.
>>>
>>> According to the wiki, this is exactly what originate_timeout is  
>>> used
>>> for. Actually the wiki gives as example for originate_timeout  
>>> exactly
>>> what I'm trying to accomplish.
>>>
>>> It seems to me that FS ignores 100 and alike messages, which are  
>>> sent
>>> as response to initial INVITE and doesn't cancel originate_timeout
>>> timer if such message is received.
>>>
>>> The more I look into this, the more I start to think that it's a  
>>> bug.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From dist.lists at gmail.com  Tue Aug 18 11:40:56 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Tue, 18 Aug 2009 21:40:56 +0300
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <A7B2812C-DA99-4284-AC44-9BCD5C78E967@freeswitch.org>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
	<A7B2812C-DA99-4284-AC44-9BCD5C78E967@freeswitch.org>
Message-ID: <2a73afe0908181140j5371cea8p79e51f2139646806@mail.gmail.com>

I've seen this parameter when I started testing, but the name suggest
that it will do icmp ping to detect failed gateways (haven't really
tried it though). I need to detect sip service failure as well - the
case where the destination server is up (ping replies are received),
but the sip process is down and there is no response to sip requests.

If the ping parameter uses sip-based ping (like OPTIONS ping) this may
be a solution for me as well.

On Tue, Aug 18, 2009 at 9:30 PM, Brian West<brian at freeswitch.org> wrote:
> setup gateways and turn on the ping option.
> /b
>
> On Aug 18, 2009, at 1:24 PM, Hristo Trendev wrote:
>
>> TCP is not really an option for me. I have tried using different sofia
>> profile for that and set:
>>
>> <param name="timer-T1X64" value="1000" />
>>
>> This works the way I want and INVITEs to dead gateways are
>> disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
>> 1. It affects all calls sent via this "custom timer" profile
>> 2. I need to use one more SIP port to bind to (I need keep the
>> "default timers" profile as well)
>> 3. The timer is used by transactions other than the initial invite
>> message and that may cause unexpected problems.
>>
>> Obviously originate_timeout doesn't work the way it's supposed to
>> (according to the wiki) so I will report it as bug.
>>
>> The profile trick above actually solves my problem and may happen to
>> be a better solution after all, but I will need to test it for some
>> time before I know.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mrene_lists at avgs.ca  Tue Aug 18 11:42:36 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 18 Aug 2009 14:42:36 -0400
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <2a73afe0908181140j5371cea8p79e51f2139646806@mail.gmail.com>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
	<A7B2812C-DA99-4284-AC44-9BCD5C78E967@freeswitch.org>
	<2a73afe0908181140j5371cea8p79e51f2139646806@mail.gmail.com>
Message-ID: <E4D7F0B9-8036-4187-99F5-171A507230F6@avgs.ca>

It does use SIP OPTIONS.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 18-Aug-09, at 2:40 PM, Hristo Trendev wrote:

> I've seen this parameter when I started testing, but the name suggest
> that it will do icmp ping to detect failed gateways (haven't really
> tried it though). I need to detect sip service failure as well - the
> case where the destination server is up (ping replies are received),
> but the sip process is down and there is no response to sip requests.
>
> If the ping parameter uses sip-based ping (like OPTIONS ping) this may
> be a solution for me as well.
>
> On Tue, Aug 18, 2009 at 9:30 PM, Brian West<brian at freeswitch.org>  
> wrote:
>> setup gateways and turn on the ping option.
>> /b
>>
>> On Aug 18, 2009, at 1:24 PM, Hristo Trendev wrote:
>>
>>> TCP is not really an option for me. I have tried using different  
>>> sofia
>>> profile for that and set:
>>>
>>> <param name="timer-T1X64" value="1000" />
>>>
>>> This works the way I want and INVITEs to dead gateways are
>>> disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
>>> 1. It affects all calls sent via this "custom timer" profile
>>> 2. I need to use one more SIP port to bind to (I need keep the
>>> "default timers" profile as well)
>>> 3. The timer is used by transactions other than the initial invite
>>> message and that may cause unexpected problems.
>>>
>>> Obviously originate_timeout doesn't work the way it's supposed to
>>> (according to the wiki) so I will report it as bug.
>>>
>>> The profile trick above actually solves my problem and may happen to
>>> be a better solution after all, but I will need to test it for some
>>> time before I know.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Tue Aug 18 11:45:11 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 18 Aug 2009 13:45:11 -0500
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <2a73afe0908181140j5371cea8p79e51f2139646806@mail.gmail.com>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
	<A7B2812C-DA99-4284-AC44-9BCD5C78E967@freeswitch.org>
	<2a73afe0908181140j5371cea8p79e51f2139646806@mail.gmail.com>
Message-ID: <11870622-2039-412A-ADB8-D58F04F4F737@freeswitch.org>

Why would it make you think its ICMP?  Weird.  Its a sip based options  
ping.

/b

On Aug 18, 2009, at 1:40 PM, Hristo Trendev wrote:

> I've seen this parameter when I started testing, but the name suggest
> that it will do icmp ping to detect failed gateways (haven't really
> tried it though). I need to detect sip service failure as well - the
> case where the destination server is up (ping replies are received),
> but the sip process is down and there is no response to sip requests.
>
> If the ping parameter uses sip-based ping (like OPTIONS ping) this may
> be a solution for me as well.




From dist.lists at gmail.com  Tue Aug 18 11:50:01 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Tue, 18 Aug 2009 21:50:01 +0300
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <C01FD4EC-8F34-4C64-8EE7-C4E18276B2C3@avgs.ca>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
	<C01FD4EC-8F34-4C64-8EE7-C4E18276B2C3@avgs.ca>
Message-ID: <2a73afe0908181150o3d868f0fgc18f1129c9cee002@mail.gmail.com>

That's exactly what I'm trying to do. This used to be implemented with
openser and on timeout it will mark destination as inactive so
subsequent calls don't get sent to dead gateways.

If originate_timeout is intended for something else, then maybe the
example given in the wiki can be changed, because may be a bit
misleading for someone trying to do failover.

On Tue, Aug 18, 2009 at 9:32 PM, Mathieu Rene<mrene_lists at avgs.ca> wrote:
> Hi,
>
> originate_timeout does exactly what it is supposed to do, cancel the
> new call if originate doesnt return within X seconds.
> In my opinion, I would keep state wether the gateway is alive or not,
> and have a reasonable t1x64 timeout value (5000 should be enough).
> This way it'll only timeout the first time the call is attempted.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 18-Aug-09, at 2:24 PM, Hristo Trendev wrote:
>
>> TCP is not really an option for me. I have tried using different sofia
>> profile for that and set:
>>
>> <param name="timer-T1X64" value="1000" />
>>
>> This works the way I want and INVITEs to dead gateways are
>> disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
>> 1. It affects all calls sent via this "custom timer" profile
>> 2. I need to use one more SIP port to bind to (I need keep the
>> "default timers" profile as well)
>> 3. The timer is used by transactions other than the initial invite
>> message and that may cause unexpected problems.
>>
>> Obviously originate_timeout doesn't work the way it's supposed to
>> (according to the wiki) so I will report it as bug.
>>
>> The profile trick above actually solves my problem and may happen to
>> be a better solution after all, but I will need to test it for some
>> time before I know.
>>
>> On Tue, Aug 18, 2009 at 9:00 PM, Brian West<brian at freeswitch.org>
>> wrote:
>>> use TCP then if you get an ICMP unreachable it'll move on instantly.
>>>
>>> /b
>>>
>>> On Aug 18, 2009, at 12:48 PM, Hristo Trendev wrote:
>>>
>>>> progress_timeout will wait for media (tried it, but I don't need
>>>> that). I want to detect the case when the destination gateway is
>>>> down
>>>> and there is no response (not even 100, 407, etc) to the initial
>>>> INVITE sent by FS.
>>>>
>>>> According to the wiki, this is exactly what originate_timeout is
>>>> used
>>>> for. Actually the wiki gives as example for originate_timeout
>>>> exactly
>>>> what I'm trying to accomplish.
>>>>
>>>> It seems to me that FS ignores 100 and alike messages, which are
>>>> sent
>>>> as response to initial INVITE and doesn't cancel originate_timeout
>>>> timer if such message is received.
>>>>
>>>> The more I look into this, the more I start to think that it's a
>>>> bug.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From dist.lists at gmail.com  Tue Aug 18 11:51:42 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Tue, 18 Aug 2009 21:51:42 +0300
Subject: [Freeswitch-users] failover based on initial INVITE timeout
In-Reply-To: <E4D7F0B9-8036-4187-99F5-171A507230F6@avgs.ca>
References: <2a73afe0908180824v2df20cc1v8ce8ee89d51877b5@mail.gmail.com>
	<45B0B757-4BCF-4328-9557-3F3A653ABA8D@freeswitch.org>
	<2a73afe0908181048l6ff6ea61n48266e5710b80005@mail.gmail.com>
	<F82A7C97-C413-47C6-94BE-C00BA0A1BE83@freeswitch.org>
	<2a73afe0908181124w1002ed86q59abcf104b64f02d@mail.gmail.com>
	<A7B2812C-DA99-4284-AC44-9BCD5C78E967@freeswitch.org>
	<2a73afe0908181140j5371cea8p79e51f2139646806@mail.gmail.com>
	<E4D7F0B9-8036-4187-99F5-171A507230F6@avgs.ca>
Message-ID: <2a73afe0908181151v6bb638bx70b86ffd30d3276b@mail.gmail.com>

Thanks. I guess I will try that as well and see which one works better
for me then.

On Tue, Aug 18, 2009 at 9:42 PM, Mathieu Rene<mrene_lists at avgs.ca> wrote:
> It does use SIP OPTIONS.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 18-Aug-09, at 2:40 PM, Hristo Trendev wrote:
>
>> I've seen this parameter when I started testing, but the name suggest
>> that it will do icmp ping to detect failed gateways (haven't really
>> tried it though). I need to detect sip service failure as well - the
>> case where the destination server is up (ping replies are received),
>> but the sip process is down and there is no response to sip requests.
>>
>> If the ping parameter uses sip-based ping (like OPTIONS ping) this may
>> be a solution for me as well.
>>
>> On Tue, Aug 18, 2009 at 9:30 PM, Brian West<brian at freeswitch.org>
>> wrote:
>>> setup gateways and turn on the ping option.
>>> /b
>>>
>>> On Aug 18, 2009, at 1:24 PM, Hristo Trendev wrote:
>>>
>>>> TCP is not really an option for me. I have tried using different
>>>> sofia
>>>> profile for that and set:
>>>>
>>>> <param name="timer-T1X64" value="1000" />
>>>>
>>>> This works the way I want and INVITEs to dead gateways are
>>>> disconnected with [RECOVERY_ON_TIMER_EXPIRE], but:
>>>> 1. It affects all calls sent via this "custom timer" profile
>>>> 2. I need to use one more SIP port to bind to (I need keep the
>>>> "default timers" profile as well)
>>>> 3. The timer is used by transactions other than the initial invite
>>>> message and that may cause unexpected problems.
>>>>
>>>> Obviously originate_timeout doesn't work the way it's supposed to
>>>> (according to the wiki) so I will report it as bug.
>>>>
>>>> The profile trick above actually solves my problem and may happen to
>>>> be a better solution after all, but I will need to test it for some
>>>> time before I know.
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From alan at chandlerfamily.org.uk  Tue Aug 18 11:59:29 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Tue, 18 Aug 2009 19:59:29 +0100
Subject: [Freeswitch-users] Stuck on my first attempt at dialplanning
Message-ID: <4A8AFA11.2080603@chandlerfamily.org.uk>

This is my first attempt at setting up the dial plan to do anything more 
than the basic default stuff.  What I am trying to do is set up some 
conferencing as follows

1. A calls B
2. B puts both A and B into a conference
3. A or B calls out from the conference (via the * set in caller 
controls) to C
4a If C hangs up or rejects the call, A/B should go back to the conference
4b If A/B types *1 before the call is answered he should go back to the 
conference
4c If A/B types *1 after the call is answered both A/B and C should go 
back into the conference.




It nearly all works except

i) A seemingly can't use the * caller control, where as B can.  The 
other caller control I set up (mute) seems to work

ii) If C rejects the call before answering, B gets hung_up (because of 
i) I can only test with B)

iii) When B gets back to the conference, the * caller control doesn't 
work (although the mute does)

Can someone tell me where I am going wrong.


----------------------

Here is the dialplan snippet that I use to handle the dialout from the 
conference (pressing the * in step 3)

     <!-- Conference Callout To Another Extension -->
     <extension name="cco">
       <condition field="destination_number" expression="^cco$">
         <action application="read" data="3 3 
'tone_stream://%(10000,0,350,440)' digits 5000 #"/>
         <action application="bind_meta_app" data="2 a s 
execute_extension::conf_transfer XML features" />
         <action application="set" data="hangup_after_bridge=false" />
         <action application="bridge" 
data="user/${digits}@${domain_name}" />
         <action application="info" />
          <action application="conference" 
data="${conference_name}@default" />
       </condition>
     </extension>

     <extension name="conf_transfer">
       <condition field="destination_number" expression="^conf_transfer$" >
         <action application="answer" />
         <action application="transfer" data="-both ${conference_name} 
XML default" />
       </condition>
     </extension>


And here is the bit in my default XML where I define the conference

     <extension name="nb_conferences">
       <condition field="destination_number" 
expression="^(2(0[1-9]|[1-9][0-9]))$">
	<action application="answer"/>
	<action application="conference" data="$1 at default"/>
       </condition>
     </extension>

The call out to the conference from the original call is done via 
changing the bind meta app in the default dialplan for local extension

	<action application="bind_meta_app" data="2 ab s execute_extension::cf 
XML features"/>

and this extension in features

     <!-- Used to transfer both legs into a conference -->
     <extension name="cf">
       <condition field="destination_number" expression="^cf$">
	<action application="answer"/>
	<action application="transfer" data="-both 2${dialed_extension:1} XML 
default"/>
       </condition>
     </extension>








I can get as far as talking to C after dialing him, but if he doesn't 
answer, or rejects the call I get lost in space somewhere.  If I type *1 
during the call with C I get back to the conference, and C appears to be 
  in the conference (hear him speak) but he gets music on hold.

What am I doing wrong?



-- 
Alan Chandler
http://www.chandlerfamily.org.uk




From alan at chandlerfamily.org.uk  Tue Aug 18 12:53:03 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Tue, 18 Aug 2009 20:53:03 +0100
Subject: [Freeswitch-users] Stuck on my first attempt at dialplanning
In-Reply-To: <4A8AFA11.2080603@chandlerfamily.org.uk>
References: <4A8AFA11.2080603@chandlerfamily.org.uk>
Message-ID: <4A8B069F.9030404@chandlerfamily.org.uk>

Ignore the very last but one paragraph of this e-mail - that's the text 
from an old attempt before I fixed some things before sending this e-mail



Alan Chandler wrote:
> This is my first attempt at setting up the dial plan to do anything more 
> than the basic default stuff.  What I am trying to do is set up some 
> conferencing as follows
> 
> 1. A calls B
> 2. B puts both A and B into a conference
> 3. A or B calls out from the conference (via the * set in caller 
> controls) to C
> 4a If C hangs up or rejects the call, A/B should go back to the conference
> 4b If A/B types *1 before the call is answered he should go back to the 
> conference
> 4c If A/B types *1 after the call is answered both A/B and C should go 
> back into the conference.
> 
> 
> 
> 
> It nearly all works except
> 
> i) A seemingly can't use the * caller control, where as B can.  The 
> other caller control I set up (mute) seems to work
> 
> ii) If C rejects the call before answering, B gets hung_up (because of 
> i) I can only test with B)
> 
> iii) When B gets back to the conference, the * caller control doesn't 
> work (although the mute does)
> 
> Can someone tell me where I am going wrong.
> 
> 
> ----------------------
> 
> Here is the dialplan snippet that I use to handle the dialout from the 
> conference (pressing the * in step 3)
> 
>      <!-- Conference Callout To Another Extension -->
>      <extension name="cco">
>        <condition field="destination_number" expression="^cco$">
>          <action application="read" data="3 3 
> 'tone_stream://%(10000,0,350,440)' digits 5000 #"/>
>          <action application="bind_meta_app" data="2 a s 
> execute_extension::conf_transfer XML features" />
>          <action application="set" data="hangup_after_bridge=false" />
>          <action application="bridge" 
> data="user/${digits}@${domain_name}" />
>          <action application="info" />
>           <action application="conference" 
> data="${conference_name}@default" />
>        </condition>
>      </extension>
> 
>      <extension name="conf_transfer">
>        <condition field="destination_number" expression="^conf_transfer$" >
>          <action application="answer" />
>          <action application="transfer" data="-both ${conference_name} 
> XML default" />
>        </condition>
>      </extension>
> 
> 
> And here is the bit in my default XML where I define the conference
> 
>      <extension name="nb_conferences">
>        <condition field="destination_number" 
> expression="^(2(0[1-9]|[1-9][0-9]))$">
> 	<action application="answer"/>
> 	<action application="conference" data="$1 at default"/>
>        </condition>
>      </extension>
> 
> The call out to the conference from the original call is done via 
> changing the bind meta app in the default dialplan for local extension
> 
> 	<action application="bind_meta_app" data="2 ab s execute_extension::cf 
> XML features"/>
> 
> and this extension in features
> 
>      <!-- Used to transfer both legs into a conference -->
>      <extension name="cf">
>        <condition field="destination_number" expression="^cf$">
> 	<action application="answer"/>
> 	<action application="transfer" data="-both 2${dialed_extension:1} XML 
> default"/>
>        </condition>
>      </extension>
> 
> 
> 
> 
> 
> 
> 
> 
> I can get as far as talking to C after dialing him, but if he doesn't 
> answer, or rejects the call I get lost in space somewhere.  If I type *1 
> during the call with C I get back to the conference, and C appears to be 
>   in the conference (hear him speak) but he gets music on hold.
> 
> What am I doing wrong?
> 
> 
> 


-- 
Alan Chandler
http://www.chandlerfamily.org.uk




From christian.loeschenkohl at xpirio.com  Tue Aug 18 14:41:41 2009
From: christian.loeschenkohl at xpirio.com (=?ISO-8859-1?Q?Christian_L=F6schenkohl?=)
Date: Tue, 18 Aug 2009 23:41:41 +0200
Subject: [Freeswitch-users] send sip options message
Message-ID: <4A8B2015.2060605@xpirio.com>

hi

does anybody know how to send a sip options message to a registered user, using the event socket
or something else build in freeswitch
i think the ping parameter does something like this for gateways.

what i want/need is the same thing that is provided in asterisk with the qualifying option,
to see how "reachable" a certain client is.

br

-- 
Ing. Christian L?schenkohl
Technische Leitung, Forschung & Entwicklung VoIP

xpirio
Telekommunikation & Service GmbH
Lakeside B04
9020 Klagenfurt
Austria

T  +43 (0) 5 77 11 - 1000
F  +43 (0) 5 77 11 - 1002
E  christian.loeschenkohl at xpirio.com



From alan at chandlerfamily.org.uk  Tue Aug 18 16:30:40 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Wed, 19 Aug 2009 00:30:40 +0100
Subject: [Freeswitch-users] Conference Status
Message-ID: <4A8B39A0.4010206@chandlerfamily.org.uk>

In the command to lists a conference it talks about member status, shown 
as hear|speak|floor

What does the floor mean as a status - I can't find any reference to it 
anywhere.


-- 
Alan Chandler
http://www.chandlerfamily.org.uk




From msc at freeswitch.org  Tue Aug 18 17:10:32 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 18 Aug 2009 17:10:32 -0700
Subject: [Freeswitch-users] Conference Status
In-Reply-To: <4A8B39A0.4010206@chandlerfamily.org.uk>
References: <4A8B39A0.4010206@chandlerfamily.org.uk>
Message-ID: <87f2f3b90908181710i4c645f06gedc00689da979677@mail.gmail.com>

On Tue, Aug 18, 2009 at 4:30 PM, Alan Chandler
<alan at chandlerfamily.org.uk>wrote:

> In the command to lists a conference it talks about member status, shown
> as hear|speak|floor
>
> What does the floor mean as a status - I can't find any reference to it
> anywhere.
>

I just spoke with Mike J about this. The floor is basically "who has the
floor" in the conference. In most cases you'll see that it is the person who
is talking or who spoke last. He said there's some algorithmic magic in
there but we don't actually use floor for much of anything...

-MC
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From mattdfong at gmail.com  Wed Aug 19 01:43:29 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Wed, 19 Aug 2009 01:43:29 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
Message-ID: <4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>

Hi Eric,
Thanks for these recommendations.

for vmd-not-panasonic-home-ans.wav changing MAX_FREQ to 1450 WORKED!

but I'm still having problems picking out the ~750Hz beep of sprint,
tmobile, and verizon. I tried first cutting POINTS and VALID in half, then
in half again, while also reducing MIN_AMPL in half but still no luck. I
assumed from the descriptions of each, that reducing the numbers would make
the algorithm less picky at finding a beep. Is this correct?

Any other recommendations on picking up these ~750Hz beeps? Thanks again for
the help.

--matt

On Mon, Aug 17, 2009 at 2:15 PM, Eric des Courtis <
eric.des.courtis at gmail.com> wrote:

> Matt,
>
> Okay the good news is vmd should be able to handle these cases. The
> bad news is for whatever reason they are not getting detected at the
> moment.
>
> vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
> MAX_FREQ to 1450 and play with MIN_AMPL  if that still doesn't help.


> The following seem to use the same beep:
>
> vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>
> You can try to play with these values:
>
> POINTS 32
> VALID 22
> MAX_CHIRP 22
>
> If that doesn't work let me know I will try to improve the algorithm
> to detect the providers.
>
> Cheers!
>
> Eric des Courtis
>
>
>
> On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong at gmail.com> wrote:
> > Hi Eric,
> > Thanks for the response. I had tried emailing you @brenbria.com and the
> > email had bounced, thanks for responding to my mail.
> > If you'd be interested I .zipped up my sample voicemail beeps
> > at http://bandcon.hellohunter.com/vmd_wav.zip
> > I'm relatively new to telephony, but can you point me in the right
> direction
> > for figuring out if the beeps are sinewaves. About as far as I've come
> with
> > audio is being able to open the .wav files in audacity. Any website
> >  recommendations I can read? Thanks so much.
> > --matt
> >
> > On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
> > <eric.des.courtis at gmail.com> wrote:
> >>
> >> Matt,
> >>
> >> You must first capture the audio beeps and verify that they are sine
> >> waves. If not, simply tweaking the algorithm will not give you better
> >> results.
> >>
> >> It might be possible to use FFT and I would be happy to help you
> >> implement such a solution but keep in mind FFT is very very demanding
> >> on the hardware. Ideally what you want to find out is what functions
> >> was use to generate the beep in the first place so that it can be
> >> detected. Is it two sines waves like in DTMF? Or something more
> >> complex?
> >>
> >> Anyway my email is eric.des.courtis at benbria.ca.
> >>
> >> Cheers.
> >>
> >> Eric des Courtis
> >>
> >> On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong at gmail.com>
> wrote:
> >> > I tried emailed Eric, seeking advice on this, but his email (the one
> in
> >> > the
> >> > source code) is bouncing email (invalid user), so thought I would ask
> >> > here
> >> > instead. If anyone has eric's new email address, I'd be interesting in
> >> > it.
> >> >
> >> > I did some tests with mod_vmd this afternoon, but I'm only finding
> about
> >> > 33%
> >> > of the voice mail beeps and did have 1 false-positive in my test of
> >> > 7 voice
> >> > mail machines. I've recorded the audio of the session in .wav files
> that
> >> > were both successful and not, as a comparison. I can upload the .wav
> >> > files
> >> > if they would be useful.
> >> > mod_vmd works great for voicemails of Skype Users, and kall8.com, but
> >> > has
> >> > issues dealing with mobile phone carriers.
> >> > sprint - not successful
> >> > tmobile - not successful
> >> > verizon - not successful
> >> > panasonic home answering machine system - not successful
> >> > kall8 - SUCCESS
> >> > skype - SUCCESS
> >> > I'm wondering if you can recommend a simple fix, like changing some of
> >> > the
> >> > constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c
> source
> >> > file, or if better success requires more complex analysis.  Do you
> have
> >> > any
> >> > recommendations on how this might be done? Listening to the .wav's
> >> > its apparent the beeps are not as loud for the mobile phone carriers
> as
> >> > they
> >> > are with skype and kall8. Any guidance would be greatly appreciated.
> >> > --matt
> >> > hello hunter
> >> > http://www.hellohunter.com
> >> > voice broadcasting & hosted dialer
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >
> >
>
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From rdenert at tng.de  Wed Aug 19 02:17:26 2009
From: rdenert at tng.de (Rudolf Denert)
Date: Wed, 19 Aug 2009 11:17:26 +0200 (CEST)
Subject: [Freeswitch-users] "mute" channel programmatically with
 mod_event_socket
In-Reply-To: <28651154.332401250673051662.JavaMail.root@zimbra.tng.de>
Message-ID: <17888468.332601250673446046.JavaMail.root@zimbra.tng.de>

You can use the caller controlls in the conference.conf.xml to implement your own features something like mute or kick. Or do you want to mute mute other conference members like a moderator can do this.

BR

----- Urspr?ngliche Mail -----
Von: "Artem Shiyanov" <shiyanov at gmail.com>
An: freeswitch-users at lists.freeswitch.org
Gesendet: Dienstag, 18. August 2009 14:54:04 GMT +01:00 Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
Betreff: [Freeswitch-users] "mute" channel programmatically with mod_event_socket


Hello all! 

I'm trying to implement "mute" feature with mod_event_socket: I want programmatically mute/unmute some channel in a call.. And I don't see any other ways except to use conference room with special rule "mute". 
Anybody knows the better way? 


Thanks, 
Artem 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



From shiyanov at gmail.com  Wed Aug 19 04:13:15 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Wed, 19 Aug 2009 15:13:15 +0400
Subject: [Freeswitch-users] "mute" channel programmatically with
	mod_event_socket
In-Reply-To: <17888468.332601250673446046.JavaMail.root@zimbra.tng.de>
References: <28651154.332401250673051662.JavaMail.root@zimbra.tng.de>
	<17888468.332601250673446046.JavaMail.root@zimbra.tng.de>
Message-ID: <d211f54a0908190413p17311b15u42f57dd3417ec634@mail.gmail.com>

The point is - a simple call flow is desired- if I have a ordinary 1-to-1
call and one of the participators decides to mute call - I don't want to put
both channels into a conference room but it looks like I have no other
choices.

BUT:
I found brilliant app - eavesdrop! If I do this for one-to-one call - mute
works!
SendMsg <uuid_channel_making_mute>
call-command: execute
execute-app-name: eavesdrop
execute-app-arg: <uuid_participator>

But the problem appears when I want to unmute.. the call! I've tried to
re-bridge channels, intercept them- nothing happens- one channel (muted one)
doesn't hear the participator. And CLI command 'show channels' shows that
channel with uui=<uuid_channel_making_mute> still process eavesdrop app.
Maybe someone know how to switch off eavesdrop app?


Artem


On Wed, Aug 19, 2009 at 1:17 PM, Rudolf Denert <rdenert at tng.de> wrote:

> You can use the caller controlls in the conference.conf.xml to implement
> your own features something like mute or kick. Or do you want to mute mute
> other conference members like a moderator can do this.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Artem Shiyanov" <shiyanov at gmail.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Dienstag, 18. August 2009 14:54:04 GMT +01:00
> Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
> Betreff: [Freeswitch-users] "mute" channel programmatically with
> mod_event_socket
>
>
> Hello all!
>
> I'm trying to implement "mute" feature with mod_event_socket: I want
> programmatically mute/unmute some channel in a call.. And I don't see any
> other ways except to use conference room with special rule "mute".
> Anybody knows the better way?
>
>
> Thanks,
> Artem
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From bruce.mcalister at blueface.ie  Wed Aug 19 05:53:05 2009
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Wed, 19 Aug 2009 13:53:05 +0100
Subject: [Freeswitch-users] Build Issue on Solaris 10 (FS v1.0.4pre9 &
 v1.0.4)
In-Reply-To: <4A891472.5060302@blueface.ie>
References: <4A8280B8.6050308@blueface.ie> <4A891472.5060302@blueface.ie>
Message-ID: <4A8BF5B1.2090801@blueface.ie>

Hi All,

JIRA FSBUILD-186 BugID has been logged for this issue.

Thanks
Bruce

Bruce McAlister wrote:
> Hi All,
> 
> Shall I log a JIRA for this issue?
> 
> Thanks
> Bruce
> 
> Bruce McAlister wrote:
>> Hi All,
>>
>> I have been having difficulty trying to build FreeSWITCH 1.0.4pre9 and 
>> 1.0.4.
>>
>> I am running on Solaris 10 Update 5 on x86 hardware (32-bit).
>>
>> The build fails with:
>>
>> --- snip ---
>> make: Fatal error: Command failed for target `all-recursive'
>> Current working directory /export/home/user/packages/BUILD/freeswitch-1.0.4
>> *** Error code 1
>> make: Fatal error: Command failed for target `all'
>> ---
>>
>> Looking back through the build I can see the following error:
>>
>> --- snip ---
>> creating libfreeswitch.la
>> (cd .libs && rm -f libfreeswitch.la && ln -s ../libfreeswitch.la 
>> libfreeswitch.la)
>> /usr/bin/cc 
>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
>> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
>> -DPATH_MAX=2048 -g -v -Xc -xc99=all -o .libs/freeswitch 
>> freeswitch-switch.o  ./.libs/libfreeswitch.so 
>> -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib 
>> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr-util/xml/expat/lib/.libs/libexpat.a 
>> /export/home/user/packages/BUILD/freeswitch-1.0.4/libs/apr/.libs/libapr-1.a 
>> -lm -L/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/srtp 
>> -L/usr/sfw/lib libs/apr/.libs/libapr-1.a -luuid -lsendfile -lrt 
>> -lpthread libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lnsl -ldl 
>> -lcurses -lsocket  -R/opt/freeswitch/lib -R/usr/sfw/lib
>> Undefined                       first referenced
>>   symbol                             in file
>> herror                              ./.libs/libfreeswitch.so
>> ld: fatal: Symbol referencing errors. No output written to .libs/freeswitch
>> *** Error code 1
>> The following command caused the error:
>> `if test -z "" ; then echo /bin/bash 
>> /export/home/user/packages/BUILD/freeswitch-1.0.4/quiet_libtool ;else 
>> echo /export/home/user/packages/BUILD/freeswitch-1.0.4/libtool; fi;` 
>> --tag=CC --mode=link /usr/bin/cc 
>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/src/include 
>> -I/export/home/user/packages/BUILD/freeswitch-1.0.4/libs/libteletone/src 
>> -KPIC -DPIC -erroff=E_END_OF_LOOP_CODE_NOT_REACHED -errtags=yes 
>> -DPATH_MAX=2048 -g -v -Xc -xc99=all  -lm -R/opt/freeswitch/lib  -o 
>> freeswitch -lm -R/opt/freeswitch/lib -rpath /opt/freeswitch/lib 
>> freeswitch-switch.o   libfreeswitch.la libs/apr/libapr-1.la 
>> libs/libedit/src/.libs/libedit.a -R/usr/sfw/lib -L/usr/sfw/lib -lssl 
>> -lcrypto -lsocket -lnsl -ldl   -lcurses -lsocket
>> --- snip ---
>>
>> Then a little above this error, there is the following warning that is 
>> displayed (I'm not sure if it is related):
>>
>> --- snip ---
>> *** Warning: Linking the shared library libfreeswitch.la against the
>> *** static library libs/libedit/src/.libs/libedit.a is not portable!
>> --- snip ---
>>
>> My configure line is as follows:
>>
>> ---
>> ./configure --prefix=/opt/freeswitch
>> ---
>>
>> I have the complete configure and make output if anyone needs them.
>>
>> Any help/pointers would be greatly appreciated.
>>
>> Thanks
>> Bruce
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 



From mrene_lists at avgs.ca  Wed Aug 19 06:02:19 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 19 Aug 2009 09:02:19 -0400
Subject: [Freeswitch-users] "mute" channel programmatically with
	mod_event_socket
In-Reply-To: <d211f54a0908190413p17311b15u42f57dd3417ec634@mail.gmail.com>
References: <28651154.332401250673051662.JavaMail.root@zimbra.tng.de>
	<17888468.332601250673446046.JavaMail.root@zimbra.tng.de>
	<d211f54a0908190413p17311b15u42f57dd3417ec634@mail.gmail.com>
Message-ID: <36B2A0E3-6139-4724-A7F3-CAD89FB98688@avgs.ca>

Hi,

Eavesdrop kind of works yeah, you can use the intercept application to  
re-bridge the channels together, like:

SendMsg <uuid_channel_making_mute>
call-command: execute
execute-app-name: intercept
execute-app-arg: <uuid_participator>


The same can be done with the uuid_bridge api.

api uuid_bridge <uuid1> <uuid2>

If you want the cleaner way you could implement a media bug that  
replaces all the audio of the channel by silence, but that'd require  
some C coding.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 19-Aug-09, at 7:13 AM, Artem Shiyanov wrote:

> The point is - a simple call flow is desired- if I have a ordinary 1- 
> to-1 call and one of the participators decides to mute call - I  
> don't want to put both channels into a conference room but it looks  
> like I have no other choices.
>
> BUT:
> I found brilliant app - eavesdrop! If I do this for one-to-one call  
> - mute works!
> SendMsg <uuid_channel_making_mute>
> call-command: execute
> execute-app-name: eavesdrop
> execute-app-arg: <uuid_participator>
>
> But the problem appears when I want to unmute.. the call! I've tried  
> to re-bridge channels, intercept them- nothing happens- one channel  
> (muted one) doesn't hear the participator. And CLI command 'show  
> channels' shows that channel with uui=<uuid_channel_making_mute>  
> still process eavesdrop app. Maybe someone know how to switch off  
> eavesdrop app?
>
>
> Artem
>
>
> On Wed, Aug 19, 2009 at 1:17 PM, Rudolf Denert <rdenert at tng.de> wrote:
> You can use the caller controlls in the conference.conf.xml to  
> implement your own features something like mute or kick. Or do you  
> want to mute mute other conference members like a moderator can do  
> this.
>
> BR
>
> ----- Urspr?ngliche Mail -----
> Von: "Artem Shiyanov" <shiyanov at gmail.com>
> An: freeswitch-users at lists.freeswitch.org
> Gesendet: Dienstag, 18. August 2009 14:54:04 GMT +01:00 Amsterdam/ 
> Berlin/Bern/Rom/Stockholm/Wien
> Betreff: [Freeswitch-users] "mute" channel programmatically with  
> mod_event_socket
>
>
> Hello all!
>
> I'm trying to implement "mute" feature with mod_event_socket: I want  
> programmatically mute/unmute some channel in a call.. And I don't  
> see any other ways except to use conference room with special rule  
> "mute".
> Anybody knows the better way?
>
>
> Thanks,
> Artem
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From ivanov.maxim at gmail.com  Wed Aug 19 02:31:30 2009
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Wed, 19 Aug 2009 13:31:30 +0400
Subject: [Freeswitch-users] playback to legB before bridge?
Message-ID: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>

Hi all!
Is it possible to execute  playback  application to legB before
bridge? I mean sequence of actions similar to this:

1. originate legB
2. playback
3. bridge legA with legB using existing session from step 1 (no new
incoming call)



From t.mahe at telemaque.fr  Wed Aug 19 06:23:42 2009
From: t.mahe at telemaque.fr (=?ISO-8859-1?Q?Tristan_Mah=E9?=)
Date: Wed, 19 Aug 2009 15:23:42 +0200
Subject: [Freeswitch-users] playback to legB before bridge?
In-Reply-To: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
References: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
Message-ID: <4A8BFCDE.4020700@telemaque.fr>

Hi Max,

Read the wiki, you'll find your answer

( here's a tip: http://wiki.freeswitch.org/wiki/Channel_Variables )

Have fun :)

Gled

Max Ivanov a ?crit :
> Hi all!
> Is it possible to execute  playback  application to legB before
> bridge? I mean sequence of actions similar to this:
>
> 1. originate legB
> 2. playback
> 3. bridge legA with legB using existing session from step 1 (no new
> incoming call)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From asannucci at gmail.com  Wed Aug 19 07:22:23 2009
From: asannucci at gmail.com (bakko)
Date: Wed, 19 Aug 2009 16:22:23 +0200
Subject: [Freeswitch-users] playback to legB before bridge?
In-Reply-To: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
References: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
Message-ID: <F0EB2D01BAF54938B619700CD0295D20@voztovoice>

http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_file

Look this variable.

I'd like know if exist similar variable for a-leg for use in the dialplan.

A call a number
A lissen a announce
A digit one or more DTMF 
The A channel is bridge with the B channel

Thank you in advance.

BR



From mrene_lists at avgs.ca  Wed Aug 19 07:29:04 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 19 Aug 2009 10:29:04 -0400
Subject: [Freeswitch-users] playback to legB before bridge?
In-Reply-To: <F0EB2D01BAF54938B619700CD0295D20@voztovoice>
References: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
	<F0EB2D01BAF54938B619700CD0295D20@voztovoice>
Message-ID: <34D37CC9-FE37-4B91-B82E-93E2312B5A1F@avgs.ca>

<action application="playback" data="file_here" />
<action application="bridge" data="dial string for b chan" />

or if you're talking about another B channel (I don't quite understand  
what you're trying to do here)
<action application="playback" data="file_here" />
<action application="intercept" data="<uuid here>" />

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 19-Aug-09, at 10:22 AM, bakko wrote:

> http://wiki.freeswitch.org/wiki/Channel_Variables#group_confirm_file
>
> Look this variable.
>
> I'd like know if exist similar variable for a-leg for use in the  
> dialplan.
>
> A call a number
> A lissen a announce
> A digit one or more DTMF
> The A channel is bridge with the B channel
>
> Thank you in advance.
>
> BR
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From asannucci at gmail.com  Wed Aug 19 07:48:12 2009
From: asannucci at gmail.com (bakko)
Date: Wed, 19 Aug 2009 16:48:12 +0200
Subject: [Freeswitch-users] playback to legB before bridge?
In-Reply-To: <34D37CC9-FE37-4B91-B82E-93E2312B5A1F@avgs.ca>
References: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com><F0EB2D01BAF54938B619700CD0295D20@voztovoice>
	<34D37CC9-FE37-4B91-B82E-93E2312B5A1F@avgs.ca>
Message-ID: <4EDE5C8143E7459AB02759B1AFFB7DC8@voztovoice>

before to bridge the call the a-leg (the caller) have to lissen a announce 
and then have to digit one or more dtmf.
If the dtmf are rights then bridge the caller with the callee, if not hangup 
the call.
I hope now my question is more clear :)
Sorry for my english.
BR 




From mrene_lists at avgs.ca  Wed Aug 19 07:51:05 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 19 Aug 2009 10:51:05 -0400
Subject: [Freeswitch-users] playback to legB before bridge?
In-Reply-To: <4EDE5C8143E7459AB02759B1AFFB7DC8@voztovoice>
References: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com><F0EB2D01BAF54938B619700CD0295D20@voztovoice>
	<34D37CC9-FE37-4B91-B82E-93E2312B5A1F@avgs.ca>
	<4EDE5C8143E7459AB02759B1AFFB7DC8@voztovoice>
Message-ID: <45F6928B-8DAE-4E6B-B801-D3909E0ABD58@avgs.ca>

Oh,

You can use embedded languages (like Lua or Javascript) to take  
control of the call and playback files / get dtmf digits, make another  
call, etc. Check on the wiki for samples.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 19-Aug-09, at 10:48 AM, bakko wrote:

> before to bridge the call the a-leg (the caller) have to lissen a  
> announce
> and then have to digit one or more dtmf.
> If the dtmf are rights then bridge the caller with the callee, if  
> not hangup
> the call.
> I hope now my question is more clear :)
> Sorry for my english.
> BR
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jmesquita at gmail.com  Wed Aug 19 09:27:17 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Wed, 19 Aug 2009 13:27:17 -0300
Subject: [Freeswitch-users] FreeSWITCH Console Released For iPhone/iPod
	Touch Users
In-Reply-To: <4A85A4E2.2030702@maxpowersoft.com>
References: <4A85A4E2.2030702@maxpowersoft.com>
Message-ID: <5a8712120908190927l15f83c6dm98d16952ad9daa37@mail.gmail.com>

No one said a thing but I really feel like initiatives like this
should be cheered.
Thank you for this app and thank you for making it free as well.

On 8/14/09, Chris Danielson <chris at maxpowersoft.com> wrote:
> Announcing the release of FreeSWITCH Console in the Apple Application
> Store.  The application is FREE and allows you to connect to a
> FreeSWITCH event socket layer module that is bound to an external
> interface.  Great for development purposes and general remote debugging.
>
> Blog announcement:
> http://www.chrisdanielson.com/2009/08/14/release-iphoneipod-touch-freeswitch-console/
>
> iTunes Store Link:
> http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221&mt=8
> <http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221&mt=8>
>
> Kind Regards,
> Chris Danielson
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device



From msc at freeswitch.org  Wed Aug 19 09:36:01 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Aug 2009 09:36:01 -0700
Subject: [Freeswitch-users] how to set different action for different
	cause code
In-Reply-To: <a44688a0908180923q4b4b8e48od86d676872c97c3a@mail.gmail.com>
References: <a44688a0908160424n3569cef1vad8fcdc68cbfb96e@mail.gmail.com>
	<87f2f3b90908170859p52259449q913b2ddab224709f@mail.gmail.com>
	<a44688a0908180923q4b4b8e48od86d676872c97c3a@mail.gmail.com>
Message-ID: <87f2f3b90908190936i50c12104vc252599fe2caf051@mail.gmail.com>

On Tue, Aug 18, 2009 at 9:23 AM, Woody Dickson <woodydickson at gmail.com>wrote:

> Hi,
>
> I have my dialplan to do some simple routing. What I need to do is when
> certain hangup code is received, route advance to the next or the route
> after next based on the hangup code received.
>
> So, I have:
>
> <condition>
>    <action application="set" data="hangup_after_bridge=true"/>
>    <action application="set" data="continue_on_fail="false"/>
>    <action application="bridge" data="sofia/internal/aaa at 123.3.3./>
>    <action application="bridge" data="sofia/internal/bbb at 123.3.3./>
>    <action application="bridge" data="sofia/internal/ccc at 123.3.3./>
>    <action application="bridge" data="sofia/internal/ddd at 123.3.3./>
>
> </condition>
>
> What I want to do is that based on the error code received from ( bridge to
> aaa), it can route to either bbb or ccc depends on the hangup code received
> from aaa.
>
> So there anyway to do it in the dialplan without any scripting?


Yes, you can use the result of the previous bridge to execute a transfer.
You'd need to use the hangup cause as the transfer destination, or part of
the destination. For example, you could try something like this:

   <action application="set" data="hangup_after_bridge=true"/>
   <action application="set" data="continue_on_fail="false"/>
   <action application="bridge" data="sofia/internal/aaa at 123.3.3./>
   <action application="transfer" data="${hangup_cause}_AAA XML default"/>
...

And then have an extension for each of the various hangup_causes
<extension name="normal clearing">
  <condition field="destination_number"
expression="^(normal_clearing_AAA)$">
    <!-- Do whatever you want for "normal clearing" bridged calls -->
  </condition>
</extension>

<extension name="user busy">
  <condition field="destination_number" expression="^(user_busy_AAA)$">
    <!-- Do whatever you want for "user busy" bridged calls -->
  </condition>
</extension>

<extension name="no answer">
  <condition field="destination_number" expression="^(no_answer_AAA)$">
    <!-- Do whatever you want for "no answer" bridged calls -->
  </condition>
</extension>

Etc., etc.

I don't claim it's pretty, but it would let you avoid the use of scripting
languages. However, I would caution you to examine just why you want/need to
avoid scripting langs. The dp-only method may indeed be best for you, or
perhaps using mod_xml_curl would be better. It does depend on your specific
needs and your environment.

-MC


>
> thannks,
> woody
>
>
>
>
>
> On Mon, Aug 17, 2009 at 11:59 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Sun, Aug 16, 2009 at 4:24 AM, Woody Dickson <woodydickson at gmail.com>wrote:
>>
>>> Hello,
>>>
>>> I find hangup_hook, but I would like to define different actions for
>>> different hangup codes.  Is there anyway to do that?
>>>
>>>
>>
>> I can think of at least two ways you could do this: one that uses only the
>> dialplan and one that uses a script. If you don't mind using a scripting
>> language then you can make it very clean:
>>
>> <action application="set" data="api_hangup_hook=luarun chancleanup.lua
>> ${hangup_cause}"/>
>>
>> Then have your Lua script handle all the if-then-else or case stuff.
>>
>> Question: are you trying to transfer the a-leg to some other destination
>> if the b-leg hangup is a specific cause, or are you just doing some external
>> cleanup stuff? Just curious...
>>
>> -MC
>>
>>
>>> Woody
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From klaus.teller at gmx.net  Wed Aug 19 09:39:41 2009
From: klaus.teller at gmx.net (Klaus Teller)
Date: Wed, 19 Aug 2009 18:39:41 +0200
Subject: [Freeswitch-users] Passing Caller ID in Junction Network
Message-ID: <20090819163941.156100@gmx.net>

Hi,

Anybody with experience in passing caller ID number in junction network? I need some help.

I have the following origination string:

{origination_caller_id_number=1514xxxxxx,ignore_early_media=true}sofia/gateway/sip.jnctn.net/1514xxxxxx

Unfortunately, with this, junctionetwork won't pass the caller ID. Any idea?

Thanks,
Klaus.
-- 
Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 -
sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser



From msc at freeswitch.org  Wed Aug 19 09:41:16 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Aug 2009 09:41:16 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>
Message-ID: <87f2f3b90908190941m49c030e4w747e053e76f9b818@mail.gmail.com>

On Wed, Aug 19, 2009 at 1:43 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> Hi Eric,
> Thanks for these recommendations.
>
> for vmd-not-panasonic-home-ans.wav changing MAX_FREQ to 1450 WORKED!
>
> but I'm still having problems picking out the ~750Hz beep of sprint,
> tmobile, and verizon. I tried first cutting POINTS and VALID in half, then
> in half again, while also reducing MIN_AMPL in half but still no luck. I
> assumed from the descriptions of each, that reducing the numbers would make
> the algorithm less picky at finding a beep. Is this correct?
>
> Any other recommendations on picking up these ~750Hz beeps? Thanks again
> for the help.
>

How close are they to 750Hz? If they're not more than say +/- 16Hz then the
tone_detect app *should* be able to detect them. At the very least I would
try it. See if tone_detect can detect those beeps. While it may not be the
most elegant solution, having mod_vmd looking for one set of tones and
tone_detect looking for the 750Hz tones might actually get the job done, at
least until you and Eric can get together to see what's happening on the
750's.

-MC


>
> --matt
>
> On Mon, Aug 17, 2009 at 2:15 PM, Eric des Courtis <
> eric.des.courtis at gmail.com> wrote:
>
>> Matt,
>>
>> Okay the good news is vmd should be able to handle these cases. The
>> bad news is for whatever reason they are not getting detected at the
>> moment.
>>
>> vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
>> MAX_FREQ to 1450 and play with MIN_AMPL  if that still doesn't help.
>
>
>> The following seem to use the same beep:
>>
>> vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
>> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>>
>> You can try to play with these values:
>>
>> POINTS 32
>> VALID 22
>> MAX_CHIRP 22
>>
>> If that doesn't work let me know I will try to improve the algorithm
>> to detect the providers.
>>
>> Cheers!
>>
>> Eric des Courtis
>>
>>
>>
>> On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong at gmail.com> wrote:
>> > Hi Eric,
>> > Thanks for the response. I had tried emailing you @brenbria.com and the
>> > email had bounced, thanks for responding to my mail.
>> > If you'd be interested I .zipped up my sample voicemail beeps
>> > at http://bandcon.hellohunter.com/vmd_wav.zip
>> > I'm relatively new to telephony, but can you point me in the right
>> direction
>> > for figuring out if the beeps are sinewaves. About as far as I've come
>> with
>> > audio is being able to open the .wav files in audacity. Any website
>> >  recommendations I can read? Thanks so much.
>> > --matt
>> >
>> > On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
>> > <eric.des.courtis at gmail.com> wrote:
>> >>
>> >> Matt,
>> >>
>> >> You must first capture the audio beeps and verify that they are sine
>> >> waves. If not, simply tweaking the algorithm will not give you better
>> >> results.
>> >>
>> >> It might be possible to use FFT and I would be happy to help you
>> >> implement such a solution but keep in mind FFT is very very demanding
>> >> on the hardware. Ideally what you want to find out is what functions
>> >> was use to generate the beep in the first place so that it can be
>> >> detected. Is it two sines waves like in DTMF? Or something more
>> >> complex?
>> >>
>> >> Anyway my email is eric.des.courtis at benbria.ca.
>> >>
>> >> Cheers.
>> >>
>> >> Eric des Courtis
>> >>
>> >> On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong at gmail.com>
>> wrote:
>> >> > I tried emailed Eric, seeking advice on this, but his email (the one
>> in
>> >> > the
>> >> > source code) is bouncing email (invalid user), so thought I would ask
>> >> > here
>> >> > instead. If anyone has eric's new email address, I'd be interesting
>> in
>> >> > it.
>> >> >
>> >> > I did some tests with mod_vmd this afternoon, but I'm only finding
>> about
>> >> > 33%
>> >> > of the voice mail beeps and did have 1 false-positive in my test of
>> >> > 7 voice
>> >> > mail machines. I've recorded the audio of the session in .wav files
>> that
>> >> > were both successful and not, as a comparison. I can upload the .wav
>> >> > files
>> >> > if they would be useful.
>> >> > mod_vmd works great for voicemails of Skype Users, and kall8.com,
>> but
>> >> > has
>> >> > issues dealing with mobile phone carriers.
>> >> > sprint - not successful
>> >> > tmobile - not successful
>> >> > verizon - not successful
>> >> > panasonic home answering machine system - not successful
>> >> > kall8 - SUCCESS
>> >> > skype - SUCCESS
>> >> > I'm wondering if you can recommend a simple fix, like changing some
>> of
>> >> > the
>> >> > constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c
>> source
>> >> > file, or if better success requires more complex analysis.  Do you
>> have
>> >> > any
>> >> > recommendations on how this might be done? Listening to the .wav's
>> >> > its apparent the beeps are not as loud for the mobile phone carriers
>> as
>> >> > they
>> >> > are with skype and kall8. Any guidance would be greatly appreciated.
>> >> > --matt
>> >> > hello hunter
>> >> > http://www.hellohunter.com
>> >> > voice broadcasting & hosted dialer
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >
>> >
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Aug 19 09:49:24 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Aug 2009 09:49:24 -0700
Subject: [Freeswitch-users] Passing Caller ID in Junction Network
In-Reply-To: <20090819163941.156100@gmx.net>
References: <20090819163941.156100@gmx.net>
Message-ID: <87f2f3b90908190949q33751864l495e463769585b7d@mail.gmail.com>

On Wed, Aug 19, 2009 at 9:39 AM, Klaus Teller <klaus.teller at gmx.net> wrote:

> Hi,
>
> Anybody with experience in passing caller ID number in junction network? I
> need some help.
>
> I have the following origination string:
>
>
> {origination_caller_id_number=1514xxxxxx,ignore_early_media=true}sofia/gateway/
> sip.jnctn.net/1514xxxxxx
>
> Unfortunately, with this, junctionetwork won't pass the caller ID. Any
> idea?


Does Junction Networks pass any caller ID info at all?  Also, are you sure
that they support customized caller ID sending?
-MC


>
> Thanks,
> Klaus.
> --
> Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla Firefox 3 -
> sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Wed Aug 19 09:50:54 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 Aug 2009 11:50:54 -0500
Subject: [Freeswitch-users] Passing Caller ID in Junction Network
In-Reply-To: <20090819163941.156100@gmx.net>
References: <20090819163941.156100@gmx.net>
Message-ID: <89755F1E-97E4-4ECA-8A6D-9D1B92B062F1@freeswitch.org>

Can you show me a sip invite of an inbound call from them?  and an  
invite you're sending to them?

/b

On Aug 19, 2009, at 11:39 AM, Klaus Teller wrote:

> Hi,
>
> Anybody with experience in passing caller ID number in junction  
> network? I need some help.
>
> I have the following origination string:
>
> {origination_caller_id_number 
> =1514xxxxxx,ignore_early_media=true}sofia/gateway/sip.jnctn.net/ 
> 1514xxxxxx
>
> Unfortunately, with this, junctionetwork won't pass the caller ID.  
> Any idea?
>
> Thanks,
> Klaus.




From mrene_lists at avgs.ca  Wed Aug 19 09:46:59 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Wed, 19 Aug 2009 12:46:59 -0400
Subject: [Freeswitch-users] Passing Caller ID in Junction Network
In-Reply-To: <20090819163941.156100@gmx.net>
References: <20090819163941.156100@gmx.net>
Message-ID: <8B74E348-B4C4-4AA0-B3A5-B0AA395DCDFF@avgs.ca>

Hi,

I would send a sip trace to your carrier and/or ask them how they like  
to receive caller id information.

PS: Are you in Montreal?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 19-Aug-09, at 12:39 PM, Klaus Teller wrote:

> Hi,
>
> Anybody with experience in passing caller ID number in junction  
> network? I need some help.
>
> I have the following origination string:
>
> {origination_caller_id_number 
> =1514xxxxxx,ignore_early_media=true}sofia/gateway/sip.jnctn.net/ 
> 1514xxxxxx
>
> Unfortunately, with this, junctionetwork won't pass the caller ID.  
> Any idea?
>
> Thanks,
> Klaus.
> -- 
> Jetzt kostenlos herunterladen: Internet Explorer 8 und Mozilla  
> Firefox 3 -
> sicherer, schneller und einfacher! http://portal.gmx.net/de/go/chbrowser
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From chris at maxpowersoft.com  Wed Aug 19 10:04:26 2009
From: chris at maxpowersoft.com (Chris Danielson)
Date: Wed, 19 Aug 2009 10:04:26 -0700
Subject: [Freeswitch-users] FreeSWITCH Console Released For iPhone/iPod
 Touch Users
In-Reply-To: <5a8712120908190927l15f83c6dm98d16952ad9daa37@mail.gmail.com>
References: <4A85A4E2.2030702@maxpowersoft.com>
	<5a8712120908190927l15f83c6dm98d16952ad9daa37@mail.gmail.com>
Message-ID: <4A8C309A.2060405@maxpowersoft.com>

Thanks!

Jo?o Mesquita wrote:
> No one said a thing but I really feel like initiatives like this
> should be cheered.
> Thank you for this app and thank you for making it free as well.
>
> On 8/14/09, Chris Danielson <chris at maxpowersoft.com> wrote:
>   
>> Announcing the release of FreeSWITCH Console in the Apple Application
>> Store.  The application is FREE and allows you to connect to a
>> FreeSWITCH event socket layer module that is bound to an external
>> interface.  Great for development purposes and general remote debugging.
>>
>> Blog announcement:
>> http://www.chrisdanielson.com/2009/08/14/release-iphoneipod-touch-freeswitch-console/
>>
>> iTunes Store Link:
>> http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221&mt=8
>> <http://itunes.apple.com/WebObjects/MZStore.woa/wa/viewSoftware?id=319105221&mt=8>
>>
>> Kind Regards,
>> Chris Danielson
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>     
>
>   
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From mike at jerris.com  Wed Aug 19 10:26:19 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 19 Aug 2009 13:26:19 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <87f2f3b90908190941m49c030e4w747e053e76f9b818@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>
	<87f2f3b90908190941m49c030e4w747e053e76f9b818@mail.gmail.com>
Message-ID: <8E7379D0-FB06-4C93-95B4-C493F8355A88@jerris.com>

my bet is if mod_vmd is not getting them that they are not going to  
work with tone detect either.  Someone needs to look at the tone and  
see what frequencies are really involved and if they change throughout  
the beep.

Mike

On Aug 19, 2009, at 12:41 PM, Michael Collins wrote:

>
>
> On Wed, Aug 19, 2009 at 1:43 AM, Matthew Fong <mattdfong at gmail.com>  
> wrote:
> Hi Eric,
>
> Thanks for these recommendations.
>
> for vmd-not-panasonic-home-ans.wav changing MAX_FREQ to 1450 WORKED!
>
> but I'm still having problems picking out the ~750Hz beep of sprint,  
> tmobile, and verizon. I tried first cutting POINTS and VALID in  
> half, then in half again, while also reducing MIN_AMPL in half but  
> still no luck. I assumed from the descriptions of each, that  
> reducing the numbers would make the algorithm less picky at finding  
> a beep. Is this correct?
>
> Any other recommendations on picking up these ~750Hz beeps? Thanks  
> again for the help.
>
> How close are they to 750Hz? If they're not more than say +/- 16Hz  
> then the tone_detect app *should* be able to detect them. At the  
> very least I would try it. See if tone_detect can detect those  
> beeps. While it may not be the most elegant solution, having mod_vmd  
> looking for one set of tones and tone_detect looking for the 750Hz  
> tones might actually get the job done, at least until you and Eric  
> can get together to see what's happening on the 750's.
>
> -MC
>
>
> --matt
>
> On Mon, Aug 17, 2009 at 2:15 PM, Eric des Courtis <eric.des.courtis at gmail.com 
> > wrote:
> Matt,
>
> Okay the good news is vmd should be able to handle these cases. The
> bad news is for whatever reason they are not getting detected at the
> moment.
>
> vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
> MAX_FREQ to 1450 and play with MIN_AMPL  if that still doesn't help.
>
> The following seem to use the same beep:
>
> vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>
> You can try to play with these values:
>
> POINTS 32
> VALID 22
> MAX_CHIRP 22
>
> If that doesn't work let me know I will try to improve the algorithm
> to detect the providers.
>
> Cheers!
>
> Eric des Courtis
>
>
>
> On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong at gmail.com>  
> wrote:
> > Hi Eric,
> > Thanks for the response. I had tried emailing you @brenbria.com  
> and the
> > email had bounced, thanks for responding to my mail.
> > If you'd be interested I .zipped up my sample voicemail beeps
> > at http://bandcon.hellohunter.com/vmd_wav.zip
> > I'm relatively new to telephony, but can you point me in the right  
> direction
> > for figuring out if the beeps are sinewaves. About as far as I've  
> come with
> > audio is being able to open the .wav files in audacity. Any website
> >  recommendations I can read? Thanks so much.
> > --matt
> >
> > On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
> > <eric.des.courtis at gmail.com> wrote:
> >>
> >> Matt,
> >>
> >> You must first capture the audio beeps and verify that they are  
> sine
> >> waves. If not, simply tweaking the algorithm will not give you  
> better
> >> results.
> >>
> >> It might be possible to use FFT and I would be happy to help you
> >> implement such a solution but keep in mind FFT is very very  
> demanding
> >> on the hardware. Ideally what you want to find out is what  
> functions
> >> was use to generate the beep in the first place so that it can be
> >> detected. Is it two sines waves like in DTMF? Or something more
> >> complex?
> >>
> >> Anyway my email is eric.des.courtis at benbria.ca.
> >>
> >> Cheers.
> >>
> >> Eric des Courtis
> >>
> >> On Mon, Aug 17, 2009 at 12:26 AM, Matthew  
> Fong<mattdfong at gmail.com> wrote:
> >> > I tried emailed Eric, seeking advice on this, but his email  
> (the one in
> >> > the
> >> > source code) is bouncing email (invalid user), so thought I  
> would ask
> >> > here
> >> > instead. If anyone has eric's new email address, I'd be  
> interesting in
> >> > it.
> >> >
> >> > I did some tests with mod_vmd this afternoon, but I'm only  
> finding about
> >> > 33%
> >> > of the voice mail beeps and did have 1 false-positive in my  
> test of
> >> > 7 voice
> >> > mail machines. I've recorded the audio of the session in .wav  
> files that
> >> > were both successful and not, as a comparison. I can upload  
> the .wav
> >> > files
> >> > if they would be useful.
> >> > mod_vmd works great for voicemails of Skype Users, and  
> kall8.com, but
> >> > has
> >> > issues dealing with mobile phone carriers.
> >> > sprint - not successful
> >> > tmobile - not successful
> >> > verizon - not successful
> >> > panasonic home answering machine system - not successful
> >> > kall8 - SUCCESS
> >> > skype - SUCCESS
> >> > I'm wondering if you can recommend a simple fix, like changing  
> some of
> >> > the
> >> > constants like MAX_FREQ, or MIN_TIME at the top of the  
> mod_vmd.c source
> >> > file, or if better success requires more complex analysis.  Do  
> you have
> >> > any
> >> > recommendations on how this might be done? Listening to  
> the .wav's
> >> > its apparent the beeps are not as loud for the mobile phone  
> carriers as
> >> > they
> >> > are with skype and kall8. Any guidance would be greatly  
> appreciated.
> >> > --matt
> >> > hello hunter
> >> > http://www.hellohunter.com
> >> > voice broadcasting & hosted dialer
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Wed Aug 19 10:30:04 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 19 Aug 2009 13:30:04 -0400
Subject: [Freeswitch-users] playback to legB before bridge?
In-Reply-To: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
References: <fbfa7b0c0908190231u4e889cf6yd5c788662d95b7dc@mail.gmail.com>
Message-ID: <78ACC57E-6AD6-4026-A715-ABB8D0F4D162@jerris.com>

you can export nolocal the execute_on_answer var to do this.  Set it  
to playback the file.

Mike

On Aug 19, 2009, at 5:31 AM, Max Ivanov wrote:

> Hi all!
> Is it possible to execute  playback  application to legB before
> bridge? I mean sequence of actions similar to this:
>
> 1. originate legB
> 2. playback
> 3. bridge legA with legB using existing session from step 1 (no new
> incoming call)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From dist.lists at gmail.com  Wed Aug 19 11:11:05 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Wed, 19 Aug 2009 21:11:05 +0300
Subject: [Freeswitch-users] custom avp in radius
Message-ID: <2a73afe0908191111h528d0a6dod257df4384318c90@mail.gmail.com>

Hi,

Does anyone know of a way to add custom information in AVP in radius
accounting messages? I can see in the radius dictionary there are many
AVPs defined, but I'd like to add some additional custom text
information. "Freeswitch-AVPair" seems like a generic AVP, but I
cannot figure out how to use it (if it's intended for something like
that at all).

Maybe there is a way to set some variable in the dial plan and to use
its contents to populate a general purpose AVP?

If there is no such functionality what will be the best way to do it?
I found similar feature added in the past
(http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=7529) and it
doesn't seem too complicated.

BR



From max.bridgewater at gmail.com  Wed Aug 19 11:14:36 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Wed, 19 Aug 2009 14:14:36 -0400
Subject: [Freeswitch-users] No Dialplan on answered channel,
	changing state to HANGUP
Message-ID: <c55d46950908191114q67f90333m4ccd53320989f965@mail.gmail.com>

Hi,

I'm trying to use les.net for termination. When i place a call from the
command line, i get the following error message:  No Dialplan on answered
channel, changing state to HANGUP. What does it mean?

My dialing string is as simple as it could be: sofia/gateway/
did.voip.les.net/13028838864 & park()
My les.net profile is located under
/usr/loca/freeswitch/conf/sip_profiles/external


<include>
  <gateway name="did.voip.les.net">
    <param name="username" value="1690xxxxxx"/>
    <param name="from-user" value="1690xxxxxx"/>
    <param name="realm" value="did.voip.les.net"/>
    <param name="password" value="ma34n2th./an4"/>
    <param name="caller-id-in-from" value="true"/>
  </gateway>
</include>

Regards,
Max.
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From brian at freeswitch.org  Wed Aug 19 11:19:01 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 19 Aug 2009 13:19:01 -0500
Subject: [Freeswitch-users] No Dialplan on answered channel,
	changing state to HANGUP
In-Reply-To: <c55d46950908191114q67f90333m4ccd53320989f965@mail.gmail.com>
References: <c55d46950908191114q67f90333m4ccd53320989f965@mail.gmail.com>
Message-ID: <17EBEEA2-A63B-44F2-84BC-CC946AC69C31@freeswitch.org>

you don't put a space after the &

/b

On Aug 19, 2009, at 1:14 PM, Max Bridgewater wrote:

> My dialing string is as simple as it could be: sofia/gateway/ 
> did.voip.les.net/13028838864 & park()

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From klaus.teller at gmx.net  Wed Aug 19 11:38:18 2009
From: klaus.teller at gmx.net (Klaus Teller)
Date: Wed, 19 Aug 2009 20:38:18 +0200
Subject: [Freeswitch-users] Passing Caller ID in Junction Network
In-Reply-To: <89755F1E-97E4-4ECA-8A6D-9D1B92B062F1@freeswitch.org>
References: <20090819163941.156100@gmx.net>
	<89755F1E-97E4-4ECA-8A6D-9D1B92B062F1@freeswitch.org>
Message-ID: <20090819183818.156130@gmx.net>

Hi Brian,

Please find my SIP trace attached. I called the Junction Networks guys and they suggest that they pass the caller id number only.

Could you confirm that everything is fine on my side so i can send the same to them?

Thanks so much,
Klaus.


-------- Original-Nachricht --------
> Datum: Wed, 19 Aug 2009 11:50:54 -0500
> Von: Brian West <brian at freeswitch.org>
> An: freeswitch-users at lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] Passing Caller ID in Junction Network

> Can you show me a sip invite of an inbound call from them?  and an  
> invite you're sending to them?
> 
> /b
> 
> On Aug 19, 2009, at 11:39 AM, Klaus Teller wrote:
> 
> > Hi,
> >
> > Anybody with experience in passing caller ID number in junction  
> > network? I need some help.
> >
> > I have the following origination string:
> >
> > {origination_caller_id_number 
> > =1514xxxxxx,ignore_early_media=true}sofia/gateway/sip.jnctn.net/ 
> > 1514xxxxxx
> >
> > Unfortunately, with this, junctionetwork won't pass the caller ID.  
> > Any idea?
> >
> > Thanks,
> > Klaus.
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-- 
GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT!
Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01
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From matt at hellohunter.com  Wed Aug 19 11:50:02 2009
From: matt at hellohunter.com (Matt Hunter)
Date: Wed, 19 Aug 2009 11:50:02 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <8E7379D0-FB06-4C93-95B4-C493F8355A88@jerris.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>
	<87f2f3b90908190941m49c030e4w747e053e76f9b818@mail.gmail.com>
	<8E7379D0-FB06-4C93-95B4-C493F8355A88@jerris.com>
Message-ID: <4256bf830908191150v5c01a829s7b5138d902f7e9e2@mail.gmail.com>

Can anyone recommend a tool to analyze the wave files to see what's causing
the sine wav not to be detected? I have them zipped at
http://bandcon.hellohunter.com/vmd_wav.zip
<http://bandcon.hellohunter.com/vmd_wav.zip>I was trying to use audacity,
but not sure how to tell the exact frequency.

--matt

On Wed, Aug 19, 2009 at 10:26 AM, Michael Jerris <mike at jerris.com> wrote:

> my bet is if mod_vmd is not getting them that they are not going to work
> with tone detect either.  Someone needs to look at the tone and see what
> frequencies are really involved and if they change throughout the beep.
> Mike
>
> On Aug 19, 2009, at 12:41 PM, Michael Collins wrote:
>
>
>
> On Wed, Aug 19, 2009 at 1:43 AM, Matthew Fong <mattdfong at gmail.com> wrote:
>
>> Hi Eric,
>> Thanks for these recommendations.
>>
>> for vmd-not-panasonic-home-ans.wav changing MAX_FREQ to 1450 WORKED!
>>
>> but I'm still having problems picking out the ~750Hz beep of sprint,
>> tmobile, and verizon. I tried first cutting POINTS and VALID in half, then
>> in half again, while also reducing MIN_AMPL in half but still no luck. I
>> assumed from the descriptions of each, that reducing the numbers would make
>> the algorithm less picky at finding a beep. Is this correct?
>>
>> Any other recommendations on picking up these ~750Hz beeps? Thanks again
>> for the help.
>>
>
> How close are they to 750Hz? If they're not more than say +/- 16Hz then the
> tone_detect app *should* be able to detect them. At the very least I would
> try it. See if tone_detect can detect those beeps. While it may not be the
> most elegant solution, having mod_vmd looking for one set of tones and
> tone_detect looking for the 750Hz tones might actually get the job done, at
> least until you and Eric can get together to see what's happening on the
> 750's.
>
> -MC
>
>
>>
>> --matt
>>
>> On Mon, Aug 17, 2009 at 2:15 PM, Eric des Courtis <
>> eric.des.courtis at gmail.com> wrote:
>>
>>> Matt,
>>>
>>> Okay the good news is vmd should be able to handle these cases. The
>>> bad news is for whatever reason they are not getting detected at the
>>> moment.
>>>
>>> vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
>>> MAX_FREQ to 1450 and play with MIN_AMPL  if that still doesn't help.
>>
>>
>>> The following seem to use the same beep:
>>>
>>> vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
>>> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>>> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>>>
>>> You can try to play with these values:
>>>
>>> POINTS 32
>>> VALID 22
>>> MAX_CHIRP 22
>>>
>>> If that doesn't work let me know I will try to improve the algorithm
>>> to detect the providers.
>>>
>>> Cheers!
>>>
>>> Eric des Courtis
>>>
>>>
>>>
>>> On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong at gmail.com>
>>> wrote:
>>> > Hi Eric,
>>> > Thanks for the response. I had tried emailing you @brenbria.com and
>>> the
>>> > email had bounced, thanks for responding to my mail.
>>> > If you'd be interested I .zipped up my sample voicemail beeps
>>> > at http://bandcon.hellohunter.com/vmd_wav.zip
>>> > I'm relatively new to telephony, but can you point me in the right
>>> direction
>>> > for figuring out if the beeps are sinewaves. About as far as I've come
>>> with
>>> > audio is being able to open the .wav files in audacity. Any website
>>> >  recommendations I can read? Thanks so much.
>>> > --matt
>>> >
>>> > On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
>>> > <eric.des.courtis at gmail.com> wrote:
>>> >>
>>> >> Matt,
>>> >>
>>> >> You must first capture the audio beeps and verify that they are sine
>>> >> waves. If not, simply tweaking the algorithm will not give you better
>>> >> results.
>>> >>
>>> >> It might be possible to use FFT and I would be happy to help you
>>> >> implement such a solution but keep in mind FFT is very very demanding
>>> >> on the hardware. Ideally what you want to find out is what functions
>>> >> was use to generate the beep in the first place so that it can be
>>> >> detected. Is it two sines waves like in DTMF? Or something more
>>> >> complex?
>>> >>
>>> >> Anyway my email is eric.des.courtis at benbria.ca.
>>> >>
>>> >> Cheers.
>>> >>
>>> >> Eric des Courtis
>>> >>
>>> >> On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong at gmail.com>
>>> wrote:
>>> >> > I tried emailed Eric, seeking advice on this, but his email (the one
>>> in
>>> >> > the
>>> >> > source code) is bouncing email (invalid user), so thought I would
>>> ask
>>> >> > here
>>> >> > instead. If anyone has eric's new email address, I'd be interesting
>>> in
>>> >> > it.
>>> >> >
>>> >> > I did some tests with mod_vmd this afternoon, but I'm only finding
>>> about
>>> >> > 33%
>>> >> > of the voice mail beeps and did have 1 false-positive in my test of
>>> >> > 7 voice
>>> >> > mail machines. I've recorded the audio of the session in .wav files
>>> that
>>> >> > were both successful and not, as a comparison. I can upload the .wav
>>> >> > files
>>> >> > if they would be useful.
>>> >> > mod_vmd works great for voicemails of Skype Users, and kall8.com,
>>> but
>>> >> > has
>>> >> > issues dealing with mobile phone carriers.
>>> >> > sprint - not successful
>>> >> > tmobile - not successful
>>> >> > verizon - not successful
>>> >> > panasonic home answering machine system - not successful
>>> >> > kall8 - SUCCESS
>>> >> > skype - SUCCESS
>>> >> > I'm wondering if you can recommend a simple fix, like changing some
>>> of
>>> >> > the
>>> >> > constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c
>>> source
>>> >> > file, or if better success requires more complex analysis.  Do you
>>> have
>>> >> > any
>>> >> > recommendations on how this might be done? Listening to the .wav's
>>> >> > its apparent the beeps are not as loud for the mobile phone carriers
>>> as
>>> >> > they
>>> >> > are with skype and kall8. Any guidance would be greatly appreciated.
>>> >> > --matt
>>> >> > hello hunter
>>> >> > http://www.hellohunter.com
>>> >> > voice broadcasting & hosted dialer
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >
>>> >
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mattdfong at gmail.com  Wed Aug 19 11:50:34 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Wed, 19 Aug 2009 11:50:34 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <8E7379D0-FB06-4C93-95B4-C493F8355A88@jerris.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>
	<87f2f3b90908190941m49c030e4w747e053e76f9b818@mail.gmail.com>
	<8E7379D0-FB06-4C93-95B4-C493F8355A88@jerris.com>
Message-ID: <4256bf830908191150x479ba3b7i35b78053c8b6123c@mail.gmail.com>

Can anyone recommend a tool to analyze the wave files to see what's causing
the sine wav not to be detected? I have them zipped at
http://bandcon.hellohunter.com/vmd_wav.zip
<http://bandcon.hellohunter.com/vmd_wav.zip>I was trying to use audacity,
but not sure how to tell the exact frequency.

sorry if this was a double post...

On Wed, Aug 19, 2009 at 10:26 AM, Michael Jerris <mike at jerris.com> wrote:

> my bet is if mod_vmd is not getting them that they are not going to work
> with tone detect either.  Someone needs to look at the tone and see what
> frequencies are really involved and if they change throughout the beep.
> Mike
>
> On Aug 19, 2009, at 12:41 PM, Michael Collins wrote:
>
>
>
> On Wed, Aug 19, 2009 at 1:43 AM, Matthew Fong <mattdfong at gmail.com> wrote:
>
>> Hi Eric,
>> Thanks for these recommendations.
>>
>> for vmd-not-panasonic-home-ans.wav changing MAX_FREQ to 1450 WORKED!
>>
>> but I'm still having problems picking out the ~750Hz beep of sprint,
>> tmobile, and verizon. I tried first cutting POINTS and VALID in half, then
>> in half again, while also reducing MIN_AMPL in half but still no luck. I
>> assumed from the descriptions of each, that reducing the numbers would make
>> the algorithm less picky at finding a beep. Is this correct?
>>
>> Any other recommendations on picking up these ~750Hz beeps? Thanks again
>> for the help.
>>
>
> How close are they to 750Hz? If they're not more than say +/- 16Hz then the
> tone_detect app *should* be able to detect them. At the very least I would
> try it. See if tone_detect can detect those beeps. While it may not be the
> most elegant solution, having mod_vmd looking for one set of tones and
> tone_detect looking for the 750Hz tones might actually get the job done, at
> least until you and Eric can get together to see what's happening on the
> 750's.
>
> -MC
>
>
>>
>> --matt
>>
>> On Mon, Aug 17, 2009 at 2:15 PM, Eric des Courtis <
>> eric.des.courtis at gmail.com> wrote:
>>
>>> Matt,
>>>
>>> Okay the good news is vmd should be able to handle these cases. The
>>> bad news is for whatever reason they are not getting detected at the
>>> moment.
>>>
>>> vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
>>> MAX_FREQ to 1450 and play with MIN_AMPL  if that still doesn't help.
>>
>>
>>> The following seem to use the same beep:
>>>
>>> vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
>>> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>>> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>>>
>>> You can try to play with these values:
>>>
>>> POINTS 32
>>> VALID 22
>>> MAX_CHIRP 22
>>>
>>> If that doesn't work let me know I will try to improve the algorithm
>>> to detect the providers.
>>>
>>> Cheers!
>>>
>>> Eric des Courtis
>>>
>>>
>>>
>>> On Mon, Aug 17, 2009 at 1:52 PM, Matthew Fong<mattdfong at gmail.com>
>>> wrote:
>>> > Hi Eric,
>>> > Thanks for the response. I had tried emailing you @brenbria.com and
>>> the
>>> > email had bounced, thanks for responding to my mail.
>>> > If you'd be interested I .zipped up my sample voicemail beeps
>>> > at http://bandcon.hellohunter.com/vmd_wav.zip
>>> > I'm relatively new to telephony, but can you point me in the right
>>> direction
>>> > for figuring out if the beeps are sinewaves. About as far as I've come
>>> with
>>> > audio is being able to open the .wav files in audacity. Any website
>>> >  recommendations I can read? Thanks so much.
>>> > --matt
>>> >
>>> > On Mon, Aug 17, 2009 at 10:29 AM, Eric des Courtis
>>> > <eric.des.courtis at gmail.com> wrote:
>>> >>
>>> >> Matt,
>>> >>
>>> >> You must first capture the audio beeps and verify that they are sine
>>> >> waves. If not, simply tweaking the algorithm will not give you better
>>> >> results.
>>> >>
>>> >> It might be possible to use FFT and I would be happy to help you
>>> >> implement such a solution but keep in mind FFT is very very demanding
>>> >> on the hardware. Ideally what you want to find out is what functions
>>> >> was use to generate the beep in the first place so that it can be
>>> >> detected. Is it two sines waves like in DTMF? Or something more
>>> >> complex?
>>> >>
>>> >> Anyway my email is eric.des.courtis at benbria.ca.
>>> >>
>>> >> Cheers.
>>> >>
>>> >> Eric des Courtis
>>> >>
>>> >> On Mon, Aug 17, 2009 at 12:26 AM, Matthew Fong<mattdfong at gmail.com>
>>> wrote:
>>> >> > I tried emailed Eric, seeking advice on this, but his email (the one
>>> in
>>> >> > the
>>> >> > source code) is bouncing email (invalid user), so thought I would
>>> ask
>>> >> > here
>>> >> > instead. If anyone has eric's new email address, I'd be interesting
>>> in
>>> >> > it.
>>> >> >
>>> >> > I did some tests with mod_vmd this afternoon, but I'm only finding
>>> about
>>> >> > 33%
>>> >> > of the voice mail beeps and did have 1 false-positive in my test of
>>> >> > 7 voice
>>> >> > mail machines. I've recorded the audio of the session in .wav files
>>> that
>>> >> > were both successful and not, as a comparison. I can upload the .wav
>>> >> > files
>>> >> > if they would be useful.
>>> >> > mod_vmd works great for voicemails of Skype Users, and kall8.com,
>>> but
>>> >> > has
>>> >> > issues dealing with mobile phone carriers.
>>> >> > sprint - not successful
>>> >> > tmobile - not successful
>>> >> > verizon - not successful
>>> >> > panasonic home answering machine system - not successful
>>> >> > kall8 - SUCCESS
>>> >> > skype - SUCCESS
>>> >> > I'm wondering if you can recommend a simple fix, like changing some
>>> of
>>> >> > the
>>> >> > constants like MAX_FREQ, or MIN_TIME at the top of the mod_vmd.c
>>> source
>>> >> > file, or if better success requires more complex analysis.  Do you
>>> have
>>> >> > any
>>> >> > recommendations on how this might be done? Listening to the .wav's
>>> >> > its apparent the beeps are not as loud for the mobile phone carriers
>>> as
>>> >> > they
>>> >> > are with skype and kall8. Any guidance would be greatly appreciated.
>>> >> > --matt
>>> >> > hello hunter
>>> >> > http://www.hellohunter.com
>>> >> > voice broadcasting & hosted dialer
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >
>>> >
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From woof at iwoof.org  Wed Aug 19 12:38:57 2009
From: woof at iwoof.org (Andy Spitzer)
Date: Wed, 19 Aug 2009 15:38:57 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4256bf830908191150x479ba3b7i35b78053c8b6123c@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4256bf830908190143i51b663efm4ccf9edd82889439@mail.gmail.com>
	<87f2f3b90908190941m49c030e4w747e053e76f9b818@mail.gmail.com>
	<8E7379D0-FB06-4C93-95B4-C493F8355A88@jerris.com>
	<4256bf830908191150x479ba3b7i35b78053c8b6123c@mail.gmail.com>
Message-ID: <op.uyxcu7rdg0dksq@woof3.fuzzy>

On Wed, 19 Aug 2009 14:50:34 -0400, Matthew Fong <mattdfong at gmail.com> wrote:

> I was trying to use audacity, but not sure how to tell the exact  
> frequency.

Audacity can do it.  Highlight the "beep" (and nothing but the beep) with the selection cursor, then click "Analyze->Plot Spectrum..."  In the "Frequency Analysis" window that opens, place the cursor on the center of the largest peak.  Near the base of that window it will show the Cursor and Peak frequencies.  For example, in the "vmd-not-panasonic-home-ans.wav" file, the "beep" is 1400 Hz (for a very short 1 second).  Then later it switches to FAX CED tone (2100 Hz) and FAX preamble.

--Woof!





From steveu at coppice.org  Wed Aug 19 13:22:57 2009
From: steveu at coppice.org (Steve Underwood)
Date: Thu, 20 Aug 2009 04:22:57 +0800
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
Message-ID: <4A8C5F21.8080401@coppice.org>

On 08/18/2009 05:15 AM, Eric des Courtis wrote:
> Matt,
>
> Okay the good news is vmd should be able to handle these cases. The
> bad news is for whatever reason they are not getting detected at the
> moment.
>
> vmd-not-panasonic-home-ans.wav is a sine at ~1400Hz you can change
> MAX_FREQ to 1450 and play with MIN_AMPL if that still doesn't help.
>
> The following seem to use the same beep:
>
> vmd-not-tmobile.wav is a sine at ~750Hz but has a bit of noise
> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
> vmd-not-sprint.wav is a sine at ~750Hz but has a bit of noise
>
> You can try to play with these values:
>
> POINTS 32
> VALID 22
> MAX_CHIRP 22
>
> If that doesn't work let me know I will try to improve the algorithm
> to detect the providers.
>    
There is no noise on those 3 beeps. In fact, for something that's been 
through ulaw/alaw compression those beeps are very clean. They are quite 
short, though.

Steve




From msc at freeswitch.org  Wed Aug 19 14:22:37 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 19 Aug 2009 14:22:37 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4A8C5F21.8080401@coppice.org>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
Message-ID: <87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>

> There is no noise on those 3 beeps. In fact, for something that's been
> through ulaw/alaw compression those beeps are very clean. They are quite
> short, though.
>

Heck yeah they're short! Steve, in your experience is there a practical way
to detect a beep that short without chewing up system resources or having
lots of false positives?
-MC
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From intralanman at freeswitch.org  Wed Aug 19 19:27:05 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Wed, 19 Aug 2009 22:27:05 -0400
Subject: [Freeswitch-users] Passing Caller ID in Junction Network
In-Reply-To: <20090819183818.156130@gmx.net>
References: <20090819163941.156100@gmx.net>
	<89755F1E-97E4-4ECA-8A6D-9D1B92B062F1@freeswitch.org>
	<20090819183818.156130@gmx.net>
Message-ID: <CE062C17-16E6-4474-87FC-29A0CFB5A7CF@freeswitch.org>

I think i remember needing to send the callerid in the From: when  
using them, try throwing
<param name="caller-id-in-from" value="true"/>
on the gateway and restarting the profile to see if it works after that

Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     

On Aug 19, 2009, at 2:38 PM, Klaus Teller wrote:

> Hi Brian,
>
> Please find my SIP trace attached. I called the Junction Networks  
> guys and they suggest that they pass the caller id number only.
>
> Could you confirm that everything is fine on my side so i can send  
> the same to them?
>
> Thanks so much,
> Klaus.
>
>
> -------- Original-Nachricht --------
>> Datum: Wed, 19 Aug 2009 11:50:54 -0500
>> Von: Brian West <brian at freeswitch.org>
>> An: freeswitch-users at lists.freeswitch.org
>> Betreff: Re: [Freeswitch-users] Passing Caller ID in Junction Network
>
>> Can you show me a sip invite of an inbound call from them?  and an
>> invite you're sending to them?
>>
>> /b
>>
>> On Aug 19, 2009, at 11:39 AM, Klaus Teller wrote:
>>
>>> Hi,
>>>
>>> Anybody with experience in passing caller ID number in junction
>>> network? I need some help.
>>>
>>> I have the following origination string:
>>>
>>> {origination_caller_id_number
>>> =1514xxxxxx,ignore_early_media=true}sofia/gateway/sip.jnctn.net/
>>> 1514xxxxxx
>>>
>>> Unfortunately, with this, junctionetwork won't pass the caller ID.
>>> Any idea?
>>>
>>> Thanks,
>>> Klaus.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> -- 
> GRATIS f?r alle GMX-Mitglieder: Die maxdome Movie-FLAT!
> Jetzt freischalten unter http://portal.gmx.net/de/go/maxdome01
> <etherXXXXFyLDxJ.cap>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From dist.lists at gmail.com  Wed Aug 19 23:43:13 2009
From: dist.lists at gmail.com (Hristo Trendev)
Date: Thu, 20 Aug 2009 09:43:13 +0300
Subject: [Freeswitch-users] custom avp in radius
In-Reply-To: <2a73afe0908191111h528d0a6dod257df4384318c90@mail.gmail.com>
References: <2a73afe0908191111h528d0a6dod257df4384318c90@mail.gmail.com>
Message-ID: <2a73afe0908192343n7c0313d6g87e650c5ecd1a457@mail.gmail.com>

I believe, I'm looking for a radius equivalent of the "userfield"
(http://wiki.freeswitch.org/wiki/Mod_cdr_csv#userfield)

br

On Wed, Aug 19, 2009 at 9:11 PM, Hristo Trendev<dist.lists at gmail.com> wrote:
> Hi,
>
> Does anyone know of a way to add custom information in AVP in radius
> accounting messages? I can see in the radius dictionary there are many
> AVPs defined, but I'd like to add some additional custom text
> information. "Freeswitch-AVPair" seems like a generic AVP, but I
> cannot figure out how to use it (if it's intended for something like
> that at all).
>
> Maybe there is a way to set some variable in the dial plan and to use
> its contents to populate a general purpose AVP?
>
> If there is no such functionality what will be the best way to do it?
> I found similar feature added in the past
> (http://fisheye.freeswitch.org/changelog/FreeSWITCH/?cs=7529) and it
> doesn't seem too complicated.
>
> BR
>



From ken at expitrans.com  Wed Aug 19 23:51:53 2009
From: ken at expitrans.com (Kenneth Shaw)
Date: Wed, 19 Aug 2009 23:51:53 -0700
Subject: [Freeswitch-users] Sharing Presence Information Across Separate
	Offices
Message-ID: <20090820065153.ac3a26cf@kms.expitrans.com>

I apologize in advance if this message duplicates a previous post or the topic has been covered elsewhere. If that's the case, a pointer would be appreciated!

I have multiple offices each running Freeswitch and a number of Polycom phones and softphones.

I would like to share the presence information between the offices, such that if a person in Office B can see if a person in Office A is on the phone. I personally don't know of any way to do this through Freeswitch as it stands. I have thought about using a VPN for the offices and just setting up the phones to all register to a single server, however I don't want calls between phones in the same office (intra-office) to have to be routed all the way through a different office, which may be on the other side of the world and costs bandwidth.

Any suggestions as to the best way to accomplish this would be greatly appreciated. Thanks!

--
Kenneth Shaw
ExpiTrans, Inc.
129 W. Wilson St., Suite 204
Costa Mesa, CA 92627
tel: 949.650.4600
fax: 949.642.6044
ken at expitrans.com
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From mattdfong at gmail.com  Thu Aug 20 00:29:08 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Thu, 20 Aug 2009 00:29:08 -0700
Subject: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with same
	Gateway
Message-ID: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>

I'm trying to get FreeSWITCH to bridge two channels together through the
same external gateway, but I'm having issues hearing audio. Both legs if
setup independently and forwarded to 5000 (test ivr) work fine for both
inbound and outbound media, but when I try to bridge them together,
everything seems fine in FreeSWITCH, but neither party can hear the other
speak. I'm thinking the external gateway might be having some issues because
I've been able to bridge 2 channels together through the same gateway on
different providers, but thought I'd also try to seek some help here.
FreeSWITCH should be handling the media for both channels, so I can't figure
out why if Leg A and Leg B work independently, but not if they are bridged
together. Is there a setting somewhere in FS that I'm missing?
Below is a ngrep of the SIP interactions if it's useful. Thanks for the
help.

--matt

interface: eth0 (172.24.200.0/255.255.255.0)
filter: (ip or ip6) and ( port 5060 )

U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060
INVITE sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>SIP/2.0.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Max-Forwards: 70.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>>.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 293.
Remote-Party-ID: "FreeSWITCH"
<sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
s=FreeSWITCH.
c=IN IP4 216.81.56.198.
t=0 0.
m=audio 24700 RTP/AVP 0 8 3 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 100 Trying.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Contact: <sip:914159927717 at 38.98.58.148:5060>.
Content-Length: 0.
.


U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 183 Session Progress.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Contact: <sip:914159927717 at 38.98.58.148:5060>.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Content-Type: application/sdp.
Content-Length: 227.
.
v=0.
o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
s=M6 Call.
c=IN IP4 38.98.58.148.
t=0 0.
m=audio 42554 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=rtcp:6461 IN IP4 10.10.24.50.


U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 200 OK.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
Contact: <sip:914159927717 at 38.98.58.148:5060>.
Session-Expires: 1800;refresher=uas.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Supported: timer.
Content-Type: application/sdp.
Content-Length: 227.
.
v=0.
o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
s=M6 Call.
c=IN IP4 38.98.58.148.
t=0 0.
m=audio 42554 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=rtcp:6461 IN IP4 10.10.24.50.


U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060
ACK sip:914159927717 at 38.98.58.148:5060 SIP/2.0.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc.
Max-Forwards: 70.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>;tag=F725.2C49.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 119257811 ACK.
Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
Content-Length: 0.
.


U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060
INVITE sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>SIP/2.0.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
Max-Forwards: 70.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>>.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257815 INVITE.
Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 293.
Remote-Party-ID: "FreeSWITCH"
<sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198.
s=FreeSWITCH.
c=IN IP4 216.81.56.198.
t=0 0.
m=audio 24798 RTP/AVP 0 8 3 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.


U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 100 Trying.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>;tag=F72E.2D4E.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257815 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
Contact: <sip:914154650027 at 38.98.58.148:5060>.
Content-Length: 0.
.


U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 183 Session Progress.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>;tag=F72E.2D4E.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257815 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
Contact: <sip:914154650027 at 38.98.58.148:5060>.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Content-Type: application/sdp.
Content-Length: 225.
.
v=0.
o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
s=M6 Call.
c=IN IP4 38.98.58.148.
t=0 0.
m=audio 46520 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=rtcp:6451 IN IP4 10.10.24.50.


U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 200 OK.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>;tag=F72E.2D4E.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257815 INVITE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
Contact: <sip:914154650027 at 38.98.58.148:5060>.
Session-Expires: 1800;refresher=uas.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Supported: timer.
Content-Type: application/sdp.
Content-Length: 225.
.
v=0.
o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
s=M6 Call.
c=IN IP4 38.98.58.148.
t=0 0.
m=audio 46520 RTP/AVP 0 101.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-15.
a=ptime:20.
a=sendrecv.
a=rtcp:6451 IN IP4 10.10.24.50.


U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060
ACK sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK.
Max-Forwards: 70.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>;tag=F72E.2D4E.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257815 ACK.
Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
Content-Length: 0.
.


U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080
BYE sip:gw+epik.com at 216.81.56.198:5080 SIP/2.0.
From: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>;tag=F725.2C49.
To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 4817 BYE.
Max-Forwards: 70.
Via: SIP/2.0/UDP 38.98.58.148:5060
;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
Contact: <sip:914159927717 at 38.98.58.148:5060>.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Content-Length: 0.
.


U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 38.98.58.148:5060
;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
From: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>;tag=F725.2C49.
To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=ZtFvjeFQmXvpp.
Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
CSeq: 4817 BYE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Content-Length: 0.
.


U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060
BYE sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
Max-Forwards: 70.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>;tag=F72E.2D4E.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257816 BYE.
Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY,
REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Reason: Q.850;cause=16;text="NORMAL_CLEARING".
Content-Length: 0.
.


U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080
SIP/2.0 200 OK.
From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>;tag=038mm9ZtH6j9H.
To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>;tag=F72E.2D4E.
Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
CSeq: 119257816 BYE.
Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
Contact: <sip:914154650027 at 38.98.58.148:5060>.
Allow:
INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
Content-Length: 0.
.
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From a.afzali2003 at gmail.com  Thu Aug 20 01:29:19 2009
From: a.afzali2003 at gmail.com (afshin afzali)
Date: Thu, 20 Aug 2009 11:59:19 +0330
Subject: [Freeswitch-users] Sharing Presence Information Across Separate
	Offices
In-Reply-To: <20090820065153.ac3a26cf@kms.expitrans.com>
References: <20090820065153.ac3a26cf@kms.expitrans.com>
Message-ID: <debff5a80908200129q60d91a76v395781da1aabd661@mail.gmail.com>

Hi Kenneth,

I'm not going to answer your question! Instead I would like to emphasize on
the thing you are going to achieve because some times ago I've post this
question in some other title but unfortunately did not get any answer. As
the SIP protocol's point of view you should be able to subscribe every
FreeSWITCH machines for external events in another one (or probably in a
central PRESENCE server such as SER) to get NOTIFY messages which indicate
the information you wish to have. Although the sofia SIP endpoint can do
this (as you can find in its features) I could not find any guidelines for
it in FreeSWITCH.

Regards,
-- Afshin Afzali

On Thu, Aug 20, 2009 at 10:21 AM, Kenneth Shaw <ken at expitrans.com> wrote:

>  I apologize in advance if this message duplicates a previous post or the
> topic has been covered elsewhere. If that's the case, a pointer would be
> appreciated!
>
> I have multiple offices each running Freeswitch and a number of Polycom
> phones and softphones.
>
> I would like to share the presence information between the offices, such
> that if a person in Office B can see if a person in Office A is on the
> phone. I personally don't know of any way to do this through Freeswitch as
> it stands. I have thought about using a VPN for the offices and just setting
> up the phones to all register to a single server, however I don't want calls
> between phones in the same office (intra-office) to have to be routed all
> the way through a different office, which may be on the other side of the
> world and costs bandwidth.
>
> Any suggestions as to the best way to accomplish this would be greatly
> appreciated. Thanks!
>
> --
> Kenneth Shaw
> ExpiTrans, Inc.
> 129 W. Wilson St., Suite 204
> Costa Mesa, CA 92627
> tel: 949.650.4600
> fax: 949.642.6044
> ken at expitrans.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dome at tel.co.th  Thu Aug 20 03:42:37 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 20 Aug 2009 17:42:37 +0700
Subject: [Freeswitch-users] FS and SS7
Message-ID: <8ccbff060908200342p53474f58m645159a191db0364@mail.gmail.com>

Dear All,
            my telco can provider  SS7 (Now i use ISDN PRI).   I have
question about SS7.
        1. Can i setup SS7 with out ss7box ?
        2. I found some tutorial about libss7 in Asterisk. Is FS
support all feature like that.


Best Regards.

Dome C.



From mrene_lists at avgs.ca  Thu Aug 20 06:00:52 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 20 Aug 2009 09:00:52 -0400
Subject: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with
	same Gateway
In-Reply-To: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>
References: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>
Message-ID: <6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>

Hi

How are you bridging the calls in FS (which api call or C function are  
you using)?

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:

> I'm trying to get FreeSWITCH to bridge two channels together through  
> the same external gateway, but I'm having issues hearing audio. Both  
> legs if setup independently and forwarded to 5000 (test ivr) work  
> fine for both inbound and outbound media, but when I try to bridge  
> them together, everything seems fine in FreeSWITCH, but neither  
> party can hear the other speak. I'm thinking the external gateway  
> might be having some issues because I've been able to bridge 2  
> channels together through the same gateway on different providers,  
> but thought I'd also try to seek some help here. FreeSWITCH should  
> be handling the media for both channels, so I can't figure out why  
> if Leg A and Leg B work independently, but not if they are bridged  
> together. Is there a setting somewhere in FS that I'm missing?
>
> Below is a ngrep of the SIP interactions if it's useful. Thanks for  
> the help.
>
> --matt
>
> interface: eth0 (172.24.200.0/255.255.255.0)
> filter: (ip or ip6) and ( port 5060 )
>
> U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060
> INVITE sip:914159927717 at 38.98.58.148 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148>.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 293.
> Remote-Party-ID: "FreeSWITCH" <sip: 
> 0000000000 at 216.81.56.198>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
> s=FreeSWITCH.
> c=IN IP4 216.81.56.198.
> t=0 0.
> m=audio 24700 RTP/AVP 0 8 3 101 13.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
>
>
> U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 100 Trying.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 183 Session Progress.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Allow:  
> INVITE 
> ,BYE 
> ,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Type: application/sdp.
> Content-Length: 227.
> .
> v=0.
> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 42554 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6461 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 200 OK.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Session-Expires: 1800;refresher=uas.
> Allow:  
> INVITE 
> ,BYE 
> ,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Supported: timer.
> Content-Type: application/sdp.
> Content-Length: 227.
> .
> v=0.
> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 42554 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6461 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060
> ACK sip:914159927717 at 38.98.58.148:5060 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 ACK.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060
> INVITE sip:914154650027 at 38.98.58.148 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 293.
> Remote-Party-ID: "FreeSWITCH" <sip: 
> 0000000000 at 216.81.56.198>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198.
> s=FreeSWITCH.
> c=IN IP4 216.81.56.198.
> t=0 0.
> m=audio 24798 RTP/AVP 0 8 3 101 13.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
>
>
> U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 100 Trying.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 183 Session Progress.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Allow:  
> INVITE 
> ,BYE 
> ,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Type: application/sdp.
> Content-Length: 225.
> .
> v=0.
> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 46520 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6451 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 200 OK.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Session-Expires: 1800;refresher=uas.
> Allow:  
> INVITE 
> ,BYE 
> ,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Supported: timer.
> Content-Type: application/sdp.
> Content-Length: 225.
> .
> v=0.
> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 46520 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6451 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060
> ACK sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 ACK.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080
> BYE sip:gw+epik.com at 216.81.56.198:5080 SIP/2.0.
> From: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 4817 BYE.
> Max-Forwards: 70.
> Via: SIP/2.0/UDP  
> 38.98.58.148:5060;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Allow:  
> INVITE 
> ,BYE 
> ,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP  
> 38.98.58.148:5060;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
> From: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 4817 BYE.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060
> BYE sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257816 BYE.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,  
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 200 OK.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257816 BYE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Allow:  
> INVITE 
> ,BYE 
> ,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Length: 0.
> .
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Thu Aug 20 06:19:50 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 08:19:50 -0500
Subject: [Freeswitch-users] FS and SS7
In-Reply-To: <8ccbff060908200342p53474f58m645159a191db0364@mail.gmail.com>
References: <8ccbff060908200342p53474f58m645159a191db0364@mail.gmail.com>
Message-ID: <D7C4EE10-6998-4FDC-BCC5-E3180F3157A8@freeswitch.org>


On Aug 20, 2009, at 5:42 AM, Dome Charoenyost wrote:

>        1. Can i setup SS7 with out ss7box ?

Sangoma's ss7box is the only solution right now thats turn key.

>        2. I found some tutorial about libss7 in Asterisk. Is FS
> support all feature like that.

You can't really use libss7 in FreeSWITCH as its not license  
compatible... neither is OpenSS7.




From matt at venturevoip.com  Thu Aug 20 02:35:31 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Thu, 20 Aug 2009 21:35:31 +1200
Subject: [Freeswitch-users] Recommendations for an interface between
	FreeSwitch and ANSI C code
Message-ID: <4A8D18E3.9090302@venturevoip.com>

Hi,

We have an application written in ANSI C which currently talks to the 
Asterisk Manager to make phone calls.

We're possibly looking at converting this to FreeSwitch.

At the moment it has an abstraction layer from Asterisk and speaks to 
between 1 and 80 Asterisk machines using round robin for distribution.

Would you recommend that I use the mod_event socket and basically work 
with FreeSwitch in the same way as I work with Asterisk or am I 
overlooking a possibly different way of doing things?

Cheers,

Matt Riddell



From mrene_lists at avgs.ca  Thu Aug 20 06:26:34 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 20 Aug 2009 09:26:34 -0400
Subject: [Freeswitch-users] Recommendations for an interface between
	FreeSwitch and ANSI C code
In-Reply-To: <4A8D18E3.9090302@venturevoip.com>
References: <4A8D18E3.9090302@venturevoip.com>
Message-ID: <85BE2921-4C4F-4FCA-B4BE-2D7C6A88B3FD@avgs.ca>

Hi,

I would recommand mod_event_socket, also look at libs/esl (event  
socket library), a C library that takes care of all the parsing to  
talk to mod_event_socket.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 20-Aug-09, at 5:35 AM, Matt Riddell wrote:

> Hi,
>
> We have an application written in ANSI C which currently talks to the
> Asterisk Manager to make phone calls.
>
> We're possibly looking at converting this to FreeSwitch.
>
> At the moment it has an abstraction layer from Asterisk and speaks to
> between 1 and 80 Asterisk machines using round robin for distribution.
>
> Would you recommend that I use the mod_event socket and basically work
> with FreeSwitch in the same way as I work with Asterisk or am I
> overlooking a possibly different way of doing things?
>
> Cheers,
>
> Matt Riddell
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Thu Aug 20 06:26:37 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 08:26:37 -0500
Subject: [Freeswitch-users] Recommendations for an interface between
	FreeSwitch and ANSI C code
In-Reply-To: <4A8D18E3.9090302@venturevoip.com>
References: <4A8D18E3.9090302@venturevoip.com>
Message-ID: <03AAD9B3-EF35-4615-892F-2A3354409A3E@freeswitch.org>

Matt,
	Event Socket would be perfect for your needs.  If you run into any  
troubles don't hesitate to join #freeswitch or ask on the list to help  
you get on the right track.

Thanks,
/b

On Aug 20, 2009, at 4:35 AM, Matt Riddell wrote:

> Would you recommend that I use the mod_event socket and basically work
> with FreeSwitch in the same way as I work with Asterisk or am I
> overlooking a possibly different way of doing things?




From matt at venturevoip.com  Thu Aug 20 06:32:05 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Fri, 21 Aug 2009 01:32:05 +1200
Subject: [Freeswitch-users] Recommendations for an interface
 between	FreeSwitch and ANSI C code
In-Reply-To: <85BE2921-4C4F-4FCA-B4BE-2D7C6A88B3FD@avgs.ca>
References: <4A8D18E3.9090302@venturevoip.com>
	<85BE2921-4C4F-4FCA-B4BE-2D7C6A88B3FD@avgs.ca>
Message-ID: <4A8D5055.9050801@venturevoip.com>

On 21/08/09 1:26 AM, Mathieu Rene wrote:
> Hi,
>
> I would recommand mod_event_socket, also look at libs/esl (event
> socket library), a C library that takes care of all the parsing to
> talk to mod_event_socket.

Awesome thanks man :)

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)



From matt at venturevoip.com  Thu Aug 20 06:32:20 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Fri, 21 Aug 2009 01:32:20 +1200
Subject: [Freeswitch-users] Recommendations for an interface
 between	FreeSwitch and ANSI C code
In-Reply-To: <03AAD9B3-EF35-4615-892F-2A3354409A3E@freeswitch.org>
References: <4A8D18E3.9090302@venturevoip.com>
	<03AAD9B3-EF35-4615-892F-2A3354409A3E@freeswitch.org>
Message-ID: <4A8D5064.60507@venturevoip.com>

On 21/08/09 1:26 AM, Brian West wrote:
> Matt,
> 	Event Socket would be perfect for your needs.  If you run into any
> troubles don't hesitate to join #freeswitch or ask on the list to help
> you get on the right track.

Cheers, will do!

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)



From Prometheus001 at gmx.net  Thu Aug 20 06:35:46 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 20 Aug 2009 15:35:46 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
Message-ID: <4A8D5132.7010807@gmx.net>

Hello,

when calling from Fritzbox to a Snom Phone , sound is fine. But when
calling an internal Freeswitch number (conference, mailbox) i hear a
very choppy voice coming from the fritzbox side. I think it may have to
do with the ptime 20msec/30msec.

Example: When calling from the fritzbox to a voicemail then the
annoucement from Freeswitch is choppy (too slow with interrups), but the
recorded message is fine.

Did anybody experience the same problem?

Best regards
Peter

Here are some SIP messages:
U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060
INVITE sip:0123456 at my.domain SIP/2.0.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 55 INVITE.
Contact:
<sip:02xxxxxxxxx at 112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3>.
Max-Forwards: 70.
Expires: 120.
User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009).
Supported: 100rel,replaces,timer.
Allow-Events: telephone-event,refer.
Allow:
INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH.
Content-Type: application/sdp.
Accept: application/sdp, multipart/mixed.
Accept-Encoding: identity.
Content-Length:   359.
.
v=0.
o=user 16423733 16423733 IN IP4 112.xxx.xx.xxx.
s=call.
c=IN IP4 112.xxx.xx.xxx.
t=0 0.
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101.
a=sendrecv.
a=rtpmap:2 G726-32/8000.
a=rtpmap:102 G726-32/8000.
a=rtpmap:100 G726-40/8000.
a=rtpmap:99 G726-24/8000.
a=rtpmap:97 iLBC/8000.
a=fmtp:97.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
a=rtcp:7079.

#
U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 55 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552.
Content-Length: 0.
.

#
U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060
SIP/2.0 407 Proxy Authentication Required.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>;tag=7t1e8BQg5B7yK.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 55 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Proxy-Authenticate: Digest realm="my.domain",
nonce="900b46a0-8d88-11de-a6a1-098738f35adb", algorithm=MD5, qop="auth".
Content-Length: 0.
.

#
U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060
ACK sip:0123456 at my.domain SIP/2.0.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK22AF53A2ECD8BEAD.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>;tag=7t1e8BQg5B7yK.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 55 ACK.
User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009).
Content-Length: 0.
.

#
U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060
INVITE sip:0123456 at my.domain SIP/2.0.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 56 INVITE.
Contact:
<sip:02xxxxxxxxx at 112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3>.
Proxy-Authorization: Digest username="02xxxxxxxxx", realm="my.domain",
nonce="900b46a0-8d88-11de-a6a1-098738f35adb", uri="sip:0123456 at my.domain",
 response="276b44e261c13bd17218adff1150f414", algorithm=MD5,
cnonce="CADBE5D624516E8A", qop=auth, nc=00000001.
Max-Forwards: 70.
Expires: 120.
User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009).
Supported: 100rel,replaces,timer.
Allow-Events: telephone-event,refer.
Allow:
INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE,PUBLISH.
Content-Type: application/sdp.
Accept: application/sdp, multipart/mixed.
Accept-Encoding: identity.
Content-Length:   359.
.
v=0.
o=user 16423733 16423733 IN IP4 112.xxx.xx.xxx.
s=call.
c=IN IP4 112.xxx.xx.xxx.
t=0 0.
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101.
a=sendrecv.
a=rtpmap:2 G726-32/8000.
a=rtpmap:102 G726-32/8000.
a=rtpmap:100 G726-40/8000.
a=rtpmap:99 G726-24/8000.
a=rtpmap:97 iLBC/8000.
a=fmtp:97 mode=30.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
a=rtcp:7079.

#
U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060
SIP/2.0 100 Trying.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 56 INVITE.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552.
Content-Length: 0.
.

#
U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP 112.xxx.xx.xxx:5060;branch=z9hG4bK7CB99A256497FBB0.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>;tag=83t7967K2mXHF.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 56 INVITE.
Contact: <sip:0123456 at 182.xxx.xx.xxx:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552.
Accept: application/sdp.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 249.
.
v=0.
o=FreeSWITCH 1250746184 1250746185 IN IP4 182.xxx.xx.xxx.
s=FreeSWITCH.
c=IN IP4 182.xxx.xx.xxx.
t=0 0.
m=audio 26670 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:20.

#
U 112.xxx.xx.xxx:5060 -> 182.xxx.xx.xxx:5060
ACK sip:0123456 at 182.xxx.xx.xxx:5060;transport=udp SIP/2.0.
Via: SIP/2.0/udp 112.xxx.xx.xxx:5060;branch=z9hG4bK39E0B6E237E408FC.
From: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
To: <sip:0123456 at my.domain>;tag=83t7967K2mXHF.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 56 ACK.
Contact:
<sip:02xxxxxxxxx at 112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3>.
Max-Forwards: 70.
User-Agent: AVM FRITZ!Box Fon WLAN 7170 29.04.70 (Feb 18 2009).
Content-Length: 0.
.

#
U 182.xxx.xx.xxx:5060 -> 112.xxx.xx.xxx:5060
INVITE sip:02xxxxxxxxx at 112.xxx.xx.xxx;uniq=2006157A4333690041A4B78E67BC3
SIP/2.0.
Via: SIP/2.0/UDP 182.xxx.xx.xxx;rport;branch=z9hG4bK7rpNeaN67y9ta.
Max-Forwards: 70.
From: <sip:0123456 at my.domain>;tag=83t7967K2mXHF.
To: <sip:02xxxxxxxxx at my.domain>;tag=9A806878F0882CFC.
Call-ID: 3125316C8A2A13B8 at 112.xxx.xx.xxx.
CSeq: 119268091 INVITE.
Contact: <sip:0123456 at 182.xxx.xx.xxx:5060;transport=udp>.
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14552.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO.
Supported: timer, precondition, path, replaces.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 249.
.
v=0.
o=FreeSWITCH 1250746184 1250746186 IN IP4 182.xxx.xx.xxx.
s=FreeSWITCH.
c=IN IP4 182.xxx.xx.xxx.
t=0 0.
m=audio 26670 RTP/AVP 8 101.
a=rtpmap:8 PCMA/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=silenceSupp:off - - - -.
a=ptime:30.








From brian at freeswitch.org  Thu Aug 20 06:38:34 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 08:38:34 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4A8D5132.7010807@gmx.net>
References: <4A8D5132.7010807@gmx.net>
Message-ID: <04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>

This is a bug in the fritzbox... you have to set your codec neg. to  
greedy on the sofia profile and that should fix it.

/b

On Aug 20, 2009, at 8:35 AM, Peter P GMX wrote:

> Hello,
>
> when calling from Fritzbox to a Snom Phone , sound is fine. But when
> calling an internal Freeswitch number (conference, mailbox) i hear a
> very choppy voice coming from the fritzbox side. I think it may have  
> to
> do with the ptime 20msec/30msec.
>
> Example: When calling from the fritzbox to a voicemail then the
> annoucement from Freeswitch is choppy (too slow with interrups), but  
> the
> recorded message is fine.
>
> Did anybody experience the same problem?
>
> Best regards
> Peter




From Prometheus001 at gmx.net  Thu Aug 20 07:27:15 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 20 Aug 2009 16:27:15 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>
References: <4A8D5132.7010807@gmx.net>
	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>
Message-ID: <4A8D5D43.3030300@gmx.net>

Hello Brian,

I have added
<param name="inbound-codec-negotiation" value="greedy"/>
to internal and external conf.
We have only allowed one codec now (G711A)

Conferencing MOH is fine. However hearing the other party (Fritzbox is
almost impossible).
Hearing voicemail announcements is also very choppy with seconds of delay.

So, in one case it works fine, in other s not.

Any more hints?

Best regards
Peter



Brian West schrieb:
> This is a bug in the fritzbox... you have to set your codec neg. to  
> greedy on the sofia profile and that should fix it.
>
> /b
>
> On Aug 20, 2009, at 8:35 AM, Peter P GMX wrote:
>
>   
>> Hello,
>>
>> when calling from Fritzbox to a Snom Phone , sound is fine. But when
>> calling an internal Freeswitch number (conference, mailbox) i hear a
>> very choppy voice coming from the fritzbox side. I think it may have  
>> to
>> do with the ptime 20msec/30msec.
>>
>> Example: When calling from the fritzbox to a voicemail then the
>> annoucement from Freeswitch is choppy (too slow with interrups), but  
>> the
>> recorded message is fine.
>>
>> Did anybody experience the same problem?
>>
>> Best regards
>> Peter
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From dave at 3c.co.uk  Thu Aug 20 07:29:09 2009
From: dave at 3c.co.uk (David Knell)
Date: Thu, 20 Aug 2009 17:29:09 +0300
Subject: [Freeswitch-users] Recommendations for an interface between
 FreeSwitch and ANSI C code
In-Reply-To: <4A8D18E3.9090302@venturevoip.com>
References: <4A8D18E3.9090302@venturevoip.com>
Message-ID: <1250778549.6309.28.camel@dk-d820>

On Thu, 2009-08-20 at 21:35 +1200, Matt Riddell wrote:
> We have an application written in ANSI C which currently talks to the 
> Asterisk Manager to make phone calls.
> 
> We're possibly looking at converting this to FreeSwitch.
> 
> At the moment it has an abstraction layer from Asterisk and speaks to 
> between 1 and 80 Asterisk machines using round robin for distribution.
> 
> Would you recommend that I use the mod_event socket and basically work 
> with FreeSwitch in the same way as I work with Asterisk or am I 
> overlooking a possibly different way of doing things?

Just to echo the sound advice from others, the event socket is almost
certainly the right place to start.  We've used it exclusively for our
stuff for getting on for two years, and I've no regrets.

--Dave

-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk




From shiyanov at gmail.com  Thu Aug 20 07:30:52 2009
From: shiyanov at gmail.com (Artem Shiyanov)
Date: Thu, 20 Aug 2009 18:30:52 +0400
Subject: [Freeswitch-users] "mute" channel programmatically with
	mod_event_socket
In-Reply-To: <36B2A0E3-6139-4724-A7F3-CAD89FB98688@avgs.ca>
References: <28651154.332401250673051662.JavaMail.root@zimbra.tng.de>
	<17888468.332601250673446046.JavaMail.root@zimbra.tng.de>
	<d211f54a0908190413p17311b15u42f57dd3417ec634@mail.gmail.com>
	<36B2A0E3-6139-4724-A7F3-CAD89FB98688@avgs.ca>
Message-ID: <d211f54a0908200730i6fe7d7ffm998cbdad601dac9b@mail.gmail.com>

Mathieu,

have to confess- you are right! uuid_bridge works as expected. Usual saying
- is didn't work last time I tried!
Anyway, thank you much!

Artem


On Wed, Aug 19, 2009 at 5:02 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> Eavesdrop kind of works yeah, you can use the intercept application to
> re-bridge the channels together, like:
>
> SendMsg <uuid_channel_making_mute>
> call-command: execute
> execute-app-name: intercept
> execute-app-arg: <uuid_participator>
>
> The same can be done with the uuid_bridge api.
>
> api uuid_bridge <uuid1> <uuid2>
>
> If you want the
> cleaner way you could implement a media bug that replaces all the audio of the channel by silence, but that'd require some C coding.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 19-Aug-09, at 7:13 AM, Artem Shiyanov wrote:
>
> The point is - a simple call flow is desired- if I have a ordinary 1-to-1
> call and one of the participators decides to mute call - I don't want to put
> both channels into a conference room but it looks like I have no other
> choices.
>
> BUT:
> I found brilliant app - eavesdrop! If I do this for one-to-one call - mute
> works!
> SendMsg <uuid_channel_making_mute>
> call-command: execute
> execute-app-name: eavesdrop
> execute-app-arg: <uuid_participator>
>
> But the problem appears when I want to unmute.. the call! I've tried to
> re-bridge channels, intercept them- nothing happens- one channel (muted one)
> doesn't hear the participator. And CLI command 'show channels' shows that
> channel with uui=<uuid_channel_making_mute> still process eavesdrop app.
> Maybe someone know how to switch off eavesdrop app?
>
>
> Artem
>
>
> On Wed, Aug 19, 2009 at 1:17 PM, Rudolf Denert <rdenert at tng.de> wrote:
>
>> You can use the caller controlls in the conference.conf.xml to implement
>> your own features something like mute or kick. Or do you want to mute mute
>> other conference members like a moderator can do this.
>>
>> BR
>>
>> ----- Urspr?ngliche Mail -----
>> Von: "Artem Shiyanov" <shiyanov at gmail.com>
>> An: freeswitch-users at lists.freeswitch.org
>> Gesendet: Dienstag, 18. August 2009 14:54:04 GMT +01:00
>> Amsterdam/Berlin/Bern/Rom/Stockholm/Wien
>> Betreff: [Freeswitch-users] "mute" channel programmatically with
>> mod_event_socket
>>
>>
>> Hello all!
>>
>> I'm trying to implement "mute" feature with mod_event_socket: I want
>> programmatically mute/unmute some channel in a call.. And I don't see any
>> other ways except to use conference room with special rule "mute".
>> Anybody knows the better way?
>>
>>
>> Thanks,
>> Artem
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Thu Aug 20 07:39:33 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 09:39:33 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4A8D5D43.3030300@gmx.net>
References: <4A8D5132.7010807@gmx.net>
	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>
	<4A8D5D43.3030300@gmx.net>
Message-ID: <C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>

Besides taking a hammer to it?  Have you tried to make sure you have  
the latest firmware?  Try setting the ptime on the fritz to 20ms?

I really can't trust a product that has fritz in its name... it might  
go on the fritz  :P  pun intended.

/b

On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:

> Any more hints?




From msc at freeswitch.org  Thu Aug 20 08:16:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Aug 2009 08:16:54 -0700
Subject: [Freeswitch-users] send sip options message
In-Reply-To: <4A8B2015.2060605@xpirio.com>
References: <4A8B2015.2060605@xpirio.com>
Message-ID: <87f2f3b90908200816x65d296dckc5052fcebad89e4a@mail.gmail.com>

2009/8/18 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>

> hi
>
> does anybody know how to send a sip options message to a registered user,
> using the event socket
> or something else build in freeswitch
> i think the ping parameter does something like this for gateways.
>
> what i want/need is the same thing that is provided in asterisk with the
> qualifying option,
> to see how "reachable" a certain client is.


Nothing exists at present but Mathieu Rene was looking at possibly adding an
API to do this. Stay tuned for more information...

-MC


> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP
>
> xpirio
> Telekommunikation & Service GmbH
> Lakeside B04
> 9020 Klagenfurt
> Austria
>
> T  +43 (0) 5 77 11 - 1000
> F  +43 (0) 5 77 11 - 1002
> E  christian.loeschenkohl at xpirio.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From boonedox at gmail.com  Thu Aug 20 08:20:17 2009
From: boonedox at gmail.com (Jeremiah Johnson)
Date: Thu, 20 Aug 2009 09:20:17 -0600
Subject: [Freeswitch-users] Media Not Heard When Bridging 2 Calls
	withsame Gateway
In-Reply-To: <6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>
References: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>
	<6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>
Message-ID: <15FBC1BB717E4A9CB3E15E01D299B672@johnson>

For my gateway, I had to set ignore_early_media=true.


From: Mathieu Rene 
Sent: Thursday, August 20, 2009 7:00 AM
To: freeswitch-users at lists.freeswitch.org 
Subject: Re: [Freeswitch-users] Media Not Heard When Bridging 2 Calls withsame Gateway


Hi


How are you bridging the calls in FS (which api call or C function are you using)? 


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca








On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:


  I'm trying to get FreeSWITCH to bridge two channels together through the same external gateway, but I'm having issues hearing audio. Both legs if setup independently and forwarded to 5000 (test ivr) work fine for both inbound and outbound media, but when I try to bridge them together, everything seems fine in FreeSWITCH, but neither party can hear the other speak. I'm thinking the external gateway might be having some issues because I've been able to bridge 2 channels together through the same gateway on different providers, but thought I'd also try to seek some help here. FreeSWITCH should be handling the media for both channels, so I can't figure out why if Leg A and Leg B work independently, but not if they are bridged together. Is there a setting somewhere in FS that I'm missing? 


  Below is a ngrep of the SIP interactions if it's useful. Thanks for the help.


  --matt



  interface: eth0 (172.24.200.0/255.255.255.0)
  filter: (ip or ip6) and ( port 5060 )


  U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060
  INVITE sip:914159927717 at 38.98.58.148 SIP/2.0.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
  Max-Forwards: 70.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  To: <sip:914159927717 at 38.98.58.148>.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257811 INVITE.
  Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
  Supported: timer, precondition, path, replaces.
  Allow-Events: talk, refer.
  Content-Type: application/sdp.
  Content-Disposition: session.
  Content-Length: 293.
  Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;party=calling;screen=yes;privacy=off.
  .
  v=0.
  o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
  s=FreeSWITCH.
  c=IN IP4 216.81.56.198.
  t=0 0.
  m=audio 24700 RTP/AVP 0 8 3 101 13.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:3 GSM/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=rtpmap:13 CN/8000.
  a=ptime:20.




  U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 100 Trying.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257811 INVITE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
  Contact: <sip:914159927717 at 38.98.58.148:5060>.
  Content-Length: 0.
  .




  U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 183 Session Progress.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257811 INVITE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
  Contact: <sip:914159927717 at 38.98.58.148:5060>.
  Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
  Content-Type: application/sdp.
  Content-Length: 227.
  .
  v=0.
  o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
  s=M6 Call.
  c=IN IP4 38.98.58.148.
  t=0 0.
  m=audio 42554 RTP/AVP 0 101.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-15.
  a=ptime:20.
  a=sendrecv.
  a=rtcp:6461 IN IP4 10.10.24.50.




  U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 200 OK.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257811 INVITE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
  Contact: <sip:914159927717 at 38.98.58.148:5060>.
  Session-Expires: 1800;refresher=uas.
  Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
  Supported: timer.
  Content-Type: application/sdp.
  Content-Length: 227.
  .
  v=0.
  o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
  s=M6 Call.
  c=IN IP4 38.98.58.148.
  t=0 0.
  m=audio 42554 RTP/AVP 0 101.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-15.
  a=ptime:20.
  a=sendrecv.
  a=rtcp:6461 IN IP4 10.10.24.50.




  U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060
  ACK sip:914159927717 at 38.98.58.148:5060 SIP/2.0.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc.
  Max-Forwards: 70.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257811 ACK.
  Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
  Content-Length: 0.
  .




  U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060
  INVITE sip:914154650027 at 38.98.58.148 SIP/2.0.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
  Max-Forwards: 70.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257815 INVITE.
  Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
  Supported: timer, precondition, path, replaces.
  Allow-Events: talk, refer.
  Content-Type: application/sdp.
  Content-Disposition: session.
  Content-Length: 293.
  Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;party=calling;screen=yes;privacy=off.
  .
  v=0.
  o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198.
  s=FreeSWITCH.
  c=IN IP4 216.81.56.198.
  t=0 0.
  m=audio 24798 RTP/AVP 0 8 3 101 13.
  a=rtpmap:0 PCMU/8000.
  a=rtpmap:8 PCMA/8000.
  a=rtpmap:3 GSM/8000.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-16.
  a=rtpmap:13 CN/8000.
  a=ptime:20.




  U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 100 Trying.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257815 INVITE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
  Contact: <sip:914154650027 at 38.98.58.148:5060>.
  Content-Length: 0.
  .




  U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 183 Session Progress.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257815 INVITE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
  Contact: <sip:914154650027 at 38.98.58.148:5060>.
  Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
  Content-Type: application/sdp.
  Content-Length: 225.
  .
  v=0.
  o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
  s=M6 Call.
  c=IN IP4 38.98.58.148.
  t=0 0.
  m=audio 46520 RTP/AVP 0 101.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-15.
  a=ptime:20.
  a=sendrecv.
  a=rtcp:6451 IN IP4 10.10.24.50.




  U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 200 OK.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257815 INVITE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
  Contact: <sip:914154650027 at 38.98.58.148:5060>.
  Session-Expires: 1800;refresher=uas.
  Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
  Supported: timer.
  Content-Type: application/sdp.
  Content-Length: 225.
  .
  v=0.
  o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
  s=M6 Call.
  c=IN IP4 38.98.58.148.
  t=0 0.
  m=audio 46520 RTP/AVP 0 101.
  a=rtpmap:101 telephone-event/8000.
  a=fmtp:101 0-15.
  a=ptime:20.
  a=sendrecv.
  a=rtcp:6451 IN IP4 10.10.24.50.




  U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060
  ACK sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK.
  Max-Forwards: 70.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257815 ACK.
  Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
  Content-Length: 0.
  .




  U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080
  BYE sip:gw+epik.com at 216.81.56.198:5080 SIP/2.0.
  From: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
  To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 4817 BYE.
  Max-Forwards: 70.
  Via: SIP/2.0/UDP 38.98.58.148:5060;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
  Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
  Contact: <sip:914159927717 at 38.98.58.148:5060>.
  Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
  Content-Length: 0.
  .




  U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060
  SIP/2.0 200 OK.
  Via: SIP/2.0/UDP 38.98.58.148:5060;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
  Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
  From: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
  To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
  Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
  CSeq: 4817 BYE.
  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
  Supported: timer, precondition, path, replaces.
  Content-Length: 0.
  .




  U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060
  BYE sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
  Max-Forwards: 70.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257816 BYE.
  Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
  User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
  Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO.
  Supported: timer, precondition, path, replaces.
  Reason: Q.850;cause=16;text="NORMAL_CLEARING".
  Content-Length: 0.
  .




  U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080
  SIP/2.0 200 OK.
  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
  To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
  Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
  CSeq: 119257816 BYE.
  Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
  Contact: <sip:914154650027 at 38.98.58.148:5060>.
  Allow: INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
  Content-Length: 0.
  .
  _______________________________________________
  FreeSWITCH-users mailing list
  FreeSWITCH-users at lists.freeswitch.org
  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
  UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
  http://www.freeswitch.org





--------------------------------------------------------------------------------


_______________________________________________
FreeSWITCH-users mailing list
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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From jan.kubr at gmail.com  Thu Aug 20 08:43:57 2009
From: jan.kubr at gmail.com (Jan Kubr)
Date: Thu, 20 Aug 2009 17:43:57 +0200
Subject: [Freeswitch-users] How to delay IVR answer during an outbound
	call
In-Reply-To: <c43084ae0908112323m1c8d5856kc05acc6664b635cd@mail.gmail.com>
References: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com> 
	<c43084ae0908112323m1c8d5856kc05acc6664b635cd@mail.gmail.com>
Message-ID: <698401620908200843p414f426cudcda3a40ac2d91ca@mail.gmail.com>

I have been experiencing this as well. It happens randomly and I
haven't been able to find out what the issue is. I think there is some
delay when the RTP ports are being negotiated/allocated. Or something.
What helped me a bit: I start with playing a file containing 1 second
of silence and only then do whatever I want to do.

Jan

On Wed, Aug 12, 2009 at 8:23 AM, Paul Li<plite2012 at gmail.com> wrote:
> I am actually doing a lua script for IVR as follows
>
> -- answer the call
> session:answer();
>
> while session:ready() == true do
> ? ? ? ?-- sleep a second
> ? ? ? ?session:sleep(1000);
>
> ? ? ? ?-- play a file
> ? ? ? ?session:streamFile("/path/to/blah.wav");
>
> ? ? ? ?-- hangup
> ? ? ? ?session:hangup();
> end
>
> The problem lies in: when I picked up my phone, blah.wav was already
> played for a while, instead of from the beginning.
>
> I shall greatly appreciate any input.
>
> On Wed, Aug 12, 2009 at 12:58 AM, Paul Li<plite2012 at gmail.com> wrote:
>> I have a dummy question. Say, you have an outbound call to the demo
>> IVR as below:
>>
>> originate sofia/gateway/myvoip/19876543210 5000
>>
>> How do I delay the IVR response until the recipient at 19876543210
>> picks up the call? I tried "ignore_early_media=true", which had no
>> effect.
>>
>> Many thanks in advance.
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From brian at freeswitch.org  Thu Aug 20 09:04:33 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 11:04:33 -0500
Subject: [Freeswitch-users] How to delay IVR answer during an outbound
	call
In-Reply-To: <698401620908200843p414f426cudcda3a40ac2d91ca@mail.gmail.com>
References: <c43084ae0908112258x341cec7dj883e27fac0179576@mail.gmail.com>
	<c43084ae0908112323m1c8d5856kc05acc6664b635cd@mail.gmail.com>
	<698401620908200843p414f426cudcda3a40ac2d91ca@mail.gmail.com>
Message-ID: <EFF7C574-B430-4DC9-BE1B-01241FDBA232@freeswitch.org>

Nope.  The issue is the far end isn't done in time.  So we send  
packets and they ignore them.  The only way to prevent this is to  
answer, silence_stream://1000, then playback.

/b

On Aug 20, 2009, at 10:43 AM, Jan Kubr wrote:

> I have been experiencing this as well. It happens randomly and I
> haven't been able to find out what the issue is. I think there is some
> delay when the RTP ports are being negotiated/allocated. Or something.
> What helped me a bit: I start with playing a file containing 1 second
> of silence and only then do whatever I want to do.
>
> Jan




From dome at tel.co.th  Thu Aug 20 09:11:08 2009
From: dome at tel.co.th (Dome Charoenyost)
Date: Thu, 20 Aug 2009 23:11:08 +0700
Subject: [Freeswitch-users] FS and SS7
In-Reply-To: <D7C4EE10-6998-4FDC-BCC5-E3180F3157A8@freeswitch.org>
References: <8ccbff060908200342p53474f58m645159a191db0364@mail.gmail.com>
	<D7C4EE10-6998-4FDC-BCC5-E3180F3157A8@freeswitch.org>
Message-ID: <8ccbff060908200911u30c7f169nb9016733aec2460f@mail.gmail.com>

2009/8/20 Brian West <brian at freeswitch.org>:
>
> On Aug 20, 2009, at 5:42 AM, Dome Charoenyost wrote:
>
>> ? ? ? ?1. Can i setup SS7 with out ss7box ?
>
> Sangoma's ss7box is the only solution right now thats turn key.

If i seting up asterisk with libss7. should be same ss7box right ?
>
>> ? ? ? ?2. I found some tutorial about libss7 in Asterisk. Is FS
>> support all feature like that.
>
> You can't really use libss7 in FreeSWITCH as its not license
> compatible... neither is OpenSS7.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mattdfong at gmail.com  Thu Aug 20 10:21:40 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Thu, 20 Aug 2009 10:21:40 -0700
Subject: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with
	same Gateway
In-Reply-To: <6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>
References: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>
	<6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>
Message-ID: <4256bf830908201021s7da0bff0wed8e060dbac0db2c@mail.gmail.com>

originate {ignore_early_media=true}sofia/gateway/epik.com/914159927717&bridge(sofia/gateway/
epik.com/914154650027)
is the string I was using from the console.

On Thu, Aug 20, 2009 at 6:00 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi
>
> How are you bridging the calls in FS (which api call or C function are you
> using)?
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:
>
> I'm trying to get FreeSWITCH to bridge two channels together through the
> same external gateway, but I'm having issues hearing audio. Both legs if
> setup independently and forwarded to 5000 (test ivr) work fine for both
> inbound and outbound media, but when I try to bridge them together,
> everything seems fine in FreeSWITCH, but neither party can hear the other
> speak. I'm thinking the external gateway might be having some issues because
> I've been able to bridge 2 channels together through the same gateway on
> different providers, but thought I'd also try to seek some help here.
> FreeSWITCH should be handling the media for both channels, so I can't figure
> out why if Leg A and Leg B work independently, but not if they are bridged
> together. Is there a setting somewhere in FS that I'm missing?
> Below is a ngrep of the SIP interactions if it's useful. Thanks for the
> help.
>
> --matt
>
> interface: eth0 (172.24.200.0/255.255.255.0)
> filter: (ip or ip6) and ( port 5060 )
>
> U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060
> INVITE sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>>.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 293.
> Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
> s=FreeSWITCH.
> c=IN IP4 216.81.56.198.
> t=0 0.
> m=audio 24700 RTP/AVP 0 8 3 101 13.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
>
>
> U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 100 Trying.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
> >;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 183 Session Progress.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
> >;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Allow:
> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Type: application/sdp.
> Content-Length: 227.
> .
> v=0.
> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 42554 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6461 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 200 OK.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
> >;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Session-Expires: 1800;refresher=uas.
> Allow:
> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Supported: timer.
> Content-Type: application/sdp.
> Content-Length: 227.
> .
> v=0.
> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 42554 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6461 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060
> ACK sip:914159927717 at 38.98.58.148:5060 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
> >;tag=F725.2C49.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257811 ACK.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060
> INVITE sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>>.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 293.
> Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198.
> s=FreeSWITCH.
> c=IN IP4 216.81.56.198.
> t=0 0.
> m=audio 24798 RTP/AVP 0 8 3 101 13.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
>
>
> U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 100 Trying.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
> >;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 183 Session Progress.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
> >;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Allow:
> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Type: application/sdp.
> Content-Length: 225.
> .
> v=0.
> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 46520 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6451 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 200 OK.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
> >;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 INVITE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Session-Expires: 1800;refresher=uas.
> Allow:
> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Supported: timer.
> Content-Type: application/sdp.
> Content-Length: 225.
> .
> v=0.
> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
> s=M6 Call.
> c=IN IP4 38.98.58.148.
> t=0 0.
> m=audio 46520 RTP/AVP 0 101.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-15.
> a=ptime:20.
> a=sendrecv.
> a=rtcp:6451 IN IP4 10.10.24.50.
>
>
> U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060
> ACK sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
> >;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257815 ACK.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080
> BYE sip:gw+epik.com at 216.81.56.198:5080 SIP/2.0.
> From: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
> >;tag=F725.2C49.
> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 4817 BYE.
> Max-Forwards: 70.
> Via: SIP/2.0/UDP 38.98.58.148:5060
> ;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
> Contact: <sip:914159927717 at 38.98.58.148:5060>.
> Allow:
> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060
> SIP/2.0 200 OK.
> Via: SIP/2.0/UDP 38.98.58.148:5060
> ;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
> From: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
> >;tag=F725.2C49.
> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=ZtFvjeFQmXvpp.
> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
> CSeq: 4817 BYE.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060
> BYE sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
> Max-Forwards: 70.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
> >;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257816 BYE.
> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO.
> Supported: timer, precondition, path, replaces.
> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
> Content-Length: 0.
> .
>
>
> U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080
> SIP/2.0 200 OK.
> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
> >;tag=038mm9ZtH6j9H.
> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
> >;tag=F72E.2D4E.
> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
> CSeq: 119257816 BYE.
> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
> Contact: <sip:914154650027 at 38.98.58.148:5060>.
> Allow:
> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
> Content-Length: 0.
> .
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From csa at nowthor.com  Thu Aug 20 10:28:56 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 20 Aug 2009 13:28:56 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
Message-ID: <4A8D87D8.8080001@nowthor.com>

Hello!

I am using Callcentric for my tests and have observed what appears to me 
a possible bug in the way Freeswitch handles DNS SRV records.

Callcentric uses DNS SRV records as a way to direct traffic to their SIP 
server. A 'srv' 'dig' of '_sip._udp.callcentric.com' returns:

_sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080 
alpha6.callcentric.com.
_sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080 
alpha7.callcentric.com.
_sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080 
alpha1.callcentric.com.
_sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080 
alpha3.callcentric.com.

Based on this information, Freeswitch appears to correctly round robin 
all available IP addresses except in a particular situation: in the 
middle of authorizations.

For example, in a registration, Freeswitch send a packet to 
alpha1.callcentric.com. Callcentric then challenges Freeswitch with a 
Proxy Authorization request. Freeswitch then sends the packet with the 
requested credentials but not necessarily to alpha1.callcentric.com! In 
many cases, instead of sticking to the 'challenging' server, Freeswitch 
round robins and sends the second packet to, one of the other servers. 
This continues for a little while and eventually, simply by luck, the 
second packet is sent to the 'challenging' Callcentric server.

Shouldn't Freeswitch stick to the same server when challenged for 
credentials? Is this a bug? Is there a way to make Freeswitch behave 
differently?

Thanks!

Carlos Antunes
Nowthor Corporation






From intralanman at freeswitch.org  Thu Aug 20 10:54:58 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 20 Aug 2009 13:54:58 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8D87D8.8080001@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>
Message-ID: <C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>

On Aug 20, 2009, at 1:28 PM, Carlos S. Antunes wrote:

> Hello!
>
> I am using Callcentric for my tests and have observed what appears  
> to me
> a possible bug in the way Freeswitch handles DNS SRV records.
>
> Callcentric uses DNS SRV records as a way to direct traffic to their  
> SIP
> server. A 'srv' 'dig' of '_sip._udp.callcentric.com' returns:
>
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha6.callcentric.com.
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha7.callcentric.com.
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha1.callcentric.com.
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha3.callcentric.com.
>
> Based on this information, Freeswitch appears to correctly round robin
> all available IP addresses except in a particular situation: in the
> middle of authorizations.
>

very true, but i've been reading over the RFCs on this, and it seems  
that FreeSWITCH isn't doing anything incorrectly.

in RFC3263 (section 4), when talking about client usage of SRV:
The procedures here MUST be done exactly once per transaction, where  
transaction is as defined in [1].
[1] being RFC3261

in RFC3261 (section 8.1.3.5), when talking about 4xx responses:
In all of the above cases, the request is retried by creating a new
    request with the appropriate modifications.  This new request
    constitutes a new transaction and SHOULD have the same value of the
    Call-ID, To, and From of the previous request, but the CSeq should
    contain a new sequence number that is one higher than the previous.

> For example, in a registration, Freeswitch send a packet to
> alpha1.callcentric.com. Callcentric then challenges Freeswitch with a
> Proxy Authorization request. Freeswitch then sends the packet with the
> requested credentials but not necessarily to alpha1.callcentric.com!  
> In
> many cases, instead of sticking to the 'challenging' server,  
> Freeswitch
> round robins and sends the second packet to, one of the other servers.
> This continues for a little while and eventually, simply by luck, the
> second packet is sent to the 'challenging' Callcentric server.
>
> Shouldn't Freeswitch stick to the same server when challenged for
> credentials?

can you show anything in the RFCs that says so?



> Is this a bug?

see above



> Is there a way to make Freeswitch behave
> differently?
disable-srv on the profile



All that said, if it has to be a "bug", then it seems to me that it's  
more of a "bug" in callcentric's service. They'd probably be better of  
actually prioritizing their SRV records. If they want load balancing  
and want to do checking for stale nonces, then they should be sharing  
nonces across all of their proxies or using a proper load balancer.


Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     

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From steveu at coppice.org  Thu Aug 20 11:06:54 2009
From: steveu at coppice.org (Steve Underwood)
Date: Fri, 21 Aug 2009 02:06:54 +0800
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
Message-ID: <4A8D90BE.2090007@coppice.org>

On 08/20/2009 05:22 AM, Michael Collins wrote:
>
>     There is no noise on those 3 beeps. In fact, for something that's been
>     through ulaw/alaw compression those beeps are very clean. They are
>     quite
>     short, though.
>
>
> Heck yeah they're short! Steve, in your experience is there a 
> practical way to detect a beep that short without chewing up system 
> resources or having lots of false positives?
> -MC
>
The tone samples I just looked at are about 130ms long. The problem is 
the detector is trying to be a very open ended detector of anything 
narrowband enough to be a single tone, and of any duration beyond some 
small minimum. Its difficult to make such a thing voice immune unless 
you can also count on a very large signal to noise ratio. With a digital 
trunk you can probably rely on a large SNR, but what happens when people 
use analogue lines? There is a reason why DTMF detectors try hard to 
work down to about 10dB SNR. :-)

Steve




From msc at freeswitch.org  Thu Aug 20 11:20:17 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Aug 2009 11:20:17 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4A8D90BE.2090007@coppice.org>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
Message-ID: <87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>

On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood <steveu at coppice.org>wrote:

> On 08/20/2009 05:22 AM, Michael Collins wrote:
> >
> >     There is no noise on those 3 beeps. In fact, for something that's
> been
> >     through ulaw/alaw compression those beeps are very clean. They are
> >     quite
> >     short, though.
> >
> >
> > Heck yeah they're short! Steve, in your experience is there a
> > practical way to detect a beep that short without chewing up system
> > resources or having lots of false positives?
> > -MC
> >
> The tone samples I just looked at are about 130ms long. The problem is
> the detector is trying to be a very open ended detector of anything
> narrowband enough to be a single tone, and of any duration beyond some
> small minimum. Its difficult to make such a thing voice immune unless
> you can also count on a very large signal to noise ratio. With a digital
> trunk you can probably rely on a large SNR, but what happens when people
> use analogue lines? There is a reason why DTMF detectors try hard to
> work down to about 10dB SNR. :-)
>
> Steve
>

Thanks for the lesson uncle Steve! I'm guessing that the OP will need a new
strategy. Possibly waiting for silence? Not sure what's the best way to go
but I'm interested in hearing if someone has a solution.

-MC
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From mike at jerris.com  Thu Aug 20 11:25:51 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 20 Aug 2009 14:25:51 -0400
Subject: [Freeswitch-users] Sharing Presence Information Across Separate
	Offices
In-Reply-To: <debff5a80908200129q60d91a76v395781da1aabd661@mail.gmail.com>
References: <20090820065153.ac3a26cf@kms.expitrans.com>
	<debff5a80908200129q60d91a76v395781da1aabd661@mail.gmail.com>
Message-ID: <4C4DB254-9F49-4181-AD9E-592571240844@jerris.com>

Check out mod_event_multicast.  I think you should be able to share  
using that to share events between boxes (although I have never tried  
it).

Mike

On Aug 20, 2009, at 4:29 AM, afshin afzali wrote:

> Hi Kenneth,
>
> I'm not going to answer your question! Instead I would like to  
> emphasize on the thing you are going to achieve because some times  
> ago I've post this question in some other title but unfortunately  
> did not get any answer. As the SIP protocol's point of view you  
> should be able to subscribe every FreeSWITCH machines for external  
> events in another one (or probably in a central PRESENCE server such  
> as SER) to get NOTIFY messages which indicate the information you  
> wish to have. Although the sofia SIP endpoint can do this (as you  
> can find in its features) I could not find any guidelines for it in  
> FreeSWITCH.
>
> Regards,
> -- Afshin Afzali   
>
> On Thu, Aug 20, 2009 at 10:21 AM, Kenneth Shaw <ken at expitrans.com>  
> wrote:
> I apologize in advance if this message duplicates a previous post or  
> the topic has been covered elsewhere. If that's the case, a pointer  
> would be appreciated!
>
> I have multiple offices each running Freeswitch and a number of  
> Polycom phones and softphones.
>
> I would like to share the presence information between the offices,  
> such that if a person in Office B can see if a person in Office A is  
> on the phone. I personally don't know of any way to do this through  
> Freeswitch as it stands. I have thought about using a VPN for the  
> offices and just setting up the phones to all register to a single  
> server, however I don't want calls between phones in the same office  
> (intra-office) to have to be routed all the way through a different  
> office, which may be on the other side of the world and costs  
> bandwidth.
>
> Any suggestions as to the best way to accomplish this would be  
> greatly appreciated. Thanks!
>
> --
> Kenneth Shaw
> ExpiTrans, Inc.
> 129 W. Wilson St., Suite 204
> Costa Mesa, CA 92627
> tel: 949.650.4600
> fax: 949.642.6044
> ken at expitrans.com

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From mike at jerris.com  Thu Aug 20 11:27:43 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 20 Aug 2009 14:27:43 -0400
Subject: [Freeswitch-users] send sip options message
In-Reply-To: <87f2f3b90908200816x65d296dckc5052fcebad89e4a@mail.gmail.com>
References: <4A8B2015.2060605@xpirio.com>
	<87f2f3b90908200816x65d296dckc5052fcebad89e4a@mail.gmail.com>
Message-ID: <B769F00C-C305-45BA-87B2-C050C5CD136B@jerris.com>

There may be a magic event you can send to do this.  I can't recall if  
we did that for options or just info/notify.  A grep for nua_options  
in mod_sofia dir should answer the question.

Mike

On Aug 20, 2009, at 11:16 AM, Michael Collins wrote:

>
>
> 2009/8/18 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
> hi
>
> does anybody know how to send a sip options message to a registered  
> user, using the event socket
> or something else build in freeswitch
> i think the ping parameter does something like this for gateways.
>
> what i want/need is the same thing that is provided in asterisk with  
> the qualifying option,
> to see how "reachable" a certain client is.
>
> Nothing exists at present but Mathieu Rene was looking at possibly  
> adding an API to do this. Stay tuned for more information...
>
> -MC
>
>
> br
>
> --
> Ing. Christian L?schenkohl
> Technische Leitung, Forschung & Entwicklung VoIP

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From mrene_lists at avgs.ca  Thu Aug 20 11:31:55 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 20 Aug 2009 14:31:55 -0400
Subject: [Freeswitch-users] send sip options message
In-Reply-To: <B769F00C-C305-45BA-87B2-C050C5CD136B@jerris.com>
References: <4A8B2015.2060605@xpirio.com>
	<87f2f3b90908200816x65d296dckc5052fcebad89e4a@mail.gmail.com>
	<B769F00C-C305-45BA-87B2-C050C5CD136B@jerris.com>
Message-ID: <09DDA10A-47F5-4DAE-8B8B-82A613BE2A53@avgs.ca>

Oh, I missed that one.

<param name="nat-options-ping" value="true" /> (in the sip profile)

It does it only when NAT is detected though.

Tschuess,

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 20-Aug-09, at 2:27 PM, Michael Jerris wrote:

> There may be a magic event you can send to do this.  I can't recall  
> if we did that for options or just info/notify.  A grep for  
> nua_options in mod_sofia dir should answer the question.
>
> Mike
>
> On Aug 20, 2009, at 11:16 AM, Michael Collins wrote:
>
>>
>>
>> 2009/8/18 Christian L?schenkohl <christian.loeschenkohl at xpirio.com>
>> hi
>>
>> does anybody know how to send a sip options message to a registered  
>> user, using the event socket
>> or something else build in freeswitch
>> i think the ping parameter does something like this for gateways.
>>
>> what i want/need is the same thing that is provided in asterisk  
>> with the qualifying option,
>> to see how "reachable" a certain client is.
>>
>> Nothing exists at present but Mathieu Rene was looking at possibly  
>> adding an API to do this. Stay tuned for more information...
>>
>> -MC
>>
>>
>> br
>>
>> --
>> Ing. Christian L?schenkohl
>> Technische Leitung, Forschung & Entwicklung VoIP
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Thu Aug 20 11:36:49 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 20 Aug 2009 14:36:49 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8D87D8.8080001@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>
Message-ID: <700500D9-4508-4848-A198-76692574AF15@jerris.com>

You can bypass the srv records if you like by passing a :port with the  
hostname where you use it in freeswitch.


On Aug 20, 2009, at 1:28 PM, Carlos S. Antunes wrote:

> Hello!
>
> I am using Callcentric for my tests and have observed what appears  
> to me
> a possible bug in the way Freeswitch handles DNS SRV records.
>
> Callcentric uses DNS SRV records as a way to direct traffic to their  
> SIP
> server. A 'srv' 'dig' of '_sip._udp.callcentric.com' returns:
>
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha6.callcentric.com.
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha7.callcentric.com.
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha1.callcentric.com.
> _sip._udp.callcentric.com. 10025 IN     SRV     20 7 5080
> alpha3.callcentric.com.
>
> Based on this information, Freeswitch appears to correctly round robin
> all available IP addresses except in a particular situation: in the
> middle of authorizations.
>
> For example, in a registration, Freeswitch send a packet to
> alpha1.callcentric.com. Callcentric then challenges Freeswitch with a
> Proxy Authorization request. Freeswitch then sends the packet with the
> requested credentials but not necessarily to alpha1.callcentric.com!  
> In
> many cases, instead of sticking to the 'challenging' server,  
> Freeswitch
> round robins and sends the second packet to, one of the other servers.
> This continues for a little while and eventually, simply by luck, the
> second packet is sent to the 'challenging' Callcentric server.
>
> Shouldn't Freeswitch stick to the same server when challenged for
> credentials? Is this a bug? Is there a way to make Freeswitch behave
> differently?
>
> Thanks!
>
> Carlos Antunes
> Nowthor Corporation




From brian at freeswitch.org  Thu Aug 20 11:39:44 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 13:39:44 -0500
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <700500D9-4508-4848-A198-76692574AF15@jerris.com>
References: <4A8D87D8.8080001@nowthor.com>
	<700500D9-4508-4848-A198-76692574AF15@jerris.com>
Message-ID: <DEBF02AF-E36C-47FA-88C7-363BCC91282F@freeswitch.org>

Or as I have argued today they should fix their SRV records to be zero  
weighted.

/b

On Aug 20, 2009, at 1:36 PM, Michael Jerris wrote:

> You can bypass the srv records if you like by passing a :port with the
> hostname where you use it in freeswitch.




From csa at nowthor.com  Thu Aug 20 12:56:43 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 20 Aug 2009 15:56:43 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>
References: <4A8D87D8.8080001@nowthor.com>
	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>
Message-ID: <4A8DAA7B.8090601@nowthor.com>

Raymond Chandler wrote:
> very true, but i've been reading over the RFCs on this, and it seems 
> that FreeSWITCH isn't doing anything incorrectly.
>
> in RFC3263 (section 4), when talking about client usage of SRV:
> The procedures here MUST be done exactly once per transaction, where transaction is as defined in [1]. 
> [1] being RFC3261
>
> in RFC3261 (section 8.1.3.5), when talking about 4xx responses:
> In all of the above cases, the request is retried by creating a new
>    request with the appropriate modifications.  This new request
>    constitutes a new transaction and SHOULD have the same value of the
>    Call-ID, To, and From of the previous request, but the CSeq should
>    contain a new sequence number that is one higher than the previous.
>

I'd say that based on my own and brief perusal of the relevant RFC's, 
you are right that Freeswitch is operating without violating the specs. 
Therefore, there is no bug in Freeswitch.


>
>> Is there a way to make Freeswitch behave
>> differently?
> disable-srv on the profile
>

In Callcentric's case, it doesn't help either. A request for 'A' recods 
for 'callcentric.com' returns a bunch. The only way appears to select 
and stick with only one.

> All that said, if it has to be a "bug", then it seems to me that it's 
> more of a "bug" in callcentric's service. They'd probably be better of 
> actually prioritizing their SRV records. If they want load balancing 
> and want to do checking for stale nonces, then they should be sharing 
> nonces across all of their proxies or using a proper load balancer.
>

Agreed. That being said, having a way to force Freeswitch to stick to 
the same IP address in the middle of authorization/authentication 
wouldn't violate any specs but would certainly make things easier when 
dealing with not so well implemented round robin scenarios. Do you think 
a new option could be added to Freeswitch to achieve round robin 
avoidance in these cases?

Thanks!

Carlos Antunes
Nowthor Corporation

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From alan at chandlerfamily.org.uk  Thu Aug 20 13:09:21 2009
From: alan at chandlerfamily.org.uk (Alan Chandler)
Date: Thu, 20 Aug 2009 21:09:21 +0100
Subject: [Freeswitch-users] Is it possible to unbind the * key?
Message-ID: <4A8DAD71.5000404@chandlerfamily.org.uk>

In a previous e-mail to this list I explained some problems I was having 
with the re-entering a conference and the caller keys not working. 
Nobody responded (See "Stuck on my first attempt at dialplanning" above).

I have made some good progress in getting things to work.  What appears 
to be the case is that as I callout from the conference the first time, 
I call bind_meta_app in order to enable me to get back to the conference 
from the callout call.

When I get back to the conference the * key no longer works as a 
conference control, presumably because bind_meta_app still has control 
of it.

Is there a way to unbind it before re-entering the conference?
-- 
Alan Chandler
http://www.chandlerfamily.org.uk




From intralanman at freeswitch.org  Thu Aug 20 13:14:36 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 20 Aug 2009 16:14:36 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8DAA7B.8090601@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>
	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>
	<4A8DAA7B.8090601@nowthor.com>
Message-ID: <CD7267C3-5FE4-407E-AA8B-D4D54D8B2152@freeswitch.org>

>
> Agreed. That being said, having a way to force Freeswitch to stick  
> to the same IP address in the middle of authorization/authentication  
> wouldn't violate any specs but would certainly make things easier  
> when dealing with not so well implemented round robin scenarios. Do  
> you think a new option could be added to Freeswitch to achieve round  
> robin avoidance in these cases?


patches gladly accepted ;-)

The code that handles the dns srv stuff is apparently buried in the  
nta code of sofia, so it's not as easy as it sounds to "just add an  
option".  I personally don't have the skillz needed to take on such an  
endeavor, but would love to see that as an option.  Callcentric isn't  
the only provider that has equally weighted SRVs and doing checks for  
stale nonces, so i'm sure we'll here more of this issue when we get  
more residential users using more various providers.


Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     

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From intralanman at freeswitch.org  Thu Aug 20 13:54:12 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 20 Aug 2009 16:54:12 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8DAA7B.8090601@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>
	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>
	<4A8DAA7B.8090601@nowthor.com>
Message-ID: <8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>

Actually, disregard my previous mail... this patch probably wouldn't  
be gladly accepted... I seem to have forgotten about the part where  
the DNS SRV lookup "MUST be done once per transaction"....

So if we don't do that, then we would be breaking spec. Just disabling  
SRV on the profile to which the gateway to the faulty carrier is  
attached should fix the problem.  You shouldn't actually need to hard  
code any IPs into your hosts file.

Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     




From Prometheus001 at gmx.net  Thu Aug 20 13:54:32 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 20 Aug 2009 22:54:32 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>
	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>
Message-ID: <4A8DB808.1030903@gmx.net>

Hello Brian,

yes we have updated to the latest Fritzbox Firmware. These FritzBoxes
are widely spread here in Germany. I know of a SIP provider who has > 5
Mio FritzBoxes out there. So overall I expect about 10Mio FritzBoxes in
Germany, and they are covering a big stake of in the market. So they
generally they work. I tested mine against my Asterisk without problems.

But in my Freeswitch environment this is not working, and we have manage
to couple of these Boxes. So any help is appreciated.

Best regards
Peter


Brian West schrieb:
> Besides taking a hammer to it?  Have you tried to make sure you have  
> the latest firmware?  Try setting the ptime on the fritz to 20ms?
>
> I really can't trust a product that has fritz in its name... it might  
> go on the fritz  :P  pun intended.
>
> /b
>
> On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:
>
>   
>> Any more hints?
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From csa at nowthor.com  Thu Aug 20 14:03:32 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 20 Aug 2009 17:03:32 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>
References: <4A8D87D8.8080001@nowthor.com>	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>	<4A8DAA7B.8090601@nowthor.com>
	<8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>
Message-ID: <4A8DBA24.6010203@nowthor.com>

Raymond Chandler wrote:
> Actually, disregard my previous mail... this patch probably wouldn't  
> be gladly accepted... I seem to have forgotten about the part where  
> the DNS SRV lookup "MUST be done once per transaction"....
>
>   

Hmm, where does it say that, after the lookup, one cannot use the same 
IP address as before? :)

Carlos Antunes
Nowthor Corporation




From intralanman at freeswitch.org  Thu Aug 20 14:13:07 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 20 Aug 2009 17:13:07 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8DBA24.6010203@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>	<4A8DAA7B.8090601@nowthor.com>
	<8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>
	<4A8DBA24.6010203@nowthor.com>
Message-ID: <4D73B669-478E-44EF-83D7-0AA745286ACB@freeswitch.org>

On Aug 20, 2009, at 5:03 PM, Carlos S. Antunes wrote:
> Hmm, where does it say that, after the lookup, one cannot use the same
> IP address as before? :)

Section 4 of RFC3263 as quoted in my first email....

"The procedures here MUST be done exactly once per transaction, where  
transaction is as defined in [1].
"

Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     

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From csa at nowthor.com  Thu Aug 20 14:25:22 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 20 Aug 2009 17:25:22 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4D73B669-478E-44EF-83D7-0AA745286ACB@freeswitch.org>
References: <4A8D87D8.8080001@nowthor.com>	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>	<4A8DAA7B.8090601@nowthor.com>	<8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>	<4A8DBA24.6010203@nowthor.com>
	<4D73B669-478E-44EF-83D7-0AA745286ACB@freeswitch.org>
Message-ID: <4A8DBF42.9050503@nowthor.com>

Raymond Chandler wrote:
> On Aug 20, 2009, at 5:03 PM, Carlos S. Antunes wrote:
>> Hmm, where does it say that, after the lookup, one cannot use the same
>> IP address as before? :)
>
> Section 4 of RFC3263 as quoted in my first email....
>
> "The procedures here MUST be done exactly once per transaction, where 
> transaction is as defined in [1]. 
> "
>

Raymond, sure. But do the "procedures here" preclude one from choosing 
the same host given that both the priorities and weights are the same 
for all the hosts?
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From msc at freeswitch.org  Thu Aug 20 14:35:08 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 20 Aug 2009 14:35:08 -0700
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4A8DB808.1030903@gmx.net>
References: <4A8D5132.7010807@gmx.net>
	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>
	<4A8D5D43.3030300@gmx.net>
	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>
	<4A8DB808.1030903@gmx.net>
Message-ID: <87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>

On Thu, Aug 20, 2009 at 1:54 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello Brian,
>
> yes we have updated to the latest Fritzbox Firmware. These FritzBoxes
> are widely spread here in Germany. I know of a SIP provider who has > 5
> Mio FritzBoxes out there. So overall I expect about 10Mio FritzBoxes in
> Germany, and they are covering a big stake of in the market. So they
> generally they work. I tested mine against my Asterisk without problems.
>
> But in my Freeswitch environment this is not working, and we have manage
> to couple of these Boxes. So any help is appreciated.
>

Just curious - if it seems to be working with Asterisk but not FreeSWITCH
then could you do some tcpdumps of working vs. non-working calls and then
analyze them with Wireshark? I think Jason Garland's ClueCon presentation(s)
might be applicable here.

Thoughts?

-MC


>
> Best regards
> Peter
>
>
> Brian West schrieb:
> > Besides taking a hammer to it?  Have you tried to make sure you have
> > the latest firmware?  Try setting the ptime on the fritz to 20ms?
> >
> > I really can't trust a product that has fritz in its name... it might
> > go on the fritz  :P  pun intended.
> >
> > /b
> >
> > On Aug 20, 2009, at 9:27 AM, Peter P GMX wrote:
> >
> >
> >> Any more hints?
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From intralanman at freeswitch.org  Thu Aug 20 14:55:40 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Thu, 20 Aug 2009 17:55:40 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8DBF42.9050503@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>	<4A8DAA7B.8090601@nowthor.com>	<8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>	<4A8DBA24.6010203@nowthor.com>
	<4D73B669-478E-44EF-83D7-0AA745286ACB@freeswitch.org>
	<4A8DBF42.9050503@nowthor.com>
Message-ID: <773E8CAB-F146-452D-A3FD-AE202157A2B2@freeswitch.org>


On Aug 20, 2009, at 5:25 PM, Carlos S. Antunes wrote:

> Raymond Chandler wrote:
>>
>> On Aug 20, 2009, at 5:03 PM, Carlos S. Antunes wrote:
>>> Hmm, where does it say that, after the lookup, one cannot use the  
>>> same
>>> IP address as before? :)
>>
>> Section 4 of RFC3263 as quoted in my first email....
>>
>> "The procedures here MUST be done exactly once per transaction,  
>> where transaction is as defined in [1].
>> "
>>
>
> Raymond, sure. But do the "procedures here" preclude one from  
> choosing the same host given that both the priorities and weights  
> are the same for all the hosts?


well, not exactly, in fact... every so often, you will end up choosing  
the same 1 out of 4 hosts twice in a row at random, but the procedures  
basically say to choose one at random if they're evenly prioritized  
and evenly weighted.... so saying "i'm gonna keep this one for later  
user" kind of goes against "random"

That said, I'm really just about done with this thread since I don't  
personally agree with the spec in this case anyway since stale nonce  
checking makes sense to avoid replay attacks, etc.  Although, I  
haven't looked at the specs to see if checking for stale nonces breaks  
any specs... that might be an interesting search ;-)

Raymond Chandler
     http://freeswitchsolutions.com
     http://cluecon.com
     
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From csa at nowthor.com  Thu Aug 20 15:18:22 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 20 Aug 2009 18:18:22 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <773E8CAB-F146-452D-A3FD-AE202157A2B2@freeswitch.org>
References: <4A8D87D8.8080001@nowthor.com>	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>	<4A8DAA7B.8090601@nowthor.com>	<8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>	<4A8DBA24.6010203@nowthor.com>	<4D73B669-478E-44EF-83D7-0AA745286ACB@freeswitch.org>	<4A8DBF42.9050503@nowthor.com>
	<773E8CAB-F146-452D-A3FD-AE202157A2B2@freeswitch.org>
Message-ID: <4A8DCBAE.60107@nowthor.com>

Raymond Chandler wrote:
>
> On Aug 20, 2009, at 5:25 PM, Carlos S. Antunes wrote:
>
>> Raymond Chandler wrote:
>>> On Aug 20, 2009, at 5:03 PM, Carlos S. Antunes wrote:
>>>> Hmm, where does it say that, after the lookup, one cannot use the same
>>>> IP address as before? :)
>>>
>>> Section 4 of RFC3263 as quoted in my first email....
>>>
>>> "The procedures here MUST be done exactly once per transaction, 
>>> where transaction is as defined in [1]. 
>>> "
>>>
>>
>> Raymond, sure. But do the "procedures here" preclude one from 
>> choosing the same host given that both the priorities and weights are 
>> the same for all the hosts?
>
>
> well, not exactly, in fact... every so often, you will end up choosing 
> the same 1 out of 4 hosts twice in a row at random

Exactly! :)

> but the procedures basically say to choose one at random

Right, but isn't the generator pseudo-random, instead? (If one takes 
"random" literally, pseudo-random would violate the spec!) What prevents 
one from using the "right" pseudo-random generator?


>
> That said, I'm really just about done with this thread since I don't 
> personally agree with the spec in this case anyway since stale nonce 
> checking makes sense to avoid replay attacks, etc.  Although, I 
> haven't looked at the specs to see if checking for stale nonces breaks 
> any specs... that might be an interesting search ;-)
>

I am going to try to find an appropriate IETF mailing list to ask some 
questions about this random SRV stuff and will repost here once I have 
some additional info.

Thanks for trying to keep me honest, though! :)

Carlos Antunes
Nowthor Corporation

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From brian at freeswitch.org  Thu Aug 20 15:24:50 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 20 Aug 2009 17:24:50 -0500
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A8DCBAE.60107@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>	<C685C86C-EC90-4568-8E31-6C05ED44FB53@freeswitch.org>	<4A8DAA7B.8090601@nowthor.com>	<8AE515D7-9C7B-4442-9A9D-DA9BFCD2C7B6@freeswitch.org>	<4A8DBA24.6010203@nowthor.com>	<4D73B669-478E-44EF-83D7-0AA745286ACB@freeswitch.org>	<4A8DBF42.9050503@nowthor.com>
	<773E8CAB-F146-452D-A3FD-AE202157A2B2@freeswitch.org>
	<4A8DCBAE.60107@nowthor.com>
Message-ID: <3CB13AD6-A2A0-448B-821A-5878961E110C@freeswitch.org>

Read RFC 2782, About the significance of a 0 weight vs weighted.  Ray  
proved that if you have your records weighted at 0 it behaves correctly.

/b

On Aug 20, 2009, at 5:18 PM, Carlos S. Antunes wrote:

> I am going to try to find an appropriate IETF mailing list to ask  
> some questions about this random SRV stuff and will repost here once  
> I have some additional info.
>
> Thanks for trying to keep me honest, though! :)




From eric.des.courtis at gmail.com  Thu Aug 20 16:51:01 2009
From: eric.des.courtis at gmail.com (Eric des Courtis)
Date: Thu, 20 Aug 2009 19:51:01 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
	<87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
Message-ID: <539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>

Matt,

For your information the tones you gave me are exactly 738Hz. If you
want to try that tone detection thing.

Cheers.

Eric des Courtis

On Thu, Aug 20, 2009 at 2:20 PM, Michael Collins<msc at freeswitch.org> wrote:
>
>
> On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood <steveu at coppice.org>
> wrote:
>>
>> On 08/20/2009 05:22 AM, Michael Collins wrote:
>> >
>> > ? ? There is no noise on those 3 beeps. In fact, for something that's
>> > been
>> > ? ? through ulaw/alaw compression those beeps are very clean. They are
>> > ? ? quite
>> > ? ? short, though.
>> >
>> >
>> > Heck yeah they're short! Steve, in your experience is there a
>> > practical way to detect a beep that short without chewing up system
>> > resources or having lots of false positives?
>> > -MC
>> >
>> The tone samples I just looked at are about 130ms long. The problem is
>> the detector is trying to be a very open ended detector of anything
>> narrowband enough to be a single tone, and of any duration beyond some
>> small minimum. Its difficult to make such a thing voice immune unless
>> you can also count on a very large signal to noise ratio. With a digital
>> trunk you can probably rely on a large SNR, but what happens when people
>> use analogue lines? There is a reason why DTMF detectors try hard to
>> work down to about 10dB SNR. :-)
>>
>> Steve
>
> Thanks for the lesson uncle Steve! I'm guessing that the OP will need a new
> strategy. Possibly waiting for silence? Not sure what's the best way to go
> but I'm interested in hearing if someone has a solution.
>
> -MC
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From eric.des.courtis at gmail.com  Thu Aug 20 16:54:52 2009
From: eric.des.courtis at gmail.com (Eric des Courtis)
Date: Thu, 20 Aug 2009 19:54:52 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
	<87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
	<539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>
Message-ID: <539da8a30908201654u3f660a43pf32100223fa1863a@mail.gmail.com>

Matt,

As is mod_vmd will not detect tones shorter then 138ms. However I
could get that value down to ~30ms at best by making a few
modifications to the algorithm.

Cheers.

Eric des Courtis


On Thu, Aug 20, 2009 at 7:51 PM, Eric des
Courtis<eric.des.courtis at gmail.com> wrote:
> Matt,
>
> For your information the tones you gave me are exactly 738Hz. If you
> want to try that tone detection thing.
>
> Cheers.
>
> Eric des Courtis
>
> On Thu, Aug 20, 2009 at 2:20 PM, Michael Collins<msc at freeswitch.org> wrote:
>>
>>
>> On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood <steveu at coppice.org>
>> wrote:
>>>
>>> On 08/20/2009 05:22 AM, Michael Collins wrote:
>>> >
>>> > ? ? There is no noise on those 3 beeps. In fact, for something that's
>>> > been
>>> > ? ? through ulaw/alaw compression those beeps are very clean. They are
>>> > ? ? quite
>>> > ? ? short, though.
>>> >
>>> >
>>> > Heck yeah they're short! Steve, in your experience is there a
>>> > practical way to detect a beep that short without chewing up system
>>> > resources or having lots of false positives?
>>> > -MC
>>> >
>>> The tone samples I just looked at are about 130ms long. The problem is
>>> the detector is trying to be a very open ended detector of anything
>>> narrowband enough to be a single tone, and of any duration beyond some
>>> small minimum. Its difficult to make such a thing voice immune unless
>>> you can also count on a very large signal to noise ratio. With a digital
>>> trunk you can probably rely on a large SNR, but what happens when people
>>> use analogue lines? There is a reason why DTMF detectors try hard to
>>> work down to about 10dB SNR. :-)
>>>
>>> Steve
>>
>> Thanks for the lesson uncle Steve! I'm guessing that the OP will need a new
>> strategy. Possibly waiting for silence? Not sure what's the best way to go
>> but I'm interested in hearing if someone has a solution.
>>
>> -MC
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>



From mattdfong at gmail.com  Thu Aug 20 17:36:43 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Thu, 20 Aug 2009 17:36:43 -0700
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <539da8a30908201654u3f660a43pf32100223fa1863a@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171029u4f4eb10bk18f2ef36d9a0bd66@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
	<87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
	<539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>
	<539da8a30908201654u3f660a43pf32100223fa1863a@mail.gmail.com>
Message-ID: <4256bf830908201736v63eb9188if1d82842a885b9d8@mail.gmail.com>

I changed

/*! Minimum time for a beep. */
#define MIN_TIME 8000
to 6500 and it seemed to work, but I'm not sure how many false positives I
will get in a real-world environment. at 4000 it fired the event like 5
times in a session, but 6500 only once. Do you think I should expect a lot
of false positives after changing this value?

--matt
http://www.hellohunter.com

On Thu, Aug 20, 2009 at 4:54 PM, Eric des Courtis <
eric.des.courtis at gmail.com> wrote:

> Matt,
>
> As is mod_vmd will not detect tones shorter then 138ms. However I
> could get that value down to ~30ms at best by making a few
> modifications to the algorithm.
>
> Cheers.
>
> Eric des Courtis
>
>
> On Thu, Aug 20, 2009 at 7:51 PM, Eric des
> Courtis<eric.des.courtis at gmail.com> wrote:
>  > Matt,
> >
> > For your information the tones you gave me are exactly 738Hz. If you
> > want to try that tone detection thing.
> >
> > Cheers.
> >
> > Eric des Courtis
> >
> > On Thu, Aug 20, 2009 at 2:20 PM, Michael Collins<msc at freeswitch.org>
> wrote:
> >>
> >>
> >> On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood <steveu at coppice.org>
> >> wrote:
> >>>
> >>> On 08/20/2009 05:22 AM, Michael Collins wrote:
> >>> >
> >>> >     There is no noise on those 3 beeps. In fact, for something that's
> >>> > been
> >>> >     through ulaw/alaw compression those beeps are very clean. They
> are
> >>> >     quite
> >>> >     short, though.
> >>> >
> >>> >
> >>> > Heck yeah they're short! Steve, in your experience is there a
> >>> > practical way to detect a beep that short without chewing up system
> >>> > resources or having lots of false positives?
> >>> > -MC
> >>> >
> >>> The tone samples I just looked at are about 130ms long. The problem is
> >>> the detector is trying to be a very open ended detector of anything
> >>> narrowband enough to be a single tone, and of any duration beyond some
> >>> small minimum. Its difficult to make such a thing voice immune unless
> >>> you can also count on a very large signal to noise ratio. With a
> digital
> >>> trunk you can probably rely on a large SNR, but what happens when
> people
> >>> use analogue lines? There is a reason why DTMF detectors try hard to
> >>> work down to about 10dB SNR. :-)
> >>>
> >>> Steve
> >>
> >> Thanks for the lesson uncle Steve! I'm guessing that the OP will need a
> new
> >> strategy. Possibly waiting for silence? Not sure what's the best way to
> go
> >> but I'm interested in hearing if someone has a solution.
> >>
> >> -MC
> >>
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >>
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From eric.des.courtis at gmail.com  Thu Aug 20 18:53:25 2009
From: eric.des.courtis at gmail.com (Eric des Courtis)
Date: Thu, 20 Aug 2009 21:53:25 -0400
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4256bf830908201736v63eb9188if1d82842a885b9d8@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
	<87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
	<539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>
	<539da8a30908201654u3f660a43pf32100223fa1863a@mail.gmail.com>
	<4256bf830908201736v63eb9188if1d82842a885b9d8@mail.gmail.com>
Message-ID: <539da8a30908201853s6d4307e7j996849a4388aa3d3@mail.gmail.com>

Matt,

I think the only way to know for sure is to try it. I would try to get
the value as high as possible while still detecting that 738Hz sine
(with a small margin of error). Lowering the value increases false
positives rapidly.

Eric des Courtis

On Thu, Aug 20, 2009 at 8:36 PM, Matthew Fong<mattdfong at gmail.com> wrote:
> I changed
>
> /*! Minimum time for a beep. */
> #define MIN_TIME 8000
> to 6500 and it seemed to work, but I'm not sure how many false positives I
> will get in a real-world environment. at 4000 it fired the event like 5
> times in a session, but 6500 only once. Do you think I should expect a lot
> of false positives after changing this value?
>
> --matt
> http://www.hellohunter.com
>
> On Thu, Aug 20, 2009 at 4:54 PM, Eric des Courtis
> <eric.des.courtis at gmail.com> wrote:
>>
>> Matt,
>>
>> As is mod_vmd will not detect tones shorter then 138ms. However I
>> could get that value down to ~30ms at best by making a few
>> modifications to the algorithm.
>>
>> Cheers.
>>
>> Eric des Courtis
>>
>>
>> On Thu, Aug 20, 2009 at 7:51 PM, Eric des
>> Courtis<eric.des.courtis at gmail.com> wrote:
>> > Matt,
>> >
>> > For your information the tones you gave me are exactly 738Hz. If you
>> > want to try that tone detection thing.
>> >
>> > Cheers.
>> >
>> > Eric des Courtis
>> >
>> > On Thu, Aug 20, 2009 at 2:20 PM, Michael Collins<msc at freeswitch.org>
>> > wrote:
>> >>
>> >>
>> >> On Thu, Aug 20, 2009 at 11:06 AM, Steve Underwood <steveu at coppice.org>
>> >> wrote:
>> >>>
>> >>> On 08/20/2009 05:22 AM, Michael Collins wrote:
>> >>> >
>> >>> > ? ? There is no noise on those 3 beeps. In fact, for something
>> >>> > that's
>> >>> > been
>> >>> > ? ? through ulaw/alaw compression those beeps are very clean. They
>> >>> > are
>> >>> > ? ? quite
>> >>> > ? ? short, though.
>> >>> >
>> >>> >
>> >>> > Heck yeah they're short! Steve, in your experience is there a
>> >>> > practical way to detect a beep that short without chewing up system
>> >>> > resources or having lots of false positives?
>> >>> > -MC
>> >>> >
>> >>> The tone samples I just looked at are about 130ms long. The problem is
>> >>> the detector is trying to be a very open ended detector of anything
>> >>> narrowband enough to be a single tone, and of any duration beyond some
>> >>> small minimum. Its difficult to make such a thing voice immune unless
>> >>> you can also count on a very large signal to noise ratio. With a
>> >>> digital
>> >>> trunk you can probably rely on a large SNR, but what happens when
>> >>> people
>> >>> use analogue lines? There is a reason why DTMF detectors try hard to
>> >>> work down to about 10dB SNR. :-)
>> >>>
>> >>> Steve
>> >>
>> >> Thanks for the lesson uncle Steve! I'm guessing that the OP will need a
>> >> new
>> >> strategy. Possibly waiting for silence? Not sure what's the best way to
>> >> go
>> >> but I'm interested in hearing if someone has a solution.
>> >>
>> >> -MC
>> >>
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From gabe at gundy.org  Thu Aug 20 20:16:42 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 20 Aug 2009 21:16:42 -0600
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <4256bf830908201736v63eb9188if1d82842a885b9d8@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<4256bf830908171052v461cac2fi71ac025bb5ddfb1a@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
	<87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
	<539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>
	<539da8a30908201654u3f660a43pf32100223fa1863a@mail.gmail.com>
	<4256bf830908201736v63eb9188if1d82842a885b9d8@mail.gmail.com>
Message-ID: <903da5680908202016w2e146c79q408146001faeecfe@mail.gmail.com>

On Thu, Aug 20, 2009 at 6:36 PM, Matthew Fong<mattdfong at gmail.com> wrote:
> /*! Minimum time for a beep. */
> #define MIN_TIME 8000
> to 6500 and it seemed to work, but I'm not sure how many false positives I
> will get in a real-world environment. at 4000 it fired the event like 5
> times in a session, but 6500 only once. Do you think I should expect a lot
> of false positives after changing this value?

YES.



From gabe at gundy.org  Thu Aug 20 20:17:02 2009
From: gabe at gundy.org (Gabriel Gunderson)
Date: Thu, 20 Aug 2009 21:17:02 -0600
Subject: [Freeswitch-users] Better results from mod_vmd
In-Reply-To: <903da5680908202016w2e146c79q408146001faeecfe@mail.gmail.com>
References: <4256bf830908162126ld91f38et1dff8f66f2901ab6@mail.gmail.com>
	<539da8a30908171415m6b7daf3cvcb6c37bf084104dd@mail.gmail.com>
	<4A8C5F21.8080401@coppice.org>
	<87f2f3b90908191422y17dc0431i1cf751afbbac3307@mail.gmail.com>
	<4A8D90BE.2090007@coppice.org>
	<87f2f3b90908201120i6a47c052j5d8f900602df088a@mail.gmail.com>
	<539da8a30908201651t630fac59o9f5bd3262256789@mail.gmail.com>
	<539da8a30908201654u3f660a43pf32100223fa1863a@mail.gmail.com>
	<4256bf830908201736v63eb9188if1d82842a885b9d8@mail.gmail.com>
	<903da5680908202016w2e146c79q408146001faeecfe@mail.gmail.com>
Message-ID: <903da5680908202017w5babe925uea912a36b63d9c2c@mail.gmail.com>

On Thu, Aug 20, 2009 at 9:16 PM, Gabriel Gunderson<gabe at gundy.org> wrote:
> On Thu, Aug 20, 2009 at 6:36 PM, Matthew Fong<mattdfong at gmail.com> wrote:
>> /*! Minimum time for a beep. */
>> #define MIN_TIME 8000
>> to 6500 and it seemed to work, but I'm not sure how many false positives I
>> will get in a real-world environment. at 4000 it fired the event like 5
>> times in a session, but 6500 only once. Do you think I should expect a lot
>> of false positives after changing this value?
>
> YES.

Err, NO.



From ahmedmunir007 at gmail.com  Thu Aug 20 21:54:44 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Fri, 21 Aug 2009 10:54:44 +0600
Subject: [Freeswitch-users] Application Variable list needed
Message-ID: <a60fed5e0908202154pd0bb1f6ga4a060be8b8e5c34@mail.gmail.com>

Hi,
I'm newbie in FreeSwitch, currently I'm replacing asterisk with FreeSwitch.
I need list of application variables so I can easily translate my asterisk
configuration in FreeSwitch. Like I don't know how I translate my dial plan
in FreeSwitch as I like to forward it to another context as sample listed
below;

exten => _X.,1,Goto(origination_incoming,${EXTEN},1)
exten => _X.,n,Hangup()

My another request is how can I change extensions.ael in asterisk to
FreeSwitch? Is there alternative way of doing this? Kindly do let me know.

-- 
Regards,

Ahmed Munir
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From diego.viola at gmail.com  Thu Aug 20 22:35:13 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 21 Aug 2009 01:35:13 -0400
Subject: [Freeswitch-users] Application Variable list needed
In-Reply-To: <a60fed5e0908202154pd0bb1f6ga4a060be8b8e5c34@mail.gmail.com>
References: <a60fed5e0908202154pd0bb1f6ga4a060be8b8e5c34@mail.gmail.com>
Message-ID: <86a32abc0908202235x6ca9cbdap8a0574429da46da@mail.gmail.com>

For dialplan apps take a look at.

http://wiki.freeswitch.org/wiki/Mod_dptools

For channel variables:

http://wiki.freeswitch.org/wiki/Channel_Variables

For FS commands:

http://wiki.freeswitch.org/wiki/Mod_commands

You could use transfer in order to send a call to another extension/context.


On Fri, Aug 21, 2009 at 12:54 AM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:

> Hi,
> I'm newbie in FreeSwitch, currently I'm replacing asterisk with FreeSwitch.
> I need list of application variables so I can easily translate my asterisk
> configuration in FreeSwitch. Like I don't know how I translate my dial plan
> in FreeSwitch as I like to forward it to another context as sample listed
> below;
>
> exten => _X.,1,Goto(origination_incoming,${EXTEN},1)
> exten => _X.,n,Hangup()
>
> My another request is how can I change extensions.ael in asterisk to
> FreeSwitch? Is there alternative way of doing this? Kindly do let me know.
>
> --
> Regards,
>
> Ahmed Munir
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Thu Aug 20 22:51:05 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 21 Aug 2009 01:51:05 -0400
Subject: [Freeswitch-users] Application Variable list needed
In-Reply-To: <86a32abc0908202235x6ca9cbdap8a0574429da46da@mail.gmail.com>
References: <a60fed5e0908202154pd0bb1f6ga4a060be8b8e5c34@mail.gmail.com>
	<86a32abc0908202235x6ca9cbdap8a0574429da46da@mail.gmail.com>
Message-ID: <86a32abc0908202251y68ee91fhb94f5b5718ba4597@mail.gmail.com>

Let us know if you have more questions or need more help :)

On Fri, Aug 21, 2009 at 1:35 AM, Diego Viola <diego.viola at gmail.com> wrote:

> For dialplan apps take a look at.
>
> http://wiki.freeswitch.org/wiki/Mod_dptools
>
> For channel variables:
>
> http://wiki.freeswitch.org/wiki/Channel_Variables
>
> For FS commands:
>
> http://wiki.freeswitch.org/wiki/Mod_commands
>
> You could use transfer in order to send a call to another
> extension/context.
>
>
> On Fri, Aug 21, 2009 at 12:54 AM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:
>
>> Hi,
>> I'm newbie in FreeSwitch, currently I'm replacing asterisk with
>> FreeSwitch. I need list of application variables so I can easily translate
>> my asterisk configuration in FreeSwitch. Like I don't know how I translate
>> my dial plan in FreeSwitch as I like to forward it to another context as
>> sample listed below;
>>
>> exten => _X.,1,Goto(origination_incoming,${EXTEN},1)
>> exten => _X.,n,Hangup()
>>
>> My another request is how can I change extensions.ael in asterisk to
>> FreeSwitch? Is there alternative way of doing this? Kindly do let me know.
>>
>> --
>> Regards,
>>
>> Ahmed Munir
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From b_ball_henry at hotmail.com  Fri Aug 21 01:06:24 2009
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Fri, 21 Aug 2009 16:06:24 +0800
Subject: [Freeswitch-users] Dialing SIP URL issue
Message-ID: <59ad9ca10908210106g2541ac79x4e511be8820ffa20@mail.gmail.com>

Hi:

I try to dial sip url from my softphone but seems like the sip address is
being processed by sofia before it pass to the dialplan. The example here is
:

*X-lite(softphone) dials -> 1009 at 4.2.2.2 (it's fake sip address, the purpose
was just to test what's being passed to dialplan)
sofia receives the invite and return with trying
sofia pass the destination number to dailplan with "1009" (without the
"sip:" in front and without the "@4.2.2.2" after it)
*
Please see pastebin for full log. http://pastebin.freeswitch.org/10089
ignore anything after line 80, because it's not my point, and the
destination is a fake address.

I would like to know how do you actually pass a full sip url to the dialplan
to do the regex match. Because from the default.xml dialplan, it comes with
an example sip url dialing extension that match's *^sip:(.*)$ *. So I assume
there must be a way of passing full sip url to the dialplan. Here is the
example dialplan expecting sofia to pass it a full sip url:

 <!-- dial via SIP uri -->
    <extension name="sip_uri">
      <condition field="destination_number" expression="^sip:(.*)$">
        <action application="bridge" data="sofia/${use_profile}/$1"/>
      </condition>
    </extension>

Thanks
-- 
Henry Huang
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From woodydickson at gmail.com  Fri Aug 21 02:54:09 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Fri, 21 Aug 2009 17:54:09 +0800
Subject: [Freeswitch-users] zombie channels
Message-ID: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>

Hi,

I am running 1.0.4 right now using latest trunk.

After a high traffic session, I do "show channels", I would find a bunch of
"CS_HIBERNATE" channels that don't get removed after all the traffic is
gone.

Does anyone know what is the case of thoes CS_HIBERNATE'd channels?  How can
I set  a timeout for those channels to be removed?

Thanks,
Woody
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From dave at 3c.co.uk  Fri Aug 21 03:07:04 2009
From: dave at 3c.co.uk (David Knell)
Date: Fri, 21 Aug 2009 13:07:04 +0300
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
References: <4A8D5132.7010807@gmx.net>
	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>
	<4A8D5D43.3030300@gmx.net>
	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>
	<4A8DB808.1030903@gmx.net>
	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
Message-ID: <1250849224.4663.68.camel@dk-d820>

On Thu, 2009-08-20 at 14:35 -0700, Michael Collins wrote:


> Just curious - if it seems to be working with Asterisk but not
> FreeSWITCH then could you do some tcpdumps of working vs. non-working
> calls and then analyze them with Wireshark? I think Jason Garland's
> ClueCon presentation(s) might be applicable here. 

Just to deepen the mystery a little, we have a FRITZ!Box here in Greece,
and it works like a little champ for us.  Firmware's 06.04.49, it's
talking to a FreeSWITCH box in London.  It's set to pick its own codec
(but the other end only supports G.711), VAD's off.

Cheers --

Dave






From fraunhofer.lists.freeswitch-001 at traced.net  Fri Aug 21 04:03:07 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Fri, 21 Aug 2009 13:03:07 +0200
Subject: [Freeswitch-users] zombie channels
In-Reply-To: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
References: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
Message-ID: <a37510170908210403o66aebaffw7cf913fec6e3615@mail.gmail.com>

Hi Woody,

2009/8/21 Woody Dickson <woodydickson at gmail.com>:

> After a high traffic session, I do "show channels", I would find a bunch of
> "CS_HIBERNATE" channels that don't get removed after all the traffic is
> gone.
>
> Does anyone know what is the case of thoes CS_HIBERNATE'd channels?? How can
> I set? a timeout for those channels to be removed?

Is this simiilar to "my" bugreport?

http://jira.freeswitch.org/browse/FSCORE-415

what do you do in your "high traffic"-scenario? We've got tons of
CS_HIBERNATE channels and even the sip stack is somehow affected
(ignoring messages) but it looks like it's the scheduler to blame.
In your example this would mean that the "garbage collection" for
finished sessions is not run.
Can you get rid of the channels using "uuid_kill" or a sweeping swipe
"fsctl hupall"?

Greetinx
  Beni.



From tparikh at gmail.com  Thu Aug 20 22:30:33 2009
From: tparikh at gmail.com (Tapan Parikh)
Date: Thu, 20 Aug 2009 22:30:33 -0700
Subject: [Freeswitch-users] Accepting google talk friend requests
Message-ID: <1ecdcb6a0908202230s7324790bvf0c5adae7397b4a2@mail.gmail.com>

Hi Folks -

Im a relative newbie freeswitch and first of all wanted to thank you
for all the great work u have done here.

My question is about mod dingaling, and specifically being able to get
incoming calls from Google Talk.  Ive got the client set up on
Freeswitch, and am able to receive calls, IMs, etc. once I accept the
friend invite.

However, new accounts cannot talk to FS until they are already
friends.  Is there a way to get FS / mod dingaling to automatically
accept incoming friend requests?

I saw something in the archives about this, but no obvious resolution.
 People seem to say this is already true, but I dont even see the
incoming invite requests coming in w/ DL in debug mode.

Thanks in advance,
Tapan



From panayotov.vd at gmail.com  Thu Aug 20 23:45:58 2009
From: panayotov.vd at gmail.com (Vassil Panayotov)
Date: Fri, 21 Aug 2009 09:45:58 +0300
Subject: [Freeswitch-users] Sangoma A500 and FreeSWITCH
Message-ID: <8a9b664c0908202345g6dd56154v2abba5d76a9e663e@mail.gmail.com>

Hi,

Is it already possible to use FreeSWITCH with A500 BRI card? I found a
thread (
http://www.mail-archive.com/freeswitch-users at lists.freeswitch.org/msg09633.html)
stating that the support is not quite ready.

Can you give me some pointers about A500 configuration with FreeSWITCH? I
need nothing fancy - just inbound/outbound calls to telco.

Best regards,
V. Panayotov
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From odermann at googlemail.com  Fri Aug 21 05:46:56 2009
From: odermann at googlemail.com (Dennis)
Date: Fri, 21 Aug 2009 14:46:56 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup cause
Message-ID: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>

hi,

we are using fs for different services, but we never used it, to
connect sip-phones directly with fs.

now we want to do so, but we encounter big problems. everything works
fine, but after 120 seconds fs hangs up with the hangup cause
RECOVERY_ON_TIMER_EXPIRE.

it seems that this has something to do with nat problems (i read about
it in nabble), but we can not figure out, how to fix the problem.

we are not using stun, because all ip-addresses are static. we opened
our firewall for our ip-addresses and opened the following ports:
5060 - 5091 TCP / UDP
10000 - 32767 UDP

we are using a quite actual version of fs - the revision ist 13783.

we set up a profile in the internal.xml and played a lot with
different settings. we can talk to each other, make outbound calls and
receive calls - but we can not figure out, how to avoid the hangup
after 120 seconds.

thanks a lot for your help
dennis



From chris.chen2004 at gmail.com  Fri Aug 21 06:16:50 2009
From: chris.chen2004 at gmail.com (Chris Chen)
Date: Fri, 21 Aug 2009 09:16:50 -0400
Subject: [Freeswitch-users] Accepting google talk friend requests
In-Reply-To: <1ecdcb6a0908202230s7324790bvf0c5adae7397b4a2@mail.gmail.com>
References: <1ecdcb6a0908202230s7324790bvf0c5adae7397b4a2@mail.gmail.com>
Message-ID: <507898380908210616p4a0f9f29v98287286fa108d6a@mail.gmail.com>

Hi Tapan, if your google talk is loaded as client mode in FreeSWITCH, you
cannot automatically accept new incoming
invite requests. But if you have google talk (mod_dingaling) loaded as
component mode (assuming you have your
own jabber server setup properly with google federation  etc, and loaded
with server.xml), for the invite requests to some
FreeSWITCH built-in accounts such as "user+bla blah at your jabber server"
"ext+blah blah@ your jabber server",
they will be automatically accepted.

Hope this helps.

Chris


On Fri, Aug 21, 2009 at 1:30 AM, Tapan Parikh <tparikh at gmail.com> wrote:

> Hi Folks -
>
> Im a relative newbie freeswitch and first of all wanted to thank you
> for all the great work u have done here.
>
> My question is about mod dingaling, and specifically being able to get
> incoming calls from Google Talk.  Ive got the client set up on
> Freeswitch, and am able to receive calls, IMs, etc. once I accept the
> friend invite.
>
> However, new accounts cannot talk to FS until they are already
> friends.  Is there a way to get FS / mod dingaling to automatically
> accept incoming friend requests?
>
> I saw something in the archives about this, but no obvious resolution.
>  People seem to say this is already true, but I dont even see the
> incoming invite requests coming in w/ DL in debug mode.
>
> Thanks in advance,
> Tapan
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Aug 21 06:20:42 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 21 Aug 2009 08:20:42 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
Message-ID: <B30F06D6-3D0E-4405-B63B-907246EEEEC8@freeswitch.org>

You have a NAT issue.

/b

On Aug 21, 2009, at 7:46 AM, Dennis wrote:

> hi,
>
> we are using fs for different services, but we never used it, to
> connect sip-phones directly with fs.
>
> now we want to do so, but we encounter big problems. everything works
> fine, but after 120 seconds fs hangs up with the hangup cause
> RECOVERY_ON_TIMER_EXPIRE.
>
> it seems that this has something to do with nat problems (i read about
> it in nabble), but we can not figure out, how to fix the problem.
>
> we are not using stun, because all ip-addresses are static. we opened
> our firewall for our ip-addresses and opened the following ports:
> 5060 - 5091 TCP / UDP
> 10000 - 32767 UDP
>
> we are using a quite actual version of fs - the revision ist 13783.
>
> we set up a profile in the internal.xml and played a lot with
> different settings. we can talk to each other, make outbound calls and
> receive calls - but we can not figure out, how to avoid the hangup
> after 120 seconds.
>
> thanks a lot for your help
> dennis
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mrene_lists at avgs.ca  Fri Aug 21 06:31:30 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 21 Aug 2009 09:31:30 -0400
Subject: [Freeswitch-users] zombie channels
In-Reply-To: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
References: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
Message-ID: <87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>

Hi,

CS_REPORTING is the state in which cdrs are written, if the channel  
gets stuck in that state, the cdr module you are using is probably  
hanging somewhere.

Use the "freeswitch-gcore" script in your source tree's scripts  
directory to generate a bug report for hanging channels.

should be like..

cd /usr/src/freeswitch # or whatever your source tree is
bash ./scripts/freeswitch-gcore > bugreport.txt

then submit it on http://jira.freeswitch.org/ so we can look at it.

As you wish, you can also hop on #freeswitch / irc.freenode.net and  
have someone look into it.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 21-Aug-09, at 5:54 AM, Woody Dickson wrote:

> Hi,
>
> I am running 1.0.4 right now using latest trunk.
>
> After a high traffic session, I do "show channels", I would find a  
> bunch of "CS_HIBERNATE" channels that don't get removed after all  
> the traffic is gone.
>
> Does anyone know what is the case of thoes CS_HIBERNATE'd channels?   
> How can I set  a timeout for those channels to be removed?
>
> Thanks,
> Woody
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From asannucci at gmail.com  Fri Aug 21 06:35:26 2009
From: asannucci at gmail.com (bakko)
Date: Fri, 21 Aug 2009 15:35:26 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
Message-ID: <E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>

Do you have those lines in switch.conf file?

    <!--RTP port range -->
    <param name="rtp-start-port" value="10000"/>
    <param name="rtp-end-port" value="32767"/>


BR



From moises.silva at gmail.com  Fri Aug 21 07:26:56 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 21 Aug 2009 10:26:56 -0400
Subject: [Freeswitch-users] Sangoma A500 and FreeSWITCH
In-Reply-To: <8a9b664c0908202345g6dd56154v2abba5d76a9e663e@mail.gmail.com>
References: <8a9b664c0908202345g6dd56154v2abba5d76a9e663e@mail.gmail.com>
Message-ID: <c4d05cbe0908210726j3b643fcapd3901fa5d92f28f6@mail.gmail.com>

>
> Can you give me some pointers about A500 configuration with FreeSWITCH? I
> need nothing fancy - just inbound/outbound calls to telco.
>

There is some progress and you should be able to get it working with this:
http://wiki.sangoma.com/boostbri

If you find any problem please contact techdesk at sangoma.com and they will
take care of helping you to get you up and running.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From msc at freeswitch.org  Fri Aug 21 08:06:56 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Fri, 21 Aug 2009 08:06:56 -0700
Subject: [Freeswitch-users] Application Variable list needed
In-Reply-To: <a60fed5e0908202154pd0bb1f6ga4a060be8b8e5c34@mail.gmail.com>
References: <a60fed5e0908202154pd0bb1f6ga4a060be8b8e5c34@mail.gmail.com>
Message-ID: <5C49E69C-5C6E-4B26-9845-CAB0CE765705@freeswitch.org>

Be sure to visit wiki.freeswitch.org and search for "rosetta stone"  
which will take you to a page that helps translate Asterisk concepts  
into FreeSWITCH concepts.

It will seem strange at first but once you get over the hump you will  
really appreciate the power of FreeSWITCH.  Be sure to visit the IRC  
channel for a friendly place to ask questions.

-MC

Sent from my iPhone

On Aug 20, 2009, at 9:54 PM, Ahmed Munir <ahmedmunir007 at gmail.com>  
wrote:

> Hi,
>
> I'm newbie in FreeSwitch, currently I'm replacing asterisk with  
> FreeSwitch. I need list of application variables so I can easily  
> translate my asterisk configuration in FreeSwitch. Like I don't know  
> how I translate my dial plan in FreeSwitch as I like to forward it  
> to another context as sample listed below;
>
> exten => _X.,1,Goto(origination_incoming,${EXTEN},1)
> exten => _X.,n,Hangup()
>
> My another request is how can I change extensions.ael in asterisk to  
> FreeSwitch? Is there alternative way of doing this? Kindly do let me  
> know.
>
> -- 
> Regards,
>
> Ahmed Munir
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From Prometheus001 at gmx.net  Fri Aug 21 08:38:31 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 21 Aug 2009 17:38:31 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>	<4A8DB808.1030903@gmx.net>
	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
Message-ID: <4A8EBF77.1080204@gmx.net>

Hello Michael,

I made some tests with Freeswitch and Fritzbox and found by Wireshark that:
within one call

    * Freeswitch starts sending 20msec packets, then after ~0,2 second
      sends 30msec packets
    * FritzBox always sends 30msec packets.

The average jitter is below 2 msec in both directions.
 
The below logs shows that Freeswitch considers the FritzBox to be broken
and starts using 30msec packets. But there is no SIP message from FS to
Fritzbox telling him that FB will use 30msec packets. SDP from FS to
Fritzbox always shows ptime:20

BTW: We can ship you a FritzBox if you need one for testing.

Best regards
Peter


Log:

2009-08-21 17:15:50.271555 [DEBUG] switch_rtp.c:1138 Starting timer
[soft] 160 bytes per 20ms
2009-08-21 17:15:50.281563 [INFO] mod_sofia.c:1499 Ring SDP:
v=0
o=FreeSWITCH 1250837460 1250837461 IN IP4 182.xxx.xx.xxx
s=FreeSWITCH
c=IN IP4 182.xxx.xx.xxx
t=0 0
m=audio 30290 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-08-21 17:15:50.281563 [NOTICE] mod_sofia.c:1502 Pre-Answer
sofia/internal/02xxxxxxxxx at fs1.my.domain!
2009-08-21 17:15:50.281563 [DEBUG] switch_core_session.c:630 Send signal
sofia/internal/02xxxxxxxxx at fs1.my.domain [BREAK]
EXECUTE sofia/internal/02xxxxxxxxx at fs1.my.domain
playback(voicemail/8000/vm-that_was_an_invalid_ext.wav)
2009-08-21 17:15:50.281563 [DEBUG] switch_ivr_play_say.c:1097 Codec
Activated L16 at 8000hz 1 channels 20ms
2009-08-21 17:15:50.281563 [DEBUG] sofia.c:3302 Channel
sofia/internal/02xxxxxxxxx at fs1.my.domain entering state [early][183]
2009-08-21 17:15:50.281563 [DEBUG] switch_core_io.c:649
sofia/internal/02xxxxxxxxx at fs1.my.domain receive message
[TRANSCODING_NECESSARY]
2009-08-21 17:15:50.531551 [WARNING] mod_sofia.c:799 We were told to use
ptime 20 but what they meant to say was 30
This issue has so far been identified to happen on the following broken
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken
who knows what will happen..








From markmorreny at gmail.com  Fri Aug 21 08:41:13 2009
From: markmorreny at gmail.com (mark morreny)
Date: Fri, 21 Aug 2009 23:41:13 +0800
Subject: [Freeswitch-users] hanging problem with switch_scheduler_add_task
Message-ID: <20ad6b920908210841l453b2833v9ac12612c174820b@mail.gmail.com>

Hi,

I am experiencing some hanging when fs is executing
switch_scheduler_add_task.

switch_scheduler_add_task(switch_epoch_time_now(NULL) , data_flush_callback,
"data_flush","core",0,NULL,SSHF_NONE|SSHF_NO_DEL);

I place switch_scheduler_add_task in my SWITCH_MODULE_LOAD_FUNCTION and
sometimes, hanging occurs on that particular line.  My data_flush_callback
is as follows.  I debug the module and it does not even enter
data_flush_callback.  It is hanging at switch_scheduler_add_task().

SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {

    switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "starting to
flush cdr record...\n");

    int last_sequence = flush_data_to_csv();
    if (last_sequence > -1 ) {

        update_last_seq(last_sequence);
    }
    task->runtime = switch_epoch_time_now(NULL) + globals.cycle_time;

}

Does anyone know why?

Thanks,
Mark
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From panayotov.vd at gmail.com  Fri Aug 21 08:56:56 2009
From: panayotov.vd at gmail.com (Vassil Panayotov)
Date: Fri, 21 Aug 2009 18:56:56 +0300
Subject: [Freeswitch-users] Sangoma A500 and FreeSWITCH
Message-ID: <8a9b664c0908210856m3b90f9e1n233b7fe7eb9467a9@mail.gmail.com>

I already tried the boostbri config, but with no success...
I will contact the support.

Thank you!

Best regards,
V. Panayotov
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From mrene_lists at avgs.ca  Fri Aug 21 08:59:04 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 21 Aug 2009 11:59:04 -0400
Subject: [Freeswitch-users] hanging problem with
	switch_scheduler_add_task
In-Reply-To: <20ad6b920908210841l453b2833v9ac12612c174820b@mail.gmail.com>
References: <20ad6b920908210841l453b2833v9ac12612c174820b@mail.gmail.com>
Message-ID: <B4BB7488-5C31-4447-BAD7-EAA062C467CC@avgs.ca>

Means another task is hanging, do a "thread apply all bt" in gdb and  
look for scheduler tasks.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 21-Aug-09, at 11:41 AM, mark morreny wrote:

> Hi,
>
> I am experiencing some hanging when fs is executing  
> switch_scheduler_add_task.
>
> switch_scheduler_add_task(switch_epoch_time_now(NULL) ,  
> data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE| 
> SSHF_NO_DEL);
>
> I place switch_scheduler_add_task in my SWITCH_MODULE_LOAD_FUNCTION  
> and sometimes, hanging occurs on that particular line.  My  
> data_flush_callback is as follows.  I debug the module and it does  
> not even enter data_flush_callback.  It is hanging at  
> switch_scheduler_add_task().
>
> SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>
>     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,  
> "starting to flush cdr record...\n");
>
>     int last_sequence = flush_data_to_csv();
>     if (last_sequence > -1 ) {
>
>         update_last_seq(last_sequence);
>     }
>     task->runtime = switch_epoch_time_now(NULL) + globals.cycle_time;
>
> }
>
> Does anyone know why?
>
> Thanks,
> Mark
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From mrene_lists at avgs.ca  Fri Aug 21 09:06:23 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Fri, 21 Aug 2009 12:06:23 -0400
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4A8EBF77.1080204@gmx.net>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>	<4A8DB808.1030903@gmx.net>
	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
	<4A8EBF77.1080204@gmx.net>
Message-ID: <50DA88E6-832E-4596-851D-CB8FFFC54EAA@avgs.ca>

Try setting that in your sip profile:

<param name="rtp-autofix-timing" value="false" />

Thats a feature to work around with devices lying about their ptime in  
their sdp payload.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 21-Aug-09, at 11:38 AM, Peter P GMX wrote:

> Hello Michael,
>
> I made some tests with Freeswitch and Fritzbox and found by  
> Wireshark that:
> within one call
>
>    * Freeswitch starts sending 20msec packets, then after ~0,2 second
>      sends 30msec packets
>    * FritzBox always sends 30msec packets.
>
> The average jitter is below 2 msec in both directions.
>
> The below logs shows that Freeswitch considers the FritzBox to be  
> broken
> and starts using 30msec packets. But there is no SIP message from FS  
> to
> Fritzbox telling him that FB will use 30msec packets. SDP from FS to
> Fritzbox always shows ptime:20
>
> BTW: We can ship you a FritzBox if you need one for testing.
>
> Best regards
> Peter
>
>
> Log:
>
> 2009-08-21 17:15:50.271555 [DEBUG] switch_rtp.c:1138 Starting timer
> [soft] 160 bytes per 20ms
> 2009-08-21 17:15:50.281563 [INFO] mod_sofia.c:1499 Ring SDP:
> v=0
> o=FreeSWITCH 1250837460 1250837461 IN IP4 182.xxx.xx.xxx
> s=FreeSWITCH
> c=IN IP4 182.xxx.xx.xxx
> t=0 0
> m=audio 30290 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 2009-08-21 17:15:50.281563 [NOTICE] mod_sofia.c:1502 Pre-Answer
> sofia/internal/02xxxxxxxxx at fs1.my.domain!
> 2009-08-21 17:15:50.281563 [DEBUG] switch_core_session.c:630 Send  
> signal
> sofia/internal/02xxxxxxxxx at fs1.my.domain [BREAK]
> EXECUTE sofia/internal/02xxxxxxxxx at fs1.my.domain
> playback(voicemail/8000/vm-that_was_an_invalid_ext.wav)
> 2009-08-21 17:15:50.281563 [DEBUG] switch_ivr_play_say.c:1097 Codec
> Activated L16 at 8000hz 1 channels 20ms
> 2009-08-21 17:15:50.281563 [DEBUG] sofia.c:3302 Channel
> sofia/internal/02xxxxxxxxx at fs1.my.domain entering state [early][183]
> 2009-08-21 17:15:50.281563 [DEBUG] switch_core_io.c:649
> sofia/internal/02xxxxxxxxx at fs1.my.domain receive message
> [TRANSCODING_NECESSARY]
> 2009-08-21 17:15:50.531551 [WARNING] mod_sofia.c:799 We were told to  
> use
> ptime 20 but what they meant to say was 30
> This issue has so far been identified to happen on the following  
> broken
> platforms/devices:
> Linksys/Sipura aka Cisco
> ShoreTel
> Sonus/L3
> We will try to fix it but some of the devices on this list are so  
> broken
> who knows what will happen..
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From markmorreny at gmail.com  Fri Aug 21 09:24:08 2009
From: markmorreny at gmail.com (mark morreny)
Date: Sat, 22 Aug 2009 00:24:08 +0800
Subject: [Freeswitch-users] hanging problem with
	switch_scheduler_add_task
In-Reply-To: <B4BB7488-5C31-4447-BAD7-EAA062C467CC@avgs.ca>
References: <20ad6b920908210841l453b2833v9ac12612c174820b@mail.gmail.com>
	<B4BB7488-5C31-4447-BAD7-EAA062C467CC@avgs.ca>
Message-ID: <20ad6b920908210924y7e11c91di9613ce74fa18ef35@mail.gmail.com>

Hi Mathieu,

Thanks for your help.

How can I intentionally crash a hanging freeswitch and obtain the core file
to run gdb?

Thanks,
Mark

On Fri, Aug 21, 2009 at 11:59 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Means another task is hanging, do a "thread apply all bt" in gdb and
> look for scheduler tasks.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 21-Aug-09, at 11:41 AM, mark morreny wrote:
>
> > Hi,
> >
> > I am experiencing some hanging when fs is executing
> > switch_scheduler_add_task.
> >
> > switch_scheduler_add_task(switch_epoch_time_now(NULL) ,
> > data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|
> > SSHF_NO_DEL);
> >
> > I place switch_scheduler_add_task in my SWITCH_MODULE_LOAD_FUNCTION
> > and sometimes, hanging occurs on that particular line.  My
> > data_flush_callback is as follows.  I debug the module and it does
> > not even enter data_flush_callback.  It is hanging at
> > switch_scheduler_add_task().
> >
> > SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
> >
> >     switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,
> > "starting to flush cdr record...\n");
> >
> >     int last_sequence = flush_data_to_csv();
> >     if (last_sequence > -1 ) {
> >
> >         update_last_seq(last_sequence);
> >     }
> >     task->runtime = switch_epoch_time_now(NULL) + globals.cycle_time;
> >
> > }
> >
> > Does anyone know why?
> >
> > Thanks,
> > Mark
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From woodydickson at gmail.com  Fri Aug 21 09:31:34 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 22 Aug 2009 00:31:34 +0800
Subject: [Freeswitch-users] zombie channels
In-Reply-To: <87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>
References: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
	<87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>
Message-ID: <a44688a0908210931ocafd373qc0465f8f24d35335@mail.gmail.com>

Hello,

Yes, I am using cdr, so I guess CS_REPORTING could be a problem.

I tried running the core, but I am getting some errors:
 ./freeswitch-gcore
/usr/local/freeswitch/log/freeswitch.gcore.fm5478:1: Error in sourced
command file:
ptrace: No such process.
gcore: failed to create /usr/local/freeswitch/log/freeswitch.gcore.16240

What is the proper way of using freeswitch-gcore?

Thanks,
Woody


On Fri, Aug 21, 2009 at 9:31 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> CS_REPORTING is the state in which cdrs are written, if the channel
> gets stuck in that state, the cdr module you are using is probably
> hanging somewhere.
>
> Use the "freeswitch-gcore" script in your source tree's scripts
> directory to generate a bug report for hanging channels.
>
> should be like..
>
> cd /usr/src/freeswitch # or whatever your source tree is
> bash ./scripts/freeswitch-gcore > bugreport.txt
>
> then submit it on http://jira.freeswitch.org/ so we can look at it.
>
> As you wish, you can also hop on #freeswitch / irc.freenode.net and
> have someone look into it.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 21-Aug-09, at 5:54 AM, Woody Dickson wrote:
>
> > Hi,
> >
> > I am running 1.0.4 right now using latest trunk.
> >
> > After a high traffic session, I do "show channels", I would find a
> > bunch of "CS_HIBERNATE" channels that don't get removed after all
> > the traffic is gone.
> >
> > Does anyone know what is the case of thoes CS_HIBERNATE'd channels?
> > How can I set  a timeout for those channels to be removed?
> >
> > Thanks,
> > Woody
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fraunhofer.lists.freeswitch-001 at traced.net  Fri Aug 21 10:02:31 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Fri, 21 Aug 2009 19:02:31 +0200
Subject: [Freeswitch-users] zombie channels
In-Reply-To: <a44688a0908210931ocafd373qc0465f8f24d35335@mail.gmail.com>
References: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
	<87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>
	<a44688a0908210931ocafd373qc0465f8f24d35335@mail.gmail.com>
Message-ID: <a37510170908211002h4ec4b4fek44518118d85e2594@mail.gmail.com>

Hello,

> I tried running the core, but I am getting some errors:
> ?./freeswitch-gcore
> /usr/local/freeswitch/log/freeswitch.gcore.fm5478:1: Error in sourced
> command file:
> ptrace: No such process.
> gcore: failed to create /usr/local/freeswitch/log/freeswitch.gcore.16240
>
> What is the proper way of using freeswitch-gcore?

I had the same problem... look at the source, it expects the pid file to be in
/usr/local/freeswitch/log/freeswitch.pid
and freeswitch to be installed at
/usr/local/freeswitch/bin/freeswitch

you can do that manualle, tough.

get the pid of freeswitch (something like "ps waux | grep freeswitch")
then run
gcore -o freeswitch_coredump1.core [pid found above]
gdb [location of freeswitch binary] -c freeswitch_coredump1.core \
        --eval-command='set pagination off' \
        --eval-command='bt' \
        --eval-command='bt full' \
        --eval-command='thread apply all bt' \
        --eval-command='thread apply all bt full' \
        --eval-command='quit' \
       >  freeswitch_report.txt

Beni.



From rupa at rupa.com  Fri Aug 21 10:15:26 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Fri, 21 Aug 2009 12:15:26 -0500
Subject: [Freeswitch-users] hanging problem with
	switch_scheduler_add_task
In-Reply-To: <20ad6b920908210924y7e11c91di9613ce74fa18ef35@mail.gmail.com>
References: <20ad6b920908210841l453b2833v9ac12612c174820b@mail.gmail.com>
	<B4BB7488-5C31-4447-BAD7-EAA062C467CC@avgs.ca>
	<20ad6b920908210924y7e11c91di9613ce74fa18ef35@mail.gmail.com>
Message-ID: <b39273530908211015u117dc01ep982f720334e3cdf9@mail.gmail.com>

Use gcore to get a core dump.  It will pause the process for the
duration of the dump, but will not kill the process.

On Fri, Aug 21, 2009 at 11:24 AM, mark morreny<markmorreny at gmail.com> wrote:
> Hi Mathieu,
>
> Thanks for your help.
>
> How can I intentionally crash a hanging freeswitch and obtain the core file
> to run gdb?
>
> Thanks,
> Mark
>
> On Fri, Aug 21, 2009 at 11:59 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>>
>> Means another task is hanging, do a "thread apply all bt" in gdb and
>> look for scheduler tasks.
>>
>> Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 21-Aug-09, at 11:41 AM, mark morreny wrote:
>>
>> > Hi,
>> >
>> > I am experiencing some hanging when fs is executing
>> > switch_scheduler_add_task.
>> >
>> > switch_scheduler_add_task(switch_epoch_time_now(NULL) ,
>> > data_flush_callback, "data_flush","core",0,NULL,SSHF_NONE|
>> > SSHF_NO_DEL);
>> >
>> > I place switch_scheduler_add_task in my SWITCH_MODULE_LOAD_FUNCTION
>> > and sometimes, hanging occurs on that particular line. ?My
>> > data_flush_callback is as follows. ?I debug the module and it does
>> > not even enter data_flush_callback. ?It is hanging at
>> > switch_scheduler_add_task().
>> >
>> > SWITCH_STANDARD_SCHED_FUNC(data_flush_callback) {
>> >
>> > ? ? switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG,
>> > "starting to flush cdr record...\n");
>> >
>> > ? ? int last_sequence = flush_data_to_csv();
>> > ? ? if (last_sequence > -1 ) {
>> >
>> > ? ? ? ? update_last_seq(last_sequence);
>> > ? ? }
>> > ? ? task->runtime = switch_epoch_time_now(NULL) + globals.cycle_time;
>> >
>> > }
>> >
>> > Does anyone know why?
>> >
>> > Thanks,
>> > Mark
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From Prometheus001 at gmx.net  Fri Aug 21 11:28:12 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Fri, 21 Aug 2009 20:28:12 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <50DA88E6-832E-4596-851D-CB8FFFC54EAA@avgs.ca>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>	<4A8DB808.1030903@gmx.net>	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>	<4A8EBF77.1080204@gmx.net>
	<50DA88E6-832E-4596-851D-CB8FFFC54EAA@avgs.ca>
Message-ID: <4A8EE73C.2070006@gmx.net>

Hello Mathieu,

thank for your help. But this however didn't change the behaviour.

I've read of a patch in mod_sofia.c which partly corrects the problem
temporarily:

When I change Line 784 to

if (switch_rtp_ready(tech_pvt->rtp_session) && codec_ms != tech_pvt->codec_ms) {

to

if (switch_rtp_ready(tech_pvt->rtp_session) && codec_ms != tech_pvt->codec_ms && 0) {

(add a "&& 0") to deactivate this expression)

the announcements are played correctly to the Fritzbox. Connections to
other SIP phones (Snom) are also fine.

However the person at the Fritzbox still sounds very choppy in a
conference, but this is another module where I do not have a patch
available.

Best regards
Peter

Mathieu Rene schrieb:
> Try setting that in your sip profile:
>
> <param name="rtp-autofix-timing" value="false" />
>
> Thats a feature to work around with devices lying about their ptime in  
> their sdp payload.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 21-Aug-09, at 11:38 AM, Peter P GMX wrote:
>
>   
>> Hello Michael,
>>
>> I made some tests with Freeswitch and Fritzbox and found by  
>> Wireshark that:
>> within one call
>>
>>    * Freeswitch starts sending 20msec packets, then after ~0,2 second
>>      sends 30msec packets
>>    * FritzBox always sends 30msec packets.
>>
>> The average jitter is below 2 msec in both directions.
>>
>> The below logs shows that Freeswitch considers the FritzBox to be  
>> broken
>> and starts using 30msec packets. But there is no SIP message from FS  
>> to
>> Fritzbox telling him that FB will use 30msec packets. SDP from FS to
>> Fritzbox always shows ptime:20
>>
>> BTW: We can ship you a FritzBox if you need one for testing.
>>
>> Best regards
>> Peter
>>
>>
>> Log:
>>
>> 2009-08-21 17:15:50.271555 [DEBUG] switch_rtp.c:1138 Starting timer
>> [soft] 160 bytes per 20ms
>> 2009-08-21 17:15:50.281563 [INFO] mod_sofia.c:1499 Ring SDP:
>> v=0
>> o=FreeSWITCH 1250837460 1250837461 IN IP4 182.xxx.xx.xxx
>> s=FreeSWITCH
>> c=IN IP4 182.xxx.xx.xxx
>> t=0 0
>> m=audio 30290 RTP/AVP 8 101
>> a=rtpmap:8 PCMA/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=silenceSupp:off - - - -
>> a=ptime:20
>> a=sendrecv
>>
>> 2009-08-21 17:15:50.281563 [NOTICE] mod_sofia.c:1502 Pre-Answer
>> sofia/internal/02xxxxxxxxx at fs1.my.domain!
>> 2009-08-21 17:15:50.281563 [DEBUG] switch_core_session.c:630 Send  
>> signal
>> sofia/internal/02xxxxxxxxx at fs1.my.domain [BREAK]
>> EXECUTE sofia/internal/02xxxxxxxxx at fs1.my.domain
>> playback(voicemail/8000/vm-that_was_an_invalid_ext.wav)
>> 2009-08-21 17:15:50.281563 [DEBUG] switch_ivr_play_say.c:1097 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>> 2009-08-21 17:15:50.281563 [DEBUG] sofia.c:3302 Channel
>> sofia/internal/02xxxxxxxxx at fs1.my.domain entering state [early][183]
>> 2009-08-21 17:15:50.281563 [DEBUG] switch_core_io.c:649
>> sofia/internal/02xxxxxxxxx at fs1.my.domain receive message
>> [TRANSCODING_NECESSARY]
>> 2009-08-21 17:15:50.531551 [WARNING] mod_sofia.c:799 We were told to  
>> use
>> ptime 20 but what they meant to say was 30
>> This issue has so far been identified to happen on the following  
>> broken
>> platforms/devices:
>> Linksys/Sipura aka Cisco
>> ShoreTel
>> Sonus/L3
>> We will try to fix it but some of the devices on this list are so  
>> broken
>> who knows what will happen..
>>
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From brian at freeswitch.org  Fri Aug 21 11:38:50 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 21 Aug 2009 13:38:50 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4A8EBF77.1080204@gmx.net>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>	<4A8DB808.1030903@gmx.net>
	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
	<4A8EBF77.1080204@gmx.net>
Message-ID: <BD3DE8E0-3363-465C-985D-EC593AE45694@freeswitch.org>

You can ship me one whois bkw.org, I can add it to my lab.

/b

On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:

>
> BTW: We can ship you a FritzBox if you need one for testing.




From pjintheusa at gmail.com  Fri Aug 21 12:53:30 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 21 Aug 2009 15:53:30 -0400
Subject: [Freeswitch-users] Loopback and bypass_media
In-Reply-To: <367751820908141217g2facd41av25f65d6321967d74@mail.gmail.com>
References: <367751820908101335h6d0f655g6be39e94ae4961f3@mail.gmail.com>
	<6E8D2069C08AA84A83D336E996AE4C6702CC4D8F61@mse17be1.mse17.exchange.ms>
	<367751820908120629s16802077yefd68bb60a86e422@mail.gmail.com>
	<b39273530908120722q322a68f9t756f4dbf6acf2ae1@mail.gmail.com>
	<367751820908120822j46ac0458ne7f34fa7eee698d4@mail.gmail.com>
	<b39273530908120921n307433eq67b8f1e54ef9c356@mail.gmail.com>
	<367751820908131559v2fae7e50saacaca4dfcc32464@mail.gmail.com>
	<FAC2A456-42E1-4C92-8697-20068C761637@avgs.ca>
	<b39273530908132333y52279d64qe926eca47e15b5eb@mail.gmail.com>
	<367751820908141217g2facd41av25f65d6321967d74@mail.gmail.com>
Message-ID: <367751820908211253j33776506y8264f6f2c6bad74a@mail.gmail.com>

Hi there,

I created a feature request to cover this issue:

http://jira.freeswitch.org/browse/FSCORE-422 - The ability to support
Call FollowMe (or Call Blast) and multiple termination carriers -
without Loopback

If anybody wants comment on its merits and/or make the request clearer
- that would be great.

Thanks for every bodies help on this.

Phillip Jones

On Fri, Aug 14, 2009 at 3:17 PM, Phillip Jones<pjintheusa at gmail.com> wrote:
> Hi Rupa,
>
> What about my suggestion above introduce a "api_after_bridge" event
> that fires when the switch_ivr_uuid_bridge() bridges to the two sofia
> channels that Mathieu mentioned?
>
> Is that suggestion just way off the mark? If possible that would allow
> me to move forward - although I agree that supporting groupings of
> carriers is would be the most elegant solution.
>
> Let me know if I am just talking rubbish re the api_after_bridge" event.
>
> Thanks!
>
>
> Phillip Jones
>
>
> On Fri, Aug 14, 2009 at 2:33 AM, Rupa Schomaker<rupa at rupa.com> wrote:
>> On Thu, Aug 13, 2009 at 6:54 PM, Mathieu Rene<mrene_lists at avgs.ca> wrote:
>>> Hi All,
>>>
>>> The reason it works when you wait 3 seconds is that mod_loopback bails
>>
>> [snip]
>>
>> Thanks for that explanation. ?It umm.. explains a lot. :)
>>
>>> On another note, mod_sofia will behave differently when it detects its
>>> being bridge with another sofia channel, providing optimizations when
>>> both call legs are SIP.
>>>
>>> My personal opinion is not to use mod_loopback unless absolutely
>>> necessary, FreeSWITCH's core is very flexible and there's often a
>>> (better) way than using mod_loopback.
>>
>> So, I think the temp solution is to use loopback+delayed no media.
>>
>> but the real "solution" is to either drive the forked dialing logic
>> externally (event socket) or consider supporting groupings in the
>> bridge which.. umm... ?is gonna be a pain and will need buy in from
>> from Tony and other core devs since that is a core (no pun intended)
>> piece of code that nearly everything uses.
>>
>> I'm not sure I want to take a wack at it.
>>
>> --
>> -Rupa
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From anthony.minessale at gmail.com  Fri Aug 21 13:09:25 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Aug 2009 15:09:25 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <BD3DE8E0-3363-465C-985D-EC593AE45694@freeswitch.org>
References: <4A8D5132.7010807@gmx.net>
	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>
	<4A8D5D43.3030300@gmx.net>
	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>
	<4A8DB808.1030903@gmx.net>
	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
	<4A8EBF77.1080204@gmx.net>
	<BD3DE8E0-3363-465C-985D-EC593AE45694@freeswitch.org>
Message-ID: <191c3a030908211309j60c16b26h1e6445891136ae7f@mail.gmail.com>

try setting FS to 30ms too

edit vars.xml and add @30i to everywhere you see  PCMU or PCMA so it looks
like PCMU at 30i

from:

  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>

to:

  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221 at 32000h
,G7221 at 16000h,G722,PCMU at 30i,PCMA at 30i,GSM"/>
  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i
,GSM"/>


On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian at freeswitch.org> wrote:

> You can ship me one whois bkw.org, I can add it to my lab.
>
> /b
>
> On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
>
> >
> > BTW: We can ship you a FritzBox if you need one for testing.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From rogelio.perez at gmail.com  Fri Aug 21 12:15:13 2009
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Fri, 21 Aug 2009 16:15:13 -0300
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
Message-ID: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>

Hi Everyone,

I'm working on a PBX project for the Sheevaplug ARM based computer,  
with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
So far I've found a big difference between Freeswitch and Asterisk  
performance times.
This is a comparison of the time it takes them to perform different  
actions:

startup Freeswitch:			3 min.
startup Asterisk:			2 sec.

call extension Freeswitch:	6 sec.
call extension Asterisk:		0 sec.

shutdown Freeswitch:		6.5 sec
shutdown Asterisk:			0 sec.

reload config Freeswitch: 	1 sec.
reload config Asterisk:		1 sec.

Both were built from sources natively (no cross-compiling), and they  
use the default startup configurations.
I have managed to lower the Freeswitch times by disabling most of the  
modules and recompiling, but it is still far away from Asterisk (i.e.  
FS startup time 2.5 min).

1. Is there any way to further improve Freeswitch performance for the  
ARM architecture?
2. Can this be related to the lack of a FPU (the Sheevalug emulates  
the floating point operations).
3. On the startup I see this error repeated many times: [ERR]  
switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be  
related?

Thanks,
Rogelio Perez

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From andrew at hijacked.us  Fri Aug 21 13:20:06 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Fri, 21 Aug 2009 16:20:06 -0400
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
Message-ID: <20090821202005.GB28302@hijacked.us>

On Fri, Aug 21, 2009 at 04:15:13PM -0300, Rogelio Perez wrote:
> Hi Everyone,
> 
> I'm working on a PBX project for the Sheevaplug ARM based computer,  
> with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
> So far I've found a big difference between Freeswitch and Asterisk  
> performance times.
> This is a comparison of the time it takes them to perform different  
> actions:
> 
> startup Freeswitch:			3 min.
> startup Asterisk:			2 sec.
> 
> call extension Freeswitch:	6 sec.
> call extension Asterisk:		0 sec.
> 
> shutdown Freeswitch:		6.5 sec
> shutdown Asterisk:			0 sec.
> 
> reload config Freeswitch: 	1 sec.
> reload config Asterisk:		1 sec.
> 
> Both were built from sources natively (no cross-compiling), and they  
> use the default startup configurations.
> I have managed to lower the Freeswitch times by disabling most of the  
> modules and recompiling, but it is still far away from Asterisk (i.e.  
> FS startup time 2.5 min).
> 
> 1. Is there any way to further improve Freeswitch performance for the  
> ARM architecture?
> 2. Can this be related to the lack of a FPU (the Sheevalug emulates  
> the floating point operations).
> 3. On the startup I see this error repeated many times: [ERR]  
> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be  
> related?
>

Try making where freeswitch stores it's sqlite databases
/usr/local/freeswitch/db (by default) a ramdisk. I've had this vastly
improve FS performance on embedded devices.

Andrew



From anthony.minessale at gmail.com  Fri Aug 21 13:20:57 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 21 Aug 2009 15:20:57 -0500
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
Message-ID: <191c3a030908211320s6feb5f10sf8670575069ec95a@mail.gmail.com>

probably disk i/o.

Is it some kind of flash drive?

make a ramdisk and simlink in /usr/local/freeswitch/db and
/usr/local/freeswitch/log to it
the default configuration uses a lot of high level features that use the
sqlite db on the disk.

We also offer commercial support where we could dig deeper into the problem
if you can't figure it out
consulting at freeswitch.org



On Fri, Aug 21, 2009 at 2:15 PM, Rogelio Perez <rogelio.perez at gmail.com>wrote:

> Hi Everyone,
>
> I'm working on a PBX project for the Sheevaplug<http://www.google.com/url?sa=t&source=web&ct=res&cd=1&url=http%253A%252F%252Fwww.marvell.com%252Fproducts%252Fembedded_processors%252Fdeveloper%252Fkirkwood%252Fsheevaplug.jsp&ei=EOOOSo6ILcyOtgeNruTOBA&usg=AFQjCNFREhfy_erj5irBWk8XFUjkQOP-aw&sig2=pAHYlI15IbZ5Kcw0n-nIvA> ARM
> based computer, with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
> So far I've found a big difference between Freeswitch and Asterisk
> performance times.
> This is a comparison of the time it takes them to perform different
> actions:
>
> startup Freeswitch: 3 min.
> startup Asterisk: 2 sec.
>
> call extension Freeswitch: 6 sec.
> call extension Asterisk: 0 sec.
>
> shutdown Freeswitch: 6.5 sec
>
> shutdown Asterisk: 0 sec.
>
>
> reload config Freeswitch: 1 sec.
> reload config Asterisk: 1 sec.
>
>
> Both were built from sources natively (no cross-compiling), and they use
> the default startup configurations.
> I have managed to lower the Freeswitch times by disabling most of the
> modules and recompiling, but it is still far away from Asterisk (i.e. FS
> startup time 2.5 min).
>
> 1. Is there any way to further improve Freeswitch performance for the ARM
> architecture?
> 2. Can this be related to the lack of a FPU (the Sheevalug emulates the
> floating point operations).
> 3. On the startup I see this error repeated many times: [ERR]
> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be related?
>
> Thanks,
> Rogelio Perez
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From moises.silva at gmail.com  Fri Aug 21 15:29:26 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Fri, 21 Aug 2009 18:29:26 -0400
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
Message-ID: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>

So, I finally took some days to put up OpenR2 working with OpenZAP, which
means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
support for. Including Mexico, Brazil, Argentina and others. The stack has
been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most
countries that users may be interested in, support for new variants will be
added on-demand only (in any case users can always tweak the advanced
configuration file to create their own variants as a last resort).
I created a wiki page to illustrate the basic setup:
http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2

<http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2>Now is time for testing. I
just did minimal testing on my development environment, no serious testing,
and I know that some stuff is not working at this point (I had some issues
with variable length DNIS and ANI) which should be fixed soon.

If anyone around happens to have an R2 link and wants to test R2 support in
OpenZAP, I can give them a hand with the configuration and any issues you
may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
#openzap as moy.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From msc at freeswitch.org  Fri Aug 21 15:59:03 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 21 Aug 2009 15:59:03 -0700
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
In-Reply-To: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
References: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
Message-ID: <87f2f3b90908211559n163aa631j9b74a9e66cf7d95a@mail.gmail.com>

On Fri, Aug 21, 2009 at 3:29 PM, Moises Silva <moises.silva at gmail.com>wrote:

> So, I finally took some days to put up OpenR2 working with OpenZAP, which
> means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
> support for. Including Mexico, Brazil, Argentina and others. The stack has
> been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most
> countries that users may be interested in, support for new variants will be
> added on-demand only (in any case users can always tweak the advanced
> configuration file to create their own variants as a last resort).
> I created a wiki page to illustrate the basic setup:
> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
>
> <http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2>Now is time for testing. I
> just did minimal testing on my development environment, no serious testing,
> and I know that some stuff is not working at this point (I had some issues
> with variable length DNIS and ANI) which should be fixed soon.
>
> If anyone around happens to have an R2 link and wants to test R2 support in
> OpenZAP, I can give them a hand with the configuration and any issues you
> may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
> #openzap as moy.
>

You rock, dude!
-MC


>
> --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at gmail.com  Fri Aug 21 18:20:12 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 21 Aug 2009 22:20:12 -0300
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
In-Reply-To: <87f2f3b90908211559n163aa631j9b74a9e66cf7d95a@mail.gmail.com>
References: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
	<87f2f3b90908211559n163aa631j9b74a9e66cf7d95a@mail.gmail.com>
Message-ID: <5a8712120908211820y4a674e81wba7af5cc532c3f69@mail.gmail.com>

Way to go moy!

On Fri, Aug 21, 2009 at 7:59 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Aug 21, 2009 at 3:29 PM, Moises Silva <moises.silva at gmail.com>wrote:
>
>> So, I finally took some days to put up OpenR2 working with OpenZAP, which
>> means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
>> support for. Including Mexico, Brazil, Argentina and others. The stack has
>> been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most
>> countries that users may be interested in, support for new variants will be
>> added on-demand only (in any case users can always tweak the advanced
>> configuration file to create their own variants as a last resort).
>> I created a wiki page to illustrate the basic setup:
>> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
>>
>> <http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2>Now is time for testing.
>> I just did minimal testing on my development environment, no serious
>> testing, and I know that some stuff is not working at this point (I had some
>> issues with variable length DNIS and ANI) which should be fixed soon.
>>
>> If anyone around happens to have an R2 link and wants to test R2 support
>> in OpenZAP, I can give them a hand with the configuration and any issues you
>> may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
>> #openzap as moy.
>>
>
> You rock, dude!
> -MC
>
>
>>
>> --
>> Moises Silva
>> Software Developer
>> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
>> 9T3 Canada
>> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From b_ball_henry at hotmail.com  Fri Aug 21 23:40:40 2009
From: b_ball_henry at hotmail.com (Henry Huang)
Date: Sat, 22 Aug 2009 14:40:40 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
Message-ID: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>

Hi:

I try to dial sip url from my softphone but seems like the sip address is
being processed by sofia before it pass to the dialplan. The example here is
:

*X-lite(softphone) dials -> 1009 at 4.2.2.2 (it's fake sip address, the purpose
was just to test what's being passed to dialplan)
sofia receives the invite and return with trying
sofia pass the destination number to dailplan with "1009" (without the
"sip:" in front and without the "@4.2.2.2" after it)
*
Please see pastebin for full log. http://pastebin.freeswitch.org/10089
ignore anything after line 80, because it's not my point, and the
destination is a fake address.

I would like to know how do you actually pass a full sip url to the dialplan
to do the regex match. Because from the default.xml dialplan, it comes with
an example sip url dialing extension that match's *^sip:(.*)$ *. So I assume
there must be a way of passing full sip url to the dialplan. Here is the
example dialplan expecting sofia to pass it a full sip url:

 <!-- dial via SIP uri -->
    <extension name="sip_uri">
      <condition field="destination_number" expression="^sip:(.*)$">
        <action application="bridge" data="sofia/${use_profile}/$1"/>
      </condition>
    </extension>

Thanks

-- 
Henry Huang
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From mattdfong at gmail.com  Fri Aug 21 23:41:09 2009
From: mattdfong at gmail.com (Matthew Fong)
Date: Fri, 21 Aug 2009 23:41:09 -0700
Subject: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with
	same Gateway
In-Reply-To: <4256bf830908201021s7da0bff0wed8e060dbac0db2c@mail.gmail.com>
References: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>
	<6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>
	<4256bf830908201021s7da0bff0wed8e060dbac0db2c@mail.gmail.com>
Message-ID: <4256bf830908212341k1c0a9c9x7c8a9c70460965f7@mail.gmail.com>

So there seems to be some sort of error when bridging directly like
originate {ignore_early_media=true}sofia/gateway/XXXX.com/91415992<http://epik.com/914159927717>
XXXX &bridge(sofia/gateway/XXXX.com/91415465 <http://epik.com/914154650027>
XXXX)

BUT

if I get FS to send media to leg A, and then bridge to leg B by using a lua
script like

session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/hh/hh-welcome.wav");
session:execute("bridge", "sofia/gateway/epik.com/91415XXXXXXX");

then the legs bridge together OK. This happens when trying to bridge two
channels via the same Broadsoft SBC. Does this sound like a bug that should
be submitted to JIRA?

--matt
http://www.hellohunter.com


On Thu, Aug 20, 2009 at 10:21 AM, Matthew Fong <mattdfong at gmail.com> wrote:

> originate {ignore_early_media=true}sofia/gateway/epik.com/914159927717
>  &bridge(sofia/gateway/epik.com/914154650027)
>
> is the string I was using from the console.
>
>
> On Thu, Aug 20, 2009 at 6:00 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>
>> Hi
>>
>> How are you bridging the calls in FS (which api call or C function are you
>> using)?
>>  Mathieu Rene
>> Avant-Garde Solutions Inc
>> Office: + 1 (514) 664-1044 x100
>> Cell: +1 (514) 664-1044 x200
>> mrene at avgs.ca
>>
>>
>>
>>
>> On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:
>>
>> I'm trying to get FreeSWITCH to bridge two channels together through the
>> same external gateway, but I'm having issues hearing audio. Both legs if
>> setup independently and forwarded to 5000 (test ivr) work fine for both
>> inbound and outbound media, but when I try to bridge them together,
>> everything seems fine in FreeSWITCH, but neither party can hear the other
>> speak. I'm thinking the external gateway might be having some issues because
>> I've been able to bridge 2 channels together through the same gateway on
>> different providers, but thought I'd also try to seek some help here.
>> FreeSWITCH should be handling the media for both channels, so I can't figure
>> out why if Leg A and Leg B work independently, but not if they are bridged
>> together. Is there a setting somewhere in FS that I'm missing?
>> Below is a ngrep of the SIP interactions if it's useful. Thanks for the
>> help.
>>
>> --matt
>>
>> interface: eth0 (172.24.200.0/255.255.255.0)
>> filter: (ip or ip6) and ( port 5060 )
>>
>> U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060
>> INVITE sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>SIP/2.0.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>> Max-Forwards: 70.
>>  From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>>.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257811 INVITE.
>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>> Supported: timer, precondition, path, replaces.
>> Allow-Events: talk, refer.
>> Content-Type: application/sdp.
>> Content-Disposition: session.
>> Content-Length: 293.
>> Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;party=calling;screen=yes;privacy=off.
>> .
>> v=0.
>> o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
>> s=FreeSWITCH.
>> c=IN IP4 216.81.56.198.
>> t=0 0.
>> m=audio 24700 RTP/AVP 0 8 3 101 13.
>> a=rtpmap:0 PCMU/8000.
>> a=rtpmap:8 PCMA/8000.
>> a=rtpmap:3 GSM/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=rtpmap:13 CN/8000.
>> a=ptime:20.
>>
>>
>> U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 100 Trying.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>> >;tag=F725.2C49.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257811 INVITE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 183 Session Progress.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>> >;tag=F725.2C49.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257811 INVITE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>> Allow:
>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>> Content-Type: application/sdp.
>> Content-Length: 227.
>> .
>> v=0.
>> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
>> s=M6 Call.
>> c=IN IP4 38.98.58.148.
>> t=0 0.
>> m=audio 42554 RTP/AVP 0 101.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>> a=ptime:20.
>> a=sendrecv.
>> a=rtcp:6461 IN IP4 10.10.24.50.
>>
>>
>> U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 200 OK.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>> >;tag=F725.2C49.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257811 INVITE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>> Session-Expires: 1800;refresher=uas.
>> Allow:
>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>> Supported: timer.
>> Content-Type: application/sdp.
>> Content-Length: 227.
>> .
>> v=0.
>> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
>> s=M6 Call.
>> c=IN IP4 38.98.58.148.
>> t=0 0.
>> m=audio 42554 RTP/AVP 0 101.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>> a=ptime:20.
>> a=sendrecv.
>> a=rtcp:6461 IN IP4 10.10.24.50.
>>
>>
>> U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060
>> ACK sip:914159927717 at 38.98.58.148:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc.
>> Max-Forwards: 70.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> To: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>> >;tag=F725.2C49.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257811 ACK.
>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060
>> INVITE sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>SIP/2.0.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>> Max-Forwards: 70.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>>.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257815 INVITE.
>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>> Supported: timer, precondition, path, replaces.
>> Allow-Events: talk, refer.
>> Content-Type: application/sdp.
>> Content-Disposition: session.
>> Content-Length: 293.
>> Remote-Party-ID: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;party=calling;screen=yes;privacy=off.
>> .
>> v=0.
>> o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198.
>> s=FreeSWITCH.
>> c=IN IP4 216.81.56.198.
>> t=0 0.
>> m=audio 24798 RTP/AVP 0 8 3 101 13.
>> a=rtpmap:0 PCMU/8000.
>> a=rtpmap:8 PCMA/8000.
>> a=rtpmap:3 GSM/8000.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-16.
>> a=rtpmap:13 CN/8000.
>> a=ptime:20.
>>
>>
>> U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 100 Trying.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>> >;tag=F72E.2D4E.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257815 INVITE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 183 Session Progress.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>> >;tag=F72E.2D4E.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257815 INVITE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>> Allow:
>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>> Content-Type: application/sdp.
>> Content-Length: 225.
>> .
>> v=0.
>> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
>> s=M6 Call.
>> c=IN IP4 38.98.58.148.
>> t=0 0.
>> m=audio 46520 RTP/AVP 0 101.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>> a=ptime:20.
>> a=sendrecv.
>> a=rtcp:6451 IN IP4 10.10.24.50.
>>
>>
>> U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 200 OK.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>> >;tag=F72E.2D4E.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257815 INVITE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>> Session-Expires: 1800;refresher=uas.
>> Allow:
>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>> Supported: timer.
>> Content-Type: application/sdp.
>> Content-Length: 225.
>> .
>> v=0.
>> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
>> s=M6 Call.
>> c=IN IP4 38.98.58.148.
>> t=0 0.
>> m=audio 46520 RTP/AVP 0 101.
>> a=rtpmap:101 telephone-event/8000.
>> a=fmtp:101 0-15.
>> a=ptime:20.
>> a=sendrecv.
>> a=rtcp:6451 IN IP4 10.10.24.50.
>>
>>
>> U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060
>> ACK sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK.
>> Max-Forwards: 70.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>> >;tag=F72E.2D4E.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257815 ACK.
>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080
>> BYE sip:gw+epik.com at 216.81.56.198:5080 SIP/2.0.
>> From: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>> >;tag=F725.2C49.
>> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 4817 BYE.
>> Max-Forwards: 70.
>> Via: SIP/2.0/UDP 38.98.58.148:5060
>> ;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
>> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>> Allow:
>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>  Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060
>> SIP/2.0 200 OK.
>> Via: SIP/2.0/UDP 38.98.58.148:5060
>> ;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
>> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
>> From: <sip:914159927717 at 38.98.58.148 <sip%3A914159927717 at 38.98.58.148>
>> >;tag=F725.2C49.
>> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=ZtFvjeFQmXvpp.
>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>> CSeq: 4817 BYE.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>> Supported: timer, precondition, path, replaces.
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060
>> BYE sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
>> Max-Forwards: 70.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>> >;tag=F72E.2D4E.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257816 BYE.
>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>> Supported: timer, precondition, path, replaces.
>> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
>> Content-Length: 0.
>> .
>>
>>
>> U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080
>> SIP/2.0 200 OK.
>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198<sip%3A0000000000 at 216.81.56.198>
>> >;tag=038mm9ZtH6j9H.
>> To: <sip:914154650027 at 38.98.58.148 <sip%3A914154650027 at 38.98.58.148>
>> >;tag=F72E.2D4E.
>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>> CSeq: 119257816 BYE.
>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>> Allow:
>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>> Content-Length: 0.
>> .
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From woodydickson at gmail.com  Sat Aug 22 01:08:12 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Sat, 22 Aug 2009 16:08:12 +0800
Subject: [Freeswitch-users] zombie channels
In-Reply-To: <87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>
References: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
	<87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>
Message-ID: <a44688a0908220108i4d9a63a1lbe4c36f4ed7c3f31@mail.gmail.com>

Hi

I checked and there is no looping in cdr.

Also, only a very small percentage of the channels become zombie.

What could cause fs to not releasing the channels?

Also, it seems to happen on under high traffic.  Could the fact that FS does
not receive BYE or BYE timing out on the uac side may cause this problem?

Thanks,
woody

On Fri, Aug 21, 2009 at 9:31 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> CS_REPORTING is the state in which cdrs are written, if the channel
> gets stuck in that state, the cdr module you are using is probably
> hanging somewhere.
>
> Use the "freeswitch-gcore" script in your source tree's scripts
> directory to generate a bug report for hanging channels.
>
> should be like..
>
> cd /usr/src/freeswitch # or whatever your source tree is
> bash ./scripts/freeswitch-gcore > bugreport.txt
>
> then submit it on http://jira.freeswitch.org/ so we can look at it.
>
> As you wish, you can also hop on #freeswitch / irc.freenode.net and
> have someone look into it.
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 21-Aug-09, at 5:54 AM, Woody Dickson wrote:
>
> > Hi,
> >
> > I am running 1.0.4 right now using latest trunk.
> >
> > After a high traffic session, I do "show channels", I would find a
> > bunch of "CS_HIBERNATE" channels that don't get removed after all
> > the traffic is gone.
> >
> > Does anyone know what is the case of thoes CS_HIBERNATE'd channels?
> > How can I set  a timeout for those channels to be removed?
> >
> > Thanks,
> > Woody
>  > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Sat Aug 22 01:35:07 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 22 Aug 2009 04:35:07 -0400
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
Message-ID: <EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>

No, you don't get the full sip uri in the dialplan like that.   You do  
have a whole bunch of variables of the parsed sip header you can use.   
Use the "info" application to see all the vars so you can see what you  
have to route the call on.

Mike

On Aug 22, 2009, at 2:40 AM, Henry Huang wrote:

> Hi:
>
> I try to dial sip url from my softphone but seems like the sip  
> address is being processed by sofia before it pass to the dialplan.  
> The example here is :
>
> X-lite(softphone) dials -> 1009 at 4.2.2.2 (it's fake sip address, the  
> purpose was just to test what's being passed to dialplan)
> sofia receives the invite and return with trying
> sofia pass the destination number to dailplan with "1009" (without  
> the "sip:" in front and without the "@4.2.2.2" after it)
>
> Please see pastebin for full log. http://pastebin.freeswitch.org/10089
> ignore anything after line 80, because it's not my point, and the  
> destination is a fake address.
>
> I would like to know how do you actually pass a full sip url to the  
> dialplan to do the regex match. Because from the default.xml  
> dialplan, it comes with an example sip url dialing extension that  
> match's ^sip:(.*)$ . So I assume there must be a way of passing full  
> sip url to the dialplan. Here is the example dialplan expecting  
> sofia to pass it a full sip url:
>
>  <!-- dial via SIP uri -->
>     <extension name="sip_uri">
>       <condition field="destination_number" expression="^sip:(.*)$">
>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>       </condition>
>     </extension>
>

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From red.rain.seven at gmail.com  Sat Aug 22 01:41:15 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 22 Aug 2009 16:41:15 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
Message-ID: <59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>

It that case, the example of dialing sip_uri in the dialplan/default.xml
should be removed to prevent confusion. Because according to what you said,
one can never be able to hit this extension:

<!-- dial via SIP uri -->
    <extension name="sip_uri">
      <condition field="destination_number" expression="^sip:(.*)$">
        <action application="bridge" data="sofia/${use_profile}/$1"/>
      </condition>
    </extension>

And thanks for the tip, I will use variable instead.


On Sat, Aug 22, 2009 at 4:35 PM, Michael Jerris <mike at jerris.com> wrote:

> No, you don't get the full sip uri in the dialplan like that.   You do have
> a whole bunch of variables of the parsed sip header you can use.  Use the
> "info" application to see all the vars so you can see what you have to route
> the call on.
> Mike
>
> On Aug 22, 2009, at 2:40 AM, Henry Huang wrote:
>
> Hi:
>
> I try to dial sip url from my softphone but seems like the sip address is
> being processed by sofia before it pass to the dialplan. The example here is
> :
>
> *X-lite(softphone) dials -> 1009 at 4.2.2.2 (it's fake sip address, the
> purpose was just to test what's being passed to dialplan)
> sofia receives the invite and return with trying
> sofia pass the destination number to dailplan with "1009" (without the
> "sip:" in front and without the "@4.2.2.2" after it)
> *
> Please see pastebin for full log. http://pastebin.freeswitch.org/10089
> ignore anything after line 80, because it's not my point, and the
> destination is a fake address.
>
> I would like to know how do you actually pass a full sip url to the
> dialplan to do the regex match. Because from the default.xml dialplan, it
> comes with an example sip url dialing extension that match's *^sip:(.*)$ *.
> So I assume there must be a way of passing full sip url to the dialplan.
> Here is the example dialplan expecting sofia to pass it a full sip url:
>
>  <!-- dial via SIP uri -->
>     <extension name="sip_uri">
>       <condition field="destination_number" expression="^sip:(.*)$">
>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>       </condition>
>     </extension>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From jason at jasonjgw.net  Sat Aug 22 03:30:05 2009
From: jason at jasonjgw.net (Jason White)
Date: Sat, 22 Aug 2009 20:30:05 +1000
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
Message-ID: <20090822103005.GA4299@jdc.jasonjgw.net>

Henry Huang <red.rain.seven at gmail.com> wrote:
> It that case, the example of dialing sip_uri in the dialplan/default.xml
> should be removed to prevent confusion. Because according to what you said,
> one can never be able to hit this extension:

It is entirely possible to reach this extension, but notice that the "sip:"
prefix is removed before the rest of the URI is used in calling the bridge
application.

If you don't understand why this dial-plan entry works, go back and read about
regular expressions and the format of destinations used with the bridge
application. There are examples and explanations on the wiki.




From red.rain.seven at gmail.com  Sat Aug 22 07:09:26 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 22 Aug 2009 22:09:26 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <20090822103005.GA4299@jdc.jasonjgw.net>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
Message-ID: <59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>

Jason:

I fully understand how the regex works in the dialplan. If you look closely
in my original email and check out the pastebin. You will see that sofia
does not pass the "sip:" to dialplan. I can do any combination of letters
that dials from my softphone, and it will pass them to the dialplan. but if
I put "sip:" in the front of my dial string. The "sip:" gets trunkated by
sofia module so does the "@xx.xx.xx.xx" gets trunkated before it reaches
dialplan to for regex matching. Therefore I say you can never reach the
example sip uri extension because sofia will trunkate "sip:" .

Here is the excert from pastebin:
3.    INVITE sip:1009 at 4.2.2.2 SIP/2.0
(line 3 , the freeswitch has successfuly received my dialying to sip:
1009 at 4.2.2.2)73.  2009-08-20 16:37:28.982772 [INFO]
mod_dialplan_xml.c:315Processing
1001->1009 in context Global
74.  Dialplan: sofia/trunkgroup_1/1001 at 192.168.1.67 parsing [
Global->number_1] continue=false
75.  Dialplan: sofia/trunkgroup_1/1001 at 192.168.1.67 Regex (FAIL)
[number_1]destination_number
(1009) =~ /^sip(.*)$/ break=on-false
(line 73~75, you can see that on line 73, sofia has trunkated the "sip: " &
"@4.2.2.2" and only leave "1009" as the destination to pass to dialplan for
regex match.)



On Sat, Aug 22, 2009 at 6:30 PM, Jason White <jason at jasonjgw.net> wrote:

> Henry Huang <red.rain.seven at gmail.com> wrote:
> > It that case, the example of dialing sip_uri in the dialplan/default.xml
> > should be removed to prevent confusion. Because according to what you
> said,
> > one can never be able to hit this extension:
>
> It is entirely possible to reach this extension, but notice that the "sip:"
> prefix is removed before the rest of the URI is used in calling the bridge
> application.
>
> If you don't understand why this dial-plan entry works, go back and read
> about
> regular expressions and the format of destinations used with the bridge
> application. There are examples and explanations on the wiki.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From mike at jerris.com  Sat Aug 22 07:27:26 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 22 Aug 2009 10:27:26 -0400
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
Message-ID: <EBDCE6C6-CB7C-4652-8A4D-850B364D0BB8@jerris.com>

a call coming from sofia would never hit that in the dialplan.  That  
extension is useful for dialing a sip url from mod_portaudio.

Mike

On Aug 22, 2009, at 10:09 AM, Henry Huang wrote:

> Jason:
>
> I fully understand how the regex works in the dialplan. If you look  
> closely in my original email and check out the pastebin. You will  
> see that sofia does not pass the "sip:" to dialplan. I can do any  
> combination of letters that dials from my softphone, and it will  
> pass them to the dialplan. but if I put "sip:" in the front of my  
> dial string. The "sip:" gets trunkated by sofia module so does the  
> "@xx.xx.xx.xx" gets trunkated before it reaches dialplan to for  
> regex matching. Therefore I say you can never reach the example sip  
> uri extension because sofia will trunkate "sip:" .
>
> Here is the excert from pastebin:
> 3.    INVITE sip:1009 at 4.2.2.2 SIP/2.0
> (line 3 , the freeswitch has successfuly received my dialying to  
> sip: 1009 at 4.2.2.2)
> 73.  2009-08-20 16:37:28.982772 [INFO] mod_dialplan_xml.c:315  
> Processing 1001->1009 in context Global
> 74.  Dialplan: sofia/trunkgroup_1/1001 at 192.168.1.67 parsing [Global- 
> >number_1] continue=false
> 75.  Dialplan: sofia/trunkgroup_1/1001 at 192.168.1.67 Regex (FAIL)  
> [number_1] destination_number(1009) =~ /^sip(.*)$/ break=on-false
> (line 73~75, you can see that on line 73, sofia has trunkated the  
> "sip: " & "@4.2.2.2" and only leave "1009" as the destination to  
> pass to dialplan for regex match.)
>
>
>
> On Sat, Aug 22, 2009 at 6:30 PM, Jason White <jason at jasonjgw.net>  
> wrote:
> Henry Huang <red.rain.seven at gmail.com> wrote:
> > It that case, the example of dialing sip_uri in the dialplan/ 
> default.xml
> > should be removed to prevent confusion. Because according to what  
> you said,
> > one can never be able to hit this extension:
>
> It is entirely possible to reach this extension, but notice that the  
> "sip:"
> prefix is removed before the rest of the URI is used in calling the  
> bridge
> application.
>
> If you don't understand why this dial-plan entry works, go back and  
> read about
> regular expressions and the format of destinations used with the  
> bridge
> application. There are examples and explanations on the wiki.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> -- 
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sat Aug 22 07:30:14 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 22 Aug 2009 09:30:14 -0500
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
Message-ID: <51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>

Remember the dialplan is agnostic... it has no clue about SIP, IAX,  
Jingle, H323... it routes... you have various other variables you can  
condition on also... route on destination_number and you'll be fine.

/b

On Aug 22, 2009, at 9:09 AM, Henry Huang wrote:

> I fully understand how the regex works in the dialplan. If you look  
> closely in my original email and check out the pastebin. You will  
> see that sofia does not pass the "sip:" to dialplan. I can do any  
> combination of letters that dials from my softphone, and it will  
> pass them to the dialplan. but if I put "sip:" in the front of my  
> dial string. The "sip:" gets trunkated by sofia module so does the  
> "@xx.xx.xx.xx" gets trunkated before it reaches dialplan to for  
> regex matching. Therefore I say you can never reach the example sip  
> uri extension because sofia will trunkate "sip:" .




From red.rain.seven at gmail.com  Sat Aug 22 07:46:14 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 22 Aug 2009 22:46:14 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
Message-ID: <59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>

Brian:

but why can't I pass "sip:" to dialplan? seems like it's being truncated by
sofia..
Can you confirm that?

On Sat, Aug 22, 2009 at 10:30 PM, Brian West <brian at freeswitch.org> wrote:

> Remember the dialplan is agnostic... it has no clue about SIP, IAX,
> Jingle, H323... it routes... you have various other variables you can
> condition on also... route on destination_number and you'll be fine.
>
> /b
>
> On Aug 22, 2009, at 9:09 AM, Henry Huang wrote:
>
> > I fully understand how the regex works in the dialplan. If you look
> > closely in my original email and check out the pastebin. You will
> > see that sofia does not pass the "sip:" to dialplan. I can do any
> > combination of letters that dials from my softphone, and it will
> > pass them to the dialplan. but if I put "sip:" in the front of my
> > dial string. The "sip:" gets trunkated by sofia module so does the
> > "@xx.xx.xx.xx" gets trunkated before it reaches dialplan to for
> > regex matching. Therefore I say you can never reach the example sip
> > uri extension because sofia will trunkate "sip:" .
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From brian at freeswitch.org  Sat Aug 22 07:59:00 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 22 Aug 2009 09:59:00 -0500
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
Message-ID: <66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>

Because the dial plan is technology agnostic... you have been told  
more than once it won't pass it to the dialplan from mod_sofia...

/b

On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:

> Brian:
>
> but why can't I pass "sip:" to dialplan? seems like it's being  
> truncated by sofia..
> Can you confirm that?




From red.rain.seven at gmail.com  Sat Aug 22 08:02:22 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 22 Aug 2009 23:02:22 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
	<66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
Message-ID: <59ad9ca10908220802n5285e36qbbdca2c3d50b5311@mail.gmail.com>

Brian:

Sorry, it's my English. I didn't understand what you meant by "agnostic"
back there. Now I know.

Thank you.

On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian at freeswitch.org> wrote:

> Because the dial plan is technology agnostic... you have been told
> more than once it won't pass it to the dialplan from mod_sofia...
>
> /b
>
> On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
>
> > Brian:
> >
> > but why can't I pass "sip:" to dialplan? seems like it's being
> > truncated by sofia..
> > Can you confirm that?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From red.rain.seven at gmail.com  Sat Aug 22 08:07:09 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sat, 22 Aug 2009 23:07:09 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
	<66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
Message-ID: <59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com>

Brian:

Oh, and again, if it's not passing it to the dialplan. I had suggested to
remove the sample "sip uri" extension in the default.xml dialplan. because
no one can reach the dialplan with prefix "sip:" because sofia is going to
remove that prefix.

<!-- dial via SIP uri -->
    <extension name="sip_uri">
      <condition field="destination_number" expression="^sip:(.*)$">
        <action application="bridge" data="sofia/${use_profile}/$1"/>
      </condition>
    </extension>

On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian at freeswitch.org> wrote:

> Because the dial plan is technology agnostic... you have been told
> more than once it won't pass it to the dialplan from mod_sofia...
>
> /b
>
> On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
>
> > Brian:
> >
> > but why can't I pass "sip:" to dialplan? seems like it's being
> > truncated by sofia..
> > Can you confirm that?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From brian at freeswitch.org  Sat Aug 22 08:15:21 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 22 Aug 2009 10:15:21 -0500
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
	<66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
	<59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com>
Message-ID: <366174E9-8213-4F20-81FC-FA26D0E4ED3F@freeswitch.org>

You were told already this was used by mod_portaudio.  So that you can  
pa call sip:blah at domain.com which portaudio passes the exact string  
you dial with pa call to the dialplan.

/b

On Aug 22, 2009, at 10:07 AM, Henry Huang wrote:

> Brian:
>
> Oh, and again, if it's not passing it to the dialplan. I had  
> suggested to remove the sample "sip uri" extension in the  
> default.xml dialplan. because no one can reach the dialplan with  
> prefix "sip:" because sofia is going to remove that prefix.
>
> <!-- dial via SIP uri -->
>     <extension name="sip_uri">
>       <condition field="destination_number" expression="^sip:(.*)$">
>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>       </condition>
>     </extension>

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From msc at freeswitch.org  Sat Aug 22 08:35:29 2009
From: msc at freeswitch.org (Michael Collins)
Date: Sat, 22 Aug 2009 08:35:29 -0700
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
	<66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
	<59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com>
Message-ID: <87f2f3b90908220835w6e5c98ecpeecdf81e6ac46f4c@mail.gmail.com>

On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang <red.rain.seven at gmail.com>wrote:

> Brian:
>
> Oh, and again, if it's not passing it to the dialplan. I had suggested to
> remove the sample "sip uri" extension in the default.xml dialplan. because
> no one can reach the dialplan with prefix "sip:" because sofia is going to
> remove that prefix.


Well, this isn't entirely accurate. Like Mike J said, if you dialed
something like this at the CLI:

pa call sip:user at domain.com <sip%3Auser at domain.com>

Then you'd need the dialplan entry that handles the SIP URI.

Going back to the original question...
X-Lite dials 1009 at 4.2.2.2 correct?
But you're saying that the dialplan simply sees "1009" as the destination
number? I'm looking at the pastebin (10089) and trying to figure out exactly
what is happening. All I can see is that you have a context named "Global"
so I'm assuming you've made at least some modifications to the default
dialplan. Can you pastebin that whole context?

The other thing that you should probably do is create an extension in this
global context that routes a call to the info application. You could do
something like this so that "9992" would do an info dump:
<extension name="info">
  <condition field="destination_number" expression="^(9992)$">
    <action application="info"/>
  </condition>
</extension>

Then reloadxml and make a call to 9992 from your X-Lite client. The CLI will
have a dump and you'll see all sorts of variables listed. Many of those are
available for you to use for condition matches and routing in the dialplan.

Let us know how the info application does in giving you information about
the A leg of the call.
-MC

>
>
> <!-- dial via SIP uri -->
>     <extension name="sip_uri">
>       <condition field="destination_number" expression="^sip:(.*)$">
>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>       </condition>
>     </extension>
>
> On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian at freeswitch.org> wrote:
>
>> Because the dial plan is technology agnostic... you have been told
>> more than once it won't pass it to the dialplan from mod_sofia...
>>
>> /b
>>
>> On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
>>
>> > Brian:
>> >
>> > but why can't I pass "sip:" to dialplan? seems like it's being
>> > truncated by sofia..
>> > Can you confirm that?
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From larclap at yahoo.com  Sat Aug 22 09:42:47 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Sat, 22 Aug 2009 09:42:47 -0700
Subject: [Freeswitch-users] Problem with cnam.js?
Message-ID: <00b901ca2347$950feee0$bf2fcca0$@com>

I think there's something wrong with the script at
http://wiki.freeswitch.org/wiki/Examples_cnam.js.

 

If you use it as is, it displays "Content-type: text/html" for the
effective_caller_id_name. In cnam.pl, the first two output lines are
generated by:

 

if (!$debug) {print "Content-type: text/html\n\n";}

 

with the actual name in the third line.

 

So I changed:

 

fd.open("read");

buff = fd.readln();

 

if(buff) {

   logger(buff, "info");

   session.setVariable("effective_caller_id_name", buff);

}

 

To:

 

fd.open("read");

buff = fd.readAll();

 

if(buff[2]) {

   logger(buff, "info");

   session.setVariable("effective_caller_id_name", buff[2]);

}

 

Or remove the print statement from cnam.pl.

 

Sorry for the code, but the page was not editable.

 

Lars

 

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From red.rain.seven at gmail.com  Sat Aug 22 10:35:01 2009
From: red.rain.seven at gmail.com (Henry Huang)
Date: Sun, 23 Aug 2009 01:35:01 +0800
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <87f2f3b90908220835w6e5c98ecpeecdf81e6ac46f4c@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com>
	<EBCE4DAF-B59C-4AF8-B4F0-5F78C18EFE07@jerris.com>
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com>
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com>
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com>
	<66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
	<59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com>
	<87f2f3b90908220835w6e5c98ecpeecdf81e6ac46f4c@mail.gmail.com>
Message-ID: <59ad9ca10908221035l448a9f91ncf26f978506487ad@mail.gmail.com>

Michael:

Thank you for making it in "for dummies" format. :P
These are really nice tips I can use. thanks.

On Sat, Aug 22, 2009 at 11:35 PM, Michael Collins <msc at freeswitch.org>wrote:

>
>
> On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>
>> Brian:
>>
>> Oh, and again, if it's not passing it to the dialplan. I had suggested to
>> remove the sample "sip uri" extension in the default.xml dialplan. because
>> no one can reach the dialplan with prefix "sip:" because sofia is going to
>> remove that prefix.
>
>
> Well, this isn't entirely accurate. Like Mike J said, if you dialed
> something like this at the CLI:
>
> pa call sip:user at domain.com <sip%3Auser at domain.com>
>
> Then you'd need the dialplan entry that handles the SIP URI.
>
> Going back to the original question...
> X-Lite dials 1009 at 4.2.2.2 correct?
> But you're saying that the dialplan simply sees "1009" as the destination
> number? I'm looking at the pastebin (10089) and trying to figure out exactly
> what is happening. All I can see is that you have a context named "Global"
> so I'm assuming you've made at least some modifications to the default
> dialplan. Can you pastebin that whole context?
>
> The other thing that you should probably do is create an extension in this
> global context that routes a call to the info application. You could do
> something like this so that "9992" would do an info dump:
> <extension name="info">
>   <condition field="destination_number" expression="^(9992)$">
>     <action application="info"/>
>   </condition>
> </extension>
>
> Then reloadxml and make a call to 9992 from your X-Lite client. The CLI
> will have a dump and you'll see all sorts of variables listed. Many of those
> are available for you to use for condition matches and routing in the
> dialplan.
>
> Let us know how the info application does in giving you information about
> the A leg of the call.
> -MC
>
>>
>>
>> <!-- dial via SIP uri -->
>>     <extension name="sip_uri">
>>       <condition field="destination_number" expression="^sip:(.*)$">
>>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>>       </condition>
>>     </extension>
>>
>> On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian at freeswitch.org>wrote:
>>
>>> Because the dial plan is technology agnostic... you have been told
>>> more than once it won't pass it to the dialplan from mod_sofia...
>>>
>>> /b
>>>
>>> On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
>>>
>>> > Brian:
>>> >
>>> > but why can't I pass "sip:" to dialplan? seems like it's being
>>> > truncated by sofia..
>>> > Can you confirm that?
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Henry Huang
>> UniC Solution - Communication Unified
>> VoIP & Open Source software Consultant
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Henry Huang
UniC Solution - Communication Unified
VoIP & Open Source software Consultant
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From anthony.minessale at gmail.com  Sat Aug 22 11:11:55 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 22 Aug 2009 13:11:55 -0500
Subject: [Freeswitch-users] zombie channels
In-Reply-To: <a44688a0908220108i4d9a63a1lbe4c36f4ed7c3f31@mail.gmail.com>
References: <a44688a0908210254j1dc98ae0n24a73abe65d9957a@mail.gmail.com>
	<87B29356-87DC-421A-A076-DD7EA7B90067@avgs.ca>
	<a44688a0908220108i4d9a63a1lbe4c36f4ed7c3f31@mail.gmail.com>
Message-ID: <191c3a030908221111m2181977dn782da42afee84726@mail.gmail.com>

Yes try enabling session timers

On Aug 22, 2009 3:10 AM, "Woody Dickson" <woodydickson at gmail.com> wrote:

Hi

I checked and there is no looping in cdr.

Also, only a very small percentage of the channels become zombie.

What could cause fs to not releasing the channels?

Also, it seems to happen on under high traffic.  Could the fact that FS does
not receive BYE or BYE timing out on the uac side may cause this problem?

Thanks,
woody

On Fri, Aug 21, 2009 at 9:31 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> > Hi, > > CS_REPORTING is the state in which cdrs are written, if the
channel > gets stuck in that...


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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From shaheryarkh at googlemail.com  Sat Aug 22 13:53:20 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Sun, 23 Aug 2009 01:53:20 +0500
Subject: [Freeswitch-users] SIP codec preference order
Message-ID: <b16156850908221353m3e219f7eyefd29870584e121c@mail.gmail.com>

Hi,

I have a FS gateway (SVN revision 14537) that is is receiving SIP calls from
different source gateways and sending it to one single destination gateway.
Now each source gateway can talk in one specific codec and FS itself is not
doing any transcoding. So i enabled all possible codecs that this FS may
receive from source gateways.

The problem is that the source gateways who are talking in codec that are in
first three preferred codecs list in sip profile are working fine, while the
codecs that are at 4th or greater preference order number do not work. The
call is received and accepted by destination gateway then it gets terminated
almost immediately.

Do note that destination gateway is not FS, its some CISCO device that is
also accepting all possible codecs.

Can you guys suggest why it is happening and what are the possible
solutions, other then transcode of course.

Thank you.


-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From brian at freeswitch.org  Sat Aug 22 14:07:09 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 22 Aug 2009 16:07:09 -0500
Subject: [Freeswitch-users] SIP codec preference order
In-Reply-To: <b16156850908221353m3e219f7eyefd29870584e121c@mail.gmail.com>
References: <b16156850908221353m3e219f7eyefd29870584e121c@mail.gmail.com>
Message-ID: <50B93B44-B096-44CD-A44F-7A229D0A7B6A@freeswitch.org>

Can you provide a little bit of log detail?

/b

On Aug 22, 2009, at 3:53 PM, Muhammad Shahzad wrote:

> Can you guys suggest why it is happening and what are the possible  
> solutions, other then transcode of course.




From shaheryarkh at googlemail.com  Sat Aug 22 15:16:01 2009
From: shaheryarkh at googlemail.com (Muhammad Shahzad)
Date: Sun, 23 Aug 2009 03:16:01 +0500
Subject: [Freeswitch-users] SIP codec preference order
In-Reply-To: <50B93B44-B096-44CD-A44F-7A229D0A7B6A@freeswitch.org>
References: <b16156850908221353m3e219f7eyefd29870584e121c@mail.gmail.com>
	<50B93B44-B096-44CD-A44F-7A229D0A7B6A@freeswitch.org>
Message-ID: <b16156850908221516h535ea11fh354c7d31ce316e87@mail.gmail.com>

i just upgraded it to 14599 and its working fine now.

Thank you.


On Sun, Aug 23, 2009 at 2:07 AM, Brian West <brian at freeswitch.org> wrote:

> Can you provide a little bit of log detail?
>
> /b
>
> On Aug 22, 2009, at 3:53 PM, Muhammad Shahzad wrote:
>
> > Can you guys suggest why it is happening and what are the possible
> > solutions, other then transcode of course.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +92 334 422 40 88
MSN: shari_786pk at hotmail.com
Email: shaheryarkh at googlemail.com
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From kevin at johnnyvoip.com  Sat Aug 22 16:24:26 2009
From: kevin at johnnyvoip.com (Kevin Green)
Date: Sat, 22 Aug 2009 19:24:26 -0400
Subject: [Freeswitch-users] can't pass full sip url to dialplan
In-Reply-To: <59ad9ca10908221035l448a9f91ncf26f978506487ad@mail.gmail.com>
References: <59ad9ca10908212340q5cd3be0ag9d5f41b58ad69831@mail.gmail.com> 
	<59ad9ca10908220141u3183b25cnf9ff827d138ea941@mail.gmail.com> 
	<20090822103005.GA4299@jdc.jasonjgw.net>
	<59ad9ca10908220709p6b671e98qd830cdaa0e14446a@mail.gmail.com> 
	<51EC3ADD-A345-4370-832C-D536B8909CBC@freeswitch.org>
	<59ad9ca10908220746u5ea8b569pd9aa805af21538b5@mail.gmail.com> 
	<66B09D4A-A6CC-47DD-94C3-80B0682FBEDA@freeswitch.org>
	<59ad9ca10908220807k227f1b03w32ec44a8ec0db297@mail.gmail.com> 
	<87f2f3b90908220835w6e5c98ecpeecdf81e6ac46f4c@mail.gmail.com> 
	<59ad9ca10908221035l448a9f91ncf26f978506487ad@mail.gmail.com>
Message-ID: <ab57a3e80908221624j7e32d5b3me9a8571f6e66dd59@mail.gmail.com>

X-lite I believe handles the sip: by itself sometime and therefore will try
and place a call to the sip address directly from x-lite without touching
FreeSWITCH. Be aware of this while testing and watch for this behavior
because it might throw off your expectations.

Regards,
   Kevin Green


On Sat, Aug 22, 2009 at 1:35 PM, Henry Huang <red.rain.seven at gmail.com>wrote:

> Michael:
>
> Thank you for making it in "for dummies" format. :P
> These are really nice tips I can use. thanks.
>
>
> On Sat, Aug 22, 2009 at 11:35 PM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Sat, Aug 22, 2009 at 8:07 AM, Henry Huang <red.rain.seven at gmail.com>wrote:
>>
>>> Brian:
>>>
>>> Oh, and again, if it's not passing it to the dialplan. I had suggested to
>>> remove the sample "sip uri" extension in the default.xml dialplan. because
>>> no one can reach the dialplan with prefix "sip:" because sofia is going to
>>> remove that prefix.
>>
>>
>> Well, this isn't entirely accurate. Like Mike J said, if you dialed
>> something like this at the CLI:
>>
>> pa call sip:user at domain.com <sip%3Auser at domain.com>
>>
>> Then you'd need the dialplan entry that handles the SIP URI.
>>
>> Going back to the original question...
>> X-Lite dials 1009 at 4.2.2.2 correct?
>> But you're saying that the dialplan simply sees "1009" as the destination
>> number? I'm looking at the pastebin (10089) and trying to figure out exactly
>> what is happening. All I can see is that you have a context named "Global"
>> so I'm assuming you've made at least some modifications to the default
>> dialplan. Can you pastebin that whole context?
>>
>> The other thing that you should probably do is create an extension in this
>> global context that routes a call to the info application. You could do
>> something like this so that "9992" would do an info dump:
>> <extension name="info">
>>   <condition field="destination_number" expression="^(9992)$">
>>     <action application="info"/>
>>   </condition>
>> </extension>
>>
>> Then reloadxml and make a call to 9992 from your X-Lite client. The CLI
>> will have a dump and you'll see all sorts of variables listed. Many of those
>> are available for you to use for condition matches and routing in the
>> dialplan.
>>
>> Let us know how the info application does in giving you information about
>> the A leg of the call.
>> -MC
>>
>>>
>>>
>>> <!-- dial via SIP uri -->
>>>     <extension name="sip_uri">
>>>       <condition field="destination_number" expression="^sip:(.*)$">
>>>         <action application="bridge" data="sofia/${use_profile}/$1"/>
>>>       </condition>
>>>     </extension>
>>>
>>> On Sat, Aug 22, 2009 at 10:59 PM, Brian West <brian at freeswitch.org>wrote:
>>>
>>>> Because the dial plan is technology agnostic... you have been told
>>>> more than once it won't pass it to the dialplan from mod_sofia...
>>>>
>>>> /b
>>>>
>>>> On Aug 22, 2009, at 9:46 AM, Henry Huang wrote:
>>>>
>>>> > Brian:
>>>> >
>>>> > but why can't I pass "sip:" to dialplan? seems like it's being
>>>> > truncated by sofia..
>>>> > Can you confirm that?
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>>
>>> --
>>> Henry Huang
>>> UniC Solution - Communication Unified
>>> VoIP & Open Source software Consultant
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Henry Huang
> UniC Solution - Communication Unified
> VoIP & Open Source software Consultant
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From diego.viola at gmail.com  Sat Aug 22 22:40:25 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 23 Aug 2009 01:40:25 -0400
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
In-Reply-To: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
References: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
Message-ID: <86a32abc0908222240r2543ebbbhea758cbae17e49cc@mail.gmail.com>

Nice work, keep up the great work :).

On Fri, Aug 21, 2009 at 6:29 PM, Moises Silva<moises.silva at gmail.com> wrote:
> So, I finally took some days to put up OpenR2 working with OpenZAP, which
> means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
> support for. Including Mexico, Brazil, Argentina and others. The stack has
> been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most
> countries that users may be interested in, support for new variants will be
> added on-demand only (in any case users can always tweak the advanced
> configuration file to create their own variants as a last resort).
> I created a wiki page to illustrate the basic
> setup:?http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
> Now is time for testing. I just did minimal testing on my development
> environment, no serious testing, and I know that some stuff is not working
> at this point (I had some issues with variable length DNIS and ANI) which
> should be fixed soon.
> If anyone around happens to have an R2 link and wants to test R2 support in
> OpenZAP, I can give them a hand with the configuration and any issues you
> may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
> #openzap as moy.
>
> --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
> Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From panayotov.vd at gmail.com  Sat Aug 22 23:21:10 2009
From: panayotov.vd at gmail.com (Vassil Panayotov)
Date: Sun, 23 Aug 2009 09:21:10 +0300
Subject: [Freeswitch-users] Yet another question about A500 + FS
Message-ID: <8a9b664c0908222321r20c5c4d8wd08721dc8955f62c@mail.gmail.com>

Hi,

I managed to get our A500 running with FreeSWITCH 1.0.4 stable using wanpipe
3.4.4 drivers. But now I have another problem...
I want to originate calls through event socket, and I only want to receive
ANSWERED(+OK) reply when the user actually answers.

Now the situation is:

====================================
originate openzap/1/a/123456 023
2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT:
CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456]
Ci=[0000000000]
2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4
2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel
OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082]
2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer
OpenZAP/1:1/123456!
API CALL [originate(openzap/1/a/123456 023)] output:
+OK f8fca2be-8fa7-11de-9076-511e29dfc082

2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer
OpenZAP/1:1/123456 to XML[023 at default]
freeswitch at emo-voip> 2009-08-23 08:44:06.743475 [INFO]
mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default
2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
[OpenZAP/1:1/123456] has been answered
2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5
2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6
2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup
OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT (N):
CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3
2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086 Session 2
(OpenZAP/1:1/123456) Ended
2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close Channel
OpenZAP/1:1/123456 [CS_DESTROY]
====================================

Extension 023 is an IVR. As you can see FreeSWITCH answers the call
(2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
[OpenZAP/1:1/123456] has been answered) 20 seconds before user actually pick
up the phone (2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX
EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5).

So Sangoma drivers/daemons report the events correctly.
How can I set FreeSWITCH to answer after receiving RX EVENT (N):
CALL_ANSWERED from the driver?

Thank you,
V. Panayotov
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From diego.viola at gmail.com  Sat Aug 22 23:38:05 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 23 Aug 2009 02:38:05 -0400
Subject: [Freeswitch-users] Inboud Call Queue
In-Reply-To: <7F5652780AE84DFFB371EF62042540A8@saeedlaptop>
References: <210A0FE754E74E5A9B223D3228DB75D3@saeedlaptop>
	<87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com>
	<77308CE88F604444863741D590835B10@saeedlaptop>
	<63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com>
	<EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop>
	<18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com>
	<191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com>
	<62BE31C473E04988BB3AA9553A56C4F8@saeedlaptop>
	<2FED90CE-0CBB-43DF-80E7-7593B236281E@gmail.com>
	<7F5652780AE84DFFB371EF62042540A8@saeedlaptop>
Message-ID: <86a32abc0908222338hc0ee765y8802f406d74f6460@mail.gmail.com>

Hi guys,

I was wondering if some of you run FreeSWITCH on a call center
environment, I ask this because I plan to do that soon and I was
wondering how well mod_fifo works for queues, etc.

Thanks,

Diego

On Thu, May 7, 2009 at 6:08 AM, Saeed Ahmed<saeedahmad1981 at gmail.com> wrote:
> Thanks Seven I?ll try it very soon.
>
>
>
> ________________________________
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of seven
> Sent: Thursday, May 07, 2009 5:42 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Inboud Call Queue
>
>
>
> See this:
>
>
>
> http://wiki.freeswitch.org/wiki/Simple_call_center_using_mod_fifo
>
>
>
>
>
> On May 6, 2009, at 10:15 PM, Saeed Ahmed wrote:
>
> Thanks Guys
>
>
>
> ________________________________
>
> From:?freeswitch-users-bounces at lists.freeswitch.org?[mailto:freeswitch-users-bounces at lists.freeswitch.org]?On
> Behalf Of?Anthony Minessale
> Sent:?Wednesday, May 06, 2009 3:57 PM
> To:?freeswitch-users at lists.freeswitch.org
> Subject:?Re: [Freeswitch-users] Inboud Call Queue
>
>
>
> I worked on the patch and added it to trunk rev 13240
>
>
> On Wed, May 6, 2009 at 7:53 AM, dujinfang <dujinfang at gmail.com> wrote:
>
> The patch haven't been merged into trunk. It should be as easy as execute
> the following command in the FS source code root dir:
>
>
>
> patch < /tmp/the_patch_file_name.diff
>
>
>
> I will post an example on the wiki when I finished, hope be soon.
>
>
>
> On May 6, 2009, at 6:45 PM, Saeed Ahmed wrote:
>
>
> Hi Seven,
>
> I am exactly looking for this functionality.
>
> Please let me know when you are finished with new queue manager app. I?ll
> try it in my call center.
>
> Regarding Patch: is it already part of SVN trunk? If not then could you help
> me how to install it, I have no programming background.
>
> Many Thanks.
>
>
>
> ________________________________
>
> From:?freeswitch-users-bounces at lists.freeswitch.org?[mailto:freeswitch-users-bounces at lists.freeswitch.org]?On
> Behalf Of?seven
> Sent:?Wednesday, May 06, 2009 4:17 AM
> To:?freeswitch-users at lists.freeswitch.org
> Subject:?Re: [Freeswitch-users] Inboud Call Queue
>
>
>
>
>
> On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote:
>
>
>
> Hi Michael,
>
> Thanks for a quick reply.
>
> I would definitely create a test environment, but my question is that will
> it work in required way?
>
> I read that in Mod_fifo agent has to call in queue but I need that all
> incoming calls are automatically distributed between available agents or if
> all are busy then should go to voicemail.
>
> I'm working on a call center like queue scenario right now, I'm pretty sure
> it call automatically distributed to available agents, but the customer will
> stay in the queue if all agents are busy by default. You can bind a key to
> the channel and play a message repeatedly to guide the customer to voicemail
> by press a key.
>
>
>
> Also maybe you need this patch to make the fifo works as desired.
>
>
>
> http://jira.freeswitch.org/browse/MODAPP-272
>
>
>
> I would join IRC for further assistance.
>
>
>
> Thanks.
>
>
>
> ________________________________
>
> From:?freeswitch-users-bounces at lists.freeswitch.org?[mailto:freeswitch-users-bounces at lists.freeswitch.org]?On
> Behalf Of?Michael Collins
> Sent:?Tuesday, May 05, 2009 7:19 PM
> To:?freeswitch-users at lists.freeswitch.org
> Subject:?Re: [Freeswitch-users] Inboud Call Queue
>
>
>
>
>
> On Tue, May 5, 2009 at 9:55 AM, Saeed Ahmed <saeedahmad1981 at gmail.com>
> wrote:
>
> Hi All,
>
> In an inbound call center scenario is it possible that customers calls in
> and calls are distributed between online (who are registered on FS and in
> idle state) agents. I saw some on going discussion on list where it looks
> that currently it?s not possible but I am newbie so maybe I didn?t
> understand it well. If it?s possible then please give me a start point that
> how can I implement it.
>
> I would start here:
> http://wiki.freeswitch.org/wiki/Mod_fifo
>
> I strongly recommend that you set up a FreeSWITCH server and play around
> with it so that you can learn the pros and cons of using the FIFO queues. It
> would be best if you could set up a few phones and set them as FIFO agents
> and then have someone help you make test calls so that you can emulate your
> CC environment.
>
> Also, you might want to join us on IRC: #freeswitch on?irc.freenode.net?-
> there are several users who've had real world experience with mod_fifo and
> they might be in a good position to answer your questions real-time.
>
> -MC
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH?http://www.freeswitch.org/
> ClueCon?http://www.cluecon.com/
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC:?irc.freenode.net?#freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From jason at jasonjgw.net  Sun Aug 23 00:08:48 2009
From: jason at jasonjgw.net (Jason White)
Date: Sun, 23 Aug 2009 17:08:48 +1000
Subject: [Freeswitch-users] Inboud Call Queue
In-Reply-To: <86a32abc0908222338hc0ee765y8802f406d74f6460@mail.gmail.com>
References: <87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com>
	<77308CE88F604444863741D590835B10@saeedlaptop>
	<63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com>
	<EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop>
	<18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com>
	<191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com>
	<62BE31C473E04988BB3AA9553A56C4F8@saeedlaptop>
	<2FED90CE-0CBB-43DF-80E7-7593B236281E@gmail.com>
	<7F5652780AE84DFFB371EF62042540A8@saeedlaptop>
	<86a32abc0908222338hc0ee765y8802f406d74f6460@mail.gmail.com>
Message-ID: <20090823070848.GA16072@jdc.jasonjgw.net>

Diego Viola <diego.viola at gmail.com> wrote:
> I was wondering if some of you run FreeSWITCH on a call center
> environment, I ask this because I plan to do that soon and I was
> wondering how well mod_fifo works for queues, etc.

This was mentioned on the list once before, and it might be what you want:
http://wiki.opencsm.org/wiki/index.php/Spice_Telephony
(Spice Telephony).




From darklion11 at yahoo.com  Sun Aug 23 00:26:44 2009
From: darklion11 at yahoo.com (Edmar Cruz)
Date: Sun, 23 Aug 2009 00:26:44 -0700 (PDT)
Subject: [Freeswitch-users]  Starting Freeswitch using Intercom
Message-ID: <25100943.post@talk.nabble.com>


Hi,

  I dont know how can I start Freeswitch using Intercom device, can you help
me on this? Is there an alternative software like X-Lite but only when I
press call on the Intercom device? 

Thanks, 
Edmar
-- 
View this message in context: http://www.nabble.com/Starting-Freeswitch-using-Intercom-tp25100943p25100943.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From diego.viola at gmail.com  Sun Aug 23 01:02:59 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sun, 23 Aug 2009 04:02:59 -0400
Subject: [Freeswitch-users] Inboud Call Queue
In-Reply-To: <20090823070848.GA16072@jdc.jasonjgw.net>
References: <87f2f3b90905051018j627c66eau87bac3a09daafa52@mail.gmail.com>
	<63626F42-2BD1-489A-B181-6E1E5676F6EC@gmail.com>
	<EC164FB6AE3E4D5CB8BC64900475737A@saeedlaptop>
	<18C9A32C-8BF2-45A7-993D-AC61D62D7ECB@gmail.com>
	<191c3a030905060657q533b3c05rea9687a80add4311@mail.gmail.com>
	<62BE31C473E04988BB3AA9553A56C4F8@saeedlaptop>
	<2FED90CE-0CBB-43DF-80E7-7593B236281E@gmail.com>
	<7F5652780AE84DFFB371EF62042540A8@saeedlaptop>
	<86a32abc0908222338hc0ee765y8802f406d74f6460@mail.gmail.com>
	<20090823070848.GA16072@jdc.jasonjgw.net>
Message-ID: <86a32abc0908230102i79bbbe7eoee7313f329f55607@mail.gmail.com>

Looks nice, is anyone running that in production?

On Sun, Aug 23, 2009 at 3:08 AM, Jason White <jason at jasonjgw.net> wrote:

> Diego Viola <diego.viola at gmail.com> wrote:
> > I was wondering if some of you run FreeSWITCH on a call center
> > environment, I ask this because I plan to do that soon and I was
> > wondering how well mod_fifo works for queues, etc.
>
> This was mentioned on the list once before, and it might be what you want:
> http://wiki.opencsm.org/wiki/index.php/Spice_Telephony
> (Spice Telephony).
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From egghunt at gmail.com  Sun Aug 23 04:25:20 2009
From: egghunt at gmail.com (Arnaldo de Moraes Pereira)
Date: Sun, 23 Aug 2009 08:25:20 -0300
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
In-Reply-To: <86a32abc0908222240r2543ebbbhea758cbae17e49cc@mail.gmail.com>
References: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com> 
	<86a32abc0908222240r2543ebbbhea758cbae17e49cc@mail.gmail.com>
Message-ID: <ff2539090908230425i345b9450gfdc0cb4d015fc93c@mail.gmail.com>

Thanks a lot, moy, this is great. I'll check to see if there's somewhere I
can test it.

On Sun, Aug 23, 2009 at 2:40 AM, Diego Viola <diego.viola at gmail.com> wrote:

> Nice work, keep up the great work :).
>
> On Fri, Aug 21, 2009 at 6:29 PM, Moises Silva<moises.silva at gmail.com>
> wrote:
> > So, I finally took some days to put up OpenR2 working with OpenZAP, which
> > means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
> > support for. Including Mexico, Brazil, Argentina and others. The stack
> has
> > been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers
> most
> > countries that users may be interested in, support for new variants will
> be
> > added on-demand only (in any case users can always tweak the advanced
> > configuration file to create their own variants as a last resort).
> > I created a wiki page to illustrate the basic
> > setup: http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
> > Now is time for testing. I just did minimal testing on my development
> > environment, no serious testing, and I know that some stuff is not
> working
> > at this point (I had some issues with variable length DNIS and ANI) which
> > should be fixed soon.
> > If anyone around happens to have an R2 link and wants to test R2 support
> in
> > OpenZAP, I can give them a hand with the configuration and any issues you
> > may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
> > #openzap as moy.
> >
> > --
> > Moises Silva
> > Software Developer
> > Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3
> > Canada
> > t. 1 905 474 1990 x 128 | e. moy at sangoma.com
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Arnaldo M Pereira
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From merul at mac.com  Sun Aug 23 10:27:45 2009
From: merul at mac.com (Merul Patel)
Date: Sun, 23 Aug 2009 18:27:45 +0100
Subject: [Freeswitch-users] FXO and analogue phones
Message-ID: <C281504B-3D79-465E-ADA9-25C140FCF50F@mac.com>

I have a Freeswitch setup working on an Alix embedded platform in  
conjunction with a USB FXO device from Sangoma. My goal is to be able  
to either answer incoming calls on a softphone or on a POTS handset  
elsewhere in the building, and to also be able to make outgoing calls  
from either. For clarity, the analogue line has two physical  
extensions, one connected to the POTS and the other to the FXO.

I can make and receive calls fine, but have problems when the call is  
answered on the POTS handset.

Here is the dialplan I initially used  in /opt/freeswitch/conf/ 
dialplans/public/01_incoming.xml:

<include>
   <extension name="public_did">
     <condition field="${strftime(%w)}" expression="^(\d)$">
       <!-- There seems to be a delay of 7 seconds from when FS starts  
dealing with the call and from when it starts ringing -->
       <action application="sleep" data="23000"/>
       <action application="set" data="domain_name=$${domain}"/>
       <action application="transfer" data="1001 XML default"/>
     </condition>
   </extension>
</include>

It's pretty basic, and if the softphone is not registered or does not  
answer then the call goes to voicemail. However the call will always  
go to voicemail, and the voicemail application will begin to execute  
after the call has been answered on the POTS handset.

I've been trying to make the dialplan more useful, by having it ring  
the softphone immediately, and only transfer the call to the voicemail  
application if the line is still ringing. I'm in the UK, hence my  
choice of frequencies in the tone_detect application:

<include>
   <extension name="public_did">
     <condition field="${strftime(%w)}" expression="^(\d)$">
       <!-- There seems to be a delay of 7 seconds from when FS starts  
dealing with the call and from when it starts ringing -->
       <action application="set" data="call_timeout=23"/>
       <action application="set" data="continue_on_fail=true"/>
       <action application="set" data="hangup_after_bridge=true"/>
       <action application="bridge" data="sofia/internal/1001%$$ 
{domain}"/>
       <action application="sleep" data="23000"/>
       <action application="tone_detect" data="ring 400,450 r +5000  
set RING=true"/>
       <action application="transfer" data="public_answer_and_email"/>
     </condition>
   </extension>

<extension name="public_answer_and_email">
     <condition field="RING" expression="true">
        <action application="answer"/>
        <action application="voicemail" data="default $${domain} 1001"/>
     </condition>
   </extension>
</include>

Unfortunately, this is not working, and the logs are not yielding  
anything the is helpful to me. Is my use of the tone_detect  
application and the basic dialplan correct?

Merul
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From help at pdscc.com  Sun Aug 23 11:51:53 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 11:51:53 -0700
Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3
Message-ID: <20090823185152.D17845FE@sinclaire.sibble.net>

Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server, 
then went to install FreePBX v3, I've gotten all the prerequisities in the 
wizard fixed except for ESL

As per

http://wiki.freeswitch.org/wiki/Event_Socket_Library
http://wiki.freeswitch.org/wiki/Event_Socket

I go into my FS source dir

/home/sibbleh/freeswitch-1.0.4/libs/esl

Run make and then "sudo make phpmod-install" 

and I get 


$ sudo make phpmod-install
make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-
I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb 
-I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-
variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes" CXXFLAGS="-
I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g -ggdb 
-I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" 
CXX_CFLAGS="" -C php
make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
g++  -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g 
-ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -
I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -
I/usr/include/php5/Zend -I/usr/include/php5/ext -
I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64 -
Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
cc1plus: warnings being treated as errors
esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1047: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1073: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int, 
zval*, zval**, zval*, int)':
esl_wrap.cpp:1111: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int, 
zval*, zval**, zval*, int)':
esl_wrap.cpp:1141: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1172: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1198: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1234: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1269: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1294: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**, zval*, 
int)':
esl_wrap.cpp:1346: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1403: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1441: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1478: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1508: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1538: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1571: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_addHeader(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1611: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_delHeader(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1644: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_firstHeader(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1674: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLevent_nextHeader(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1704: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_0(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1744: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_1(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1770: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_new_ESLconnection(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1803: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_socketDescriptor(int, 
zval*, zval**, zval*, int)':
esl_wrap.cpp:1846: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_connected(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1872: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_getInfo(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1898: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_send(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:1931: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendRecv(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:1964: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_api(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:2007: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_bgapi(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:2050: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendEvent(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2082: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEvent(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2108: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEventTimed(int, 
zval*, zval**, zval*, int)':
esl_wrap.cpp:2141: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_filter(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2181: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_events(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2221: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_execute(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2272: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_executeAsync(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2323: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_setAsyncExecute(int, 
zval*, zval**, zval*, int)':
esl_wrap.cpp:2356: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_setEventLock(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2389: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_ESLconnection_disconnect(int, zval*, 
zval**, zval*, int)':
esl_wrap.cpp:2415: error: format not a string literal and no format arguments
esl_wrap.cpp: In function 'void _wrap_eslSetLogLevel(int, zval*, zval**, 
zval*, int)':
esl_wrap.cpp:2438: error: format not a string literal and no format arguments
make[1]: *** [esl_wrap.o] Error 1
make[1]: Leaving directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
make: *** [phpmod] Error 2

Same thing happens if I try sudo make everymod

Checking the list archives I found this thread

http://www.nabble.com/ESL-Wrapper-td22209991.html#a22222338

I've made sure that the php-dev packages are installed. Any suggestions on 
what to do next?
-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From Prometheus001 at gmx.net  Sun Aug 23 13:20:16 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Sun, 23 Aug 2009 22:20:16 +0200
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
 freeswitch (ptime:30 msecs)
In-Reply-To: <191c3a030908211309j60c16b26h1e6445891136ae7f@mail.gmail.com>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>	<4A8DB808.1030903@gmx.net>	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>	<4A8EBF77.1080204@gmx.net>	<BD3DE8E0-3363-465C-985D-EC593AE45694@freeswitch.org>
	<191c3a030908211309j60c16b26h1e6445891136ae7f@mail.gmail.com>
Message-ID: <4A91A480.7080307@gmx.net>

Hello Anthony,

I set PCMA at 30i,PCMU at 30i and I can see in the logs that PCMA is used.
However ptime is set to 20 msec as shown in the Logs:

2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP:
v=0
o=user 2075230 2075230 IN IP4 217.xx.xx.xxx
s=call
c=IN IP4 217.xx.xx.xxx
t=0 0
m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
a=rtpmap:2 G726-32/8000
a=rtpmap:102 G726-32/8000
a=rtpmap:100 G726-40/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=rtcp:7079

2009-08-23 22:11:29.530301 [DEBUG] switch_core_state_machine.c:404
(sofia/internal/02xxxxxxxxx at fs1.my.domain) State NEW
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare
[PCMA:8:8000:0]/[G722:9:8000:20]
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare
[PCMA:8:8000:0]/[PCMU:0:8000:20]
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec Compare
[PCMA:8:8000:0]/[PCMA:8:8000:20]
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:2090 Set Codec
sofia/internal/02xxxxxxxxx at fs1.my.domain PCMA/8000 20 ms 160 samples
2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
payload to 101

Sound from FS to Fritzbox is fine. Sound from Fritzbox to FS is horrible.

Best regards
Peter

Anthony Minessale schrieb:
> try setting FS to 30ms too
>
> edit vars.xml and add @30i to everywhere you see  PCMU or PCMA so it
> looks like PCMU at 30i
>
> from:
>
>   <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>   <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>
> to:
>
>   <X-PRE-PROCESS cmd="set"
> data="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU at 30i,PCMA at 30i,GSM"/>
>   <X-PRE-PROCESS cmd="set"
> data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i,GSM"/>
>
>
> On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian at freeswitch.org
> <mailto:brian at freeswitch.org>> wrote:
>
>     You can ship me one whois bkw.org <http://bkw.org>, I can add it
>     to my lab.
>
>     /b
>
>     On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
>
>     >
>     > BTW: We can ship you a FritzBox if you need one for testing.
>
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From help at pdscc.com  Sun Aug 23 14:37:47 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 14:37:47 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
Message-ID: <20090823213745.D81819D5@sinclaire.sibble.net>

I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp capable 
endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone client, 
2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i (symbian 
s60) and an O2 Xda Flame (windows mobile 5).

All 3 endpoints are registered with FS using the default extensions of 1000-
1003

With global_setvar zrtp_secure_media=true the zrtp negotiation between end 
points happens but the SAS never matches,below is console output for a call 
between 2 of the endpoints 


2009-08-23 14:10:17.643073 [NOTICE] mod_sofia.c:1509 Pre-Answer 
sofia/internal/1003 at 10.12.13.45!
2009-08-23 14:10:21.257568 [NOTICE] sofia.c:3794 Channel 
[sofia/internal/sip:1000 at 10.12.13.166:5062] has been answered
2009-08-23 14:10:21.275521 [NOTICE] switch_ivr_originate.c:2015 Channel 
[sofia/internal/1003 at 10.12.13.45] has been answered
2009-08-23 14:10:22.232053 [WARNING] mod_sofia.c:810 We were told to use 
ptime 20 but what they meant to say was 80
This issue has so far been identified to happen on the following broken 
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who 
knows what will happen..
2009-08-23 14:11:34.496118 [NOTICE] sofia.c:322 Hangup 
sofia/internal/sip:1000 at 10.12.13.166:5062 [CS_EXCHANGE_MEDIA] 
[NORMAL_CLEARING]
2009-08-23 14:11:34.512100 [NOTICE] switch_ivr_bridge.c:1016 Hangup 
sofia/internal/1003 at 10.12.13.45 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1086 Session 16 
(sofia/internal/sip:1000 at 10.12.13.166:5062) Ended
2009-08-23 14:11:34.552158 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/sip:1000 at 10.12.13.166:5062 [CS_DESTROY]
2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1086 Session 15 
(sofia/internal/1003 at 10.12.13.45) Ended
2009-08-23 14:11:34.556441 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/1003 at 10.12.13.45 [CS_DESTROY]

Of note, with the endpoints registered through the Ekiga sip server, the sas 
DOES match on both ends.

With global_setvar zrtp_secure_media=false, the endpoints can't detect a zrtp 
peer.  

Reading the list archives hasn't enlightened me.

I see this comment from 2008

http://www.nabble.com/Freeswitch-and-Twinkle-and-ZRTP-
td18518140.html#a18518343

    On Jul 17, 2008, at 4:23 PM, Michael Jerris wrote:

    > it should in bypass_media or proxy_media modes.  in the other modes we
    > are in the media path and would not know how to handle the encrypted
    > packets.
    >
    > Mike

Is this still relevant? Or is there some other setting not covered here

http://wiki.freeswitch.org/wiki/ZRTP

to make this work properly? I ask firstly about  this in the context of a 
peer 2 peer zrtp communication between the endpoints, then secondly in the 
case of FS acting as a trusted middleman as in section 2 here

http://www.zfoneproject.com/docs/asterisk/man/html/u_guide.html#passthrough

Lastly how does one implement the security enrollment as noted above with FS
-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From brian at freeswitch.org  Sun Aug 23 15:09:56 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 17:09:56 -0500
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090823213745.D81819D5@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>
Message-ID: <FC5818F6-C173-4D47-9C2A-4CCC24A46D6B@freeswitch.org>

This is because you didn't install the zrtpagent.lua script and dial  
zrtp on your keypad to enroll the FS box as a trusted man in the  
middle... which btw will only work with the unreleased zfone3.

/b

On Aug 23, 2009, at 4:37 PM, Harondel J. Sibble wrote:

> I've got 1.0.4 running with zrtp on ubuntu 9.0.4. I have 3 zrtp  
> capable
> endpoints: an xp desktop running ekiga with the 0.92 build 218 zfone  
> client,
> 2 cell phones running ver 2.0.5 of the Tivi softphone: a nokia e61i  
> (symbian
> s60) and an O2 Xda Flame (windows mobile 5).
>
> All 3 endpoints are registered with FS using the default extensions  
> of 1000-
> 1003
>
> With global_setvar zrtp_secure_media=true the zrtp negotiation  
> between end
> points happens but the SAS never matches,below is console output for  
> a call
> between 2 of the endpoints




From help at pdscc.com  Sun Aug 23 15:39:38 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 15:39:38 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <FC5818F6-C173-4D47-9C2A-4CCC24A46D6B@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<FC5818F6-C173-4D47-9C2A-4CCC24A46D6B@freeswitch.org>
Message-ID: <20090823223936.84B1373E@sinclaire.sibble.net>

Brian, okay, that answers the case with FS acting as a trusted man in the 
middle, but what about in the peer to peer case? Shouldn't FS just be passing 
the ztrp traffic through to the endpoints? Or am I misunderstanding how it's 
supposed to work?

Secondly where would I find info about zrtpagent.lua? Doing a search for that 
term on the wiki returns no results, ditto for a search of the nabble list 
archives (other than your response today of course). Also ditto for a search 
of the box running FS.

On 23 Aug 2009 at 17:09, Brian West wrote:

> This is because you didn't install the zrtpagent.lua script and dial  
> zrtp on your keypad to enroll the FS box as a trusted man in the  
> middle... which btw will only work with the unreleased zfone3.
> 
> /b

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From brian at freeswitch.org  Sun Aug 23 15:48:27 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 17:48:27 -0500
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090823223936.84B1373E@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<FC5818F6-C173-4D47-9C2A-4CCC24A46D6B@freeswitch.org>
	<20090823223936.84B1373E@sinclaire.sibble.net>
Message-ID: <B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>


On Aug 23, 2009, at 5:39 PM, Harondel J. Sibble wrote:

> Brian, okay, that answers the case with FS acting as a trusted man  
> in the
> middle, but what about in the peer to peer case? Shouldn't FS just  
> be passing
> the ztrp traffic through to the endpoints? Or am I misunderstanding  
> how it's
> supposed to work?

Nope.  FreeSWITCH is a b2bua so the traffic is decrypted... and  
relayed and encrypted again to the far side.  Which is why you have to  
use the trusted man in the middle stuff but you can't use it fully  
unless you have zfone3 beta or release.

> Secondly where would I find info about zrtpagent.lua? Doing a search  
> for that
> term on the wiki returns no results, ditto for a search of the  
> nabble list
> archives (other than your response today of course). Also ditto for  
> a search
> of the box running FS.

scripts/lua/ (in src tree)

/b




From help at pdscc.com  Sun Aug 23 16:16:11 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 16:16:11 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090823223936.84B1373E@sinclaire.sibble.net>,
	<B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>
Message-ID: <20090823231610.3398273E@sinclaire.sibble.net>



On 23 Aug 2009 at 17:48, Brian West wrote:

> Nope.  FreeSWITCH is a b2bua so the traffic is decrypted... and  

Hadn't heard that term before

http://en.wikipedia.org/wiki/Back-to-back_user_agent

that clears it up.  Any plans to offer straight proxy/passthru?

> relayed and encrypted again to the far side.  Which is why you have to  
> use the trusted man in the middle stuff but you can't use it fully  
> unless you have zfone3 beta or release.

Which explains the mismatched sas on each end. Okay got it.
 
> scripts/lua/ (in src tree)

ahh, it's zrtp_agent.lua which is why I didn't find it. I'll add that into 
the mix and cross my fingers that zfone3 gets released soon along with it's 
inclusion into the softphones I have on my smartphone devices.
-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From intralanman at freeswitch.org  Sun Aug 23 16:23:23 2009
From: intralanman at freeswitch.org (Raymond Chandler)
Date: Sun, 23 Aug 2009 19:23:23 -0400
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
Message-ID: <3BF035D5-ED50-406E-8D19-FAAD314BB045@freeswitch.org>

Could you load freeswitch with a couple hundred calls then run the  
test again.. and do the same to asterisk and see how the numbers stack  
up then? I'm just curious to see what happens at that point.

-Ray

On Aug 21, 2009, at 3:15 PM, Rogelio Perez wrote:

> Hi Everyone,
>
> I'm working on a PBX project for the Sheevaplug ARM based computer,  
> with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
> So far I've found a big difference between Freeswitch and Asterisk  
> performance times.
> This is a comparison of the time it takes them to perform different  
> actions:
>
> startup Freeswitch:			3 min.
> startup Asterisk:			2 sec.
>
> call extension Freeswitch:	6 sec.
> call extension Asterisk:		0 sec.
>
> shutdown Freeswitch:		6.5 sec
> shutdown Asterisk:			0 sec.
>
> reload config Freeswitch: 	1 sec.
> reload config Asterisk:		1 sec.
>
> Both were built from sources natively (no cross-compiling), and they  
> use the default startup configurations.
> I have managed to lower the Freeswitch times by disabling most of  
> the modules and recompiling, but it is still far away from Asterisk  
> (i.e. FS startup time 2.5 min).
>
> 1. Is there any way to further improve Freeswitch performance for  
> the ARM architecture?
> 2. Can this be related to the lack of a FPU (the Sheevalug emulates  
> the floating point operations).
> 3. On the startup I see this error repeated many times: [ERR]  
> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be  
> related?
>
> Thanks,
> Rogelio Perez
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From matt at venturevoip.com  Sun Aug 23 16:27:56 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Mon, 24 Aug 2009 11:27:56 +1200
Subject: [Freeswitch-users] Couple of questions
Message-ID: <4A91D07C.1080805@venturevoip.com>

Hi,

I don't see how I can read some responses to command using esl.

I.E. esl_send_recv(&handle, "api show calls count\n\n");

and

printf("Header Test %s\n", esl_event_get_header(event, "API-Command"));
printf("Body Test %s\n", esl_event_get_body(event));

the header details are returned.

The body is null.

Also, I can originate a call and set the account code for it, but how do 
I get a list of calls with their account codes?

Do I get a list of calls then go through them one by one and get the 
variables for those calls by uuid?

Does anyone have any documentation for the esl api?

Even if I could read some comments from a usage of it would be useful.

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)



From sprice at gmail.com  Sun Aug 23 16:31:16 2009
From: sprice at gmail.com (SP)
Date: Sun, 23 Aug 2009 18:31:16 -0500
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <3BF035D5-ED50-406E-8D19-FAAD314BB045@freeswitch.org>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
	<3BF035D5-ED50-406E-8D19-FAAD314BB045@freeswitch.org>
Message-ID: <7e2ac3270908231631y5cc4d0bbmc0ceedcb070a04fa@mail.gmail.com>

Don't forget to press tab at the asterisk console!  :)

On Sun, Aug 23, 2009 at 18:23, Raymond Chandler
<intralanman at freeswitch.org>wrote:

> Could you load freeswitch with a couple hundred calls then run the test
> again.. and do the same to asterisk and see how the numbers stack up then?
> I'm just curious to see what happens at that point.
> -Ray
>
> On Aug 21, 2009, at 3:15 PM, Rogelio Perez wrote:
>
> Hi Everyone,
>
> I'm working on a PBX project for the Sheevaplug<http://www.google.com/url?sa=t&source=web&ct=res&cd=1&url=http%253A%252F%252Fwww.marvell.com%252Fproducts%252Fembedded_processors%252Fdeveloper%252Fkirkwood%252Fsheevaplug.jsp&ei=EOOOSo6ILcyOtgeNruTOBA&usg=AFQjCNFREhfy_erj5irBWk8XFUjkQOP-aw&sig2=pAHYlI15IbZ5Kcw0n-nIvA> ARM
> based computer, with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
> So far I've found a big difference between Freeswitch and Asterisk
> performance times.
> This is a comparison of the time it takes them to perform different
> actions:
>
> startup Freeswitch: 3 min.
> startup Asterisk: 2 sec.
>
> call extension Freeswitch: 6 sec.
> call extension Asterisk: 0 sec.
>
> shutdown Freeswitch: 6.5 sec
>
> shutdown Asterisk: 0 sec.
>
>
> reload config Freeswitch: 1 sec.
> reload config Asterisk: 1 sec.
>
>
> Both were built from sources natively (no cross-compiling), and they use
> the default startup configurations.
> I have managed to lower the Freeswitch times by disabling most of the
> modules and recompiling, but it is still far away from Asterisk (i.e. FS
> startup time 2.5 min).
>
> 1. Is there any way to further improve Freeswitch performance for the ARM
> architecture?
> 2. Can this be related to the lack of a FPU (the Sheevalug emulates the
> floating point operations).
> 3. On the startup I see this error repeated many times: [ERR]
> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be related?
>
> Thanks,
> Rogelio Perez
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Shannon
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From brian at freeswitch.org  Sun Aug 23 16:43:31 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 18:43:31 -0500
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090823231610.3398273E@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090823223936.84B1373E@sinclaire.sibble.net>,
	<B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>
	<20090823231610.3398273E@sinclaire.sibble.net>
Message-ID: <A4BF66BF-AE4F-4D1B-9102-569EBF16A08C@freeswitch.org>

I can confirm that it works 100% correct passing the SAS across the  
bridge correctly once you trust the switch in the middle.

/b

On Aug 23, 2009, at 6:16 PM, Harondel J. Sibble wrote:

>
> ahh, it's zrtp_agent.lua which is why I didn't find it. I'll add  
> that into
> the mix and cross my fingers that zfone3 gets released soon along  
> with it's
> inclusion into the softphones I have on my smartphone devices.




From help at pdscc.com  Sun Aug 23 16:50:32 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 16:50:32 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090823231610.3398273E@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>,
	<20090823231610.3398273E@sinclaire.sibble.net>
Message-ID: <20090823235030.A95BF5FE@sinclaire.sibble.net>



On 23 Aug 2009 at 16:16, Harondel J. Sibble wrote:

> ahh, it's zrtp_agent.lua which is why I didn't find it. I'll add that into
> the mix and cross my fingers that zfone3 gets released soon along with it's
> inclusion into the softphones I have on my smartphone devices.

Well good news for the Tiviphone client

    Dear Harondel J. Sibble,

    We support zfone3 starting from Tiviphone release 2.0.6 it is available for
    Symbian try upgrading.

    -- 
    Harondel J. Sibble 
    Sibble Computer Consulting
    Creating Solutions for the small and medium business computer user.
    help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
    (604) 739-3709 (voice)




From brian at freeswitch.org  Sun Aug 23 16:50:27 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 18:50:27 -0500
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <4A91D07C.1080805@venturevoip.com>
References: <4A91D07C.1080805@venturevoip.com>
Message-ID: <6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>


On Aug 23, 2009, at 6:27 PM, Matt Riddell wrote:

> Hi,
>
> I don't see how I can read some responses to command using esl.
>
> I.E. esl_send_recv(&handle, "api show calls count\n\n");
>
> and
>
> printf("Header Test %s\n", esl_event_get_header(event, "API- 
> Command"));
> printf("Body Test %s\n", esl_event_get_body(event));
>
> the header details are returned.
>
> The body is null.

I'm not too sure about using ESL in C, I have used it pretty much  
exclusively in perl.

> Also, I can originate a call and set the account code for it, but  
> how do
> I get a list of calls with their account codes?

originate {account_code=1234}sofia/profile/target at ip ....

You can get the list of the channels via "show channels" or bridged  
calls with "show calls"

 From there you have the UUID's you can call uuid_dump on them to get  
all the variables.

> Do I get a list of calls then go through them one by one and get the
> variables for those calls by uuid?

You could do this or setup a listener to get the events as they happen  
and keep the info you need.

> Does anyone have any documentation for the esl api?

http://docs.freeswitch.org/ (this should help, its under files list  
see esl.h)

> Even if I could read some comments from a usage of it would be useful.

I just find it interesting you're doing this with C.

> -- 
> Cheers,
>
> Matt Riddell
> Director
> _______________________________________________
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From brian at freeswitch.org  Sun Aug 23 16:53:36 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 18:53:36 -0500
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090823235030.A95BF5FE@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>,
	<20090823231610.3398273E@sinclaire.sibble.net>
	<20090823235030.A95BF5FE@sinclaire.sibble.net>
Message-ID: <09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org>

Wish they would send me one for my E63 for testing... only been  
working with zfone 3 so far.

/b

On Aug 23, 2009, at 6:50 PM, Harondel J. Sibble wrote:

> Well good news for the Tiviphone client




From matt at venturevoip.com  Sun Aug 23 16:57:37 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Mon, 24 Aug 2009 11:57:37 +1200
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
References: <4A91D07C.1080805@venturevoip.com>
	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
Message-ID: <4A91D771.9080609@venturevoip.com>

On 24/08/09 11:50 AM, Brian West wrote:

>> Even if I could read some comments from a usage of it would be useful.
>
> I just find it interesting you're doing this with C.

:)  The rest of the application is in C, so it makes sense to use 
FreeSwitch's esl in C.

Thanks for your help man will let you know if I have any other questions :)

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)



From help at pdscc.com  Sun Aug 23 17:20:12 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 17:20:12 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090823235030.A95BF5FE@sinclaire.sibble.net>,
	<09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org>
Message-ID: <20090824002011.04DDE9CF@sinclaire.sibble.net>

They have a trial version of the full client available from their website, I 
think it's only for 10 days though. I suspect if you approached them, they'd 
probably give you a full client for permanent use for interoperability 
testing.

http://www.tivi.com/en/download/credit_paypal.php

Opps, my bad, it's only for 3 days :-(

You can bid on the price you want to pay if you are into buying a copy.

Just got the word back from them, the 2.0.6 client is available for Windows 
Mobile but not officially released since it has some issues with video 
encryption. Since I'm not using video yet, I am going to upgrade the windows 
mobile phones with this version.

So I popped the zrtp_agent.lua script into

/usr/local/freeswitch/scripts/

added following line to 

/usr/local/freeswitch/conf/autoload_configs/lua.conf.xml

   <!--<param name="startup-script" value="zrtp_agent.lua"/>-->

under this section

    <!--
        The following options identifies a lua script that is launched
        at startup and may live forever in the background.
        You can define multiple lines, one for each script you
        need to run.

And restarted FS, when I try to connect and type zrtp on the keypad it 
doesn't work and I see

2009-08-23 17:06:46.272135 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1001 at 10.12.13.45 [03265312-9042-11de-8c5d-d333d780ffc7]
2009-08-23 17:06:46.288730 [INFO] mod_dialplan_xml.c:315 Processing 1001-
>zrtp in context default
2009-08-23 17:06:46.301899 [NOTICE] switch_ivr.c:1349 Transfer 
sofia/internal/1001 at 10.12.13.45 to enum[zrtp at default]
2009-08-23 17:06:46.495944 [INFO] switch_core_state_machine.c:136 No Route, 
Aborting
2009-08-23 17:06:46.495944 [NOTICE] switch_core_state_machine.c:137 Hangup 
sofia/internal/1001 at 10.12.13.45 [CS_ROUTING] [NO_ROUTE_DESTINATION]
2009-08-23 17:06:46.503261 [NOTICE] switch_core_session.c:1086 Session 3 
(sofia/internal/1001 at 10.12.13.45) Ended
2009-08-23 17:06:46.503261 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/1001 at 10.12.13.45 [CS_DESTROY]

Were you being literal when you said to type in zrtp on the phones keypad? I 
assume you meant dial the phone number equivalent to that.

I also assume this is not working because 
1) I should put the zrtp_agent.lua script in one of the designated script 
dirs like 

    <!-- <param name="module-directory" value="/usr/lib/lua/5.1/?.so"/> -->
    <!-- <param name="module-directory" value="/usr/local/lib/lua/5.1/?.so"/> 
-->

or 

2) I need to wait until zfone3 is released and recompile FS with the new tar 
file from the zfoneproject site

or

3) both of the above apply?


On 23 Aug 2009 at 18:53, Brian West wrote:

> Wish they would send me one for my E63 for testing... only been  
> working with zfone 3 so far.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From brian at freeswitch.org  Sun Aug 23 17:53:14 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 19:53:14 -0500
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090824002011.04DDE9CF@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090823235030.A95BF5FE@sinclaire.sibble.net>,
	<09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org>
	<20090824002011.04DDE9CF@sinclaire.sibble.net>
Message-ID: <C9F0463C-F22E-4339-879C-FF5B35C3FE43@freeswitch.org>


On Aug 23, 2009, at 7:20 PM, Harondel J. Sibble wrote:

> added following line to
>
> /usr/local/freeswitch/conf/autoload_configs/lua.conf.xml

WRONG.

>
>
>   <!--<param name="startup-script" value="zrtp_agent.lua"/>-->
>

Don't touch this.

> under this section
>
>    <!--
>        The following options identifies a lua script that is launched
>        at startup and may live forever in the background.
>        You can define multiple lines, one for each script you
>        need to run.
>
> And restarted FS, when I try to connect and type zrtp on the keypad it
> doesn't work and I see

Just put it into the scripts folder and run it from the dialplan see  
default configs.

     <!-- install zrtp_agent.lua into scripts (ZRTP == 9787) -->
     <extension name="zrtp_enrollement">
       <condition field="destination_number" expression="^9787$">
	<action application="answer"/>
	<action application="lua" data="zrtp_agent.lua"/>
       </condition>
     </extension>


/b

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From brian at freeswitch.org  Sun Aug 23 17:54:54 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 19:54:54 -0500
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <4A91D771.9080609@venturevoip.com>
References: <4A91D07C.1080805@venturevoip.com>
	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
	<4A91D771.9080609@venturevoip.com>
Message-ID: <DE14F3F4-C7A7-481D-B537-2B9D283AF600@freeswitch.org>


On Aug 23, 2009, at 6:57 PM, Matt Riddell wrote:

> :)  The rest of the application is in C, so it makes sense to use
> FreeSwitch's esl in C.

I'm a perl monkey when it comes to using ESL  :P  I like reusing all  
the modules already out there plus I'm a little lazy :P

> Thanks for your help man will let you know if I have any other  
> questions :)

kewl.

/b




From help at pdscc.com  Sun Aug 23 19:05:25 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 19:05:25 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <C9F0463C-F22E-4339-879C-FF5B35C3FE43@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090824002011.04DDE9CF@sinclaire.sibble.net>,
	<C9F0463C-F22E-4339-879C-FF5B35C3FE43@freeswitch.org>
Message-ID: <20090824020523.A44439CF@sinclaire.sibble.net>



On 23 Aug 2009 at 19:53, Brian West wrote:

> Just put it into the scripts folder and run it from the dialplan see  
> default configs.
> 
>      <!-- install zrtp_agent.lua into scripts (ZRTP == 9787) -->
>      <extension name="zrtp_enrollement">
>        <condition field="destination_number" expression="^9787$">
> 	<action application="answer"/>
> 	<action application="lua" data="zrtp_agent.lua"/>
>        </condition>
>      </extension>

Okay, where do I get the audio files?

2009-08-23 18:54:08.991629 [INFO] mod_dialplan_xml.c:315 Processing 1001-
>9787 in context default
2009-08-23 18:54:09.10863 [NOTICE] mod_dptools.c:649 Channel 
[sofia/internal/1001 at 10.12.13.45] has been answered
2009-08-23 18:54:09.215288 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr                  tp/zrtp-
status_securing.wav] [System error : No such file or directory.]
2009-08-23 18:54:12.233267 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr                  tp/zrtp-
status_secure.wav] [System error : No such file or directory.]
2009-08-23 18:54:12.234565 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr                  tp/zrtp-
enroll_welcome.wav] [System error : No such file or directory.]
2009-08-23 18:54:13.252933 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr                  tp/zrtp-
enroll_confirmed.wav] [System error : No such file or directory.]
2009-08-23 18:54:15.293476 [WARNING] mod_sofia.c:810 We were told to use 
ptime 20 but what they meant to say was 80
This issue has so far been identified to happen on the following broken 
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who 
knows what will happen..
2009-08-23 18:54:17.333543 [ERR] mod_sndfile.c:194 Error Opening File 
[/usr/local/freeswitch/sounds/en/us/callie/zr                  tp/zrtp-
thankyou_goodbye.wav] [System error : No such file or directory.]


my /usr/local/freeswitch/sounds is bereft of any files whatsoever, searching 
the full filesystem for callie only gets me

/home/sibbleh/freeswitch-1.0.4/debian/freeswitch-sounds-en-us-callie-
16000.install
/home/sibbleh/freeswitch-1.0.4/debian/freeswitch-sounds-en-us-callie-
32000.install
/home/sibbleh/freeswitch-1.0.4/debian/freeswitch-sounds-en-us-callie-
8000.install





-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From brian at freeswitch.org  Sun Aug 23 19:30:11 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 21:30:11 -0500
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090824020523.A44439CF@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090824002011.04DDE9CF@sinclaire.sibble.net>,
	<C9F0463C-F22E-4339-879C-FF5B35C3FE43@freeswitch.org>
	<20090824020523.A44439CF@sinclaire.sibble.net>
Message-ID: <CD6A6C1A-9CEC-4663-8C6C-C7A5237BBC30@freeswitch.org>

If you reinstall the latest sound files you'll have them.

/b

On Aug 23, 2009, at 9:05 PM, Harondel J. Sibble wrote:

>>     </extension>
>
> Okay, where do I get the audio files?




From help at pdscc.com  Sun Aug 23 19:38:51 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 19:38:51 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <CD6A6C1A-9CEC-4663-8C6C-C7A5237BBC30@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090824020523.A44439CF@sinclaire.sibble.net>,
	<CD6A6C1A-9CEC-4663-8C6C-C7A5237BBC30@freeswitch.org>
Message-ID: <20090824023850.05CEE79C@sinclaire.sibble.net>

Ahh, I didn't quite clue-in that I had to run the additional installers as 
below when the main compile finished, I thought it was saying it had already 
done that.

Makes perfect sense in hindsite ;-)

    +-------- FreeSWITCH install Complete ----------+
     + FreeSWITCH has been successfully installed.   +
     +                                               +
     +       Install sounds:                         +
     +       (uhd-sounds includes hd-sounds, sounds) +
     +       (hd-sounds includes sounds)             +
     +       ------------------------------------    +
     +               make cd-sounds-install          +
     +               make cd-moh-install             +
     +                                               +
     +               make uhd-sounds-install         +
     +               make uhd-moh-install            +
     +                                               +
     +               make hd-sounds-install          +
     +               make hd-moh-install             +
     +                                               +
     +               make sounds-install             +
     +               make moh-install                +
     +                                               +
     +       Install non english sounds:             +
     +       replace XX with language                +
     +       (ru : Russian)                          +
     +       ------------------------------------    +
     +               make cd-sounds-XX-install       +
     +               make uhd-sounds-XX-install      +
     +               make hd-sounds-XX-install       +
     +               make sounds-XX-install          +


On 23 Aug 2009 at 21:30, Brian West wrote:

> If you reinstall the latest sound files you'll have them.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From help at pdscc.com  Sun Aug 23 20:22:02 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 20:22:02 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <CD6A6C1A-9CEC-4663-8C6C-C7A5237BBC30@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<20090824020523.A44439CF@sinclaire.sibble.net>,
	<CD6A6C1A-9CEC-4663-8C6C-C7A5237BBC30@freeswitch.org>
Message-ID: <20090824032200.B213D79C@sinclaire.sibble.net>

Whoah.....

2009-08-23 20:07:52.583524 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1001 at 192.168.73.45 [4ff9d452-905b-11de-8c5d-d333d780ffc7]
2009-08-23 20:07:52.740094 [INFO] mod_dialplan_xml.c:315 Processing 1001-
>9787 in context default
2009-08-23 20:07:52.980164 [NOTICE] mod_dptools.c:649 Channel 
[sofia/internal/1001 at 192.168.73.45] has been answered
2009-08-23 20:07:54.333362 [WARNING] mod_sofia.c:810 We were told to use 
ptime 20 but what they meant to say was 80
This issue has so far been identified to happen on the following broken 
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who 
knows what will happen..
2009-08-23 20:08:16.912027 [NOTICE] sofia.c:322 Hangup 
sofia/internal/1001 at 192.168.73.45 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 20:08:22.54838 [NOTICE] switch_core_session.c:1086 Session 12 
(sofia/internal/1001 at 192.168.73.45) Ended
2009-08-23 20:08:22.54838 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/1001 at 192.168.73.45 [CS_DESTROY]

I get audio now, but it's running really slooooooooooooooowly. I'd say about 
1/4 to 1/8 normal speech speed

I did the following

 make cd-sounds-install;make uhd-sounds-install;make uhd-moh-install

Probably relevant, box this is running on is a Celeron 1ghz with 512mb ram.

Weird, it starts off okay, but when this pops up

2009-08-23 20:21:10.93004 [WARNING] mod_sofia.c:810 We were told to use ptime 
20 but what they meant to say was 80
This issue has so far been identified to happen on the following broken 
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who 
knows what will happen..

the audio starts to crawl...
On 23 Aug 2009 at 21:30, Brian West wrote:

> If you reinstall the latest sound files you'll have them.
> 
> /b
> 
> On Aug 23, 2009, at 9:05 PM, Harondel J. Sibble wrote:
> 
> >>     </extension>
> >
> > Okay, where do I get the audio files?
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org


-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From scott.torr.fs at letterboxes.org  Sun Aug 23 20:26:17 2009
From: scott.torr.fs at letterboxes.org (Scott Torr)
Date: Mon, 24 Aug 2009 13:26:17 +1000
Subject: [Freeswitch-users] Screaming monkeys on ext 5000
Message-ID: <1251084377.461.1331260223@webmail.messagingengine.com>

Just a quick note, and I'm sure why, but screaming monkeys does not play
on the the default installation.

I have not looked into why, but thought I would just quickly let you
know.

Perhaps I have not done something?

regards,
sbt 



From brian at freeswitch.org  Sun Aug 23 20:31:14 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 22:31:14 -0500
Subject: [Freeswitch-users] Screaming monkeys on ext 5000
In-Reply-To: <1251084377.461.1331260223@webmail.messagingengine.com>
References: <1251084377.461.1331260223@webmail.messagingengine.com>
Message-ID: <A7FF6EFA-5CBD-4419-BC0A-656B43B34686@freeswitch.org>


On Aug 23, 2009, at 10:26 PM, Scott Torr wrote:

> Just a quick note, and I'm sure why, but screaming monkeys does not  
> play
> on the the default installation.

It requires internet connectivity.  It calls a remote system to play  
which is out of our control.

> I have not looked into why, but thought I would just quickly let you
> know.

Thanks,

> Perhaps I have not done something?
>
> regards,
> sbt

/b




From jim at evolutiontel.net  Sun Aug 23 21:14:15 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 24 Aug 2009 14:14:15 +1000
Subject: [Freeswitch-users] Media Not Heard When Bridging 2 Calls with
	same Gateway
In-Reply-To: <4256bf830908212341k1c0a9c9x7c8a9c70460965f7@mail.gmail.com>
References: <4256bf830908200029q6d5aabecuf83854c8db28d131@mail.gmail.com>
	<6E34B15B-31E8-436A-A6BB-0D7157A181F1@avgs.ca>
	<4256bf830908201021s7da0bff0wed8e060dbac0db2c@mail.gmail.com>
	<4256bf830908212341k1c0a9c9x7c8a9c70460965f7@mail.gmail.com>
Message-ID: <ddad80610908232114y159e70cew499138d0261f35b5@mail.gmail.com>

If I understand your issue correctly, it sounds to me like FS is not
set to anchor the RTP media stream.  Experience suggests that most
SBC's do not like trying to loopback RTP traffic to themselves.

Check to see what IP address's are getting used in the
c=xxx.xxx.xxx.xxx for the INVITE and 200OK messages.  If you see the
SBC IP address in messages on both sides of the FS box this will be
your issue.

You could try setting bypass_media=false in your dialplan.



On Sat, Aug 22, 2009 at 4:41 PM, Matthew Fong<mattdfong at gmail.com> wrote:
> So there seems to be some sort of error when bridging directly like
> originate
> {ignore_early_media=true}sofia/gateway/XXXX.com/91415992XXXX?&bridge(sofia/gateway/XXXX.com/91415465XXXX)
> BUT
> if I get FS to send media to leg A, and then bridge to leg B by using a lua
> script like
> session:streamFile("/usr/local/freeswitch/sounds/en/us/callie/hh/hh-welcome.wav");
> session:execute("bridge", "sofia/gateway/epik.com/91415XXXXXXX");
> then the legs bridge together OK. This happens when trying to bridge two
> channels via the same Broadsoft SBC. Does this sound like a bug that should
> be submitted to JIRA?
> --matt
> http://www.hellohunter.com
>
> On Thu, Aug 20, 2009 at 10:21 AM, Matthew Fong <mattdfong at gmail.com> wrote:
>>
>> originate
>> {ignore_early_media=true}sofia/gateway/epik.com/914159927717?&bridge(sofia/gateway/epik.com/914154650027)
>>
>> is the string I was using from the console.
>>
>> On Thu, Aug 20, 2009 at 6:00 AM, Mathieu Rene <mrene_lists at avgs.ca> wrote:
>>>
>>> Hi
>>> How are you bridging the calls in FS (which api call or C function are
>>> you using)?
>>> Mathieu Rene
>>> Avant-Garde Solutions Inc
>>> Office: + 1 (514) 664-1044 x100
>>> Cell: +1 (514) 664-1044 x200
>>> mrene at avgs.ca
>>>
>>>
>>>
>>> On 20-Aug-09, at 3:29 AM, Matthew Fong wrote:
>>>
>>> I'm trying to get FreeSWITCH to bridge two channels together through the
>>> same external gateway, but I'm having issues hearing audio. Both legs if
>>> setup independently and forwarded to 5000 (test ivr) work fine for both
>>> inbound and outbound media, but when I try to bridge them together,
>>> everything seems fine in FreeSWITCH, but neither party can hear the other
>>> speak. I'm thinking the external gateway might be having some issues because
>>> I've been able to bridge 2 channels together through the same gateway on
>>> different providers, but thought I'd also try to seek some help here.
>>> FreeSWITCH should be handling the media for both channels, so I can't figure
>>> out why if Leg A and Leg B work independently, but not if they are bridged
>>> together. Is there a setting somewhere in FS that I'm missing?
>>> Below is a ngrep of the SIP interactions if it's useful. Thanks for the
>>> help.
>>> --matt
>>>
>>> interface: eth0 (172.24.200.0/255.255.255.0)
>>> filter: (ip or ip6) and ( port 5060 )
>>> U 2009/08/20 07:11:34.038686 216.81.56.198:5080 -> 38.98.58.148:5060
>>> INVITE sip:914159927717 at 38.98.58.148 SIP/2.0.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>>> Max-Forwards: 70.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> To: <sip:914159927717 at 38.98.58.148>.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257811 INVITE.
>>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>>> Supported: timer, precondition, path, replaces.
>>> Allow-Events: talk, refer.
>>> Content-Type: application/sdp.
>>> Content-Disposition: session.
>>> Content-Length: 293.
>>> Remote-Party-ID: "FreeSWITCH"
>>> <sip:0000000000 at 216.81.56.198>;party=calling;screen=yes;privacy=off.
>>> .
>>> v=0.
>>> o=FreeSWITCH 1250727594 1250727595 IN IP4 216.81.56.198.
>>> s=FreeSWITCH.
>>> c=IN IP4 216.81.56.198.
>>> t=0 0.
>>> m=audio 24700 RTP/AVP 0 8 3 101 13.
>>> a=rtpmap:0 PCMU/8000.
>>> a=rtpmap:8 PCMA/8000.
>>> a=rtpmap:3 GSM/8000.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-16.
>>> a=rtpmap:13 CN/8000.
>>> a=ptime:20.
>>>
>>> U 2009/08/20 07:11:34.128331 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 100 Trying.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257811 INVITE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:34.338105 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 183 Session Progress.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257811 INVITE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>>> Allow:
>>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>> Content-Type: application/sdp.
>>> Content-Length: 227.
>>> .
>>> v=0.
>>> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
>>> s=M6 Call.
>>> c=IN IP4 38.98.58.148.
>>> t=0 0.
>>> m=audio 42554 RTP/AVP 0 101.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-15.
>>> a=ptime:20.
>>> a=sendrecv.
>>> a=rtcp:6461 IN IP4 10.10.24.50.
>>>
>>> U 2009/08/20 07:11:42.239312 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 200 OK.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257811 INVITE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK2gvHUU5aXjXZg.
>>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>>> Session-Expires: 1800;refresher=uas.
>>> Allow:
>>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>> Supported: timer.
>>> Content-Type: application/sdp.
>>> Content-Length: 227.
>>> .
>>> v=0.
>>> o=BroadSoft 23178 23178 IN IP4 10.10.10.11.
>>> s=M6 Call.
>>> c=IN IP4 38.98.58.148.
>>> t=0 0.
>>> m=audio 42554 RTP/AVP 0 101.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-15.
>>> a=ptime:20.
>>> a=sendrecv.
>>> a=rtcp:6461 IN IP4 10.10.24.50.
>>>
>>> U 2009/08/20 07:11:42.240828 216.81.56.198:5080 -> 38.98.58.148:5060
>>> ACK sip:914159927717 at 38.98.58.148:5060 SIP/2.0.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK3SNaXppetUKjc.
>>> Max-Forwards: 70.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> To: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257811 ACK.
>>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:42.245678 216.81.56.198:5080 -> 38.98.58.148:5060
>>> INVITE sip:914154650027 at 38.98.58.148 SIP/2.0.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>>> Max-Forwards: 70.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257815 INVITE.
>>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>>> Supported: timer, precondition, path, replaces.
>>> Allow-Events: talk, refer.
>>> Content-Type: application/sdp.
>>> Content-Disposition: session.
>>> Content-Length: 293.
>>> Remote-Party-ID: "FreeSWITCH"
>>> <sip:0000000000 at 216.81.56.198>;party=calling;screen=yes;privacy=off.
>>> .
>>> v=0.
>>> o=FreeSWITCH 1250727504 1250727505 IN IP4 216.81.56.198.
>>> s=FreeSWITCH.
>>> c=IN IP4 216.81.56.198.
>>> t=0 0.
>>> m=audio 24798 RTP/AVP 0 8 3 101 13.
>>> a=rtpmap:0 PCMU/8000.
>>> a=rtpmap:8 PCMA/8000.
>>> a=rtpmap:3 GSM/8000.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-16.
>>> a=rtpmap:13 CN/8000.
>>> a=ptime:20.
>>>
>>> U 2009/08/20 07:11:42.333184 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 100 Trying.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257815 INVITE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:42.514501 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 183 Session Progress.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257815 INVITE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>>> Allow:
>>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>> Content-Type: application/sdp.
>>> Content-Length: 225.
>>> .
>>> v=0.
>>> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
>>> s=M6 Call.
>>> c=IN IP4 38.98.58.148.
>>> t=0 0.
>>> m=audio 46520 RTP/AVP 0 101.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-15.
>>> a=ptime:20.
>>> a=sendrecv.
>>> a=rtcp:6451 IN IP4 10.10.24.50.
>>>
>>> U 2009/08/20 07:11:46.190607 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 200 OK.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257815 INVITE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK42e3yH7HQ494Q.
>>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>>> Session-Expires: 1800;refresher=uas.
>>> Allow:
>>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>> Supported: timer.
>>> Content-Type: application/sdp.
>>> Content-Length: 225.
>>> .
>>> v=0.
>>> o=BroadSoft 2035 2035 IN IP4 10.10.10.11.
>>> s=M6 Call.
>>> c=IN IP4 38.98.58.148.
>>> t=0 0.
>>> m=audio 46520 RTP/AVP 0 101.
>>> a=rtpmap:101 telephone-event/8000.
>>> a=fmtp:101 0-15.
>>> a=ptime:20.
>>> a=sendrecv.
>>> a=rtcp:6451 IN IP4 10.10.24.50.
>>>
>>> U 2009/08/20 07:11:46.191161 216.81.56.198:5080 -> 38.98.58.148:5060
>>> ACK sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK5B8U0crNmD0QK.
>>> Max-Forwards: 70.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257815 ACK.
>>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:55.139274 38.98.58.148:5060 -> 216.81.56.198:5080
>>> BYE sip:gw+epik.com at 216.81.56.198:5080 SIP/2.0.
>>> From: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
>>> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 4817 BYE.
>>> Max-Forwards: 70.
>>> Via: SIP/2.0/UDP
>>> 38.98.58.148:5060;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
>>> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
>>> Contact: <sip:914159927717 at 38.98.58.148:5060>.
>>> Allow:
>>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:55.140390 216.81.56.198:5080 -> 38.98.58.148:5060
>>> SIP/2.0 200 OK.
>>> Via: SIP/2.0/UDP
>>> 38.98.58.148:5060;branch=z9hG4bK4A8C.F725.2C494819.943A6226.00000BC3.
>>> Via: SIP/2.0/UDP 10.10.10.11:5060;branch=z9hG4bK4A8C.F725.2C494819.
>>> From: <sip:914159927717 at 38.98.58.148>;tag=F725.2C49.
>>> To: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=ZtFvjeFQmXvpp.
>>> Call-ID: 88c12fa5-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 4817 BYE.
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>>> Supported: timer, precondition, path, replaces.
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:55.145438 216.81.56.198:5080 -> 38.98.58.148:5060
>>> BYE sip:914154650027 at 38.98.58.148:5060 SIP/2.0.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
>>> Max-Forwards: 70.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257816 BYE.
>>> Contact: <sip:gw+epik.com at 216.81.56.198:5080;transport=udp>.
>>> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-14417.
>>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>>> NOTIFY, REFER, UPDATE, REGISTER, INFO.
>>> Supported: timer, precondition, path, replaces.
>>> Reason: Q.850;cause=16;text="NORMAL_CLEARING".
>>> Content-Length: 0.
>>> .
>>>
>>> U 2009/08/20 07:11:55.232064 38.98.58.148:5060 -> 216.81.56.198:5080
>>> SIP/2.0 200 OK.
>>> From: "FreeSWITCH" <sip:0000000000 at 216.81.56.198>;tag=038mm9ZtH6j9H.
>>> To: <sip:914154650027 at 38.98.58.148>;tag=F72E.2D4E.
>>> Call-ID: 8da57c9a-07fb-122d-cc93-00e0813368fe.
>>> CSeq: 119257816 BYE.
>>> Via: SIP/2.0/UDP 216.81.56.198:5080;rport;branch=z9hG4bK6m1m278rHppaF.
>>> Contact: <sip:914154650027 at 38.98.58.148:5060>.
>>> Allow:
>>> INVITE,BYE,INFO,PRACK,CANCEL,ACK,OPTIONS,SUBSCRIBE,NOTIFY,REGISTER,REFER,UPDATE.
>>> Content-Length: 0.
>>> .
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net



From jim at evolutiontel.net  Sun Aug 23 21:19:55 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 24 Aug 2009 14:19:55 +1000
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
Message-ID: <ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>

In your SIP profiles this could be set.  I beleive 120 is the default setting.

param name="session-timeout" value="120"

On Fri, Aug 21, 2009 at 11:35 PM, bakko<asannucci at gmail.com> wrote:
> Do you have those lines in switch.conf file?
>
> ? ?<!--RTP port range -->
> ? ?<param name="rtp-start-port" value="10000"/>
> ? ?<param name="rtp-end-port" value="32767"/>
>
>
> BR
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net



From jmesquita at gmail.com  Sun Aug 23 21:27:16 2009
From: jmesquita at gmail.com (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 24 Aug 2009 01:27:16 -0300
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
References: <4A91D07C.1080805@venturevoip.com>
	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
Message-ID: <5a8712120908232127n6cb2f0bdq7e05b3d7778f0e70@mail.gmail.com>

Hey there, FsGui uses ESL a lot and I had to go through the code to document
it so here is a few hints inline ...

Don't hesitate to keep the questions coming. I will fill in whenever I can.

jmesquita

On Sun, Aug 23, 2009 at 8:50 PM, Brian West <brian at freeswitch.org> wrote:

>
> On Aug 23, 2009, at 6:27 PM, Matt Riddell wrote:
>
> > Hi,
> >
> > I don't see how I can read some responses to command using esl.
> >
> > I.E. esl_send_recv(&handle, "api show calls count\n\n");
> >
> > and
> >
> > printf("Header Test %s\n", esl_event_get_header(event, "API-
> > Command"));
> > printf("Body Test %s\n", esl_event_get_body(event));
> >
> > the header details are returned.
> >
> > The body is null.
>

Body is null on every event that does not use headers to output information.
A good example would be console logs. I haven't seen too many default events
besides log that have body besides a application custom events.


>
>
> I'm not too sure about using ESL in C, I have used it pretty much
> exclusively in perl.
>
> > Also, I can originate a call and set the account code for it, but
> > how do
> > I get a list of calls with their account codes?
>
> originate {account_code=1234}sofia/profile/target at ip ....
>
> You can get the list of the channels via "show channels" or bridged
> calls with "show calls"
>
>  From there you have the UUID's you can call uuid_dump on them to get
> all the variables.


>
> > Do I get a list of calls then go through them one by one and get the
> > variables for those calls by uuid?
>

All ESL does is output events to socket and expose the API commands. It does
not maintain any kind of list of calls or anything like that so it is up to
you to maintain that yourself if you don't want to parse API output every
time.


>
>
> You could do this or setup a listener to get the events as they happen
> and keep the info you need.
>
> > Does anyone have any documentation for the esl api?
>
> http://docs.freeswitch.org/ (this should help, its under files list
> see esl.h)
>

I need to work a little bit more on that documentation as well.. I saw a few
conflicts with the core documentation too. Will get there once I have some
more time left.


>
> > Even if I could read some comments from a usage of it would be useful.
>
> I just find it interesting you're doing this with C.
>
> > --
> > Cheers,
> >
> > Matt Riddell
> > Director
> > _______________________________________________
> >
> > http://www.venturevoip.com/news.php (Daily Asterisk News)
> > http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> > http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jim at evolutiontel.net  Sun Aug 23 21:32:26 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 24 Aug 2009 14:32:26 +1000
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<2B75328D-8763-4C89-A15E-ED5168BE5926@freeswitch.org>
	<10128ef10908130704md1bab3bj696e3f36fbe020d8@mail.gmail.com>
	<43E9016A-1B93-437F-8883-D6F6CCDF276E@freeswitch.org>
	<10128ef10908130737q2541ec87u102bd0a38f753d60@mail.gmail.com>
	<45042DEA-A9F6-4C61-960B-A2DC9529C4A8@freeswitch.org>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
Message-ID: <ddad80610908232132v3442c353s6cc09362cf71952c@mail.gmail.com>

X-Lite does support TCP, however you need to have NAPTR and SRV DNS
entries.  It used to support TLS, but this seems to have been removed
:(

sip.mydomain.net. IN NAPTR 0 0 "s" "SIPS+D2T"  "" _sips._tcp.sip.mydomain.net.
sip.mydomain.net. IN NAPTR 1 0 "s" "SIP+D2T"  "" _sip._tcp.sip.mydomain.net.
sip.mydomain.net. IN NAPTR 2 0 "s" "SIP+D2U"  "" _sip._udp.sip.mydomain.net.


_sip._udp.sip.mydomain.net.   43200 IN SRV 1 10 5060  sip.mydomain.net.
_sip._tcp.sip.mydomain.net.   43200 IN SRV 1 10 5060  sip.mydomain.net.
_sips._tcp.sip.mydomain.net.   43200 IN SRV 1 10 5061  sip.mydomain.net.


On Mon, Aug 17, 2009 at 10:18 PM, bakko<asannucci at gmail.com> wrote:
> Y make my tests with eyebeam.
>
> I thing X-lite dont't support TCP transport.
>
> BR
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net



From rogelio.perez at gmail.com  Sun Aug 23 21:45:06 2009
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Mon, 24 Aug 2009 01:45:06 -0300
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <191c3a030908211320s6feb5f10sf8670575069ec95a@mail.gmail.com>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
	<191c3a030908211320s6feb5f10sf8670575069ec95a@mail.gmail.com>
Message-ID: <9D12BA51-0C76-45B1-A8DF-3A1FB74E005B@gmail.com>

Thanks Andrew and Anthony,

I created a ramdisk for the db and log directories using tmpfs and now  
I see better performance times:

startup:		15.6 sec.
call extension: 	0 sec.
shutdown:	7.5 sec
reload config:	0 sec.

I have noticed that during the startup there is a 12 sec. pause while  
checking for UPnP:

2009-08-24 04:39:29.910459 [ERR] switch_nat.c:183 Error checking for  
PMP [general error]
2009-08-24 04:39:29.910694 [DEBUG] switch_nat.c:397 Checking for UPnP
2009-08-24 04:39:41.906029 [INFO] switch_nat.c:411 No PMP or UPnP NAT  
detected!

Is there any way to lower this time? Maybe disabling the check?

Thanks,
Rogelio

On Aug 21, 2009, at 5:20 PM, Anthony Minessale wrote:

> probably disk i/o.
>
> Is it some kind of flash drive?
>
> make a ramdisk and simlink in /usr/local/freeswitch/db and /usr/ 
> local/freeswitch/log to it
> the default configuration uses a lot of high level features that use  
> the sqlite db on the disk.
>
> We also offer commercial support where we could dig deeper into the  
> problem if you can't figure it out
> consulting at freeswitch.org
>
>
>
> On Fri, Aug 21, 2009 at 2:15 PM, Rogelio Perez <rogelio.perez at gmail.com 
> > wrote:
> Hi Everyone,
>
> I'm working on a PBX project for the Sheevaplug ARM based computer,  
> with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
> So far I've found a big difference between Freeswitch and Asterisk  
> performance times.
> This is a comparison of the time it takes them to perform different  
> actions:
>
> startup Freeswitch:			3 min.
> startup Asterisk:			2 sec.
>
> call extension Freeswitch:	6 sec.
> call extension Asterisk:		0 sec.
>
> shutdown Freeswitch:		6.5 sec
> shutdown Asterisk:			0 sec.
>
> reload config Freeswitch: 	1 sec.
> reload config Asterisk:		1 sec.
>
> Both were built from sources natively (no cross-compiling), and they  
> use the default startup configurations.
> I have managed to lower the Freeswitch times by disabling most of  
> the modules and recompiling, but it is still far away from Asterisk  
> (i.e. FS startup time 2.5 min).
>
> 1. Is there any way to further improve Freeswitch performance for  
> the ARM architecture?
> 2. Can this be related to the lack of a FPU (the Sheevalug emulates  
> the floating point operations).
> 3. On the startup I see this error repeated many times: [ERR]  
> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be  
> related?
>
> Thanks,
> Rogelio Perez
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From jim at evolutiontel.net  Sun Aug 23 21:49:01 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 24 Aug 2009 14:49:01 +1000
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <9B65BFD4-C93F-4E26-82F1-AF8D687DCF52@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
	<CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>
	<10128ef10908170706p414098faq5d9d8024ba0cb3ef@mail.gmail.com>
	<9B65BFD4-C93F-4E26-82F1-AF8D687DCF52@freeswitch.org>
Message-ID: <ddad80610908232149k39ac88d6qee647b78f1d19078@mail.gmail.com>

"Well if you append ;transport=tcp on the bridge lines it will use TCP"

IMHO this statement needs some clarification based on the context of
this thread.

If the destination is another PBX or Freeswitch box then this is ok as
FS will be initiating the TCP connection.

For terminating calls to registered User Agents (UA) the decision to
use TCP or UDP should be made using information collected when the UA
registered.  i.e if the UA registers using TCP, FS should use TCP to
send and receive messages, if the UA registers with TLS, then FS
should use TLS, same story for UDP.

On Tue, Aug 18, 2009 at 12:10 AM, Brian West<brian at freeswitch.org> wrote:
> Well if you append ;transport=tcp on the bridge lines it will use TCP .
>
> /b
>
> On Aug 17, 2009, at 9:06 AM, Tzury Bar Yochay wrote:
>
>>> FreeSWITCH works very well as a client :P
>> I am currently porting it into iPhone and Symbian, I am almost
>> done ;-)
>>
>> anyway, seriously now, can one point to a wiki page about this?
>> How do I do that?
>> I would need 3 server instances to place a call, right?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net



From brian at freeswitch.org  Sun Aug 23 21:52:51 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 23:52:51 -0500
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <9D12BA51-0C76-45B1-A8DF-3A1FB74E005B@gmail.com>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
	<191c3a030908211320s6feb5f10sf8670575069ec95a@mail.gmail.com>
	<9D12BA51-0C76-45B1-A8DF-3A1FB74E005B@gmail.com>
Message-ID: <2439A4F0-2969-4C9F-ACBC-B56788B15410@freeswitch.org>

freeswitch -nonat

/b

On Aug 23, 2009, at 11:45 PM, Rogelio Perez wrote:

> 2009-08-24 04:39:29.910459 [ERR] switch_nat.c:183 Error checking for  
> PMP [general error]
> 2009-08-24 04:39:29.910694 [DEBUG] switch_nat.c:397 Checking for UPnP
> 2009-08-24 04:39:41.906029 [INFO] switch_nat.c:411 No PMP or UPnP  
> NAT detected!




From brian at freeswitch.org  Sun Aug 23 21:53:39 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 23 Aug 2009 23:53:39 -0500
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <ddad80610908232149k39ac88d6qee647b78f1d19078@mail.gmail.com>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908160425q14f81721pd27123ec0fd50342@mail.gmail.com>
	<01C3FB81-804F-4E0B-85D6-896D8A7FD31C@freeswitch.org>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
	<CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>
	<10128ef10908170706p414098faq5d9d8024ba0cb3ef@mail.gmail.com>
	<9B65BFD4-C93F-4E26-82F1-AF8D687DCF52@freeswitch.org>
	<ddad80610908232149k39ac88d6qee647b78f1d19078@mail.gmail.com>
Message-ID: <8A373389-2CE4-4606-8073-1D763BA2C644@freeswitch.org>

It already does exactly this.

/b

On Aug 23, 2009, at 11:49 PM, Jim Burke wrote:

> For terminating calls to registered User Agents (UA) the decision to
> use TCP or UDP should be made using information collected when the UA
> registered.  i.e if the UA registers using TCP, FS should use TCP to
> send and receive messages, if the UA registers with TLS, then FS
> should use TLS, same story for UDP.




From rogelio.perez at gmail.com  Sun Aug 23 22:03:40 2009
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Mon, 24 Aug 2009 02:03:40 -0300
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <2439A4F0-2969-4C9F-ACBC-B56788B15410@freeswitch.org>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
	<191c3a030908211320s6feb5f10sf8670575069ec95a@mail.gmail.com>
	<9D12BA51-0C76-45B1-A8DF-3A1FB74E005B@gmail.com>
	<2439A4F0-2969-4C9F-ACBC-B56788B15410@freeswitch.org>
Message-ID: <CC428124-7379-4C78-8122-88ED5573C464@gmail.com>

Thanks Brian, now the startup time is 3 sec.

On Aug 24, 2009, at 1:52 AM, Brian West wrote:

> freeswitch -nonat
>
> /b
>
> On Aug 23, 2009, at 11:45 PM, Rogelio Perez wrote:
>
>> 2009-08-24 04:39:29.910459 [ERR] switch_nat.c:183 Error checking for
>> PMP [general error]
>> 2009-08-24 04:39:29.910694 [DEBUG] switch_nat.c:397 Checking for UPnP
>> 2009-08-24 04:39:41.906029 [INFO] switch_nat.c:411 No PMP or UPnP
>> NAT detected!
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From jim at evolutiontel.net  Sun Aug 23 22:05:42 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 24 Aug 2009 15:05:42 +1000
Subject: [Freeswitch-users] Transporting SIP over TCP
In-Reply-To: <8A373389-2CE4-4606-8073-1D763BA2C644@freeswitch.org>
References: <10128ef10908130140s5085502av7360eca8487efb3a@mail.gmail.com>
	<10128ef10908170014m34a0dc2ayf35bbd5315b4f12e@mail.gmail.com>
	<C62C40FE5A6547719969BCD0653D1A4E@voztovoice>
	<3CE97EA6-AA60-482B-998F-D26FB3775E83@freeswitch.org>
	<10128ef10908170653g1b66e7bfv597b083854d5d783@mail.gmail.com>
	<CC7A8BCE-4E12-46AF-9333-9641411A221A@freeswitch.org>
	<10128ef10908170706p414098faq5d9d8024ba0cb3ef@mail.gmail.com>
	<9B65BFD4-C93F-4E26-82F1-AF8D687DCF52@freeswitch.org>
	<ddad80610908232149k39ac88d6qee647b78f1d19078@mail.gmail.com>
	<8A373389-2CE4-4606-8073-1D763BA2C644@freeswitch.org>
Message-ID: <ddad80610908232205h3d720c1ai35a68c749cde9cfa@mail.gmail.com>

I always expected it did :)

My point was that you cannot put transport=TCP on a bridge statement
line to an internal registered client and expect it to use a protocol
that was not used at registration.

Hence the clarification based on the context of the thread :)

On Mon, Aug 24, 2009 at 2:53 PM, Brian West<brian at freeswitch.org> wrote:
> It already does exactly this.
>
> /b
>
> On Aug 23, 2009, at 11:49 PM, Jim Burke wrote:
>
>> For terminating calls to registered User Agents (UA) the decision to
>> use TCP or UDP should be made using information collected when the UA
>> registered. ?i.e if the UA registers using TCP, FS should use TCP to
>> send and receive messages, if the UA registers with TLS, then FS
>> should use TLS, same story for UDP.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net



From help at pdscc.com  Sun Aug 23 22:49:03 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Sun, 23 Aug 2009 22:49:03 -0700
Subject: [Freeswitch-users] zrtp endpoints have different sas through fs
	1.0.4
In-Reply-To: <20090824032200.B213D79C@sinclaire.sibble.net>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<CD6A6C1A-9CEC-4663-8C6C-C7A5237BBC30@freeswitch.org>,
	<20090824032200.B213D79C@sinclaire.sibble.net>
Message-ID: <20090824054901.A614079C@sinclaire.sibble.net>



On 23 Aug 2009 at 20:22, Harondel J. Sibble wrote:

> Whoah.....
> I get audio now, but it's running really slooooooooooooooowly. I'd say about
> 1/4 to 1/8 normal speech speed

Hmmm, using one of my hardphones, specifically the Integrated Networks IN-
1002

2009-08-23 20:46:07.334596 [NOTICE] switch_channel.c:602 New Channel 
sofia/internal/1004 at 10.12.13.45 [a7c10746-9060-11de-8c5d-d333d780ffc7]
2009-08-23 20:46:07.348834 [INFO] mod_dialplan_xml.c:315 Processing 1004-
>5000 in context default
2009-08-23 20:46:07.364484 [NOTICE] mod_dptools.c:649 Channel 
[sofia/internal/1004 at 10.12.13.45] has been answered
2009-08-23 20:46:08.33173 [WARNING] mod_sofia.c:810 We were told to use ptime 
20 but what they meant to say was 10
This issue has so far been identified to happen on the following broken 
platforms/devices:
Linksys/Sipura aka Cisco
ShoreTel
Sonus/L3
We will try to fix it but some of the devices on this list are so broken who 
knows what will happen..
2009-08-23 20:47:30.512970 [NOTICE] switch_core_state_machine.c:179 Hangup 
sofia/internal/1004 at 10.12.13.45 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-23 20:47:30.541157 [NOTICE] switch_core_session.c:1086 Session 20 
(sofia/internal/1004 at 10.12.13.45) Ended
2009-08-23 20:47:30.541157 [NOTICE] switch_core_session.c:1088 Close Channel 
sofia/internal/1004 at 10.12.13.45 [CS_DESTROY]

gets me much better results and no slowdown of audio, although there are some 
drop outs.  Running Ekiga on xp machine with zfone client works okay, but 
being it's an older verison of zfone, I'm guessing it doesn't support the 
enrollment packet from FS as the voice says I have to choose to enroll and 
then says goodbye ;-)

>From looking through the list archives, I am guessing I need to talk to the 
Tivi developers, question is what should I say to them, there's nothing 
really exposed in the client, other than adjusting the codecs, 3 choices
GSM, uLaw and aLaw can be enabled or disabled.

Both phones are running over a wifi link and there is (randomly) a fair bit 
of interference affecting speeds.

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From juanbackson at gmail.com  Sun Aug 23 23:00:43 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Mon, 24 Aug 2009 14:00:43 +0800
Subject: [Freeswitch-users] unable to download non-US sound files
Message-ID: <27c25bc40908232300h7ae3cbbbkcd5ce5bbddb885f2@mail.gmail.com>

Hi

I tried to download non-US sound files, but I am getting this error:

--21:50:09--  http://files.freeswitch.org/freeswitch-sounds-fr-1.0.10.tar.gz
Resolving files.freeswitch.org... 69.174.57.101
Connecting to files.freeswitch.org|69.174.57.101|:80... connected.
HTTP request sent, awaiting response... 404 Not Found
21:50:12 ERROR 404: Not Found.

The command I used is make sounds-fr-install

Any idea what's wrong?

Thanks,
JB
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From matt at venturevoip.com  Sun Aug 23 23:07:19 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Mon, 24 Aug 2009 18:07:19 +1200
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <5a8712120908232127n6cb2f0bdq7e05b3d7778f0e70@mail.gmail.com>
References: <4A91D07C.1080805@venturevoip.com>	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
	<5a8712120908232127n6cb2f0bdq7e05b3d7778f0e70@mail.gmail.com>
Message-ID: <4A922E17.2040401@venturevoip.com>

On 24/08/09 4:27 PM, Jo?o Mesquita wrote:
>
> I need to work a little bit more on that documentation as well.. I saw a
> few conflicts with the core documentation too. Will get there once I
> have some more time left.

Awesome man - thanks for these.

So, one more question - in the show calls (or show channels) api 
response, which header would I use to retrieve the response?

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)



From matt at venturevoip.com  Sun Aug 23 23:15:06 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Mon, 24 Aug 2009 18:15:06 +1200
Subject: [Freeswitch-users] unable to download non-US sound files
In-Reply-To: <27c25bc40908232300h7ae3cbbbkcd5ce5bbddb885f2@mail.gmail.com>
References: <27c25bc40908232300h7ae3cbbbkcd5ce5bbddb885f2@mail.gmail.com>
Message-ID: <4A922FEA.3020203@venturevoip.com>

On 24/08/09 6:00 PM, Juan Backson wrote:
> Hi
>
> I tried to download non-US sound files, but I am getting this error:
>
> --21:50:09-- http://files.freeswitch.org/freeswitch-sounds-fr-1.0.10.tar.gz
> Resolving files.freeswitch.org... 69.174.57.101
> Connecting to files.freeswitch.org
> <http://files.freeswitch.org>|69.174.57.101|:80... connected.
> HTTP request sent, awaiting response... 404 Not Found
> 21:50:12 ERROR 404: Not Found.
>
> The command I used is make sounds-fr-install

It appears there are only sounds for en-us and ru-RU in:

http://files.freeswitch.org/

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)



From bruce.mcalister at blueface.ie  Mon Aug 24 01:59:24 2009
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Mon, 24 Aug 2009 09:59:24 +0100
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>
	<4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>
Message-ID: <4A92566C.3030200@blueface.ie>

Hi Vladimir,

Did you get an update on this at all?

Thanks


vmorales wrote:
> Hi Michal,
> 
> Just checking in to see if you've been able to take a stab at this.
> 
> Thanks,
> Vladimir
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michal Bielicki
> Sent: Tuesday, August 11, 2009 5:33 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
> 
> I'll retst it later today and give you a link with instructions
> 
> Am 10.08.2009 um 20:14 schrieb vmorales:
> 
>> By "./compile" I was referring to "./configure"
>>
>> Vladimir
>>
>> -----Original Message-----
>> From: vmorales [mailto:email.list.subscriber at gmail.com]
>> Sent: Monday, August 10, 2009 11:49 AM
>> To: 'freeswitch-users at lists.freeswitch.org'
>> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> Thanks for the response(s):
>>
>> I ran the "./compile" script with a set PREFIX.  This took a few
>> attempts with errors before it was able to complete error-free, as I
>> had to install libtool.
>>
>> Since then, I have tried running 'make', 'gmake', and
>> '/opt/gnu/bin/make', but each results with an error.  This is the
>> error when running 'make' or 'gmake':
>>
>> <snip>
>> make: Fatal error: Command failed for target `all-recursive'
>> Current working directory /home/vmorales/freeswitch-1.0.4
>> *** Error code 1
>> make: Fatal error: Command failed for target `all'
>> </snip>
>>
>>
>> This is the error when running '/opt/gnu/bin/make':
>>
>> <snip>
>> make[5]: *** [mod_amr.so] Error 1
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_amr-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> Making all in build
>> +-------- FreeSWITCH Build Complete -----------+
>> + FreeSWITCH has been successfully built.      +
>> + Install by running:                          +
>> +                                              +
>> +               /opt/gnu/bin/make install                   +
>> +----------------------------------------------+
>> make[1]: *** [all-recursive] Error 1
>> make: *** [all] Error 2
>> </snip>
>>
>>
>> I re-untar'd before each compile/make attempt.  Let me know if this
> is
>> something that I can resolve.
>>
>> Vladimir
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Michael Jerris
>> Sent: Saturday, August 08, 2009 12:37 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> This is not currently a supported platform, it only builds on 64 bit
>> right now I think on solaris.
>>
>> Mike
>>
>> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>>
>>> Hello,
>>>
>>> Does anyone have, or know where to get, a pre-compiled copy of
>>> FreeSwitch for Solaris 10/x86?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>> http://www.freeswitch.org
> 
> Michal Bielicki
> Leiter der Niederlassung
> HaloKwadrat Sp. z o.o.
> Niederlassung Kleinmachnow
> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
> Ust.Id.: DE261885536
> Geschaeftsfuehrer: Aleksander Wiercinski
> Meiereifeld 2b, 14532 Kleinmachnow
> t. +49 33203 263220
> f. +49 33203 263229 sip. info at halokwadrat.de
> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
> Hauptgesch?ftsstelle:
> Halo Kwadrat Sp. z o.o.
> ul. Polna 46/14
> 00-644 Warszawa, Polen
> EIngetragen im HRB Warszawa, KRS 0000153539
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 



From gmaruzz at celliax.org  Mon Aug 24 03:03:17 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Mon, 24 Aug 2009 12:03:17 +0200
Subject: [Freeswitch-users] Screaming monkeys on ext 5000
In-Reply-To: <A7FF6EFA-5CBD-4419-BC0A-656B43B34686@freeswitch.org>
References: <1251084377.461.1331260223@webmail.messagingengine.com> 
	<A7FF6EFA-5CBD-4419-BC0A-656B43B34686@freeswitch.org>
Message-ID: <7b197bef0908240303r61d89baar351450e488dc51a3@mail.gmail.com>

On Mon, Aug 24, 2009 at 5:31 AM, Brian West<brian at freeswitch.org> wrote:
> It requires internet connectivity. ?It calls a remote system to play
> which is out of our control.
>

Yeah, I noted this too, since a couple weeks at least...

Maybe let's Todd know it's monkeys are out of voice?

-giovanni



From Prometheus001 at gmx.net  Mon Aug 24 04:25:16 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 24 Aug 2009 13:25:16 +0200
Subject: [Freeswitch-users] XML-RPC on different ip than 0.0.0.0
Message-ID: <4A92789C.5040900@gmx.net>

Hello,

is there any chance to limit the listening ips of the xml-rpc server
(which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)?

Best regards
Peter



From asannucci at gmail.com  Mon Aug 24 06:15:49 2009
From: asannucci at gmail.com (bakko)
Date: Mon, 24 Aug 2009 15:15:49 +0200
Subject: [Freeswitch-users] zrtp endpoints have different sas through
	fs1.0.4
In-Reply-To: <09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org>
References: <20090823213745.D81819D5@sinclaire.sibble.net>,
	<B24122B7-9B6A-4779-82AB-E48BB8363E7B@freeswitch.org>,
	<20090823231610.3398273E@sinclaire.sibble.net><20090823235030.A95BF5FE@sinclaire.sibble.net>
	<09D6850B-2F59-4C80-B1F5-ED731E6D213C@freeswitch.org>
Message-ID: <7541BAF571934E12B19428744AE5FB96@voztovoice>

Me too :)

----- Original Message ----- 
From: "Brian West" <brian at freeswitch.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Monday, August 24, 2009 1:53 AM
Subject: Re: [Freeswitch-users] zrtp endpoints have different sas through 
fs1.0.4


> Wish they would send me one for my E63 for testing... only been
> working with zfone 3 so far.
>
> /b
>
> On Aug 23, 2009, at 6:50 PM, Harondel J. Sibble wrote:
>
>> Well good news for the Tiviphone client
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org 




From m.krivushin at imarto.net  Sun Aug 23 22:57:19 2009
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Mon, 24 Aug 2009 12:57:19 +0700
Subject: [Freeswitch-users] CHANNEL_CREATE and full information
Message-ID: <5be734a50908232257s1950f82esf8d26cc6c12441be@mail.gmail.com>

I put some variable in originate command for identify created channel for my
app using event socket.

Is any ability to get all channel variables without uuid_show on
channel_create? Can I ask FS to take me all information on channel creating?
Can this be in trunk, if it cannt be get without source editing? May be we
can add custom event - CHANNEL_CREATE_FULL_INFO?

I just think, that is smarty to push channel information for controller in
channel create. I can call api uuid_call, but in some situations, why I call
api, channel destroyed. I can use bgapi result, BACKGROUND_JOB, but not in
all cases.

Sometimes we have situation when call initiated not from my caller app. In
this case we *cant* get information about destroyed channel.

Please, point me to right side - may be I need to write something module?
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From jmesquita at freeswitch.org  Mon Aug 24 00:10:33 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 24 Aug 2009 04:10:33 -0300
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <4A922E17.2040401@venturevoip.com>
References: <4A91D07C.1080805@venturevoip.com>
	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
	<5a8712120908232127n6cb2f0bdq7e05b3d7778f0e70@mail.gmail.com>
	<4A922E17.2040401@venturevoip.com>
Message-ID: <ebbc2d550908240010i17ac47b8r44a2465cab2f673@mail.gmail.com>

For api responses you have to use body. I should have reminded that as
well, sorry...

Jmesquita

On 8/24/09, Matt Riddell <matt at venturevoip.com> wrote:
> On 24/08/09 4:27 PM, Jo?o Mesquita wrote:
>>
>> I need to work a little bit more on that documentation as well.. I saw a
>> few conflicts with the core documentation too. Will get there once I
>> have some more time left.
>
> Awesome man - thanks for these.
>
> So, one more question - in the show calls (or show channels) api
> response, which header would I use to retrieve the response?
>
> --
> Cheers,
>
> Matt Riddell
> Director
> _______________________________________________
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device



From brian at freeswitch.org  Mon Aug 24 07:05:26 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 09:05:26 -0500
Subject: [Freeswitch-users] unable to download non-US sound files
In-Reply-To: <27c25bc40908232300h7ae3cbbbkcd5ce5bbddb885f2@mail.gmail.com>
References: <27c25bc40908232300h7ae3cbbbkcd5ce5bbddb885f2@mail.gmail.com>
Message-ID: <C426BDB1-7BB8-4FAE-BD9F-661D2F8A8849@freeswitch.org>

No french files exist yet. ;)

/b

On Aug 24, 2009, at 1:00 AM, Juan Backson wrote:

> Any idea what's wrong?




From anthony.minessale at gmail.com  Mon Aug 24 07:05:31 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 09:05:31 -0500
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <4A922E17.2040401@venturevoip.com>
References: <4A91D07C.1080805@venturevoip.com>
	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>
	<5a8712120908232127n6cb2f0bdq7e05b3d7778f0e70@mail.gmail.com>
	<4A922E17.2040401@venturevoip.com>
Message-ID: <191c3a030908240705m5003a26bpd97b2c041d7b1526@mail.gmail.com>

I updated testclient.c so you can see how now.

#include <stdio.h>
#include <stdlib.h>
#include <esl.h>


int main(void)
{
    esl_handle_t handle = {{0}};

    esl_connect(&handle, "localhost", 8021, "ClueCon");

    esl_send_recv(&handle, "api status\n\n");

    if (handle.last_sr_event && handle.last_sr_event->body) {
        printf("%s\n", handle.last_sr_event->body);
    } else {
        // this is unlikely to happen with api or bgapi (which is hardcoded
above) but prefix but may be true for other commands
        printf("%s\n", handle.last_sr_reply);
    }

    esl_disconnect(&handle);

    return 0;
}


On Mon, Aug 24, 2009 at 1:07 AM, Matt Riddell <matt at venturevoip.com> wrote:

> On 24/08/09 4:27 PM, Jo?o Mesquita wrote:
> >
> > I need to work a little bit more on that documentation as well.. I saw a
> > few conflicts with the core documentation too. Will get there once I
> > have some more time left.
>
> Awesome man - thanks for these.
>
> So, one more question - in the show calls (or show channels) api
> response, which header would I use to retrieve the response?
>
> --
> Cheers,
>
> Matt Riddell
> Director
> _______________________________________________
>
> http://www.venturevoip.com/news.php (Daily Asterisk News)
> http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
> http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Aug 24 07:07:10 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 09:07:10 -0500
Subject: [Freeswitch-users] CHANNEL_CREATE and full information
In-Reply-To: <5be734a50908232257s1950f82esf8d26cc6c12441be@mail.gmail.com>
References: <5be734a50908232257s1950f82esf8d26cc6c12441be@mail.gmail.com>
Message-ID: <191c3a030908240707o2a8a534axf1ac789dd65d25a2@mail.gmail.com>

try looking for the channel_originate events which is fired as soon as any
outbound call is mature enough to have variables.


On Mon, Aug 24, 2009 at 12:57 AM, Mikhail Krivushin
<m.krivushin at imarto.net>wrote:

> I put some variable in originate command for identify created channel for
> my app using event socket.
>
> Is any ability to get all channel variables without uuid_show on
> channel_create? Can I ask FS to take me all information on channel creating?
> Can this be in trunk, if it cannt be get without source editing? May be we
> can add custom event - CHANNEL_CREATE_FULL_INFO?
>
> I just think, that is smarty to push channel information for controller in
> channel create. I can call api uuid_call, but in some situations, why I call
> api, channel destroyed. I can use bgapi result, BACKGROUND_JOB, but not in
> all cases.
>
> Sometimes we have situation when call initiated not from my caller app. In
> this case we *cant* get information about destroyed channel.
>
> Please, point me to right side - may be I need to write something module?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From juanbackson at gmail.com  Mon Aug 24 07:25:01 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Mon, 24 Aug 2009 22:25:01 +0800
Subject: [Freeswitch-users] fifo question
Message-ID: <27c25bc40908240725k5e0c92a2u2def5bd679b7b666@mail.gmail.com>

Hi,

Does anyone know the purpose of fifo_orbit_announce?

When does fifo_orbit_announce get played?

Thanks,
JB
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From houndd at ymail.com  Mon Aug 24 07:45:29 2009
From: houndd at ymail.com (Hound Dog)
Date: Mon, 24 Aug 2009 07:45:29 -0700 (PDT)
Subject: [Freeswitch-users] topology hiding leaking information in SDP data
Message-ID: <510213.5618.qm@web44709.mail.sp1.yahoo.com>

carriers need topology hiding , its an important feature for both security and also to hide you business partners from each other 

freeSwitch talks about it and also does a good job in hiding the signalling topology 

there is however a hole in the SDP manipulation that I am trying to plug and would love to get some help , obviously once resolved I am also happy to add documentation for all to use 


I was thinking that the best way would be to build the SDP message from scratch based on the incoming info , and maybe in special cases have the SDP copied over from the original message. 

is there a way to have FS build a clean SDP message ?





see and example of Bria softphone making a call via freeswitch , note that the SDP to leg B contains original addresses and even internal ones 


incoming Invite message 
-------------------
INVITE sip:442078562101 at pbx.rilcomm.com SIP/2.0
Via: SIP/2.0/UDP 82.80.130.222:26762;branch=z9hG4bK-d8754z-8fba35a96b6d949f-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:44558678567378 at 82.80.130.222:26762>
To: <sip:442078562101 at pbx.rilcomm.com>
From: <sip:44558678567378 at pbx.rilcomm.com>;tag=47360c64
Call-ID: YTY4NjMwMjg4MWRmODY5NDlhOWQ4MDg5MWIwN2Y3MTY.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: Bria release 2.5.4 stamp 53956
Content-Length: 325

v=0
o=- 0 2 IN IP4 82.80.130.222
s=CounterPath Bria
c=IN IP4 82.80.130.222
t=0 0
m=audio 27848 RTP/AVP 18 101
a=sendrecv
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 2 : XlhJjZhm LLmd1Kyz 192.168.1.50 24514
a=alt:2 1 : GOsgLipv VjRq6zYk 192.168.1.60 24514

-------------------


message to leg B see the original IP addresses in the SDP  ( anything that is not 81.89.136.231)

note 82.80.130.222 which is the original address being visible in the SDP fields

-------------------
INVITE sip:1001 at 82.80.130.222:20014 SIP/2.0
Via: SIP/2.0/UDP 81.89.136.231:5080;rport;branch=z9hG4bKBBgvK7Sap23tN
Max-Forwards: 67
From: "44558678567378" <sip:44558678567378 at 81.89.136.231>;tag=1FUrD0t1gF55a
To: <sip:1001 at 82.80.130.222:20014>
Call-ID: 8eed1cbc-9030-11de-87e7-27e3a7b5c9da
CSeq: 119414118 INVITE
Contact: <sip:mod_sofia at 81.89.136.231:5080>
User-Agent: FreeSWITCH-mod_sofia/1.0.4-hacked
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 331
Remote-Party-ID: "44558678567378" <sip:44558678567378 at 81.89.136.231>;party=calling;screen=yes;privacy=off

v=0
o=- 2301954626585387485 2 IN IP4 82.80.130.222
s=CounterPath Bria
c=IN IP4 81.89.136.231
t=0 0
m=audio 25974 RTP/AVP 18 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=alt:1 2 : XlhJjZhm LLmd1Kyz 192.168.1.50 24514
a=alt:2 1 : GOsgLipv VjRq6zYk 192.168.1.60 24514

-------------------



thank you
Ori


      
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From brian at freeswitch.org  Mon Aug 24 07:53:15 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 09:53:15 -0500
Subject: [Freeswitch-users] topology hiding leaking information in SDP
	data
In-Reply-To: <510213.5618.qm@web44709.mail.sp1.yahoo.com>
References: <510213.5618.qm@web44709.mail.sp1.yahoo.com>
Message-ID: <6D897008-2E65-4314-B890-18264A615EBC@freeswitch.org>

You're using Proxy Media and the only clean way to do this is not use  
proxy media that way a complete clean SDP will be generated for the B- 
Leg.

/b

On Aug 24, 2009, at 9:45 AM, Hound Dog wrote:

> is there a way to have FS build a clean SDP message ?

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From vhatz at kinetix.gr  Mon Aug 24 08:28:05 2009
From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI))
Date: Mon, 24 Aug 2009 18:28:05 +0300
Subject: [Freeswitch-users] topology hiding leaking information in SDP
 data
In-Reply-To: <6D897008-2E65-4314-B890-18264A615EBC@freeswitch.org>
References: <510213.5618.qm@web44709.mail.sp1.yahoo.com>
	<6D897008-2E65-4314-B890-18264A615EBC@freeswitch.org>
Message-ID: <4A92B185.7030401@kinetix.gr>

But that would allow tell FS to also do transcoding in some cases, correct?

Is there a way to avoid transcoding and still build the required SDP in 
a clean manner?


Best regards,
Vlasis Hatzistavrou.

Brian West wrote:
> You're using Proxy Media and the only clean way to do this is not use 
> proxy media that way a complete clean SDP will be generated for the B-Leg.
> 
> /b
> 
> On Aug 24, 2009, at 9:45 AM, Hound Dog wrote:
> 
>> is there a way to have FS build a clean SDP message ?
> 
> 
> ------------------------------------------------------------------------
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org



From brian at freeswitch.org  Mon Aug 24 08:37:57 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 10:37:57 -0500
Subject: [Freeswitch-users] topology hiding leaking information in SDP
	data
In-Reply-To: <4A92B185.7030401@kinetix.gr>
References: <510213.5618.qm@web44709.mail.sp1.yahoo.com>
	<6D897008-2E65-4314-B890-18264A615EBC@freeswitch.org>
	<4A92B185.7030401@kinetix.gr>
Message-ID: <057646D2-5A8D-4877-8C2B-E72D2D902446@freeswitch.org>

You can disable transcoding on the sofia profiles... see defaults.

/b

On Aug 24, 2009, at 10:28 AM, Vlasis Hatzistavrou (KTI) wrote:

> But that would allow tell FS to also do transcoding in some cases,  
> correct?
>
> Is there a way to avoid transcoding and still build the required SDP  
> in
> a clean manner?
>
>
> Best regards,
> Vlasis Hatzistavrou.




From d_hound at ymail.com  Mon Aug 24 09:33:14 2009
From: d_hound at ymail.com (Hound Dog)
Date: Mon, 24 Aug 2009 09:33:14 -0700 (PDT)
Subject: [Freeswitch-users]  topology hiding leaking information in SDP
Message-ID: <508126.46278.qm@web111920.mail.gq1.yahoo.com>

Brian,

I set proxy_media to false ad now getting clean SDP without all the incoming data

thank you 



this also means that the packets have to go all the way in and out of the codec chain  ( instead of being taken in and thrown out with no processing )   , is that the case ?

what is the performance hit of such change , on a modern strong machine - what is the estimated streams that FS can take ?



BTW - sorry for not answering the list as I should have , I think I registered wrong somewhere so now have a new user 


      
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From anthony.minessale at gmail.com  Mon Aug 24 09:51:29 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 11:51:29 -0500
Subject: [Freeswitch-users] XML-RPC on different ip than 0.0.0.0
In-Reply-To: <4A92789C.5040900@gmx.net>
References: <4A92789C.5040900@gmx.net>
Message-ID: <191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>

there is a configuration option in the xml file to control which ip it binds
to.



On Mon, Aug 24, 2009 at 6:25 AM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Hello,
>
> is there any chance to limit the listening ips of the xml-rpc server
> (which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)?
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Mon Aug 24 09:58:47 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 11:58:47 -0500
Subject: [Freeswitch-users] topology hiding leaking information in SDP
In-Reply-To: <508126.46278.qm@web111920.mail.gq1.yahoo.com>
References: <508126.46278.qm@web111920.mail.gq1.yahoo.com>
Message-ID: <191c3a030908240958q45f44fd6jb2eeaa6e6a95bbea@mail.gmail.com>

if you establish an audio path that is using the same codec there is no
difference in performance as the codecs
do not do anything when both sides are the same.


On Mon, Aug 24, 2009 at 11:33 AM, Hound Dog <d_hound at ymail.com> wrote:

> Brian,
>
> I set proxy_media to false ad now getting clean SDP without all the
> incoming data
>
> thank you
>
>
>
> this also means that the packets have to go all the way in and out of the
> codec chain  ( instead of being taken in and thrown out with no processing )
>   , is that the case ?
>
> what is the performance hit of such change , on a modern strong machine -
> what is the estimated streams that FS can take ?
>
>
>
> BTW - sorry for not answering the list as I should have , I think I
> registered wrong somewhere so now have a new user
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From tculjaga at gmail.com  Mon Aug 24 10:19:01 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 24 Aug 2009 19:19:01 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK
	message
In-Reply-To: <65d96fc80908241006p7b5aef46n1475a1fc0710ca4@mail.gmail.com>
References: <65d96fc80908241006p7b5aef46n1475a1fc0710ca4@mail.gmail.com>
Message-ID: <65d96fc80908241019p365b563eu4507913d3fcb7ecd@mail.gmail.com>

Hello,

I've been with freeswittch for a while now.. and i can say it is worth
developing it.

anyhow i got into a strange issue... I'm tryng to see what load FS on my
server can take. The Call flow is like this:

SIPp                   FS

INVITE -------->
           <------- 100 Trying
           <------- 302 Moved Temporary
ACK    --------->



I use a dummy dialplan for that. All custom functions i've build are
disabled and i'm not using it here. Also custom modules are not loaded as
well.


   <extension name="ServiceLookup">
      <condition field="destination_number" expression="(^300030)(.*)">
         <!--action application="lookup_service_destination" data="in
${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, i
n $1, in pgw01.ot.hr:5060, out red_contact, out authResult"/-->
         <action application="log" data="INFO ########################
ServiceLookup ########################\n"/>
         <action application="log" data="INFO ########################
contact = '${red_contact}' ##############\n"/>
         <action application="log" data="INFO ########################
CallerNum = '${caller_id_number:6:16}' ##########\n"/>
         <action application="log" data="INFO ########################
RADIUS auth = '${authResult}' ##########\n"/>
         <action application="execute_extension" data="doRedirect XML
public"/>
        </condition>
   </extension>


   <extension name="doRedirect">
      <condition field="destination_number" expression="^doRedirect$"/>
      <condition field="${authResult}" expression="^0$|^60$">
         <action application="log" data="INFO ########################
RADIUS auth OK!!!' ##########\n"/>
         <!--action application="redirect" data="sip:${red_contact}"/-->
         <!--action application="answer"/-->
         <action application="redirect" data="
sip:12345616094191500 at pgw01.ot.hr:5060"/>
         <!--anti-action application="answer"/-->
         <!--anti-action application="sleep" data="2000"/-->
         <action application="hangup" data="USER_BUSY"/>
         <anti-action application="redirect" data="
sip:12345616094191500 at pgw01.ot.hr:5060"/>
         <anti-action application="log" data="INFO ########################
RADIUS auth NOK!! ##########\n"/>
         <!--anti-action application="respond" data="403 Forbidden"/-->
         <anti-action application="hangup" data="USER_BUSY"/>
      </condition>
   </extension>


When i place a call from x-lite everything works fine ... x-lite sends an
invite, gets SIP 302 and ACKs it correctly... FS is happy.

When i place a call from SIPp i have the same scenario except FS seems not
understand ACK message from SIPp and re-sends SIP 302 multiple times untill
it gives up.


I beleive this is due to 302 resend issue but; when i load FS with 100 CPS,
i can see high CPU usage (just one thread taking most load... the rest does
almost nothing) on FS. Also, starting from 40 CPS there is a big delay in
receiving SIP 302 messages meaning i've sent 6000 calls and so far only for
half of them got 302 response.


Does anybody have a clue ?





Here is a trace taken on FS for calls originated from SIPp (sipp -sn uac
10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100
-trace_msg -inf test.txt -m 1 -l 4000):

freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
16:44:26.527236:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Max-Forwards: 70
   Subject: Performance Test
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 ACK
   Contact: sip:sipp at 10.4.4.252:5060
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0


Tihomir.
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freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 16:44:26.527236:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Max-Forwards: 70
   Subject: Performance Test
   Content-Type: application/sdp
   Content-Length:   131
   
   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-08-24 18:44:26.525364 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22222238515000403 at 10.4.4.251 [62a395f8-90cd-11de-99d5-cb0c6bd8522b]
2009-08-24 18:44:26.533096 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->30003016094191500 in context public
2009-08-24 18:44:26.533096 [INFO] mod_dptools.c:932 ######################## ServiceLookup ########################\n
2009-08-24 18:44:26.533096 [INFO] mod_dptools.c:932 ######################## contact = '' ##############\n
2009-08-24 18:44:26.533096 [INFO] mod_dptools.c:932 ######################## CallerNum = '38515000403' ##########\n
2009-08-24 18:44:26.533096 [INFO] mod_dptools.c:932 ######################## RADIUS auth = '' ##########\n
2009-08-24 18:44:26.533096 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->doRedirect in context public
2009-08-24 18:44:26.533096 [NOTICE] switch_core_session.c:1576 Execute redirect(sip:12345616094191500 at pgw01.ot.hr:5060)
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 ACK
   Contact: sip:sipp at 10.4.4.252:5060
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-08-24 18:44:26.533096 [NOTICE] sofia.c:3863 Hangup sofia/internal/22222238515000403 at 10.4.4.251 [CS_EXECUTE] [NORMAL_UNSPECIFIED]
2009-08-24 18:44:26.537029 [NOTICE] switch_core_session.c:1086 Session 3 (sofia/internal/22222238515000403 at 10.4.4.251) Ended
2009-08-24 18:44:26.537029 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22222238515000403 at 10.4.4.251 [CS_DESTROY]
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:30.037065:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:34.037063:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:38.037068:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:42.037064:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:46.037064:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:50.037066:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:54.037067:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:58.037067:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
-------------- next part --------------
freeswitch at l01sipindir1> 
freeswitch at l01sipindir1> 
freeswitch at l01sipindir1> recv 855 bytes from udp/[10.1.14.30]:27454 at 16:42:56.522659:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251 SIP/2.0
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-b9056b2ac55b4e54-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.1.14.30:27454>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 1 INVITE
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
   Content-Type: application/sdp
   User-Agent: X-Lite release 1103k stamp 53621
   Content-Length: 258
   
   v=0
   o=- 0 2 IN IP4 10.1.14.30
   s=CounterPath X-Lite 3.0
   c=IN IP4 10.1.14.30
   t=0 0
   m=audio 44488 RTP/AVP 107 0 8 101
   a=alt:1 1 : 5KgV8ECY o9MWrE7h 10.1.14.30 44488
   a=fmtp:101 0-15
   a=rtpmap:107 BV32/16000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   ------------------------------------------------------------------------
send 396 bytes to udp/[10.1.14.30]:27454 at 16:42:56.523076:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-b9056b2ac55b4e54-1---d8754z-;rport=27454
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-08-24 18:42:56.521041 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22222238515000403 at 10.4.4.251 [2cfdfdbc-90cd-11de-99d5-cb0c6bd8522b]
2009-08-24 18:42:56.529053 [CONSOLE] sofia_presence.c:680 Event Thread Started
2009-08-24 18:42:56.529053 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->30003016094191500 in context public
2009-08-24 18:42:56.529053 [INFO] mod_dptools.c:932 ######################## ServiceLookup ########################\n
2009-08-24 18:42:56.529053 [INFO] mod_dptools.c:932 ######################## contact = '' ##############\n
2009-08-24 18:42:56.529053 [INFO] mod_dptools.c:932 ######################## CallerNum = '38515000403' ##########\n
2009-08-24 18:42:56.529053 [INFO] mod_dptools.c:932 ######################## RADIUS auth = '' ##########\n
2009-08-24 18:42:56.529053 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->doRedirect in context public
2009-08-24 18:42:56.529053 [NOTICE] switch_core_session.c:1576 Execute redirect(sip:12345616094191500 at pgw01.ot.hr:5060)
send 790 bytes to udp/[10.1.14.30]:27454 at 16:42:56.531535:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-b9056b2ac55b4e54-1---d8754z-;rport=27454
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=F6H3DQS7g78ra
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces2009-08-24 18:42:56.529053 [NOTICE] switch_core_session.c:1576 Execute log(INFO ######################## RADIUS auth NOK!! ##########\n)

   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-08-24 18:42:56.529053 [NOTICE] sofia.c:3863 Hangup sofia/internal/22222238515000403 at 10.4.4.251 [CS_EXECUTE] [NORMAL_UNSPECIFIED]
2009-08-24 18:42:56.529053 [INFO] mod_dptools.c:932 ######################## RADIUS auth NOK!! ##########\n
2009-08-24 18:42:56.529053 [NOTICE] switch_core_session.c:1086 Session 1 (sofia/internal/22222238515000403 at 10.4.4.251) Ended
2009-08-24 18:42:56.529053 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22222238515000403 at 10.4.4.251 [CS_DESTROY]
recv 379 bytes from udp/[10.1.14.30]:27454 at 16:42:56.533966:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251 SIP/2.0
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-b9056b2ac55b4e54-1---d8754z-;rport
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=F6H3DQS7g78ra
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 1 ACK
   Content-Length: 0
   
   ------------------------------------------------------------------------
recv 861 bytes from udp/[10.1.14.30]:27454 at 16:42:56.638930:
   ------------------------------------------------------------------------
   INVITE sip:12345616094191500 at pgw01.ot.hr:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-213d2e12ad57b353-1---d8754z-;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.1.14.30:27454>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 2 INVITE
   Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
   Content-Type: application/sdp
   User-Agent: X-Lite release 1103k stamp 53621
   Content-Length: 258
   
   v=0
   o=- 0 2 IN IP4 10.1.14.30
   s=CounterPath X-Lite 3.0
   c=IN IP4 10.1.14.30
   t=0 0
   m=audio 44488 RTP/AVP 107 0 8 101
   a=alt:1 1 : 5KgV8ECY o9MWrE7h 10.1.14.30 44488
   a=fmtp:101 0-15
   a=rtpmap:107 BV32/16000
   a=rtpmap:101 telephone-event/8000
   a=sendrecv
   ------------------------------------------------------------------------
send 396 bytes to udp/[10.1.14.30]:27454 at 16:42:56.639220:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-213d2e12ad57b353-1---d8754z-;rport=27454
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 2 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-08-24 18:42:56.637047 [NOTICE] switch_channel.c:602 New Channel sofia/internal/22222238515000403 at 10.4.4.251 [2d0fb3fe-90cd-11de-99d5-cb0c6bd8522b]
2009-08-24 18:42:56.641071 [INFO] mod_dialplan_xml.c:315 Processing 22222238515000403->12345616094191500 in context public
2009-08-24 18:42:56.641071 [INFO] switch_core_state_machine.c:136 No Route, Aborting
2009-08-24 18:42:56.641071 [NOTICE] switch_core_state_machine.c:137 Hangup sofia/internal/22222238515000403 at 10.4.4.251 [CS_ROUTING] [NO_ROUTE_DESTINATION]
send 782 bytes to udp/[10.1.14.30]:27454 at 16:42:56.642591:
   ------------------------------------------------------------------------
   SIP/2.0 404 Not Found
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-213d2e12ad57b353-1---d8754z-;rport=27454
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=gFBvFjaBegZBp
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 2 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Reason: Q.850;cause=3;text="NO_ROUTE_DESTINATION"
   Content-Length: 0
   
   ------------------------------------------------------------------------
2009-08-24 18:42:56.641071 [NOTICE] switch_core_session.c:1086 Session 2 (sofia/internal/22222238515000403 at 10.4.4.251) Ended
2009-08-24 18:42:56.641071 [NOTICE] switch_core_session.c:1088 Close Channel sofia/internal/22222238515000403 at 10.4.4.251 [CS_DESTROY]
recv 385 bytes from udp/[10.1.14.30]:27454 at 16:42:56.646734:
   ------------------------------------------------------------------------
   ACK sip:12345616094191500 at pgw01.ot.hr:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.1.14.30:27454;branch=z9hG4bK-d8754z-213d2e12ad57b353-1---d8754z-;rport
   To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251>;tag=gFBvFjaBegZBp
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=a669bf79
   Call-ID: ZjBmNWQwN2EzZjNmZmM0MzUxMDRlMTkyNjM2MzQ2MTc.
   CSeq: 2 ACK
   Content-Length: 0
   
   ------------------------------------------------------------------------
-------------- next part --------------
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Size: 3927 bytes
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Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090824/ef7ebc73/attachment-0002.xml 
-------------- next part --------------
SEQUENTIAL
22222238515000403;22222238515000403

From anthony.minessale at gmail.com  Mon Aug 24 10:32:27 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 12:32:27 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241019p365b563eu4507913d3fcb7ecd@mail.gmail.com>
References: <65d96fc80908241006p7b5aef46n1475a1fc0710ca4@mail.gmail.com>
	<65d96fc80908241019p365b563eu4507913d3fcb7ecd@mail.gmail.com>
Message-ID: <191c3a030908241032i7d8eef27v5baf67b909d01e95@mail.gmail.com>

Your ACK message must not be valid (dialog matching or something else)
so every 1 call will generate 30 retries that are queued up in the sip
stack.

at 100cps you will be generating this problem 100 times per second and queue
up countless unfinished dialogs thus
eating up the cpu.






On Mon, Aug 24, 2009 at 12:19 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> Hello,
>
> I've been with freeswittch for a while now.. and i can say it is worth
> developing it.
>
> anyhow i got into a strange issue... I'm tryng to see what load FS on my
> server can take. The Call flow is like this:
>
> SIPp                   FS
>
> INVITE -------->
>            <------- 100 Trying
>            <------- 302 Moved Temporary
> ACK    --------->
>
>
>
> I use a dummy dialplan for that. All custom functions i've build are
> disabled and i'm not using it here. Also custom modules are not loaded as
> well.
>
>
>    <extension name="ServiceLookup">
>       <condition field="destination_number" expression="(^300030)(.*)">
>          <!--action application="lookup_service_destination" data="in
> ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, i
> n $1, in pgw01.ot.hr:5060, out red_contact, out authResult"/-->
>          <action application="log" data="INFO ########################
> ServiceLookup ########################\n"/>
>          <action application="log" data="INFO ########################
> contact = '${red_contact}' ##############\n"/>
>          <action application="log" data="INFO ########################
> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
>          <action application="log" data="INFO ########################
> RADIUS auth = '${authResult}' ##########\n"/>
>          <action application="execute_extension" data="doRedirect XML
> public"/>
>         </condition>
>    </extension>
>
>
>    <extension name="doRedirect">
>       <condition field="destination_number" expression="^doRedirect$"/>
>       <condition field="${authResult}" expression="^0$|^60$">
>          <action application="log" data="INFO ########################
> RADIUS auth OK!!!' ##########\n"/>
>          <!--action application="redirect" data="sip:${red_contact}"/-->
>          <!--action application="answer"/-->
>          <action application="redirect" data="
> sip:12345616094191500 at pgw01.ot.hr:5060"/>
>          <!--anti-action application="answer"/-->
>          <!--anti-action application="sleep" data="2000"/-->
>          <action application="hangup" data="USER_BUSY"/>
>          <anti-action application="redirect" data="
> sip:12345616094191500 at pgw01.ot.hr:5060"/>
>          <anti-action application="log" data="INFO ########################
> RADIUS auth NOK!! ##########\n"/>
>          <!--anti-action application="respond" data="403 Forbidden"/-->
>          <anti-action application="hangup" data="USER_BUSY"/>
>       </condition>
>    </extension>
>
>
> When i place a call from x-lite everything works fine ... x-lite sends an
> invite, gets SIP 302 and ACKs it correctly... FS is happy.
>
> When i place a call from SIPp i have the same scenario except FS seems not
> understand ACK message from SIPp and re-sends SIP 302 multiple times untill
> it gives up.
>
>
> I beleive this is due to 302 resend issue but; when i load FS with 100 CPS,
> i can see high CPU usage (just one thread taking most load... the rest does
> almost nothing) on FS. Also, starting from 40 CPS there is a big delay in
> receiving SIP 302 messages meaning i've sent 6000 calls and so far only for
> half of them got 302 response.
>
>
> Does anybody have a clue ?
>
>
>
>
>
> Here is a trace taken on FS for calls originated from SIPp (sipp -sn uac
> 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100
> -trace_msg -inf test.txt -m 1 -l 4000):
>
> freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
> 16:44:26.527236:
>    ------------------------------------------------------------------------
>    INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
>    Max-Forwards: 70
>    Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
> >
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
> >
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
> >;tag=1
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Type: application/sdp
>    Content-Length:   131
>
>    v=0
>    o=user1 53655765 2353687637 IN IP4 10.4.4.252
>    s=-
>    c=IN IP4 10.4.4.252
>    t=0 0
>    m=audio 6000 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
>    ------------------------------------------------------------------------
> send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
> >;tag=1
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
> >
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
>    ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
> >;tag=1
>    To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
> >;tag=Hr4mHDUeBSNyH
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
>    ------------------------------------------------------------------------
>    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
> >
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
> >;tag=1
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 ACK
>    Contact: sip:sipp at 10.4.4.252:5060
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070:
>    ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
> >;tag=1
>    To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
> >;tag=Hr4mHDUeBSNyH
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063:
>    ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
> >;tag=1
>    To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
> >;tag=Hr4mHDUeBSNyH
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>
> Tihomir.
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From raffaele.p.guidi at gmail.com  Mon Aug 24 11:24:28 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Mon, 24 Aug 2009 20:24:28 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
Message-ID: <b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>

Actually I did that and it worked fine. I had the problem the SECOND time I
run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were
not

On Sun, Aug 16, 2009 at 16:04, Carlos Talbot <carlos.talbot at gmail.com>wrote:

> When you configure FreePBX for the first time it wipes out the sip_profiles
> directory. If you follow the FreePBX shortcut on your desktop it'll mention
> this on the last screen of the configuration. You might see something such
> as the following below. If you plan to use FreePBX you'll need to define
> trunk groups, trunks, etc in order to have the sip_profiles directory
> populated.
> regards,
>
> Carlos
>
>
>  Incompatible Configuration WARNING: THE FOLLOWING FILES WILL BE DELETED!
>
>    - D:/FreeSWITCH/conf/sip_profiles/external.xml
>    - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
>    - D:/FreeSWITCH/conf/sip_profiles/internal.xml
>
>
> On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi <
> raffaele.p.guidi at gmail.com> wrote:
>
>> I had the sweet surprise to find the installer packaged with FreePBX...
>> really great! Why it has not been advertised as it deserves? It worked like
>> a breeze once launched, with the automatic configuration and all of that.,
>> Only thing: once stopped I cannot get it to load sofia profiles anymore -
>> issueing sofia status doesn't show anything. I had to copy internal.xml and
>> default.xml from a previous installation and everything got to work again -
>> but no FreePBX anymore :( I'm sure I'm missing something important.
>> Can you give me a hint? Should sofia profiles be served by curl or
>> something?
>>
>> Thanks,
>>    Raffaele
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From raffaele.p.guidi at gmail.com  Mon Aug 24 11:27:12 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Mon, 24 Aug 2009 20:27:12 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
Message-ID: <b0a8792f0908241127k54ac14bfo8f441244ddc838b9@mail.gmail.com>

actually I meant "the SECOND time I *RAN* FS and freepbx."

On Mon, Aug 24, 2009 at 20:24, Raffaele P. Guidi <raffaele.p.guidi at gmail.com
> wrote:

> Actually I did that and it worked fine. I had the problem the SECOND time I
> run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were
> not
>
>
> On Sun, Aug 16, 2009 at 16:04, Carlos Talbot <carlos.talbot at gmail.com>wrote:
>
>> When you configure FreePBX for the first time it wipes out the
>> sip_profiles directory. If you follow the FreePBX shortcut on your desktop
>> it'll mention this on the last screen of the configuration. You might see
>> something such as the following below. If you plan to use FreePBX you'll
>> need to define trunk groups, trunks, etc in order to have the sip_profiles
>> directory populated.
>> regards,
>>
>> Carlos
>>
>>
>>  Incompatible Configuration WARNING: THE FOLLOWING FILES WILL BE DELETED!
>>
>>    - D:/FreeSWITCH/conf/sip_profiles/external.xml
>>    - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
>>    - D:/FreeSWITCH/conf/sip_profiles/internal.xml
>>
>>
>> On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi <
>> raffaele.p.guidi at gmail.com> wrote:
>>
>>> I had the sweet surprise to find the installer packaged with FreePBX...
>>> really great! Why it has not been advertised as it deserves? It worked like
>>> a breeze once launched, with the automatic configuration and all of that.,
>>> Only thing: once stopped I cannot get it to load sofia profiles anymore -
>>> issueing sofia status doesn't show anything. I had to copy internal.xml and
>>> default.xml from a previous installation and everything got to work again -
>>> but no FreePBX anymore :( I'm sure I'm missing something important.
>>> Can you give me a hint? Should sofia profiles be served by curl or
>>> something?
>>>
>>> Thanks,
>>>    Raffaele
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From brian at freeswitch.org  Mon Aug 24 11:30:43 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 13:30:43 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
Message-ID: <22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>

If you installed FreePBX then it would be that softwares job to manage  
the sofia profiles... wouldn't it?

/b

On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:

> Actually I did that and it worked fine. I had the problem the SECOND  
> time I run FS and freepbx. And (@Brian) mod_sofia was loaded but  
> sip_profiles were not




From d_hound at ymail.com  Mon Aug 24 11:37:53 2009
From: d_hound at ymail.com (Hound Dog)
Date: Mon, 24 Aug 2009 11:37:53 -0700 (PDT)
Subject: [Freeswitch-users] topology hiding leaking information in SDP
In-Reply-To: <191c3a030908240958q45f44fd6jb2eeaa6e6a95bbea@mail.gmail.com>
References: <508126.46278.qm@web111920.mail.gq1.yahoo.com>
	<191c3a030908240958q45f44fd6jb2eeaa6e6a95bbea@mail.gmail.com>
Message-ID: <920686.47099.qm@web111920.mail.gq1.yahoo.com>

so problem solved , Thanks you anthony and Brian



________________________________
From: Anthony Minessale <anthony.minessale at gmail.com>
To: freeswitch-users at lists.freeswitch.org
Sent: Monday, August 24, 2009 7:58:47 PM
Subject: Re: [Freeswitch-users] topology hiding leaking information in SDP

if you establish an audio path that is using the same codec there is no difference in performance as the codecs
do not do anything when both sides are the same.



On Mon, Aug 24, 2009 at 11:33 AM, Hound Dog <d_hound at ymail.com> wrote:

Brian,
>
>
>I set proxy_media to false ad now getting clean SDP without all the incoming data
>
>
>thank you 
>
>
>
>
>
>
>this also means that the packets have to go all the way in and out of the codec chain  ( instead of being taken in and thrown out with no processing )   , is that the case ?
>
>
>what is the performance hit of such change , on a modern strong machine - what is the estimated streams that FS can take ?
>
>
>
>
>
>
>BTW - sorry for not answering the list as I should have , I think I registered wrong somewhere so now have a new user 
>
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org
pstn:213-799-1400



      
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From tculjaga at gmail.com  Mon Aug 24 12:03:01 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 24 Aug 2009 21:03:01 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK
	message
Message-ID: <65d96fc80908241203m29abc2c6t7678689a3402c31f@mail.gmail.com>

Hi Anthony,

I'm aware it is generating 30 retries per a call and this is killing me ...

I lost my entire working day to figure out what is missing in the damn ACK
message SIPp is sending back... ACK looks quite ok to me.

pls can you help ?


freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
16:44:26.527236:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Max-Forwards: 70
   Subject: Performance Test
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=Hr4mHDUeBSNyH
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0


   ------------------------------------------------------------------------
recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-6962 at 10.4.4.252
   CSeq: 1 ACK
   Contact: sip:sipp at 10.4.4.252:5060
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0




What m'i missing ?



> Your ACK message must not be valid (dialog matching or something else)
> so every 1 call will generate 30 retries that are queued up in the sip
> stack.
>
> at 100cps you will be generating this problem 100 times per second and
> queue up countless unfinished dialogs thus
> eating up the cpu.
>
>
>
>
>
>
> On Mon, Aug 24, 2009 at 12:19 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> Hello,
>>
>> I've been with freeswittch for a while now.. and i can say it is worth
>> developing it.
>>
>> anyhow i got into a strange issue... I'm tryng to see what load FS on my
>> server can take. The Call flow is like this:
>>
>> SIPp                   FS
>>
>> INVITE -------->
>>            <------- 100 Trying
>>            <------- 302 Moved Temporary
>> ACK    --------->
>>
>>
>>
>> I use a dummy dialplan for that. All custom functions i've build are
>> disabled and i'm not using it here. Also custom modules are not loaded as
>> well.
>>
>>
>>    <extension name="ServiceLookup">
>>       <condition field="destination_number" expression="(^300030)(.*)">
>>          <!--action application="lookup_service_destination" data="in
>> ${caller_id_number:6:16}, in ${caller_id_number:0:6}, in $2, i
>> n $1, in pgw01.ot.hr:5060, out red_contact, out authResult"/-->
>>          <action application="log" data="INFO ########################
>> ServiceLookup ########################\n"/>
>>          <action application="log" data="INFO ########################
>> contact = '${red_contact}' ##############\n"/>
>>          <action application="log" data="INFO ########################
>> CallerNum = '${caller_id_number:6:16}' ##########\n"/>
>>          <action application="log" data="INFO ########################
>> RADIUS auth = '${authResult}' ##########\n"/>
>>          <action application="execute_extension" data="doRedirect XML
>> public"/>
>>         </condition>
>>    </extension>
>>
>>
>>    <extension name="doRedirect">
>>       <condition field="destination_number" expression="^doRedirect$"/>
>>       <condition field="${authResult}" expression="^0$|^60$">
>>          <action application="log" data="INFO ########################
>> RADIUS auth OK!!!' ##########\n"/>
>>          <!--action application="redirect" data="sip:${red_contact}"/-->
>>          <!--action application="answer"/-->
>>          <action application="redirect" data="
>> sip:12345616094191500 at pgw01.ot.hr:5060"/>
>>          <!--anti-action application="answer"/-->
>>          <!--anti-action application="sleep" data="2000"/-->
>>          <action application="hangup" data="USER_BUSY"/>
>>          <anti-action application="redirect" data="
>> sip:12345616094191500 at pgw01.ot.hr:5060"/>
>>          <anti-action application="log" data="INFO
>> ######################## RADIUS auth NOK!! ##########\n"/>
>>          <!--anti-action application="respond" data="403 Forbidden"/-->
>>          <anti-action application="hangup" data="USER_BUSY"/>
>>       </condition>
>>    </extension>
>>
>>
>> When i place a call from x-lite everything works fine ... x-lite sends an
>> invite, gets SIP 302 and ACKs it correctly... FS is happy.
>>
>> When i place a call from SIPp i have the same scenario except FS seems not
>> understand ACK message from SIPp and re-sends SIP 302 multiple times untill
>> it gives up.
>>
>>
>> I beleive this is due to 302 resend issue but; when i load FS with 100
>> CPS, i can see high CPU usage (just one thread taking most load... the rest
>> does almost nothing) on FS. Also, starting from 40 CPS there is a big delay
>> in receiving SIP 302 messages meaning i've sent 6000 calls and so far only
>> for half of them got 302 response.
>>
>>
>> Does anybody have a clue ?
>>
>>
>>
>>
>>
>> Here is a trace taken on FS for calls originated from SIPp (sipp -sn uac
>> 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100
>> -trace_msg -inf test.txt -m 1 -l 4000):
>>
>> freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
>> 16:44:26.527236:
>>
>> ------------------------------------------------------------------------
>>    INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
>>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
>>    Max-Forwards: 70
>>    Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>> >
>>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >
>>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>    Call-ID: 1-6962 at 10.4.4.252
>>    CSeq: 1 INVITE
>>    Max-Forwards: 70
>>    Subject: Performance Test
>>    Content-Type: application/sdp
>>    Content-Length:   131
>>
>>    v=0
>>    o=user1 53655765 2353687637 IN IP4 10.4.4.252
>>    s=-
>>    c=IN IP4 10.4.4.252
>>    t=0 0
>>    m=audio 6000 RTP/AVP 0
>>    a=rtpmap:0 PCMU/8000
>>
>> ------------------------------------------------------------------------
>> send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 100 Trying
>>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >
>>    Call-ID: 1-6962 at 10.4.4.252
>>    CSeq: 1 INVITE
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 302 Moved Temporarily
>>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>    To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >;tag=Hr4mHDUeBSNyH
>>    Call-ID: 1-6962 at 10.4.4.252
>>    CSeq: 1 INVITE
>>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>>    Supported: timer, precondition, path, replaces
>>    Allow-Events: talk, presence, dialog, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
>>
>> ------------------------------------------------------------------------
>>    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>>    Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >
>>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>    Call-ID: 1-6962 at 10.4.4.252
>>    CSeq: 1 ACK
>>    Contact: sip:sipp at 10.4.4.252:5060
>>    Max-Forwards: 70
>>    Subject: Performance Test
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 302 Moved Temporarily
>>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>    To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >;tag=Hr4mHDUeBSNyH
>>    Call-ID: 1-6962 at 10.4.4.252
>>    CSeq: 1 INVITE
>>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>>    Supported: timer, precondition, path, replaces
>>    Allow-Events: talk, presence, dialog, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>>    Content-Length: 0
>>
>>
>> ------------------------------------------------------------------------
>> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063:
>>
>> ------------------------------------------------------------------------
>>    SIP/2.0 302 Moved Temporarily
>>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>    To: "30003016094191500" <sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >;tag=Hr4mHDUeBSNyH
>>    Call-ID: 1-6962 at 10.4.4.252
>>    CSeq: 1 INVITE
>>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>>    Accept: application/sdp
>>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
>> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>>    Supported: timer, precondition, path, replaces
>>    Allow-Events: talk, presence, dialog, call-info, sla,
>> include-session-description, presence.winfo, message-summary, refer
>>    Content-Length: 0
>>
>>
>> Tihomir.
>>
>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
> ---------- Forwarded message ----------
> From: "Raffaele P. Guidi" <raffaele.p.guidi at gmail.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 20:24:28 +0200
> Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
> but I have a little problem
> Actually I did that and it worked fine. I had the problem the SECOND time I
> run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were
> not
>
> On Sun, Aug 16, 2009 at 16:04, Carlos Talbot <carlos.talbot at gmail.com>wrote:
>
>> When you configure FreePBX for the first time it wipes out the
>> sip_profiles directory. If you follow the FreePBX shortcut on your desktop
>> it'll mention this on the last screen of the configuration. You might see
>> something such as the following below. If you plan to use FreePBX you'll
>> need to define trunk groups, trunks, etc in order to have the sip_profiles
>> directory populated.
>> regards,
>>
>> Carlos
>>
>>
>>  Incompatible Configuration WARNING: THE FOLLOWING FILES WILL BE DELETED!
>>
>>    - D:/FreeSWITCH/conf/sip_profiles/external.xml
>>    - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
>>    - D:/FreeSWITCH/conf/sip_profiles/internal.xml
>>
>>
>> On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi <
>> raffaele.p.guidi at gmail.com> wrote:
>>
>>> I had the sweet surprise to find the installer packaged with FreePBX...
>>> really great! Why it has not been advertised as it deserves? It worked like
>>> a breeze once launched, with the automatic configuration and all of that.,
>>> Only thing: once stopped I cannot get it to load sofia profiles anymore -
>>> issueing sofia status doesn't show anything. I had to copy internal.xml and
>>> default.xml from a previous installation and everything got to work again -
>>> but no FreePBX anymore :( I'm sure I'm missing something important.
>>> Can you give me a hint? Should sofia profiles be served by curl or
>>> something?
>>>
>>> Thanks,
>>>    Raffaele
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Mon Aug 24 12:05:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 24 Aug 2009 15:05:58 -0400
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <4A91A480.7080307@gmx.net>
References: <4A8D5132.7010807@gmx.net>	<04CA910F-7FEE-43C4-9B7F-B566FEE8F7BE@freeswitch.org>	<4A8D5D43.3030300@gmx.net>	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>	<4A8DB808.1030903@gmx.net>	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>	<4A8EBF77.1080204@gmx.net>	<BD3DE8E0-3363-465C-985D-EC593AE45694@freeswitch.org>
	<191c3a030908211309j60c16b26h1e6445891136ae7f@mail.gmail.com>
	<4A91A480.7080307@gmx.net>
Message-ID: <D8BD24FC-9B07-4045-B6FD-A6D00156F877@jerris.com>

That is the remote sdp, not the local sdp.  They are sending ptime 20,  
not us.  Are they actually sending 20 ms packets or are they sending 30?

MIke

On Aug 23, 2009, at 4:20 PM, Peter P GMX wrote:

> Hello Anthony,
>
> I set PCMA at 30i,PCMU at 30i and I can see in the logs that PCMA is used.
> However ptime is set to 20 msec as shown in the Logs:
>
> 2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP:
> v=0
> o=user 2075230 2075230 IN IP4 217.xx.xx.xxx
> s=call
> c=IN IP4 217.xx.xx.xxx
> t=0 0
> m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
> a=rtpmap:2 G726-32/8000
> a=rtpmap:102 G726-32/8000
> a=rtpmap:100 G726-40/8000
> a=rtpmap:99 G726-24/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-11
> a=rtcp:7079
>
> 2009-08-23 22:11:29.530301 [DEBUG] switch_core_state_machine.c:404
> (sofia/internal/02xxxxxxxxx at fs1.my.domain) State NEW
> 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec  
> Compare
> [PCMA:8:8000:0]/[G722:9:8000:20]
> 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec  
> Compare
> [PCMA:8:8000:0]/[PCMU:0:8000:20]
> 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec  
> Compare
> [PCMA:8:8000:0]/[PCMA:8:8000:20]
> 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:2090 Set Codec
> sofia/internal/02xxxxxxxxx at fs1.my.domain PCMA/8000 20 ms 160 samples
> 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
> payload to 101
>
> Sound from FS to Fritzbox is fine. Sound from Fritzbox to FS is  
> horrible.
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
>> try setting FS to 30ms too
>>
>> edit vars.xml and add @30i to everywhere you see  PCMU or PCMA so it
>> looks like PCMU at 30i
>>
>> from:
>>
>>  <X-PRE-PROCESS cmd="set"
>> data 
>> ="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
>>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
>>
>> to:
>>
>>  <X-PRE-PROCESS cmd="set"
>> data 
>> = 
>> "global_codec_prefs 
>> =G7221 at 32000h,G7221 at 16000h,G722,PCMU at 30i,PCMA at 30i,GSM"/>
>>  <X-PRE-PROCESS cmd="set"
>> data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i,GSM"/>
>>
>>
>> On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian at freeswitch.org
>> <mailto:brian at freeswitch.org>> wrote:
>>
>>    You can ship me one whois bkw.org <http://bkw.org>, I can add it
>>    to my lab.
>>
>>    /b
>>
>>    On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
>>
>>>
>>> BTW: We can ship you a FritzBox if you need one for testing.
>>
>>
>>    _______________________________________________
>>    FreeSWITCH-users mailing list
>>    FreeSWITCH-users at lists.freeswitch.org
>>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>> freeswitch-users
>>    http://www.freeswitch.org
>>
>>
>>
>>
>> -- 
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> <mailto:sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> <http://iax:guest at conference.freeswitch.org/888>
>> googletalk:conf+888 at conference.freeswitch.org
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From brian at freeswitch.org  Mon Aug 24 12:15:40 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 14:15:40 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241203m29abc2c6t7678689a3402c31f@mail.gmail.com>
References: <65d96fc80908241203m29abc2c6t7678689a3402c31f@mail.gmail.com>
Message-ID: <A972ED9E-9EA2-481C-818F-98437DF5FC09@freeswitch.org>

In your scenario you need to add [peer_tag_param] at the end of the to  
on the Ack.

/b

On Aug 24, 2009, at 2:03 PM, Tihomir Culjaga wrote:

>
>     
> ------------------------------------------------------------------------
> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
>     
> ------------------------------------------------------------------------
>    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 ACK
>    Contact: sip:sipp at 10.4.4.252:5060
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Length: 0




From jerry.richards at teotech.com  Mon Aug 24 12:24:42 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 24 Aug 2009 12:24:42 -0700
Subject: [Freeswitch-users] Cannot create outgoing channel type [error]
	cause: [FACILITY_NOT_SUBSCRIBED]
Message-ID: <16C552FB75474E8D82974FCF182B06D3@greyhawk.tonecommander.com>

Hello All,

I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP machine
for the first time using the Getting Started Guide.  I can register three
lines (1000, 1001, and 1002), but when I attempt to call one phone to the
other I hear the operator say:

"The person at extension 1000 is not available..."

Also, the Freeswitch log shows:

Cannot create outgoing channel type [error] cause:
[FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause:
[FACILITY_NOT_SUBSCRIBED]

Does anyone know why I get this error?

Best Regards,
Jerry




From brian at freeswitch.org  Mon Aug 24 12:33:22 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 14:33:22 -0500
Subject: [Freeswitch-users] Cannot create outgoing channel type [error]
	cause: [FACILITY_NOT_SUBSCRIBED]
In-Reply-To: <16C552FB75474E8D82974FCF182B06D3@greyhawk.tonecommander.com>
References: <16C552FB75474E8D82974FCF182B06D3@greyhawk.tonecommander.com>
Message-ID: <C61B5EA8-345C-471E-BBA5-534A1726F396@freeswitch.org>

Are you trying to test everything on the same machine?

/b

On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote:

> Hello All,
>
> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP  
> machine
> for the first time using the Getting Started Guide.  I can register  
> three
> lines (1000, 1001, and 1002), but when I attempt to call one phone  
> to the
> other I hear the operator say:
>
> "The person at extension 1000 is not available..."
>
> Also, the Freeswitch log shows:
>
> Cannot create outgoing channel type [error] cause:
> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user]  
> cause:
> [FACILITY_NOT_SUBSCRIBED]
>
> Does anyone know why I get this error?
>
> Best Regards,
> Jerry




From mike at jerris.com  Mon Aug 24 12:44:18 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 24 Aug 2009 15:44:18 -0400
Subject: [Freeswitch-users] Problem with cnam.js?
In-Reply-To: <00b901ca2347$950feee0$bf2fcca0$@com>
References: <00b901ca2347$950feee0$bf2fcca0$@com>
Message-ID: <2FCFC745-86FC-4A66-B919-200FC865C019@jerris.com>

Every page on the wiki should be editable.  If you don't already have  
an account, go to:

http://wiki.freeswitch.org/index.php?title=Special:UserLogin&type=signup

Mike

On Aug 22, 2009, at 12:42 PM, Lars Zeb wrote:

> I think there?s something wrong with the script at http://wiki.freeswitch.org/wiki/Examples_cnam.js 
> .
>
> If you use it as is, it displays ?Content-type: text/html? for the  
> effective_caller_id_name. In cnam.pl, the first two output lines are  
> generated by:
>
> if (!$debug) {print "Content-type: text/html\n\n";}
>
> with the actual name in the third line.
>
> So I changed:
>
> fd.open("read");
> buff = fd.readln();
>
> if(buff) {
>    logger(buff, "info");
>    session.setVariable("effective_caller_id_name", buff);
> }
>
> To:
>
> fd.open("read");
> buff = fd.readAll();
>
> if(buff[2]) {
>    logger(buff, "info");
>    session.setVariable("effective_caller_id_name", buff[2]);
> }
>
> Or remove the print statement from cnam.pl.
>
> Sorry for the code, but the page was not editable.
>
> Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mike at jerris.com  Mon Aug 24 12:46:58 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 24 Aug 2009 15:46:58 -0400
Subject: [Freeswitch-users] Yet another question about A500 + FS
In-Reply-To: <8a9b664c0908222321r20c5c4d8wd08721dc8955f62c@mail.gmail.com>
References: <8a9b664c0908222321r20c5c4d8wd08721dc8955f62c@mail.gmail.com>
Message-ID: <56CFD144-94D0-42CE-B59D-7DC524C1919D@jerris.com>

Do you have an answer in the dialplan for that extension?  Also, check  
out the ignore_early_media variable.

Mike

On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote:

> Hi,
>
> I managed to get our A500 running with FreeSWITCH 1.0.4 stable using  
> wanpipe 3.4.4 drivers. But now I have another problem...
> I want to originate calls through event socket, and I only want to  
> receive ANSWERED(+OK) reply when the user actually answers.
>
> Now the situation is:
>
> ====================================
> originate openzap/1/a/123456 023
> 2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT:  
> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456] Ci= 
> [0000000000]
> 2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT  
> (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4
> 2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel  
> OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082]
> 2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer  
> OpenZAP/1:1/123456!
> API CALL [originate(openzap/1/a/123456 023)] output:
> +OK f8fca2be-8fa7-11de-9076-511e29dfc082
>
> 2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer  
> OpenZAP/1:1/123456 to XML[023 at default]
> freeswitch at emo-voip> 2009-08-23 08:44:06.743475 [INFO]  
> mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default
> 2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel  
> [OpenZAP/1:1/123456] has been answered
> 2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT  
> (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5
> 2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT  
> (N): CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6
> 2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup  
> OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT  
> (N): CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3
> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086  
> Session 2 (OpenZAP/1:1/123456) Ended
> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close  
> Channel OpenZAP/1:1/123456 [CS_DESTROY]
> ====================================
>
> Extension 023 is an IVR. As you can see FreeSWITCH answers the call  
> (2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel  
> [OpenZAP/1:1/123456] has been answered) 20 seconds before user  
> actually pick up the phone (2009-08-23 08:44:20.206010 [WARNING]  
> ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0  
> CSid=2 Seq=5).
>
> So Sangoma drivers/daemons report the events correctly.
> How can I set FreeSWITCH to answer after receiving RX EVENT (N):  
> CALL_ANSWERED from the driver?
>
> Thank you,
> V. Panayotov
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Mon Aug 24 12:50:57 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 14:50:57 -0500
Subject: [Freeswitch-users] Choppy voive in conference from fritzbox to
	freeswitch (ptime:30 msecs)
In-Reply-To: <D8BD24FC-9B07-4045-B6FD-A6D00156F877@jerris.com>
References: <4A8D5132.7010807@gmx.net> <4A8D5D43.3030300@gmx.net>
	<C4E1AFB8-12E3-45DE-ABF5-B1320B06D481@freeswitch.org>
	<4A8DB808.1030903@gmx.net>
	<87f2f3b90908201435r7adbdc83vd7240e0df8394292@mail.gmail.com>
	<4A8EBF77.1080204@gmx.net>
	<BD3DE8E0-3363-465C-985D-EC593AE45694@freeswitch.org>
	<191c3a030908211309j60c16b26h1e6445891136ae7f@mail.gmail.com>
	<4A91A480.7080307@gmx.net>
	<D8BD24FC-9B07-4045-B6FD-A6D00156F877@jerris.com>
Message-ID: <191c3a030908241250o5284e986hc9cfa6093495a42c@mail.gmail.com>

mr fritz is lying somewhere

get a pcap of the traffic from fritz to FS and look at the size of the audio
packets
if they are 160(172 with headers) bytes then it's 20ms if it's 240 (252)
then it's 30ms

if it's saying 20 but it means 30 you should leave the last change in place
and also add in

<param name="rtp-autofix-timing" value="true" />
and
<param name="inbound-codec-negotiation" value="scrooge"/>

to overcome the bug on their end.


On Mon, Aug 24, 2009 at 2:05 PM, Michael Jerris <mike at jerris.com> wrote:

> That is the remote sdp, not the local sdp.  They are sending ptime 20,
> not us.  Are they actually sending 20 ms packets or are they sending 30?
>
> MIke
>
> On Aug 23, 2009, at 4:20 PM, Peter P GMX wrote:
>
> > Hello Anthony,
> >
> > I set PCMA at 30i,PCMU at 30i and I can see in the logs that PCMA is used.
> > However ptime is set to 20 msec as shown in the Logs:
> >
> > 2009-08-23 22:11:29.530301 [DEBUG] sofia.c:3309 Remote SDP:
> > v=0
> > o=user 2075230 2075230 IN IP4 217.xx.xx.xxx
> > s=call
> > c=IN IP4 217.xx.xx.xxx
> > t=0 0
> > m=audio 7078 RTP/AVP 8 0 2 102 100 99 97 101
> > a=rtpmap:2 G726-32/8000
> > a=rtpmap:102 G726-32/8000
> > a=rtpmap:100 G726-40/8000
> > a=rtpmap:99 G726-24/8000
> > a=rtpmap:97 iLBC/8000
> > a=fmtp:97 mode=30
> > a=rtpmap:101 telephone-event/8000
> > a=fmtp:101 0-11
> > a=rtcp:7079
> >
> > 2009-08-23 22:11:29.530301 [DEBUG] switch_core_state_machine.c:404
> > (sofia/internal/02xxxxxxxxx at fs1.my.domain) State NEW
> > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec
> > Compare
> > [PCMA:8:8000:0]/[G722:9:8000:20]
> > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec
> > Compare
> > [PCMA:8:8000:0]/[PCMU:0:8000:20]
> > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3132 Audio Codec
> > Compare
> > [PCMA:8:8000:0]/[PCMA:8:8000:20]
> > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:2090 Set Codec
> > sofia/internal/02xxxxxxxxx at fs1.my.domain PCMA/8000 20 ms 160 samples
> > 2009-08-23 22:11:29.530301 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
> > payload to 101
> >
> > Sound from FS to Fritzbox is fine. Sound from Fritzbox to FS is
> > horrible.
> >
> > Best regards
> > Peter
> >
> > Anthony Minessale schrieb:
> >> try setting FS to 30ms too
> >>
> >> edit vars.xml and add @30i to everywhere you see  PCMU or PCMA so it
> >> looks like PCMU at 30i
> >>
> >> from:
> >>
> >>  <X-PRE-PROCESS cmd="set"
> >> data
> >> ="global_codec_prefs=G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM"/>
> >>  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/>
> >>
> >> to:
> >>
> >>  <X-PRE-PROCESS cmd="set"
> >> data
> >> =
> >> "global_codec_prefs
> >> =G7221 at 32000h,G7221 at 16000h,G722,PCMU at 30i,PCMA at 30i,GSM"/>
> >>  <X-PRE-PROCESS cmd="set"
> >> data="outbound_codec_prefs=PCMU at 30i,PCMA at 30i,GSM"/>
> >>
> >>
> >> On Fri, Aug 21, 2009 at 1:38 PM, Brian West <brian at freeswitch.org
> >> <mailto:brian at freeswitch.org>> wrote:
> >>
> >>    You can ship me one whois bkw.org <http://bkw.org>, I can add it
> >>    to my lab.
> >>
> >>    /b
> >>
> >>    On Aug 21, 2009, at 10:38 AM, Peter P GMX wrote:
> >>
> >>>
> >>> BTW: We can ship you a FritzBox if you need one for testing.
> >>
> >>
> >>    _______________________________________________
> >>    FreeSWITCH-users mailing list
> >>    FreeSWITCH-users at lists.freeswitch.org
> >>    <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >>    http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>    UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
> >> freeswitch-users
> >>    http://www.freeswitch.org
> >>
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> >> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> >> iax:guest at conference.freeswitch.org/888
> >> <http://iax:guest at conference.freeswitch.org/888>
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> >> pstn:213-799-1400
> >> ------------------------------------------------------------------------
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> >> users
> >> http://www.freeswitch.org
> >>
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jerry.richards at teotech.com  Mon Aug 24 13:25:59 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Mon, 24 Aug 2009 13:25:59 -0700
Subject: [Freeswitch-users] Cannot create outgoing channel type
	[error]cause: [FACILITY_NOT_SUBSCRIBED]
In-Reply-To: <C61B5EA8-345C-471E-BBA5-534A1726F396@freeswitch.org>
References: <16C552FB75474E8D82974FCF182B06D3@greyhawk.tonecommander.com>
	<C61B5EA8-345C-471E-BBA5-534A1726F396@freeswitch.org>
Message-ID: <12329F13B58D4DE2AC8AC895EBA94074@greyhawk.tonecommander.com>

Yes.  This a stand-alone Windows XP machine.

Jerry 

-----Original Message-----
From: Brian West [mailto:brian at freeswitch.org] 
Sent: Monday, August 24, 2009 12:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Cannot create outgoing channel type
[error]cause: [FACILITY_NOT_SUBSCRIBED]

Are you trying to test everything on the same machine?

/b

On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote:

> Hello All,
>
> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP 
> machine for the first time using the Getting Started Guide.  I can 
> register three lines (1000, 1001, and 1002), but when I attempt to 
> call one phone to the other I hear the operator say:
>
> "The person at extension 1000 is not available..."
>
> Also, the Freeswitch log shows:
>
> Cannot create outgoing channel type [error] cause:
> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user]
> cause:
> [FACILITY_NOT_SUBSCRIBED]
>
> Does anyone know why I get this error?
>
> Best Regards,
> Jerry







From brian at freeswitch.org  Mon Aug 24 13:29:09 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 15:29:09 -0500
Subject: [Freeswitch-users] Cannot create outgoing channel type
	[error]cause: [FACILITY_NOT_SUBSCRIBED]
In-Reply-To: <12329F13B58D4DE2AC8AC895EBA94074@greyhawk.tonecommander.com>
References: <16C552FB75474E8D82974FCF182B06D3@greyhawk.tonecommander.com>
	<C61B5EA8-345C-471E-BBA5-534A1726F396@freeswitch.org>
	<12329F13B58D4DE2AC8AC895EBA94074@greyhawk.tonecommander.com>
Message-ID: <E4FC66FB-73A5-48F0-BEC3-E9A71403B31B@freeswitch.org>

Make sure your x-lite is not on port 5060

/b

On Aug 24, 2009, at 3:25 PM, Jerry Richards wrote:

> Yes.  This a stand-alone Windows XP machine.
>
> Jerry




From tculjaga at gmail.com  Mon Aug 24 13:37:24 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 24 Aug 2009 22:37:24 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
Message-ID: <65d96fc80908241337m3c2ea93bv935a5b1cfc3f0c9a@mail.gmail.com>

Hello Brian,

it doesn't work .. tried this today as well:



freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
20:28:09.367300:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 INVITE
   Max-Forwards: 70
   Subject: Performance Test
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 328 bytes to udp/[10.4.4.252]:5060 at 20:28:09.367634:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.371759:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=ygQBtp6QpKtcD
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 401 bytes from udp/[10.4.4.252]:5060 at 20:28:09.371989:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7019-1-3;rport
   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=ygQBtp6QpKtcD
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 ACK
   Contact: sip:sipp at 10.4.4.252:5060
   Max-Forwards: 70
   Subject: Performance Test
   Content-Length: 0

   ------------------------------------------------------------------------
send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.873045:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>;tag=1
   To: "30003016094191500"
<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>;tag=ygQBtp6QpKtcD
   Call-ID: 1-7019 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0



This thing is driving me crazy, pls help.

T.



>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 14:15:40 -0500
> Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand
> ACK message
> In your scenario you need to add [peer_tag_param] at the end of the to on
> the Ack.
>
> /b
>
> On Aug 24, 2009, at 2:03 PM, Tihomir Culjaga wrote:
>
>
>>   ------------------------------------------------------------------------
>> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
>>   ------------------------------------------------------------------------
>>   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>>   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>>   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>
>> >
>>   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251<sip%3A22222238515000403 at 10.4.4.251>
>> >;tag=1
>>   Call-ID: 1-6962 at 10.4.4.252
>>   CSeq: 1 ACK
>>   Contact: sip:sipp at 10.4.4.252:5060
>>   Max-Forwards: 70
>>   Subject: Performance Test
>>   Content-Length: 0
>>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: "Jerry Richards" <jerry.richards at teotech.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 24 Aug 2009 12:24:42 -0700
> Subject: [Freeswitch-users] Cannot create outgoing channel type [error]
> cause: [FACILITY_NOT_SUBSCRIBED]
> Hello All,
>
> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP
> machine
> for the first time using the Getting Started Guide.  I can register three
> lines (1000, 1001, and 1002), but when I attempt to call one phone to the
> other I hear the operator say:
>
> "The person at extension 1000 is not available..."
>
> Also, the Freeswitch log shows:
>
> Cannot create outgoing channel type [error] cause:
> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause:
> [FACILITY_NOT_SUBSCRIBED]
>
> Does anyone know why I get this error?
>
> Best Regards,
> Jerry
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 14:33:22 -0500
> Subject: Re: [Freeswitch-users] Cannot create outgoing channel type [error]
> cause: [FACILITY_NOT_SUBSCRIBED]
> Are you trying to test everything on the same machine?
>
> /b
>
> On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote:
>
>  Hello All,
>>
>> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP
>> machine
>> for the first time using the Getting Started Guide.  I can register three
>> lines (1000, 1001, and 1002), but when I attempt to call one phone to the
>> other I hear the operator say:
>>
>> "The person at extension 1000 is not available..."
>>
>> Also, the Freeswitch log shows:
>>
>> Cannot create outgoing channel type [error] cause:
>> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause:
>> [FACILITY_NOT_SUBSCRIBED]
>>
>> Does anyone know why I get this error?
>>
>> Best Regards,
>> Jerry
>>
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 15:44:18 -0400
> Subject: Re: [Freeswitch-users] Problem with cnam.js?
> Every page on the wiki should be editable.  If you don't already have an
> account, go to:
> http://wiki.freeswitch.org/index.php?title=Special:UserLogin&type=signup
>
> Mike
>
> On Aug 22, 2009, at 12:42 PM, Lars Zeb wrote:
>
> I think there?s something wrong with the script at
> http://wiki.freeswitch.org/wiki/Examples_cnam.js.
>
> If you use it as is, it displays ?Content-type: text/html? for the
> effective_caller_id_name. In cnam.pl, the first two output lines are
> generated by:
>
> if (!$debug) {print "Content-type: text/html\n\n";}
>
> with the actual name in the third line.
>
> So I changed:
>
> fd.open("read");
> buff = fd.readln();
>
> if(buff) {
>    logger(buff, "info");
>    session.setVariable("effective_caller_id_name", buff);
> }
>
> To:
>
> fd.open("read");
> buff = fd.readAll();
>
> if(buff[2]) {
>    logger(buff, "info");
>    session.setVariable("effective_caller_id_name", buff[2]);
> }
>
> Or remove the print statement from cnam.pl.
>
> Sorry for the code, but the page was not editable.
>
> Lars
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 15:46:58 -0400
> Subject: Re: [Freeswitch-users] Yet another question about A500 + FS
> Do you have an answer in the dialplan for that extension?  Also, check out
> the ignore_early_media variable.
>
> Mike
>
> On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote:
>
>  Hi,
>>
>> I managed to get our A500 running with FreeSWITCH 1.0.4 stable using
>> wanpipe 3.4.4 drivers. But now I have another problem...
>> I want to originate calls through event socket, and I only want to receive
>> ANSWERED(+OK) reply when the user actually answers.
>>
>> Now the situation is:
>>
>> ====================================
>> originate openzap/1/a/123456 023
>> 2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT:
>> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456]
>> Ci=[0000000000]
>> 2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
>> CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4
>> 2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel
>> OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082]
>> 2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer
>> OpenZAP/1:1/123456!
>> API CALL [originate(openzap/1/a/123456 023)] output:
>> +OK f8fca2be-8fa7-11de-9076-511e29dfc082
>>
>> 2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer
>> OpenZAP/1:1/123456 to XML[023 at default]
>> freeswitch at emo-voip> 2009-08-23 08:44:06.743475 [INFO]
>> mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default
>> 2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
>> [OpenZAP/1:1/123456] has been answered
>> 2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
>> CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5
>> 2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N):
>> CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6
>> 2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup
>> OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT (N):
>> CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3
>> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086 Session 2
>> (OpenZAP/1:1/123456) Ended
>> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close
>> Channel OpenZAP/1:1/123456 [CS_DESTROY]
>> ====================================
>>
>> Extension 023 is an IVR. As you can see FreeSWITCH answers the call
>> (2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
>> [OpenZAP/1:1/123456] has been answered) 20 seconds before user actually pick
>> up the phone (2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX
>> EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5).
>>
>> So Sangoma drivers/daemons report the events correctly.
>> How can I set FreeSWITCH to answer after receiving RX EVENT (N):
>> CALL_ANSWERED from the driver?
>>
>> Thank you,
>> V. Panayotov
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dave at 3c.co.uk  Mon Aug 24 13:41:23 2009
From: dave at 3c.co.uk (David Knell)
Date: Mon, 24 Aug 2009 23:41:23 +0300
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
 ACK message
In-Reply-To: <65d96fc80908241203m29abc2c6t7678689a3402c31f@mail.gmail.com>
References: <65d96fc80908241203m29abc2c6t7678689a3402c31f@mail.gmail.com>
Message-ID: <1251146483.16312.36.camel@dk-d820>

Hi Tihomir -

I'm no SIP guru, but the things which look suspicious about the ACK to
me are:
- Via header - different branch
- Contact header - differs from INVITE

--Dave

> Hi Anthony,
> 
> I'm aware it is generating 30 retries per a call and this is killing
> me ...
> 
> I lost my entire working day to figure out what is missing in the damn
> ACK message SIPp is sending back... ACK looks quite ok to me.
> 
> pls can you help ?
> 
> 
> freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at
> 16:44:26.527236:
>    ------------------------------
> ------------------------------------------
>    INVITE sip:30003016094191500 at 10.4.4.251 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
>    Max-Forwards: 70
>    Contact: <sip:22222238515000403 at 10.4.4.252>
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Type: application/sdp
>    Content-Length:   131
>    
>    v=0
>    o=user1 53655765 2353687637 IN IP4 10.4.4.252
>    s=-
>    c=IN IP4 10.4.4.252
>    t=0 0
>    m=audio 6000 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
> 
> ------------------------------------------------------------------------
> send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566:
> 
> ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Content-Length: 0
>    
> 
> ------------------------------------------------------------------------
> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582:
> 
> ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    To: "30003016094191500"
> <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,
> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
> 
> 
> 
> ------------------------------------------------------------------------
> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
> 
> ------------------------------------------------------------------------
>    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    Call-ID: 1-6962 at 10.4.4.252
>    CSeq: 1 ACK
>    Contact: sip:sipp at 10.4.4.252:5060
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Length: 0
> 
> 
> 
> 
> 
> What m'i missing ?
> 
> 
> 
>         
>         Your ACK message must not be valid (dialog matching or
>         something else)
>         so every 1 call will generate 30 retries that are queued up in
>         the sip stack.
>         
>         at 100cps you will be generating this problem 100 times per
>         second and queue up countless unfinished dialogs thus
>         eating up the cpu.
>         
>         
>         
>         
>         
>         
>         On Mon, Aug 24, 2009 at 12:19 PM, Tihomir Culjaga
>         <tculjaga at gmail.com> wrote:
>                 Hello,
>                 
>                 I've been with freeswittch for a while now.. and i can
>                 say it is worth developing it.
>                 
>                 anyhow i got into a strange issue... I'm tryng to see
>                 what load FS on my server can take. The Call flow is
>                 like this:
>                 
>                 SIPp                   FS
>                 
>                 INVITE --------> 
>                            <------- 100 Trying
>                            <------- 302 Moved Temporary
>                 ACK    --------->
>                 
>                 
>                 
>                 I use a dummy dialplan for that. All custom functions
>                 i've build are disabled and i'm not using it here.
>                 Also custom modules are not loaded as well.
>                 
>                 
>                    <extension name="ServiceLookup">
>                       <condition field="destination_number"
>                 expression="(^300030)(.*)">
>                          <!--action
>                 application="lookup_service_destination" data="in
>                 ${caller_id_number:6:16}, in ${caller_id_number:0:6},
>                 in $2, i
>                 n $1, in pgw01.ot.hr:5060, out red_contact, out
>                 authResult"/-->
>                          <action application="log" data="INFO
>                 ######################## ServiceLookup
>                 ########################\n"/>
>                          <action application="log" data="INFO
>                 ######################## contact = '${red_contact}'
>                 ##############\n"/>
>                          <action application="log" data="INFO
>                 ######################## CallerNum =
>                 '${caller_id_number:6:16}' ##########\n"/>
>                          <action application="log" data="INFO
>                 ######################## RADIUS auth = '${authResult}'
>                 ##########\n"/>
>                          <action application="execute_extension"
>                 data="doRedirect XML public"/>
>                         </condition>
>                    </extension>
>                 
>                 
>                    <extension name="doRedirect">
>                       <condition field="destination_number"
>                 expression="^doRedirect$"/>
>                       <condition field="${authResult}" expression="^0
>                 $|^60$">
>                          <action application="log" data="INFO
>                 ######################## RADIUS auth OK!!!' ##########
>                 \n"/>
>                          <!--action application="redirect" data="sip:
>                 ${red_contact}"/-->
>                          <!--action application="answer"/-->
>                          <action application="redirect"
>                 data="sip:12345616094191500 at pgw01.ot.hr:5060"/>
>                          <!--anti-action application="answer"/-->
>                          <!--anti-action application="sleep"
>                 data="2000"/-->
>                          <action application="hangup"
>                 data="USER_BUSY"/>
>                          <anti-action application="redirect"
>                 data="sip:12345616094191500 at pgw01.ot.hr:5060"/>
>                          <anti-action application="log" data="INFO
>                 ######################## RADIUS auth NOK!! ##########
>                 \n"/>
>                          <!--anti-action application="respond"
>                 data="403 Forbidden"/-->
>                          <anti-action application="hangup"
>                 data="USER_BUSY"/>
>                       </condition>
>                    </extension>
>                 
>                 
>                 When i place a call from x-lite everything works
>                 fine ... x-lite sends an invite, gets SIP 302 and ACKs
>                 it correctly... FS is happy.
>                 
>                 When i place a call from SIPp i have the same scenario
>                 except FS seems not understand ACK message from SIPp
>                 and re-sends SIP 302 multiple times untill it gives
>                 up.
>                 
>                 
>                 I beleive this is due to 302 resend issue but; when i
>                 load FS with 100 CPS, i can see high CPU usage (just
>                 one thread taking most load... the rest does almost
>                 nothing) on FS. Also, starting from 40 CPS there is a
>                 big delay in receiving SIP 302 messages meaning i've
>                 sent 6000 calls and so far only for half of them got
>                 302 response.
>                 
>                 
>                 Does anybody have a clue ?
>                 
>                 
>                 
>                 
>                 
>                 Here is a trace taken on FS for calls originated from
>                 SIPp (sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
>                 30003016094191500 -trace_err -r 1 -rp 100 -trace_msg
>                 -inf test.txt -m 1 -l 4000):
>                 
>                 freeswitch at l01sipindir1> recv 573 bytes from
>                 udp/[10.4.4.252]:5060 at 16:44:26.527236:
>                 
>                 ------------------------------------------------------------------------
>                    INVITE sip:30003016094191500 at 10.4.4.251 SIP/2.0
>                    Via: SIP/2.0/UDP
>                 10.4.4.252;branch=z9hG4bK-6962-1-0;rport
>                    Max-Forwards: 70
>                    Contact: <sip:22222238515000403 at 10.4.4.252>
>                    To:
>                 "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>                    From:
>                 "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>                    Call-ID: 1-6962 at 10.4.4.252
>                    CSeq: 1 INVITE
>                    Max-Forwards: 70
>                    Subject: Performance Test
>                    Content-Type: application/sdp
>                    Content-Length:   131
>                    
>                    v=0
>                    o=user1 53655765 2353687637 IN IP4 10.4.4.252
>                    s=-
>                    c=IN IP4 10.4.4.252
>                    t=0 0
>                    m=audio 6000 RTP/AVP 0
>                    a=rtpmap:0 PCMU/8000
>                 
>                 ------------------------------------------------------------------------
>                 send 328 bytes to udp/[10.4.4.252]:5060 at
>                 16:44:26.527566:
>                 
>                 ------------------------------------------------------------------------
>                    SIP/2.0 100 Trying
>                    Via: SIP/2.0/UDP
>                 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>                    From:
>                 "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>                    To:
>                 "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>                    Call-ID: 1-6962 at 10.4.4.252
>                    CSeq: 1 INVITE
>                    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>                    Content-Length: 0
>                    
>                 
>                 ------------------------------------------------------------------------
>                 send 722 bytes to udp/[10.4.4.252]:5060 at
>                 16:44:26.535582:
>                 
>                 ------------------------------------------------------------------------
>                    SIP/2.0 302 Moved Temporarily
>                    Via: SIP/2.0/UDP
>                 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>                    From:
>                 "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>                    To: "30003016094191500"
>                 <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
>                    Call-ID: 1-6962 at 10.4.4.252
>                    CSeq: 1 INVITE
>                    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>                    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>                    Accept: application/sdp
>                    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
>                 MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER,
>                 INFO, PUBLISH
>                    Supported: timer, precondition, path, replaces
>                    Allow-Events: talk, presence, dialog, call-info,
>                 sla, include-session-description, presence.winfo,
>                 message-summary, refer
>                    Content-Length: 0
>                    
>                 
>                 ------------------------------------------------------------------------
>                 recv 383 bytes from udp/[10.4.4.252]:5060 at
>                 16:44:26.535809:
>                 
>                 ------------------------------------------------------------------------
>                    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>                    Via: SIP/2.0/UDP
>                 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>                    To:
>                 "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>                    From:
>                 "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>                    Call-ID: 1-6962 at 10.4.4.252
>                    CSeq: 1 ACK
>                    Contact: sip:sipp at 10.4.4.252:5060
>                    Max-Forwards: 70
>                    Subject: Performance Test
>                    Content-Length: 0
>                    
>                 
>                 ------------------------------------------------------------------------
>                 send 722 bytes to udp/[10.4.4.252]:5060 at
>                 16:44:27.037070:
>                 
>                 ------------------------------------------------------------------------
>                    SIP/2.0 302 Moved Temporarily
>                    Via: SIP/2.0/UDP
>                 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>                    From:
>                 "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>                    To: "30003016094191500"
>                 <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
>                    Call-ID: 1-6962 at 10.4.4.252
>                    CSeq: 1 INVITE
>                    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>                    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>                    Accept: application/sdp
>                    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
>                 MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER,
>                 INFO, PUBLISH
>                    Supported: timer, precondition, path, replaces
>                    Allow-Events: talk, presence, dialog, call-info,
>                 sla, include-session-description, presence.winfo,
>                 message-summary, refer
>                    Content-Length: 0
>                    
>                 
>                 ------------------------------------------------------------------------
>                 send 722 bytes to udp/[10.4.4.252]:5060 at
>                 16:44:28.037063:
>                 
>                 ------------------------------------------------------------------------
>                    SIP/2.0 302 Moved Temporarily
>                    Via: SIP/2.0/UDP
>                 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060
>                    From:
>                 "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>                    To: "30003016094191500"
>                 <sip:30003016094191500 at 10.4.4.251>;tag=Hr4mHDUeBSNyH
>                    Call-ID: 1-6962 at 10.4.4.252
>                    CSeq: 1 INVITE
>                    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>                    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>                    Accept: application/sdp
>                    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK,
>                 MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER,
>                 INFO, PUBLISH
>                    Supported: timer, precondition, path, replaces
>                    Allow-Events: talk, presence, dialog, call-info,
>                 sla, include-session-description, presence.winfo,
>                 message-summary, refer
>                    Content-Length: 0
>                 
>                 
>                 Tihomir.
>                 
>                 
>                 
>                 
>                 
>                 
>                 _______________________________________________
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at lists.freeswitch.org
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 http://www.freeswitch.org
>                 
>         
>         
>         
>         -- 
>         Anthony Minessale II
>         
>         FreeSWITCH http://www.freeswitch.org/
>         ClueCon http://www.cluecon.com/
>         Twitter: http://twitter.com/FreeSWITCH_wire
>         
>         AIM: anthm
>         MSN:anthony_minessale at hotmail.com
>         GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>         IRC: irc.freenode.net #freeswitch
>         
>         FreeSWITCH Developer Conference
>         sip:888 at conference.freeswitch.org
>         iax:guest at conference.freeswitch.org/888
>         googletalk:conf+888 at conference.freeswitch.org
>         pstn:213-799-1400
>         
>         
>         ---------- Forwarded message ----------
>         From: "Raffaele P. Guidi" <raffaele.p.guidi at gmail.com>
>         To: freeswitch-users at lists.freeswitch.org
>         Date: Mon, 24 Aug 2009 20:24:28 +0200
>         Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows
>         installer - great but I have a little problem
>         Actually I did that and it worked fine. I had the problem the
>         SECOND time I run FS and freepbx. And (@Brian) mod_sofia was
>         loaded but sip_profiles were not
>         
>         On Sun, Aug 16, 2009 at 16:04, Carlos Talbot
>         <carlos.talbot at gmail.com> wrote:
>                 When you configure FreePBX for the first time it wipes
>                 out the sip_profiles directory. If you follow the
>                 FreePBX shortcut on your desktop it'll mention this on
>                 the last screen of the configuration. You might see
>                 something such as the following below. If you plan to
>                 use FreePBX you'll need to define trunk groups,
>                 trunks, etc in order to have the sip_profiles
>                 directory populated.
>                 
>                 
>                 regards,
>                 
>                 
>                 Carlos
>                 
>                 
>                 
>                 
>                               Incompatible Configuration
>                     WARNING: THE FOLLOWING FILES WILL BE DELETED!
>                 
>                       * D:/FreeSWITCH/conf/sip_profiles/external.xml
>                       * D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
>                       * D:/FreeSWITCH/conf/sip_profiles/internal.xml
>                 
>                 
>                 On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi
>                 <raffaele.p.guidi at gmail.com> wrote:
>                 
>                         
>                         I had the sweet surprise to find the installer
>                         packaged with FreePBX... really great! Why it
>                         has not been advertised as it deserves? It
>                         worked like a breeze once launched, with the
>                         automatic configuration and all of that., Only
>                         thing: once stopped I cannot get it to load
>                         sofia profiles anymore - issueing sofia status
>                         doesn't show anything. I had to copy
>                         internal.xml and default.xml from a previous
>                         installation and everything got to work again
>                         - but no FreePBX anymore :( I'm sure I'm
>                         missing something important.
>                         
>                         
>                         Can you give me a hint? Should sofia profiles
>                         be served by curl or something?
>                         
>                         
>                         Thanks,
>                            Raffaele
>                         
>                         
>                         _______________________________________________
>                         FreeSWITCH-users mailing list
>                         FreeSWITCH-users at lists.freeswitch.org
>                         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                         http://www.freeswitch.org
>                         
>                 
>                 
>                 
>                 _______________________________________________
>                 FreeSWITCH-users mailing list
>                 FreeSWITCH-users at lists.freeswitch.org
>                 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>                 UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>                 http://www.freeswitch.org
>                 
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk




From brian at freeswitch.org  Mon Aug 24 13:42:31 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 15:42:31 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241337m3c2ea93bv935a5b1cfc3f0c9a@mail.gmail.com>
References: <65d96fc80908241337m3c2ea93bv935a5b1cfc3f0c9a@mail.gmail.com>
Message-ID: <FEB84EFF-305E-4964-BBAB-2C95B2EAB865@freeswitch.org>

ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

Use that.. your scenario has some hard coded IP's in the fields that  
shouldn't be there.

/b

On Aug 24, 2009, at 3:37 PM, Tihomir Culjaga wrote:

> Hello Brian,
>
> it doesn't work .. tried this today as well:
>
>
>
> freeswitch at l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060  
> at 20:28:09.367300:
>     
> ------------------------------------------------------------------------
>    INVITE sip:30003016094191500 at 10.4.4.251 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport
>    Max-Forwards: 70
>    Contact: <sip:22222238515000403 at 10.4.4.252>
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    Call-ID: 1-7019 at 10.4.4.252
>    CSeq: 1 INVITE
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Type: application/sdp
>    Content-Length:   131
>
>    v=0
>    o=user1 53655765 2353687637 IN IP4 10.4.4.252
>    s=-
>    c=IN IP4 10.4.4.252
>    t=0 0
>    m=audio 6000 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
>     
> ------------------------------------------------------------------------
> send 328 bytes to udp/[10.4.4.252]:5060 at 20:28:09.367634:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>    Call-ID: 1-7019 at 10.4.4.252
>    CSeq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
> send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.371759:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    To: "30003016094191500" <sip: 
> 30003016094191500 at 10.4.4.251>;tag=ygQBtp6QpKtcD
>    Call-ID: 1-7019 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla, include- 
> session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
> recv 401 bytes from udp/[10.4.4.252]:5060 at 20:28:09.371989:
>     
> ------------------------------------------------------------------------
>    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7019-1-3;rport
>    To: "30003016094191500"<sip: 
> 30003016094191500 at 10.4.4.251>;tag=ygQBtp6QpKtcD
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    Call-ID: 1-7019 at 10.4.4.252
>    CSeq: 1 ACK
>    Contact: sip:sipp at 10.4.4.252:5060
>    Max-Forwards: 70
>    Subject: Performance Test
>    Content-Length: 0
>
>     
> ------------------------------------------------------------------------
> send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.873045:
>     
> ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060
>    From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>    To: "30003016094191500" <sip: 
> 30003016094191500 at 10.4.4.251>;tag=ygQBtp6QpKtcD
>    Call-ID: 1-7019 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE,  
> SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla, include- 
> session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>
>
> This thing is driving me crazy, pls help.
>
> T.
>
>
>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 14:15:40 -0500
> Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't  
> understand ACK message
> In your scenario you need to add [peer_tag_param] at the end of the  
> to on the Ack.
>
> /b
>
> On Aug 24, 2009, at 2:03 PM, Tihomir Culjaga wrote:
>
>
>    
> ------------------------------------------------------------------------
> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809:
>    
> ------------------------------------------------------------------------
>   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport
>   To: "30003016094191500"<sip:30003016094191500 at 10.4.4.251>
>   From: "22222238515000403"<sip:22222238515000403 at 10.4.4.251>;tag=1
>   Call-ID: 1-6962 at 10.4.4.252
>   CSeq: 1 ACK
>   Contact: sip:sipp at 10.4.4.252:5060
>   Max-Forwards: 70
>   Subject: Performance Test
>   Content-Length: 0
>
>
>
>
>
> ---------- Forwarded message ----------
> From: "Jerry Richards" <jerry.richards at teotech.com>
> To: <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 24 Aug 2009 12:24:42 -0700
> Subject: [Freeswitch-users] Cannot create outgoing channel type  
> [error] cause: [FACILITY_NOT_SUBSCRIBED]
> Hello All,
>
> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP  
> machine
> for the first time using the Getting Started Guide.  I can register  
> three
> lines (1000, 1001, and 1002), but when I attempt to call one phone  
> to the
> other I hear the operator say:
>
> "The person at extension 1000 is not available..."
>
> Also, the Freeswitch log shows:
>
> Cannot create outgoing channel type [error] cause:
> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user]  
> cause:
> [FACILITY_NOT_SUBSCRIBED]
>
> Does anyone know why I get this error?
>
> Best Regards,
> Jerry
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Brian West <brian at freeswitch.org>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 14:33:22 -0500
> Subject: Re: [Freeswitch-users] Cannot create outgoing channel type  
> [error] cause: [FACILITY_NOT_SUBSCRIBED]
> Are you trying to test everything on the same machine?
>
> /b
>
> On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote:
>
> Hello All,
>
> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP  
> machine
> for the first time using the Getting Started Guide.  I can register  
> three
> lines (1000, 1001, and 1002), but when I attempt to call one phone  
> to the
> other I hear the operator say:
>
> "The person at extension 1000 is not available..."
>
> Also, the Freeswitch log shows:
>
> Cannot create outgoing channel type [error] cause:
> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user]  
> cause:
> [FACILITY_NOT_SUBSCRIBED]
>
> Does anyone know why I get this error?
>
> Best Regards,
> Jerry
>
>
>
>
>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 15:44:18 -0400
> Subject: Re: [Freeswitch-users] Problem with cnam.js?
> Every page on the wiki should be editable.  If you don't already  
> have an account, go to:
>
> http://wiki.freeswitch.org/index.php?title=Special:UserLogin&type=signup
>
> Mike
>
> On Aug 22, 2009, at 12:42 PM, Lars Zeb wrote:
>
>> I think there?s something wrong with the script at http://wiki.freeswitch.org/wiki/Examples_cnam.js 
>> .
>>
>> If you use it as is, it displays ?Content-type: text/html? for the  
>> effective_caller_id_name. In cnam.pl, the first two output lines  
>> are generated by:
>>
>> if (!$debug) {print "Content-type: text/html\n\n";}
>>
>> with the actual name in the third line.
>>
>> So I changed:
>>
>> fd.open("read");
>> buff = fd.readln();
>>
>> if(buff) {
>>    logger(buff, "info");
>>    session.setVariable("effective_caller_id_name", buff);
>> }
>>
>> To:
>>
>> fd.open("read");
>> buff = fd.readAll();
>>
>> if(buff[2]) {
>>    logger(buff, "info");
>>    session.setVariable("effective_caller_id_name", buff[2]);
>> }
>>
>> Or remove the print statement from cnam.pl.
>>
>> Sorry for the code, but the page was not editable.
>>
>> Lars
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
>
>
> ---------- Forwarded message ----------
> From: Michael Jerris <mike at jerris.com>
> To: freeswitch-users at lists.freeswitch.org
> Date: Mon, 24 Aug 2009 15:46:58 -0400
> Subject: Re: [Freeswitch-users] Yet another question about A500 + FS
> Do you have an answer in the dialplan for that extension?  Also,  
> check out the ignore_early_media variable.
>
> Mike
>
> On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote:
>
> Hi,
>
> I managed to get our A500 running with FreeSWITCH 1.0.4 stable using  
> wanpipe 3.4.4 drivers. But now I have another problem...
> I want to originate calls through event socket, and I only want to  
> receive ANSWERED(+OK) reply when the user actually answers.
>
> Now the situation is:
>
> ====================================
> originate openzap/1/a/123456 023
> 2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT:  
> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456] Ci= 
> [0000000000]
> 2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT  
> (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4
> 2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel  
> OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082]
> 2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer  
> OpenZAP/1:1/123456!
> API CALL [originate(openzap/1/a/123456 023)] output:
> +OK f8fca2be-8fa7-11de-9076-511e29dfc082
>
> 2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer  
> OpenZAP/1:1/123456 to XML[023 at default]
> freeswitch at emo-voip> 2009-08-23 08:44:06.743475 [INFO]  
> mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default
> 2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel  
> [OpenZAP/1:1/123456] has been answered
> 2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT  
> (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5
> 2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT  
> (N): CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6
> 2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup  
> OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT  
> (N): CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3
> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086  
> Session 2 (OpenZAP/1:1/123456) Ended
> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close  
> Channel OpenZAP/1:1/123456 [CS_DESTROY]
> ====================================
>
> Extension 023 is an IVR. As you can see FreeSWITCH answers the call  
> (2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel  
> [OpenZAP/1:1/123456] has been answered) 20 seconds before user  
> actually pick up the phone (2009-08-23 08:44:20.206010 [WARNING]  
> ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0  
> CSid=2 Seq=5).
>
> So Sangoma drivers/daemons report the events correctly.
> How can I set FreeSWITCH to answer after receiving RX EVENT (N):  
> CALL_ANSWERED from the driver?
>
> Thank you,
> V. Panayotov
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From MPeace at edcogroupinc.com  Mon Aug 24 13:42:10 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Mon, 24 Aug 2009 15:42:10 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>


I just installed FS on a fresh Centos 5.3 install, everything went perfect. I loaded x0lite up on a couple of hosts and they registered and worked fine. Then I moved the Centos server to a new subnet on out network that we have tons of boxes on and have all the routes already configured on and now the Sip phones will no longer register. No other changes have been made on the FreeSwitch server other than changing the IP.  BTW I can ping to and from the FS server and the hosts and the firewall is off on the FS server and the hosts.

I plugged a laptop with X-lite into a switch on the same network so the packets don't even need to route and it still won't register!

BTW  I also downloaded 3CX VoIP phone and it behaves the same way.

Any ideas?

TIA

=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From brian at freeswitch.org  Mon Aug 24 13:45:30 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 24 Aug 2009 15:45:30 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
Message-ID: <8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>

Did you happen to hard code an IP in your vars.xml?

/b

On Aug 24, 2009, at 3:42 PM, Mike Peace wrote:

>
> I just installed FS on a fresh Centos 5.3 install, everything went  
> perfect. I loaded x0lite up on a couple of hosts and they registered  
> and worked fine. Then I moved the Centos server to a new subnet on  
> out network that we have tons of boxes on and have all the routes  
> already configured on and now the Sip phones will no longer  
> register. No other changes have been made on the FreeSwitch server  
> other than changing the IP.  BTW I can ping to and from the FS  
> server and the hosts and the firewall is off on the FS server and  
> the hosts.
>
> I plugged a laptop with X-lite into a switch on the same network so  
> the packets don?t even need to route and it still won?t register!
>
> BTW  I also downloaded 3CX VoIP phone and it behaves the same way.
>
> Any ideas?
>
> TIA
>

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From javieraristizabal at gmail.com  Mon Aug 24 13:46:09 2009
From: javieraristizabal at gmail.com (=?ISO-8859-1?Q?Javier_Aristiz=E1bal?=)
Date: Mon, 24 Aug 2009 15:46:09 -0500
Subject: [Freeswitch-users] pass-thru issue (G729)
Message-ID: <b6f11f300908241346k7526b910rf21a7cc30db363fa@mail.gmail.com>

Hello All,
I'm having an issue with g.729 pass-thru. On an older 1.0.3 install it's all
good, but on a newer machine, more cores, etc. the quality of voice on g.729
is very poor. Anyone here dealt with something like this?

With all other codecs are perfect, even in pass-thru...


Thanks in ddvance
Javier
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From matt at venturevoip.com  Mon Aug 24 13:47:42 2009
From: matt at venturevoip.com (Matt Riddell)
Date: Tue, 25 Aug 2009 08:47:42 +1200
Subject: [Freeswitch-users] Couple of questions
In-Reply-To: <191c3a030908240705m5003a26bpd97b2c041d7b1526@mail.gmail.com>
References: <4A91D07C.1080805@venturevoip.com>	<6A2860CD-3166-4C9E-8649-C39799CA34EB@freeswitch.org>	<5a8712120908232127n6cb2f0bdq7e05b3d7778f0e70@mail.gmail.com>	<4A922E17.2040401@venturevoip.com>
	<191c3a030908240705m5003a26bpd97b2c041d7b1526@mail.gmail.com>
Message-ID: <4A92FC6E.5050005@venturevoip.com>

On 25/08/09 2:05 AM, Anthony Minessale wrote:
> I updated testclient.c so you can see how now.

Awesome awesome awesome!

Thanks heaps man - everybody here has been great!

-- 
Cheers,

Matt Riddell
Director
_______________________________________________

http://www.venturevoip.com/news.php (Daily Asterisk News)
http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer)
http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)



From MPeace at edcogroupinc.com  Mon Aug 24 14:01:00 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Mon, 24 Aug 2009 16:01:00 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>

I haven't changed any of the conf files.

Mike Peace
Network Analyst
EDCO, The Document People
Direct 417-447-3367

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West
Sent: Monday, August 24, 2009 3:46 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

Did you happen to hard code an IP in your vars.xml?

/b

On Aug 24, 2009, at 3:42 PM, Mike Peace wrote:



I just installed FS on a fresh Centos 5.3 install, everything went perfect. I loaded x0lite up on a couple of hosts and they registered and worked fine. Then I moved the Centos server to a new subnet on out network that we have tons of boxes on and have all the routes already configured on and now the Sip phones will no longer register. No other changes have been made on the FreeSwitch server other than changing the IP.  BTW I can ping to and from the FS server and the hosts and the firewall is off on the FS server and the hosts.

I plugged a laptop with X-lite into a switch on the same network so the packets don't even need to route and it still won't register!

BTW  I also downloaded 3CX VoIP phone and it behaves the same way.

Any ideas?

TIA



=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From Prometheus001 at gmx.net  Mon Aug 24 14:17:15 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 24 Aug 2009 23:17:15 +0200
Subject: [Freeswitch-users] XML-RPC on different ip than 0.0.0.0
In-Reply-To: <191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
Message-ID: <4A93035B.5010907@gmx.net>

Any clue which one? I could not identify it, also looked into the sources.

Best regards
Peter

Anthony Minessale schrieb:
> there is a configuration option in the xml file to control which ip it
> binds to.
>
>
>
> On Mon, Aug 24, 2009 at 6:25 AM, Peter P GMX <Prometheus001 at gmx.net
> <mailto:Prometheus001 at gmx.net>> wrote:
>
>     Hello,
>
>     is there any chance to limit the listening ips of the xml-rpc server
>     (which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)?
>
>     Best regards
>     Peter
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From msc at freeswitch.org  Mon Aug 24 14:18:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Aug 2009 14:18:10 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
Message-ID: <87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>

On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com> wrote:

>  I haven?t changed any of the conf files.
>
>
What happens when you try to register? Do you get 480? (timeout) Or
something else?
-MC
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From MPeace at edcogroupinc.com  Mon Aug 24 14:26:46 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Mon, 24 Aug 2009 16:26:46 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>

"Registration error: 408-Request Timeout"

Mike Peace
Network Analyst
EDCO, The Document People
Direct 417-447-3367

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:18 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

I haven't changed any of the conf files.

What happens when you try to register? Do you get 480? (timeout) Or something else?
-MC

=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From msc at freeswitch.org  Mon Aug 24 14:34:29 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Aug 2009 14:34:29 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
Message-ID: <87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>

On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com> wrote:

>  ?Registration error: 408-Request Timeout?
>
Sorry for the the typo - 408 = timeout. (480 = temp unavail)

Try stopping iptables in Linux and try again. Sounds like something is
interfering with your packets getting from here to there...
Try:

/etc/init.d/iptables stop

And then see if your packets can move again.

-MC

>
>
> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:18 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> I haven?t changed any of the conf files.
>
>
>
> What happens when you try to register? Do you get 480? (timeout) Or
> something else?
> -MC
>
> ------------------------------
>
>  *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>  Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Aug 24 14:41:05 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 16:41:05 -0500
Subject: [Freeswitch-users] XML-RPC on different ip than 0.0.0.0
In-Reply-To: <4A93035B.5010907@gmx.net>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
Message-ID: <191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>

oh yeah, it's only the port they let you pick

it looks like they don't let you pick ip in the abyss code, it would require
an intrusive patch into that depend lib
to allow you to set that.


On Mon, Aug 24, 2009 at 4:17 PM, Peter P GMX <Prometheus001 at gmx.net> wrote:

> Any clue which one? I could not identify it, also looked into the sources.
>
> Best regards
> Peter
>
> Anthony Minessale schrieb:
> > there is a configuration option in the xml file to control which ip it
> > binds to.
> >
> >
> >
> > On Mon, Aug 24, 2009 at 6:25 AM, Peter P GMX <Prometheus001 at gmx.net
> > <mailto:Prometheus001 at gmx.net>> wrote:
> >
> >     Hello,
> >
> >     is there any chance to limit the listening ips of the xml-rpc server
> >     (which is currently 0.0.0.0) to another one (e.g. 127.0.0.1)?
> >
> >     Best regards
> >     Peter
> >
> >     _______________________________________________
> >     FreeSWITCH-users mailing list
> >     FreeSWITCH-users at lists.freeswitch.org
> >     <mailto:FreeSWITCH-users at lists.freeswitch.org>
> >     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >     UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >     http://www.freeswitch.org
> >
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > <mailto:MSN%3Aanthony_minessale at hotmail.com<MSN%253Aanthony_minessale at hotmail.com>
> >
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > <mailto:PAYPAL%3Aanthony.minessale at gmail.com<PAYPAL%253Aanthony.minessale at gmail.com>
> >
> > IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > <mailto:sip%3A888 at conference.freeswitch.org<sip%253A888 at conference.freeswitch.org>
> >
> > iax:guest at conference.freeswitch.org/888
> > <http://iax:guest at conference.freeswitch.org/888>
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org<googletalk%253Aconf%252B888 at conference.freeswitch.org>
> >
> > pstn:213-799-1400
> > ------------------------------------------------------------------------
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From MPeace at edcogroupinc.com  Mon Aug 24 14:49:37 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Mon, 24 Aug 2009 16:49:37 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>

It does the same thing, does anything get set during the install that would remember or cache the old network settings? I can access anything from the FS server on any of several networks and vice-versa but the SIP will not register, again no firewalls are upon any of the test hosts.
Doesn't make sense to me.

Mike Peace
Network Analyst
EDCO, The Document People
Direct 417-447-3367

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:34 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

"Registration error: 408-Request Timeout"
Sorry for the the typo - 408 = timeout. (480 = temp unavail)

Try stopping iptables in Linux and try again. Sounds like something is interfering with your packets getting from here to there...
Try:

/etc/init.d/iptables stop

And then see if your packets can move again.

-MC



Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:18 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.





On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

I haven't changed any of the conf files.



What happens when you try to register? Do you get 480? (timeout) Or something else?
-MC

________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From telles-listas at devel-it.com.br  Mon Aug 24 15:00:11 2009
From: telles-listas at devel-it.com.br (Rodrigo P. Telles)
Date: Mon, 24 Aug 2009 19:00:11 -0300
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
In-Reply-To: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
References: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
Message-ID: <4A930D6B.2040202@devel-it.com.br>

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From raffaele.p.guidi at gmail.com  Mon Aug 24 15:06:18 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Tue, 25 Aug 2009 00:06:18 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
Message-ID: <b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>

This is what I was asking! :D When the installer finished it started the
whole thing and everything got loaded fine, but when I restarted my system
it didn't (and did not anymore). Well, I will try to install everything from
scratch again and see...

On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org> wrote:

> If you installed FreePBX then it would be that softwares job to manage
> the sofia profiles... wouldn't it?
>
> /b
>
> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>
> > Actually I did that and it worked fine. I had the problem the SECOND
> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
> > sip_profiles were not
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Mon Aug 24 15:13:47 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 00:13:47 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
Message-ID: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>

Hello Brian, Dave


Still nothing... i've changed ip_addresses (remote_ip, local_ip) and changed
branch within ACK message to meet INVITE's one....but it is still not
enough...
Also i checked RFC and this is how should it be ... (ACK without contact
taking care to have correct TAGs and branch)... what can it be?


   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 325 bytes to udp/[10.4.4.252]:5060 at 21:56:08.152812:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
send 718 bytes to udp/[10.4.4.252]:5060 at 21:56:08.159929:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=cFS6jHj9DgjjF
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 342 bytes from udp/[10.4.4.252]:5060 at 21:56:08.160166:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7079-1-0
   From: 22222238515000403 <sip:22222238515000403 at 110.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=cFS6jHj9DgjjF
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 ACK
   Max-Forwards: 70
   Content-Length: 0

   ------------------------------------------------------------------------
send 718 bytes to udp/[10.4.4.252]:5060 at 21:56:08.661299:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7079-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=cFS6jHj9DgjjF
   Call-ID: 1-7079 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0





here is a scenario i use:


<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">


<scenario name="Basic Sipstone UAC">
  <send retrans="500">
    <![CDATA[

      INVITE sip:[service]@[remote_ip] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip];branch=[branch]
      Max-Forwards: 70
      Contact: <sip:[field1]@[local_ip]>
      From: [field1]
<sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
      To: [service] <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000

    ]]>
  </send>

  <recv response="100"
        optional="true">
  </recv>


  <recv response="302" rtd="true">
  </recv>

  <send>
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
      From: [field1]
<sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]
      To: [service]
<sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Max-Forwards: 70
      Content-Length: 0

    ]]>
  </send>

  <!-- definition of the response time repartition table (unit is ms)   -->
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

  <!-- definition of the call length repartition table (unit is ms)     -->
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>






---------- Forwarded message ----------
From: Brian West <brian at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Date: Mon, 24 Aug 2009 15:42:31 -0500
Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand
ACK message
ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
Call-ID: [call_id]
CSeq: 1 ACK
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Performance Test
Content-Length: 0

Use that.. your scenario has some hard coded IP's in the fields that
shouldn't be there.

/b

On Aug 24, 2009, at 3:37 PM, Tihomir Culjaga wrote:

Hello Brian,

it doesn't work .. tried this today as well:
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From anthony.minessale at gmail.com  Mon Aug 24 15:20:45 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 17:20:45 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
Message-ID: <191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>

What is your exact sipp scenerio file and dialplan to this point after the
changes you were suggested to use.
please send both.

If we have to stop what we are doing to prove this works are you prepared to
offer your soul to help
document and other project maintenance?


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Mon Aug 24 15:31:11 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 00:31:11 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
Message-ID: <65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>

sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l
4000
scenario file:      uac_redirect.xml
FS dialplan:       public.xml
SIP trace:          trace.log


Here it is... sorry for not including at first...

T.



On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> What is your exact sipp scenerio file and dialplan to this point after the
> changes you were suggested to use.
> please send both.
>
> If we have to stop what we are doing to prove this works are you prepared
> to offer your soul to help
> document and other project maintenance?
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Aug 24 15:31:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 24 Aug 2009 15:31:54 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
Message-ID: <87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>

On Mon, Aug 24, 2009 at 2:49 PM, Mike Peace <MPeace at edcogroupinc.com> wrote:

>  It does the same thing, does anything get set during the install that
> would remember or cache the old network settings? I can access anything from
> the FS server on any of several networks and vice-versa but the SIP will not
> register, again no firewalls are upon any of the test hosts.
>
> Doesn?t make sense to me.
>
>
>
Time to bust out tcpdump and/or wireshark to make 100% certain you know
what's happening with all those SIP packets.

> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:34 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> ?Registration error: 408-Request Timeout?
>
> Sorry for the the typo - 408 = timeout. (480 = temp unavail)
>
> Try stopping iptables in Linux and try again. Sounds like something is
> interfering with your packets getting from here to there...
> Try:
>
> /etc/init.d/iptables stop
>
> And then see if your packets can move again.
>
> -MC
>
>
>
> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:18 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> I haven?t changed any of the conf files.
>
>
>
> What happens when you try to register? Do you get 480? (timeout) Or
> something else?
> -MC
>
>
>  ------------------------------
>
>
>
> *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>
>
> Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
>
>  *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>  Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon Aug 24 15:48:42 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 00:48:42 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
Message-ID: <65d96fc80908241548g2a83b718y550efd87e431a700@mail.gmail.com>

... documentation hate that :)) ... but thats my life actually... thats what
i do for living :)

of course i can go forward and document a lot of things... a lot of
docummentation is pending to completing the project i currently have using
FS.

well ... my soul? .. is it really necessary?

T.


On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> What is your exact sipp scenerio file and dialplan to this point after the
> changes you were suggested to use.
> please send both.
>
> If we have to stop what we are doing to prove this works are you prepared
> to offer your soul to help
> document and other project maintenance?
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Mon Aug 24 15:55:22 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Mon, 24 Aug 2009 17:55:22 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241548g2a83b718y550efd87e431a700@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241548g2a83b718y550efd87e431a700@mail.gmail.com>
Message-ID: <191c3a030908241555x14edc34bu41bda4d3a9a48f4e@mail.gmail.com>

Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]

shouldn't that be

Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]


On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> ... documentation hate that :)) ... but thats my life actually... thats
> what i do for living :)
>
> of course i can go forward and document a lot of things... a lot of
> docummentation is pending to completing the project i currently have using
> FS.
>
> well ... my soul? .. is it really necessary?
>
> T.
>
>
> On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> What is your exact sipp scenerio file and dialplan to this point after the
>> changes you were suggested to use.
>> please send both.
>>
>> If we have to stop what we are doing to prove this works are you prepared
>> to offer your soul to help
>> document and other project maintenance?
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From tculjaga at gmail.com  Mon Aug 24 16:14:23 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 01:14:23 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <191c3a030908241555x14edc34bu41bda4d3a9a48f4e@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241548g2a83b718y550efd87e431a700@mail.gmail.com>
	<191c3a030908241555x14edc34bu41bda4d3a9a48f4e@mail.gmail.com>
Message-ID: <65d96fc80908241614l60a9b6bbq3c77ae5caa2c11e5@mail.gmail.com>

this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0



http://sipp.sourceforge.net/doc3.0/reference.html

*[branch]* - Provide a branch value which is a concatenation of magic cookie
(z9hG4bK) + call number + message index in scenario.
An offset (like [branch-N]) can be appended if you need to have the same
branch value as a previous message.



http://www.ietf.org/rfc/rfc3665.txt :according to examples in RFC i se
branch being same for INVITE, 302 and ACK messages...



[branch-3] makes branch of ACK message same as INVITE....



here is a test with just [branch] ... pls note: Via: SIP/2.0/UDP
10.4.4.252:5060;branch=z9hG4bK-7093-1-3 from ACK message .... the 3rd
message...



freeswitch at l01sipindir1> recv 530 bytes from udp/[10.4.4.252]:5060 at
23:11:34.506681:
   ------------------------------------------------------------------------
   INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   Max-Forwards: 70
   Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
>
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
   Call-ID: 1-7093 at 10.4.4.252
   CSeq: 1 INVITE
   Content-Type: application/sdp
   Content-Length:   131

   v=0
   o=user1 53655765 2353687637 IN IP4 10.4.4.252
   s=-
   c=IN IP4 10.4.4.252
   t=0 0
   m=audio 6000 RTP/AVP 0
   a=rtpmap:0 PCMU/8000
   ------------------------------------------------------------------------
send 325 bytes to udp/[10.4.4.252]:5060 at 23:11:34.506957:
   ------------------------------------------------------------------------
   SIP/2.0 100 Trying
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
   Call-ID: 1-7093 at 10.4.4.252
   CSeq: 1 INVITE
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Content-Length: 0

   ------------------------------------------------------------------------
2009-08-25 01:11:34.505272 [NOTICE] switch_channel.c:602 New Channel
sofia/internal/22222238515000403 at 10.4.4.252:5060[7797f5ae-9103-11de-95e3-cda626a03c4b]
2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing
22222238515000403->30003016094191500 in context public
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ########################
ServiceLookup ########################\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ########################
contact = '' ##############\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ########################
CallerNum = '38515000403' ##########\n
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ########################
RADIUS auth = '' ##########\n
2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing
22222238515000403->doRedirect in context public
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute
redirect(sip:12345616094191500 at pgw01.ot.hr:5060)
send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:34.515143:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0

   ------------------------------------------------------------------------
recv 342 bytes from udp/[10.4.4.252]:5060 at 23:11:34.515369:
   ------------------------------------------------------------------------
   ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
   Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3
   From: 22222238515000403 <sip:22222238515000403 at 110.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093 at 10.4.4.252
   CSeq: 1 ACK
   Max-Forwards: 70
   Content-Length: 0

   ------------------------------------------------------------------------
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute
log(INFO ######################## RADIUS auth NOK!! ##########\n)
2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932 ########################
RADIUS auth NOK!! ##########\n
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute
hangup(USER_BUSY)
2009-08-25 01:11:34.513275 [NOTICE] mod_dptools.c:633 Hangup sofia/internal/
22222238515000403 at 10.4.4.252:5060 [CS_EXECUTE] [USER_BUSY]
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1086 Session 3
(sofia/internal/22222238515000403 at 10.4.4.252:5060) Ended
2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1088 Close Channel
sofia/internal/22222238515000403 at 10.4.4.252:5060 [CS_DESTROY]
send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:35.017059:
   ------------------------------------------------------------------------
   SIP/2.0 302 Moved Temporarily
   Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
   From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
   To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
>;tag=y38ac8m1m6e0K
   Call-ID: 1-7093 at 10.4.4.252
   CSeq: 1 INVITE
   Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
   User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
   Accept: application/sdp
   Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
   Supported: timer, precondition, path, replaces
   Allow-Events: talk, presence, dialog, call-info, sla,
include-session-description, presence.winfo, message-summary, refer
   Content-Length: 0




On Tue, Aug 25, 2009 at 12:55 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>
> shouldn't that be
>
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>
>
>
> On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> ... documentation hate that :)) ... but thats my life actually... thats
>> what i do for living :)
>>
>> of course i can go forward and document a lot of things... a lot of
>> docummentation is pending to completing the project i currently have using
>> FS.
>>
>> well ... my soul? .. is it really necessary?
>>
>> T.
>>
>>
>> On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <
>> anthony.minessale at gmail.com> wrote:
>>
>>> What is your exact sipp scenerio file and dialplan to this point after
>>> the changes you were suggested to use.
>>> please send both.
>>>
>>> If we have to stop what we are doing to prove this works are you prepared
>>> to offer your soul to help
>>> document and other project maintenance?
>>>
>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
>>> ClueCon http://www.cluecon.com/
>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>
>>> AIM: anthm
>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> IRC: irc.freenode.net #freeswitch
>>>
>>> FreeSWITCH Developer Conference
>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>> iax:guest at conference.freeswitch.org/888
>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> pstn:213-799-1400
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon Aug 24 16:17:18 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 01:17:18 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241614l60a9b6bbq3c77ae5caa2c11e5@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241548g2a83b718y550efd87e431a700@mail.gmail.com>
	<191c3a030908241555x14edc34bu41bda4d3a9a48f4e@mail.gmail.com>
	<65d96fc80908241614l60a9b6bbq3c77ae5caa2c11e5@mail.gmail.com>
Message-ID: <65d96fc80908241617g169f261r706ef2fec4514650@mail.gmail.com>

Is there any way to make FS complain about what header is wrong ?

T.

On Tue, Aug 25, 2009 at 1:14 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> this is original: Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7085-1-0
>
>
>
> http://sipp.sourceforge.net/doc3.0/reference.html
>
> *[branch]* - Provide a branch value which is a concatenation of magic
> cookie (z9hG4bK) + call number + message index in scenario.
> An offset (like [branch-N]) can be appended if you need to have the same
> branch value as a previous message.
>
>
>
> http://www.ietf.org/rfc/rfc3665.txt :according to examples in RFC i se
> branch being same for INVITE, 302 and ACK messages...
>
>
>
> [branch-3] makes branch of ACK message same as INVITE....
>
>
>
> here is a test with just [branch] ... pls note: Via: SIP/2.0/UDP
> 10.4.4.252:5060;branch=z9hG4bK-7093-1-3 from ACK message .... the 3rd
> message...
>
>
>
> freeswitch at l01sipindir1> recv 530 bytes from udp/[10.4.4.252]:5060 at
> 23:11:34.506681:
>    ------------------------------------------------------------------------
>    INVITE sip:30003016094191500 at 10.4.4.251<sip%3A30003016094191500 at 10.4.4.251>SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
>    Max-Forwards: 70
>    Contact: <sip:22222238515000403 at 10.4.4.252<sip%3A22222238515000403 at 10.4.4.252>
> >
>    From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
>    To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
>    Call-ID: 1-7093 at 10.4.4.252
>    CSeq: 1 INVITE
>    Content-Type: application/sdp
>    Content-Length:   131
>
>    v=0
>    o=user1 53655765 2353687637 IN IP4 10.4.4.252
>    s=-
>    c=IN IP4 10.4.4.252
>    t=0 0
>    m=audio 6000 RTP/AVP 0
>    a=rtpmap:0 PCMU/8000
>    ------------------------------------------------------------------------
> send 325 bytes to udp/[10.4.4.252]:5060 at 23:11:34.506957:
>    ------------------------------------------------------------------------
>    SIP/2.0 100 Trying
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
>    From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
>    To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060>
>    Call-ID: 1-7093 at 10.4.4.252
>    CSeq: 1 INVITE
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2009-08-25 01:11:34.505272 [NOTICE] switch_channel.c:602 New Channel
> sofia/internal/22222238515000403 at 10.4.4.252:5060[7797f5ae-9103-11de-95e3-cda626a03c4b]
> 2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing
> 22222238515000403->30003016094191500 in context public
> 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932
> ######################## ServiceLookup ########################\n
> 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932
> ######################## contact = '' ##############\n
> 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932
> ######################## CallerNum = '38515000403' ##########\n
> 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932
> ######################## RADIUS auth = '' ##########\n
> 2009-08-25 01:11:34.513275 [INFO] mod_dialplan_xml.c:315 Processing
> 22222238515000403->doRedirect in context public
> 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute
> redirect(sip:12345616094191500 at pgw01.ot.hr:5060)
> send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:34.515143:
>    ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
>    From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
>    To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
> >;tag=y38ac8m1m6e0K
>    Call-ID: 1-7093 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> recv 342 bytes from udp/[10.4.4.252]:5060 at 23:11:34.515369:
>    ------------------------------------------------------------------------
>    ACK sip:30003016094191500 at 10.4.4.251:5060 SIP/2.0
>    Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7093-1-3
>    From: 22222238515000403 <sip:22222238515000403 at 110.4.4.252:5060>;tag=1
>    To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
> >;tag=y38ac8m1m6e0K
>    Call-ID: 1-7093 at 10.4.4.252
>    CSeq: 1 ACK
>    Max-Forwards: 70
>    Content-Length: 0
>
>    ------------------------------------------------------------------------
> 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute
> log(INFO ######################## RADIUS auth NOK!! ##########\n)
> 2009-08-25 01:11:34.513275 [INFO] mod_dptools.c:932
> ######################## RADIUS auth NOK!! ##########\n
> 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1576 Execute
> hangup(USER_BUSY)
> 2009-08-25 01:11:34.513275 [NOTICE] mod_dptools.c:633 Hangup
> sofia/internal/22222238515000403 at 10.4.4.252:5060 [CS_EXECUTE] [USER_BUSY]
> 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1086 Session 3
> (sofia/internal/22222238515000403 at 10.4.4.252:5060) Ended
> 2009-08-25 01:11:34.513275 [NOTICE] switch_core_session.c:1088 Close
> Channel sofia/internal/22222238515000403 at 10.4.4.252:5060 [CS_DESTROY]
> send 718 bytes to udp/[10.4.4.252]:5060 at 23:11:35.017059:
>    ------------------------------------------------------------------------
>    SIP/2.0 302 Moved Temporarily
>    Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7093-1-0
>    From: 22222238515000403 <sip:22222238515000403 at 10.4.4.252:5060>;tag=1
>    To: 30003016094191500 <sip:30003016094191500 at 10.4.4.251:5060
> >;tag=y38ac8m1m6e0K
>    Call-ID: 1-7093 at 10.4.4.252
>    CSeq: 1 INVITE
>    Contact: <sip:12345616094191500 at pgw01.ot.hr:5060>
>    User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported
>    Accept: application/sdp
>    Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE,
> NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
>    Supported: timer, precondition, path, replaces
>    Allow-Events: talk, presence, dialog, call-info, sla,
> include-session-description, presence.winfo, message-summary, refer
>    Content-Length: 0
>
>
>
>
> On Tue, Aug 25, 2009 at 12:55 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>>
>> shouldn't that be
>>
>> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>
>>
>>
>> On Mon, Aug 24, 2009 at 5:48 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> ... documentation hate that :)) ... but thats my life actually... thats
>>> what i do for living :)
>>>
>>> of course i can go forward and document a lot of things... a lot of
>>> docummentation is pending to completing the project i currently have using
>>> FS.
>>>
>>> well ... my soul? .. is it really necessary?
>>>
>>> T.
>>>
>>>
>>> On Tue, Aug 25, 2009 at 12:20 AM, Anthony Minessale <
>>> anthony.minessale at gmail.com> wrote:
>>>
>>>> What is your exact sipp scenerio file and dialplan to this point after
>>>> the changes you were suggested to use.
>>>> please send both.
>>>>
>>>> If we have to stop what we are doing to prove this works are you
>>>> prepared to offer your soul to help
>>>> document and other project maintenance?
>>>>
>>>>
>>>>
>>>> --
>>>> Anthony Minessale II
>>>>
>>>> FreeSWITCH http://www.freeswitch.org/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
>>>> FreeSWITCH Developer Conference
>>>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>>>> iax:guest at conference.freeswitch.org/888
>>>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>>> pstn:213-799-1400
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From rogelio.perez at gmail.com  Mon Aug 24 16:52:47 2009
From: rogelio.perez at gmail.com (Rogelio Perez)
Date: Mon, 24 Aug 2009 20:52:47 -0300
Subject: [Freeswitch-users] Freeswitch vs. Asterisk speed on ARM
In-Reply-To: <3BF035D5-ED50-406E-8D19-FAAD314BB045@freeswitch.org>
References: <0DE26D3F-27B9-4082-93CB-A07A5F60875F@gmail.com>
	<3BF035D5-ED50-406E-8D19-FAAD314BB045@freeswitch.org>
Message-ID: <9C2E3DC8-3A13-4D2F-A55C-4AAA88FC90CC@gmail.com>

Hi Raymond,

I'm not planning to have more than 10 concurrent calls on this  
devices, but I'm also curious as you about how many calls can it handle.
When I get to that point I will post the test results on this list.

Regards,
Rogelio


On Aug 23, 2009, at 8:23 PM, Raymond Chandler wrote:

> Could you load freeswitch with a couple hundred calls then run the  
> test again.. and do the same to asterisk and see how the numbers  
> stack up then? I'm just curious to see what happens at that point.
>
> -Ray
>
> On Aug 21, 2009, at 3:15 PM, Rogelio Perez wrote:
>
>> Hi Everyone,
>>
>> I'm working on a PBX project for the Sheevaplug ARM based computer,  
>> with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU.
>> So far I've found a big difference between Freeswitch and Asterisk  
>> performance times.
>> This is a comparison of the time it takes them to perform different  
>> actions:
>>
>> startup Freeswitch:			3 min.
>> startup Asterisk:			2 sec.
>>
>> call extension Freeswitch:	6 sec.
>> call extension Asterisk:		0 sec.
>>
>> shutdown Freeswitch:		6.5 sec
>> shutdown Asterisk:			0 sec.
>>
>> reload config Freeswitch: 	1 sec.
>> reload config Asterisk:		1 sec.
>>
>> Both were built from sources natively (no cross-compiling), and  
>> they use the default startup configurations.
>> I have managed to lower the Freeswitch times by disabling most of  
>> the modules and recompiling, but it is still far away from Asterisk  
>> (i.e. FS startup time 2.5 min).
>>
>> 1. Is there any way to further improve Freeswitch performance for  
>> the ARM architecture?
>> 2. Can this be related to the lack of a FPU (the Sheevalug emulates  
>> the floating point operations).
>> 3. On the startup I see this error repeated many times: [ERR]  
>> switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be  
>> related?
>>
>> Thanks,
>> Rogelio Perez
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mayamatakeshi at gmail.com  Mon Aug 24 18:52:27 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 25 Aug 2009 10:52:27 +0900
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
Message-ID: <15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>

On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com> wrote:
>
> sipp_cmd:???????? sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l
> 4000
> scenario file:????? uac_redirect.xml
> FS dialplan:?????? public.xml
> SIP trace:????????? trace.log

The Via definition in your SIPp scenario differs between the INVITE and the ACK:

INVITE:
Via: SIP/2.0/[transport] [local_ip];branch=[branch]

ACK:
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]


In the INVITE, you are not adding the [local_port] as you do in the ACK.
Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK.
So it seems FS is not checking just the ACK's branch against the
INVITE's; it seems it is checking the whole Via header.
I don't know if this is in accordance to SIP specs.

Another thing, about the way you are calling SIPp: do no use "-sn uac"
and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
means "use the internal (embedded) scenario named xxx". So this
conflicts with the other parameter "-sf" which specifies an external
profile. It seems this doesn't cause any problem (probably because in
the sipp startup, -sf overrides -sn), but it is misleading.

regards,
takeshi



From mayamatakeshi at gmail.com  Mon Aug 24 18:57:36 2009
From: mayamatakeshi at gmail.com (mayamatakeshi)
Date: Tue, 25 Aug 2009 10:57:36 +0900
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
	<15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>
Message-ID: <15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com>

On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com> wrote:
> On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com> wrote:
>>
>> sipp_cmd:???????? sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
>> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1 -l
>> 4000
>> scenario file:????? uac_redirect.xml
>> FS dialplan:?????? public.xml
>> SIP trace:????????? trace.log
>
> The Via definition in your SIPp scenario differs between the INVITE and the ACK:
>
> INVITE:
> Via: SIP/2.0/[transport] [local_ip];branch=[branch]
>
> ACK:
> Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>
>
> In the INVITE, you are not adding the [local_port] as you do in the ACK.
> Just adding the [local_port] in the INVITE makes FreeSWITCH accept the ACK.
> So it seems FS is not checking just the ACK's branch against the
> INVITE's; it seems it is checking the whole Via header.
> I don't know if this is in accordance to SIP specs.
> Another thing, about the way you are calling SIPp: do no use "-sn uac"
> and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
> means "use the internal (embedded) scenario named xxx". So this
> conflicts with the other parameter "-sf" which specifies an external
> profile.

I mean, an external scenario (file).

 It seems this doesn't cause any problem (probably because in
> the sipp startup, -sf overrides -sn), but it is misleading.
>
> regards,
> takeshi
>



From gmaruzz at celliax.org  Mon Aug 24 19:56:45 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 04:56:45 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com> 
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com> 
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com> 
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
Message-ID: <7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>

Windows installer does not work for me.

I've reinstalled various times, same results.

I can correctly create a number, but when I try to create a device for
that number, it tells me that cannot locate the device, and the
password for vicemail will be invalid.

After that, it begins to give the php error page, it cannot find the
start < tag in directory/default.xml

Also for me there are no sofia profiles...

So, I cannot start to test it (eg: I would like to add mod_skypiax
support to it).


Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
Guidi<raffaele.p.guidi at gmail.com> wrote:
> This is what I was asking! :D When the installer finished it started the
> whole thing and everything got loaded fine, but when I restarted my system
> it didn't (and did not anymore). Well, I will try to install everything from
> scratch again and see...
>
> On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org> wrote:
>>
>> If you installed FreePBX then it would be that softwares job to manage
>> the sofia profiles... wouldn't it?
>>
>> /b
>>
>> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>>
>> > Actually I did that and it worked fine. I had the problem the SECOND
>> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
>> > sip_profiles were not
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From tparikh at gmail.com  Mon Aug 24 21:56:45 2009
From: tparikh at gmail.com (Tapan Parikh)
Date: Mon, 24 Aug 2009 21:56:45 -0700
Subject: [Freeswitch-users] compile freeswitch, lua, luasql for 64-bit arch
Message-ID: <1ecdcb6a0908242156l65cb60fk6c096760da0a2d8e@mail.gmail.com>

Hi -

Does anyone know how to compile freeswitch for a 64-bit architecture
on mac os x?

MySQL on MAC OS X only comes pre-compiled for 64-bit.  I compiled
luasql against this libmysqlclient.dylib.  Now when I call lua from
freeswitch I get:

2009-08-24 21:06:44.370439 [ERR] mod_lua.cpp:182 error loading module
'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so':
	dlopen(/usr/local/lib/lua/5.1/luasql/mysql.so, 2): no suitable image
found.  Did find:
	/usr/local/lib/lua/5.1/luasql/mysql.so: mach-o, but wrong architecture
stack traceback:
	[C]: ?
	[C]: in function 'require'
	/usr/local/freeswitch/scripts/otalo.lua:3: in main chun

Any pointers?



From panayotov.vd at gmail.com  Mon Aug 24 23:22:37 2009
From: panayotov.vd at gmail.com (Vassil Panayotov)
Date: Tue, 25 Aug 2009 09:22:37 +0300
Subject: [Freeswitch-users] Yet another question about A500 + FS
In-Reply-To: <56CFD144-94D0-42CE-B59D-7DC524C1919D@jerris.com>
References: <8a9b664c0908222321r20c5c4d8wd08721dc8955f62c@mail.gmail.com>
	<56CFD144-94D0-42CE-B59D-7DC524C1919D@jerris.com>
Message-ID: <8a9b664c0908242322t652c7120ob76bb984597402d@mail.gmail.com>

Thank you for you reply Mike!

'ignore_early_media=true' variable setting is the solution, but I figured it
out shortly after posting to the ML.

Best regards,
V. Panayotov

On Mon, Aug 24, 2009 at 10:46 PM, Michael Jerris <mike at jerris.com> wrote:

> Do you have an answer in the dialplan for that extension?  Also, check
> out the ignore_early_media variable.
>
> Mike
>
> On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote:
>
> > Hi,
> >
> > I managed to get our A500 running with FreeSWITCH 1.0.4 stable using
> > wanpipe 3.4.4 drivers. But now I have another problem...
> > I want to originate calls through event socket, and I only want to
> > receive ANSWERED(+OK) reply when the user actually answers.
> >
> > Now the situation is:
> >
> > ====================================
> > originate openzap/1/a/123456 023
> > 2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT:
> > CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456] Ci=
> > [0000000000]
> > 2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT
> > (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4
> > 2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel
> > OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082]
> > 2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer
> > OpenZAP/1:1/123456!
> > API CALL [originate(openzap/1/a/123456 023)] output:
> > +OK f8fca2be-8fa7-11de-9076-511e29dfc082
> >
> > 2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer
> > OpenZAP/1:1/123456 to XML[023 at default]
> > freeswitch at emo-voip> 2009-08-23 08:44:06.743475 [INFO]
> > mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default
> > 2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
> > [OpenZAP/1:1/123456] has been answered
> > 2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT
> > (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5
> > 2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT
> > (N): CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6
> > 2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup
> > OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING]
> > 2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT
> > (N): CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3
> > 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086
> > Session 2 (OpenZAP/1:1/123456) Ended
> > 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close
> > Channel OpenZAP/1:1/123456 [CS_DESTROY]
> > ====================================
> >
> > Extension 023 is an IVR. As you can see FreeSWITCH answers the call
> > (2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel
> > [OpenZAP/1:1/123456] has been answered) 20 seconds before user
> > actually pick up the phone (2009-08-23 08:44:20.206010 [WARNING]
> > ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0
> > CSid=2 Seq=5).
> >
> > So Sangoma drivers/daemons report the events correctly.
> > How can I set FreeSWITCH to answer after receiving RX EVENT (N):
> > CALL_ANSWERED from the driver?
> >
> > Thank you,
> > V. Panayotov
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Mon Aug 24 23:51:22 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 08:51:22 +0200
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
	<15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>
	<15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com>
Message-ID: <65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com>

Hello Takeshi,

Thanks for your hint... it worked out... so to be precise:

VIA header of both INVITE and ACK messages MUST be identical (IP:PORT +
branch)... and you are right... it might not be according to SIP
specification. Anyhow, i get FS understand my ACK message.


Finally, here is what i used and I'm getting some poor results .. but this
is another topic :)


Thanks for your help.
Tihomir.


sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1
-rp 100 -trace_msg -inf test.txt -m 20000 -l 4000


<?xml version="1.0" encoding="ISO-8859-1" ?>
<!DOCTYPE scenario SYSTEM "sipp.dtd">


<scenario name="Basic Sipstone UAC">
  <send retrans="500" start_rtd="1" start_rtd="2">

    <![CDATA[

      INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
      Max-Forwards: 70
      Contact: <sip:[field1]@[local_ip]>
      From: [field1]
<sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
      To: [service] <sip:[service]@[remote_ip]:[remote_port]>
      Call-ID: [call_id]
      CSeq: 1 INVITE
      Content-Type: application/sdp
      Content-Length: [len]

      v=0
      o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
      s=-
      c=IN IP[media_ip_type] [media_ip]
      t=0 0
      m=audio [media_port] RTP/AVP 0
      a=rtpmap:0 PCMU/8000

    ]]>
  </send>

  <recv response="100"
        optional="true" rtd="1">
  </recv>


  <recv response="302" rtd="2">
  </recv>

  <send>
    <![CDATA[

      ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
      Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
      From: [field1]
<sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]
      To: [service]
<sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
      Call-ID: [call_id]
      CSeq: 1 ACK
      Max-Forwards: 70
      Content-Length: 0

    ]]>
  </send>

  <!-- definition of the response time repartition table (unit is ms)   -->
  <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>

  <!-- definition of the call length repartition table (unit is ms)     -->
  <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>

</scenario>



On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:

> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com>
> wrote:
> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com>
> wrote:
> >>
> >> sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m 1
> -l
> >> 4000
> >> scenario file:      uac_redirect.xml
> >> FS dialplan:       public.xml
> >> SIP trace:          trace.log
> >
> > The Via definition in your SIPp scenario differs between the INVITE and
> the ACK:
> >
> > INVITE:
> > Via: SIP/2.0/[transport] [local_ip];branch=[branch]
> >
> > ACK:
> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
> >
> >
> > In the INVITE, you are not adding the [local_port] as you do in the ACK.
> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept the
> ACK.
> > So it seems FS is not checking just the ACK's branch against the
> > INVITE's; it seems it is checking the whole Via header.
> > I don't know if this is in accordance to SIP specs.
> > Another thing, about the way you are calling SIPp: do no use "-sn uac"
> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
> > means "use the internal (embedded) scenario named xxx". So this
> > conflicts with the other parameter "-sf" which specifies an external
> > profile.
>
> I mean, an external scenario (file).
>
>  It seems this doesn't cause any problem (probably because in
> > the sipp startup, -sf overrides -sn), but it is misleading.
> >
> > regards,
> > takeshi
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From juanbackson at gmail.com  Tue Aug 25 01:21:58 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Tue, 25 Aug 2009 16:21:58 +0800
Subject: [Freeswitch-users] reload user data
Message-ID: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>

Hello,

I would like to dynamically add user to freeswitch.  If I add a new file to
the directory dir, is there anyway to have freeswitch to read the new user
xml file without having to restart freeswitch?

Other than using flat file, is there anyway to add user to freeswitch user
api command?

Thanks,
JB
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From krice at freeswitch.org  Tue Aug 25 01:30:17 2009
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 25 Aug 2009 03:30:17 -0500
Subject: [Freeswitch-users] reload user data
In-Reply-To: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>
Message-ID: <C6B90B49.5D5AC%krice@freeswitch.org>

You just need to reloadxml you don?t have to restart the whole thing. You
can also use xml_curl to feed the users from a database see my contrib
directory (contrib/swk) for some example scripts and db code



From: Juan Backson <juanbackson at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Tue, 25 Aug 2009 16:21:58 +0800
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] reload user data

Hello,
?
I would like to dynamically add user to freeswitch.? If I add a new file to
the directory dir, is there anyway to have freeswitch to read the new user
xml file without having to restart freeswitch?
?
Other than using flat file, is there anyway to add user to freeswitch user
api command?
?
Thanks,
JB


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From tculjaga at gmail.com  Tue Aug 25 01:54:07 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 10:54:07 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting server
Message-ID: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>

Hello,

i'm trying to use freeswitch as a redirecting server meaning FS has to
receive an INVITE and according to some rules it will redirect calls to
other destinations.


CALLING_USER                FREESWITCH                        SOMEWHERE

INVITE ------------------------------->
           <------------------------------ 100 Trying
           <------------------------------ 302 Moved Temporary
ACK    ------------------------------->
INVITE--------------------------------------------------------------------------------->



Well, wverything works well except i have perfromance issues .... on my HW
FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary). When
i increase the rate, FS starts delaying 302 response. Right at 50 CPS i see
"calls" being build up in FS and the delay begining to grow.

When i observe the machine, load average is almost nothing (load average:
1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread taking
most load... all others are just sitting there with 1-5 % CPU time.
This looks to me as FS handles 302 messages in a single thread?!?!


tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H

top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41, 0.61, 0.60
Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,  0.2%si,
0.0%st
Mem:   2074520k total,   571244k used,  1503276k free,   259604k buffers
Swap:  2650684k total,     3020k used,  2647664k free,   153868k cached

  PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
COMMAND
 4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
freeswitch
 4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
freeswitch
 4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
freeswitch
 4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
freeswitch
 4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
freeswitch
 4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
freeswitch
 4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
freeswitch
 4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
freeswitch
 4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
freeswitch
 4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50 freeswitch



cat /proc/cpuinfo
processor       : 0
vendor_id       : GenuineIntel
cpu family      : 6
model           : 15
model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
stepping        : 6
cpu MHz         : 2333.560
cache size      : 4096 KB
physical id     : 0
siblings        : 2
core id         : 0
cpu cores       : 2
apicid          : 0
initial apicid  : 0
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16
xtpr dca lahf_lm
bogomips        : 4670.78
clflush size    : 64
power management:

processor       : 1
vendor_id       : GenuineIntel
cpu family      : 6
model           : 15
model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
stepping        : 6
cpu MHz         : 2333.560
cache size      : 4096 KB
physical id     : 0
siblings        : 2
core id         : 1
cpu cores       : 2
apicid          : 1
initial apicid  : 1
fdiv_bug        : no
hlt_bug         : no
f00f_bug        : no
coma_bug        : no
fpu             : yes
fpu_exception   : yes
cpuid level     : 10
wp              : yes
flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16
xtpr dca lahf_lm
bogomips        : 4666.82
clflush size    : 64
power management:



uname -a
Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009 i686
GNU/Linux



Of course, i've tuned the machine up

ulimit -c unlimited
ulimit -d unlimited
ulimit -f unlimited
ulimit -i unlimited
ulimit -n 999999
ulimit -q unlimited
ulimit -u unlimited
ulimit -v unlimited
ulimit -x unlimited
ulimit -s 240
ulimit -l unlimited
ulimit -a


Started FS with minimum modules but still 40 CPS seems to be the limit.


So, is there any way to improve performance?


Tihomir.
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From ivanov.maxim at gmail.com  Tue Aug 25 01:57:08 2009
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Tue, 25 Aug 2009 12:57:08 +0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
Message-ID: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>

Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
,
the whole sting is tooo long, is there any shorter way to write same thing? Like
"sofia/gateway/panas*/1000" will try all gateways matching the pattern.



From gmaruzz at celliax.org  Tue Aug 25 02:00:06 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 11:00:06 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
Message-ID: <7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>

Maybe your load comes from disk access?

Try putting the sql and log directories on a ramdisk.

OTH,

-giovanni

On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com> wrote:
> Hello,
>
> i'm trying to use freeswitch as a redirecting server meaning FS has to
> receive an INVITE and according to some rules it will redirect calls to
> other destinations.
>
>
> CALLING_USER??????????????? FREESWITCH??????????????????????? SOMEWHERE
>
> INVITE ------------------------------->
> ?????????? <------------------------------ 100 Trying
> ?????????? <------------------------------ 302 Moved Temporary
> ACK ?? ------------------------------->
> INVITE--------------------------------------------------------------------------------->
>
>
>
> Well, wverything works well except i have perfromance issues .... on my HW
> FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary). When
> i increase the rate, FS starts delaying 302 response. Right at 50 CPS i see
> "calls" being build up in FS and the delay begining to grow.
>
> When i observe the machine, load average is almost nothing (load average:
> 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread taking
> most load... all others are just sitting there with 1-5 % CPU time.
> This looks to me as FS handles 302 messages in a single thread?!?!
>
>
> tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>
> top - 10:41:37 up 167 days, 20:42,? 3 users,? load average: 1.41, 0.61, 0.60
> Tasks:? 83 total,?? 2 running,? 81 sleeping,?? 0 stopped,?? 0 zombie
> Cpu(s): 25.3%us,? 1.5%sy,? 0.0%ni, 30.3%id, 42.7%wa,? 0.0%hi,? 0.2%si,
> 0.0%st
> Mem:?? 2074520k total,?? 571244k used,? 1503276k free,?? 259604k buffers
> Swap:? 2650684k total,???? 3020k used,? 2647664k free,?? 153868k cached
>
> ? PID USER????? PR? NI? VIRT? RES? SHR S %CPU %MEM??? TIME+
> COMMAND
> ?4814 root????? 20?? 0 34188? 20m 3780 S?? 38? 1.0?? 3:10.29
> freeswitch
> ?4800 root????? 20?? 0 34188? 20m 3780 S??? 6? 1.0?? 0:08.26
> freeswitch
> ?4798 root????? 20?? 0 34188? 20m 3780 R??? 5? 1.0?? 0:24.46
> freeswitch
> ?4787 root????? 20?? 0 34188? 20m 3780 S??? 2? 1.0?? 0:11.24
> freeswitch
> ?4794 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.42
> freeswitch
> ?4803 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.74
> freeswitch
> ?4788 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.96
> freeswitch
> ?4804 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.64
> freeswitch
> ?4807 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.68
> freeswitch
> ?4811 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.50 freeswitch
>
>
>
> cat /proc/cpuinfo
> processor?????? : 0
> vendor_id?????? : GenuineIntel
> cpu family????? : 6
> model?????????? : 15
> model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
> stepping??????? : 6
> cpu MHz???????? : 2333.560
> cache size????? : 4096 KB
> physical id???? : 0
> siblings??????? : 2
> core id???????? : 0
> cpu cores?????? : 2
> apicid????????? : 0
> initial apicid? : 0
> fdiv_bug??????? : no
> hlt_bug???????? : no
> f00f_bug??????? : no
> coma_bug??????? : no
> fpu???????????? : yes
> fpu_exception?? : yes
> cpuid level???? : 10
> wp????????????? : yes
> flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
> cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16
> xtpr dca lahf_lm
> bogomips??????? : 4670.78
> clflush size??? : 64
> power management:
>
> processor?????? : 1
> vendor_id?????? : GenuineIntel
> cpu family????? : 6
> model?????????? : 15
> model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
> stepping??????? : 6
> cpu MHz???????? : 2333.560
> cache size????? : 4096 KB
> physical id???? : 0
> siblings??????? : 2
> core id???????? : 1
> cpu cores?????? : 2
> apicid????????? : 1
> initial apicid? : 1
> fdiv_bug??????? : no
> hlt_bug???????? : no
> f00f_bug??????? : no
> coma_bug??????? : no
> fpu???????????? : yes
> fpu_exception?? : yes
> cpuid level???? : 10
> wp????????????? : yes
> flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge mca
> cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3 cx16
> xtpr dca lahf_lm
> bogomips??????? : 4666.82
> clflush size??? : 64
> power management:
>
>
>
> uname -a
> Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009 i686
> GNU/Linux
>
>
>
> Of course, i've tuned the machine up
>
> ulimit -c unlimited
> ulimit -d unlimited
> ulimit -f unlimited
> ulimit -i unlimited
> ulimit -n 999999
> ulimit -q unlimited
> ulimit -u unlimited
> ulimit -v unlimited
> ulimit -x unlimited
> ulimit -s 240
> ulimit -l unlimited
> ulimit -a
>
>
> Started FS with minimum modules but still 40 CPS seems to be the limit.
>
>
> So, is there any way to improve performance?
>
>
> Tihomir.
>
>
>
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From juanbackson at gmail.com  Tue Aug 25 02:22:54 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Tue, 25 Aug 2009 17:22:54 +0800
Subject: [Freeswitch-users] reload user data
In-Reply-To: <C6B90B49.5D5AC%krice@freeswitch.org>
References: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>
	<C6B90B49.5D5AC%krice@freeswitch.org>
Message-ID: <27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>

Hi Ken,

xml_curl is a great idea.  Is there anyway to not having to setup another
HTTP server?  For instance, can I have freeswitch to call an api or call a
lua or php or c script that will return the xml response?  That way, I don't
need to maintain yet another service.

Thanks,
JB

On Tue, Aug 25, 2009 at 4:30 PM, Ken Rice <krice at freeswitch.org> wrote:

>  You just need to reloadxml you don?t have to restart the whole thing. You
> can also use xml_curl to feed the users from a database see my contrib
> directory (contrib/swk) for some example scripts and db code
>
>
> ------------------------------
> *From: *Juan Backson <juanbackson at gmail.com>
> *Reply-To: *<freeswitch-users at lists.freeswitch.org>
> *Date: *Tue, 25 Aug 2009 16:21:58 +0800
> *To: *<freeswitch-users at lists.freeswitch.org>
> *Subject: *[Freeswitch-users] reload user data
>
>
> Hello,
>
> I would like to dynamically add user to freeswitch.  If I add a new file to
> the directory dir, is there anyway to have freeswitch to read the new user
> xml file without having to restart freeswitch?
>
> Other than using flat file, is there anyway to add user to freeswitch user
> api command?
>
> Thanks,
> JB
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jim at evolutiontel.net  Tue Aug 25 02:30:37 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Tue, 25 Aug 2009 19:30:37 +1000
Subject: [Freeswitch-users] reload user data
In-Reply-To: <27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>
References: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>
	<C6B90B49.5D5AC%krice@freeswitch.org>
	<27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>
Message-ID: <ddad80610908250230y373207d3j85f201e73a834dc5@mail.gmail.com>

Is this what you are after?

http://wiki.freeswitch.org/wiki/Mod_xml_odbc

Cheers,
Jim

On Tue, Aug 25, 2009 at 7:22 PM, Juan Backson<juanbackson at gmail.com> wrote:
> Hi Ken,
>
> xml_curl is a great idea.? Is there anyway to not having to setup another
> HTTP server?? For instance, can I have freeswitch to call an api or call a
> lua or php or c script that will return the xml response?? That way, I don't
> need to maintain yet another service.
>
> Thanks,
> JB
>
> On Tue, Aug 25, 2009 at 4:30 PM, Ken Rice <krice at freeswitch.org> wrote:
>>
>> You just need to reloadxml you don?t have to restart the whole thing. You
>> can also use xml_curl to feed the users from a database see my contrib
>> directory (contrib/swk) for some example scripts and db code
>>
>>
>> ________________________________
>> From: Juan Backson <juanbackson at gmail.com>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Tue, 25 Aug 2009 16:21:58 +0800
>> To: <freeswitch-users at lists.freeswitch.org>
>> Subject: [Freeswitch-users] reload user data
>>
>> Hello,
>>
>> I would like to dynamically add user to freeswitch.? If I add a new file
>> to the directory dir, is there anyway to have freeswitch to read the new
>> user xml file without having to restart freeswitch?
>>
>> Other than using flat file, is there anyway to add user to freeswitch user
>> api command?
>>
>> Thanks,
>> JB
>>
>> ________________________________
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net



From leon at scarlet-internet.nl  Tue Aug 25 03:00:57 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Tue, 25 Aug 2009 12:00:57 +0200
Subject: [Freeswitch-users] reload user data
In-Reply-To: <ddad80610908250230y373207d3j85f201e73a834dc5@mail.gmail.com>
References: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>
	<C6B90B49.5D5AC%krice@freeswitch.org>
	<27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>
	<ddad80610908250230y373207d3j85f201e73a834dc5@mail.gmail.com>
Message-ID: <8685DDD8-F246-4F3C-91FF-420BA108680F@scarlet-internet.nl>

Hi,

I wrote that module, but been on vacation for a while :-) It's not  
really finished yet, but it worked well for generating user directory  
xml..

Some things that still need to be done:
- Fix it so that reloadxml works
- Don't write the generated xml always to disk before returning it to fs
- Surely there are bugs that need to be fixed

Did anyone try it yet ? If so, what is your experience ?

Regards,

Leon




On Aug 25, 2009, at 11:30 AM, Jim Burke wrote:

> Is this what you are after?
>
> http://wiki.freeswitch.org/wiki/Mod_xml_odbc
>
> Cheers,
> Jim
>
> On Tue, Aug 25, 2009 at 7:22 PM, Juan Backson<juanbackson at gmail.com>  
> wrote:
>> Hi Ken,
>>
>> xml_curl is a great idea.  Is there anyway to not having to setup  
>> another
>> HTTP server?  For instance, can I have freeswitch to call an api or  
>> call a
>> lua or php or c script that will return the xml response?  That  
>> way, I don't
>> need to maintain yet another service.
>>
>> Thanks,
>> JB
>>
>> On Tue, Aug 25, 2009 at 4:30 PM, Ken Rice <krice at freeswitch.org>  
>> wrote:
>>>
>>> You just need to reloadxml you don?t have to restart the whole  
>>> thing. You
>>> can also use xml_curl to feed the users from a database see my  
>>> contrib
>>> directory (contrib/swk) for some example scripts and db code
>>>
>>>
>>> ________________________________
>>> From: Juan Backson <juanbackson at gmail.com>
>>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>>> Date: Tue, 25 Aug 2009 16:21:58 +0800
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Subject: [Freeswitch-users] reload user data
>>>
>>> Hello,
>>>
>>> I would like to dynamically add user to freeswitch.  If I add a  
>>> new file
>>> to the directory dir, is there anyway to have freeswitch to read  
>>> the new
>>> user xml file without having to restart freeswitch?
>>>
>>> Other than using flat file, is there anyway to add user to  
>>> freeswitch user
>>> api command?
>>>
>>> Thanks,
>>> JB
>>>
>>> ________________________________
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> -- 
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From tculjaga at gmail.com  Tue Aug 25 06:19:41 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 15:19:41 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
Message-ID: <65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>

Hey Giovanni,

thanks for the tip... indeed the db files were heavily used regardless if i
started freeswitch with nosql option (freeswitch -nosql)... FS was not
writing anything into that files ... instead it was just accessing it....
This behaviour leads to a waste of 40% CPU time... waiting for other
processes (mainly disk access) to finish!!!

I moved freeswitch/db/ to a ramdisk and the performance got a boost to 140
CPS with a CPU load of 80%. I was keeping the machine for a while (20 - 30
minutes) on that rate when i sow CPU suddenly went to 100% and FS becoming
irresponsive :).


What can be wrong?
What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
cross?
What fine tuning do we need in order to asure a long high load run?



Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have sense
to move my OS to 64 bit? ... will FS gain more preformance ?... I mean will
FS perofomr drastically better 20%+ ?


Tihomir.


On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> Maybe your load comes from disk access?
>
> Try putting the sql and log directories on a ramdisk.
>
> OTH,
>
> -giovanni
>
> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com>
> wrote:
> > Hello,
> >
> > i'm trying to use freeswitch as a redirecting server meaning FS has to
> > receive an INVITE and according to some rules it will redirect calls to
> > other destinations.
> >
> >
> > CALLING_USER                FREESWITCH                        SOMEWHERE
> >
> > INVITE ------------------------------->
> >            <------------------------------ 100 Trying
> >            <------------------------------ 302 Moved Temporary
> > ACK    ------------------------------->
> >
> INVITE--------------------------------------------------------------------------------->
> >
> >
> >
> > Well, wverything works well except i have perfromance issues .... on my
> HW
> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary).
> When
> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS i
> see
> > "calls" being build up in FS and the delay begining to grow.
> >
> > When i observe the machine, load average is almost nothing (load average:
> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
> taking
> > most load... all others are just sitting there with 1-5 % CPU time.
> > This looks to me as FS handles 302 messages in a single thread?!?!
> >
> >
> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
> >
> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41, 0.61,
> 0.60
> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,  0.2%si,
> > 0.0%st
> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k buffers
> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k cached
> >
> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
> > COMMAND
> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
> > freeswitch
> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
> > freeswitch
> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
> > freeswitch
> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
> > freeswitch
> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
> > freeswitch
> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
> > freeswitch
> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
> > freeswitch
> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
> > freeswitch
> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
> > freeswitch
> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50 freeswitch
> >
> >
> >
> > cat /proc/cpuinfo
> > processor       : 0
> > vendor_id       : GenuineIntel
> > cpu family      : 6
> > model           : 15
> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> > stepping        : 6
> > cpu MHz         : 2333.560
> > cache size      : 4096 KB
> > physical id     : 0
> > siblings        : 2
> > core id         : 0
> > cpu cores       : 2
> > apicid          : 0
> > initial apicid  : 0
> > fdiv_bug        : no
> > hlt_bug         : no
> > f00f_bug        : no
> > coma_bug        : no
> > fpu             : yes
> > fpu_exception   : yes
> > cpuid level     : 10
> > wp              : yes
> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
> mca
> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
> cx16
> > xtpr dca lahf_lm
> > bogomips        : 4670.78
> > clflush size    : 64
> > power management:
> >
> > processor       : 1
> > vendor_id       : GenuineIntel
> > cpu family      : 6
> > model           : 15
> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> > stepping        : 6
> > cpu MHz         : 2333.560
> > cache size      : 4096 KB
> > physical id     : 0
> > siblings        : 2
> > core id         : 1
> > cpu cores       : 2
> > apicid          : 1
> > initial apicid  : 1
> > fdiv_bug        : no
> > hlt_bug         : no
> > f00f_bug        : no
> > coma_bug        : no
> > fpu             : yes
> > fpu_exception   : yes
> > cpuid level     : 10
> > wp              : yes
> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
> mca
> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
> cx16
> > xtpr dca lahf_lm
> > bogomips        : 4666.82
> > clflush size    : 64
> > power management:
> >
> >
> >
> > uname -a
> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009 i686
> > GNU/Linux
> >
> >
> >
> > Of course, i've tuned the machine up
> >
> > ulimit -c unlimited
> > ulimit -d unlimited
> > ulimit -f unlimited
> > ulimit -i unlimited
> > ulimit -n 999999
> > ulimit -q unlimited
> > ulimit -u unlimited
> > ulimit -v unlimited
> > ulimit -x unlimited
> > ulimit -s 240
> > ulimit -l unlimited
> > ulimit -a
> >
> >
> > Started FS with minimum modules but still 40 CPS seems to be the limit.
> >
> >
> > So, is there any way to improve performance?
> >
> >
> > Tihomir.
> >
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jaybinks at gmail.com  Tue Aug 25 06:39:07 2009
From: jaybinks at gmail.com (Jay Binks)
Date: Tue, 25 Aug 2009 23:39:07 +1000
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
Message-ID: <BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>

Everytime someone asks this ,  the resounding answer is use a 64bit os..

No question

Jay



On 25/08/2009, at 23:19, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Hey Giovanni,
>
> thanks for the tip... indeed the db files were heavily used  
> regardless if i started freeswitch with nosql option (freeswitch - 
> nosql)... FS was not writing anything into that files ... instead it  
> was just accessing it....
> This behaviour leads to a waste of 40% CPU time... waiting for other  
> processes (mainly disk access) to finish!!!
>
> I moved freeswitch/db/ to a ramdisk and the performance got a boost  
> to 140 CPS with a CPU load of 80%. I was keeping the machine for a  
> while (20 - 30 minutes) on that rate when i sow CPU suddenly went to  
> 100% and FS becoming irresponsive :).
>
>
> What can be wrong?
> What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should  
> not cross?
> What fine tuning do we need in order to asure a long high load run?
>
>
>
> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it  
> have sense to move my OS to 64 bit? ... will FS gain more  
> preformance ?... I mean will FS perofomr drastically better 20%+ ?
>
>
> Tihomir.
>
>
> On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli <gmaruzz at celliax.org 
> > wrote:
> Maybe your load comes from disk access?
>
> Try putting the sql and log directories on a ramdisk.
>
> OTH,
>
> -giovanni
>
> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir  
> Culjaga<tculjaga at gmail.com> wrote:
> > Hello,
> >
> > i'm trying to use freeswitch as a redirecting server meaning FS  
> has to
> > receive an INVITE and according to some rules it will redirect  
> calls to
> > other destinations.
> >
> >
> > CALLING_USER                FREESWITCH                         
> SOMEWHERE
> >
> > INVITE ------------------------------->
> >            <------------------------------ 100 Trying
> >            <------------------------------ 302 Moved Temporary
> > ACK    ------------------------------->
> >  
> INVITE- 
> --- 
> --- 
> --- 
> --- 
> -------------------------------------------------------------------->
> >
> >
> >
> > Well, wverything works well except i have perfromance issues ....  
> on my HW
> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved  
> Temporary). When
> > i increase the rate, FS starts delaying 302 response. Right at 50  
> CPS i see
> > "calls" being build up in FS and the delay begining to grow.
> >
> > When i observe the machine, load average is almost nothing (load  
> average:
> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one  
> thread taking
> > most load... all others are just sitting there with 1-5 % CPU time.
> > This looks to me as FS handles 302 messages in a single thread?!?!
> >
> >
> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
> >
> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41,  
> 0.61, 0.60
> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,   
> 0.2%si,
> > 0.0%st
> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k  
> buffers
> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k  
> cached
> >
> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
> > COMMAND
> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
> > freeswitch
> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
> > freeswitch
> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
> > freeswitch
> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
> > freeswitch
> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
> > freeswitch
> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
> > freeswitch
> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
> > freeswitch
> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
> > freeswitch
> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
> > freeswitch
> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50  
> freeswitch
> >
> >
> >
> > cat /proc/cpuinfo
> > processor       : 0
> > vendor_id       : GenuineIntel
> > cpu family      : 6
> > model           : 15
> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> > stepping        : 6
> > cpu MHz         : 2333.560
> > cache size      : 4096 KB
> > physical id     : 0
> > siblings        : 2
> > core id         : 0
> > cpu cores       : 2
> > apicid          : 0
> > initial apicid  : 0
> > fdiv_bug        : no
> > hlt_bug         : no
> > f00f_bug        : no
> > coma_bug        : no
> > fpu             : yes
> > fpu_exception   : yes
> > cpuid level     : 10
> > wp              : yes
> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr  
> pge mca
> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2  
> ssse3 cx16
> > xtpr dca lahf_lm
> > bogomips        : 4670.78
> > clflush size    : 64
> > power management:
> >
> > processor       : 1
> > vendor_id       : GenuineIntel
> > cpu family      : 6
> > model           : 15
> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> > stepping        : 6
> > cpu MHz         : 2333.560
> > cache size      : 4096 KB
> > physical id     : 0
> > siblings        : 2
> > core id         : 1
> > cpu cores       : 2
> > apicid          : 1
> > initial apicid  : 1
> > fdiv_bug        : no
> > hlt_bug         : no
> > f00f_bug        : no
> > coma_bug        : no
> > fpu             : yes
> > fpu_exception   : yes
> > cpuid level     : 10
> > wp              : yes
> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr  
> pge mca
> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2  
> ssse3 cx16
> > xtpr dca lahf_lm
> > bogomips        : 4666.82
> > clflush size    : 64
> > power management:
> >
> >
> >
> > uname -a
> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC  
> 2009 i686
> > GNU/Linux
> >
> >
> >
> > Of course, i've tuned the machine up
> >
> > ulimit -c unlimited
> > ulimit -d unlimited
> > ulimit -f unlimited
> > ulimit -i unlimited
> > ulimit -n 999999
> > ulimit -q unlimited
> > ulimit -u unlimited
> > ulimit -v unlimited
> > ulimit -x unlimited
> > ulimit -s 240
> > ulimit -l unlimited
> > ulimit -a
> >
> >
> > Started FS with minimum modules but still 40 CPS seems to be the  
> limit.
> >
> >
> > So, is there any way to improve performance?
> >
> >
> > Tihomir.
> >
> >
> >
> >
> >
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From gmaruzz at celliax.org  Tue Aug 25 06:41:56 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 15:41:56 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com> 
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com> 
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
Message-ID: <7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com>

Definitely go for 64 bit OS.

If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
used both for development and for heavy duty production.

Also Ubuntu 8.04 is good.

Other versions/distros are less used by the community.

Adding RAM and CPUs helps to scale up.

-gm



Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com> wrote:
> Hey Giovanni,
>
> thanks for the tip... indeed the db files were heavily used regardless if i
> started freeswitch with nosql option (freeswitch -nosql)... FS was not
> writing anything into that files ... instead it was just accessing it....
> This behaviour leads to a waste of 40% CPU time... waiting for other
> processes (mainly disk access) to finish!!!
>
> I moved freeswitch/db/ to a ramdisk and the performance got a boost to 140
> CPS with a CPU load of 80%. I was keeping the machine for a while (20 - 30
> minutes) on that rate when i sow CPU suddenly went to 100% and FS becoming
> irresponsive :).
>
>
> What can be wrong?
> What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
> cross?
> What fine tuning do we need in order to asure a long high load run?
>
>
>
> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have sense
> to move my OS to 64 bit? ... will FS gain more preformance ?... I mean will
> FS perofomr drastically better 20%+ ?
>
>
> Tihomir.
>
>
> On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> Maybe your load comes from disk access?
>>
>> Try putting the sql and log directories on a ramdisk.
>>
>> OTH,
>>
>> -giovanni
>>
>> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com>
>> wrote:
>> > Hello,
>> >
>> > i'm trying to use freeswitch as a redirecting server meaning FS has to
>> > receive an INVITE and according to some rules it will redirect calls to
>> > other destinations.
>> >
>> >
>> > CALLING_USER??????????????? FREESWITCH??????????????????????? SOMEWHERE
>> >
>> > INVITE ------------------------------->
>> > ?????????? <------------------------------ 100 Trying
>> > ?????????? <------------------------------ 302 Moved Temporary
>> > ACK ?? ------------------------------->
>> >
>> > INVITE--------------------------------------------------------------------------------->
>> >
>> >
>> >
>> > Well, wverything works well except i have perfromance issues .... on my
>> > HW
>> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary).
>> > When
>> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS i
>> > see
>> > "calls" being build up in FS and the delay begining to grow.
>> >
>> > When i observe the machine, load average is almost nothing (load
>> > average:
>> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
>> > taking
>> > most load... all others are just sitting there with 1-5 % CPU time.
>> > This looks to me as FS handles 302 messages in a single thread?!?!
>> >
>> >
>> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>> >
>> > top - 10:41:37 up 167 days, 20:42,? 3 users,? load average: 1.41, 0.61,
>> > 0.60
>> > Tasks:? 83 total,?? 2 running,? 81 sleeping,?? 0 stopped,?? 0 zombie
>> > Cpu(s): 25.3%us,? 1.5%sy,? 0.0%ni, 30.3%id, 42.7%wa,? 0.0%hi,? 0.2%si,
>> > 0.0%st
>> > Mem:?? 2074520k total,?? 571244k used,? 1503276k free,?? 259604k buffers
>> > Swap:? 2650684k total,???? 3020k used,? 2647664k free,?? 153868k cached
>> >
>> > ? PID USER????? PR? NI? VIRT? RES? SHR S %CPU %MEM??? TIME+
>> > COMMAND
>> > ?4814 root????? 20?? 0 34188? 20m 3780 S?? 38? 1.0?? 3:10.29
>> > freeswitch
>> > ?4800 root????? 20?? 0 34188? 20m 3780 S??? 6? 1.0?? 0:08.26
>> > freeswitch
>> > ?4798 root????? 20?? 0 34188? 20m 3780 R??? 5? 1.0?? 0:24.46
>> > freeswitch
>> > ?4787 root????? 20?? 0 34188? 20m 3780 S??? 2? 1.0?? 0:11.24
>> > freeswitch
>> > ?4794 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.42
>> > freeswitch
>> > ?4803 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.74
>> > freeswitch
>> > ?4788 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.96
>> > freeswitch
>> > ?4804 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.64
>> > freeswitch
>> > ?4807 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.68
>> > freeswitch
>> > ?4811 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.50 freeswitch
>> >
>> >
>> >
>> > cat /proc/cpuinfo
>> > processor?????? : 0
>> > vendor_id?????? : GenuineIntel
>> > cpu family????? : 6
>> > model?????????? : 15
>> > model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
>> > stepping??????? : 6
>> > cpu MHz???????? : 2333.560
>> > cache size????? : 4096 KB
>> > physical id???? : 0
>> > siblings??????? : 2
>> > core id???????? : 0
>> > cpu cores?????? : 2
>> > apicid????????? : 0
>> > initial apicid? : 0
>> > fdiv_bug??????? : no
>> > hlt_bug???????? : no
>> > f00f_bug??????? : no
>> > coma_bug??????? : no
>> > fpu???????????? : yes
>> > fpu_exception?? : yes
>> > cpuid level???? : 10
>> > wp????????????? : yes
>> > flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
>> > mca
>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
>> > cx16
>> > xtpr dca lahf_lm
>> > bogomips??????? : 4670.78
>> > clflush size??? : 64
>> > power management:
>> >
>> > processor?????? : 1
>> > vendor_id?????? : GenuineIntel
>> > cpu family????? : 6
>> > model?????????? : 15
>> > model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
>> > stepping??????? : 6
>> > cpu MHz???????? : 2333.560
>> > cache size????? : 4096 KB
>> > physical id???? : 0
>> > siblings??????? : 2
>> > core id???????? : 1
>> > cpu cores?????? : 2
>> > apicid????????? : 1
>> > initial apicid? : 1
>> > fdiv_bug??????? : no
>> > hlt_bug???????? : no
>> > f00f_bug??????? : no
>> > coma_bug??????? : no
>> > fpu???????????? : yes
>> > fpu_exception?? : yes
>> > cpuid level???? : 10
>> > wp????????????? : yes
>> > flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
>> > mca
>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
>> > cx16
>> > xtpr dca lahf_lm
>> > bogomips??????? : 4666.82
>> > clflush size??? : 64
>> > power management:
>> >
>> >
>> >
>> > uname -a
>> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009 i686
>> > GNU/Linux
>> >
>> >
>> >
>> > Of course, i've tuned the machine up
>> >
>> > ulimit -c unlimited
>> > ulimit -d unlimited
>> > ulimit -f unlimited
>> > ulimit -i unlimited
>> > ulimit -n 999999
>> > ulimit -q unlimited
>> > ulimit -u unlimited
>> > ulimit -v unlimited
>> > ulimit -x unlimited
>> > ulimit -s 240
>> > ulimit -l unlimited
>> > ulimit -a
>> >
>> >
>> > Started FS with minimum modules but still 40 CPS seems to be the limit.
>> >
>> >
>> > So, is there any way to improve performance?
>> >
>> >
>> > Tihomir.
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From tculjaga at gmail.com  Tue Aug 25 06:42:24 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 15:42:24 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
Message-ID: <65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>

thanks for the feedback... this is something im going to do tomorrow...

what about other things?


On Tue, Aug 25, 2009 at 3:39 PM, Jay Binks <jaybinks at gmail.com> wrote:

> Everytime someone asks this ,  the resounding answer is use a 64bit os..
>
> No question
>
> Jay
>
>
>
> On 25/08/2009, at 23:19, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>
> Hey Giovanni,
>
> thanks for the tip... indeed the db files were heavily used regardless if i
> started freeswitch with nosql option (freeswitch -nosql)... FS was not
> writing anything into that files ... instead it was just accessing it....
> This behaviour leads to a waste of 40% CPU time... waiting for other
> processes (mainly disk access) to finish!!!
>
> I moved freeswitch/db/ to a ramdisk and the performance got a boost to 140
> CPS with a CPU load of 80%. I was keeping the machine for a while (20 - 30
> minutes) on that rate when i sow CPU suddenly went to 100% and FS becoming
> irresponsive :).
>
>
> What can be wrong?
> What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
> cross?
> What fine tuning do we need in order to asure a long high load run?
>
>
>
> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
> sense to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
> will FS perofomr drastically better 20%+ ?
>
>
> Tihomir.
>
>
> On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli <<gmaruzz at celliax.org>
> gmaruzz at celliax.org> wrote:
>
>> Maybe your load comes from disk access?
>>
>> Try putting the sql and log directories on a ramdisk.
>>
>> OTH,
>>
>> -giovanni
>>
>> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga< <tculjaga at gmail.com>
>> tculjaga at gmail.com> wrote:
>> > Hello,
>> >
>> > i'm trying to use freeswitch as a redirecting server meaning FS has to
>> > receive an INVITE and according to some rules it will redirect calls to
>> > other destinations.
>> >
>> >
>> > CALLING_USER                FREESWITCH                        SOMEWHERE
>> >
>> > INVITE ------------------------------->
>> >            <------------------------------ 100 Trying
>> >            <------------------------------ 302 Moved Temporary
>> > ACK    ------------------------------->
>> >
>> INVITE--------------------------------------------------------------------------------->
>> >
>> >
>> >
>> > Well, wverything works well except i have perfromance issues .... on my
>> HW
>> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary).
>> When
>> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS i
>> see
>> > "calls" being build up in FS and the delay begining to grow.
>> >
>> > When i observe the machine, load average is almost nothing (load
>> average:
>> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
>> taking
>> > most load... all others are just sitting there with 1-5 % CPU time.
>> > This looks to me as FS handles 302 messages in a single thread?!?!
>> >
>> >
>> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>> >
>> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41, 0.61,
>> 0.60
>> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
>> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,  0.2%si,
>> > 0.0%st
>> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k buffers
>> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k cached
>> >
>> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
>> > COMMAND
>> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
>> > freeswitch
>> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
>> > freeswitch
>> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
>> > freeswitch
>> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
>> > freeswitch
>> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
>> > freeswitch
>> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
>> > freeswitch
>> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
>> > freeswitch
>> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
>> > freeswitch
>> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
>> > freeswitch
>> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50 freeswitch
>> >
>> >
>> >
>> > cat /proc/cpuinfo
>> > processor       : 0
>> > vendor_id       : GenuineIntel
>> > cpu family      : 6
>> > model           : 15
>> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
>> > stepping        : 6
>> > cpu MHz         : 2333.560
>> > cache size      : 4096 KB
>> > physical id     : 0
>> > siblings        : 2
>> > core id         : 0
>> > cpu cores       : 2
>> > apicid          : 0
>> > initial apicid  : 0
>> > fdiv_bug        : no
>> > hlt_bug         : no
>> > f00f_bug        : no
>> > coma_bug        : no
>> > fpu             : yes
>> > fpu_exception   : yes
>> > cpuid level     : 10
>> > wp              : yes
>> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
>> mca
>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
>> cx16
>> > xtpr dca lahf_lm
>> > bogomips        : 4670.78
>> > clflush size    : 64
>> > power management:
>> >
>> > processor       : 1
>> > vendor_id       : GenuineIntel
>> > cpu family      : 6
>> > model           : 15
>> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
>> > stepping        : 6
>> > cpu MHz         : 2333.560
>> > cache size      : 4096 KB
>> > physical id     : 0
>> > siblings        : 2
>> > core id         : 1
>> > cpu cores       : 2
>> > apicid          : 1
>> > initial apicid  : 1
>> > fdiv_bug        : no
>> > hlt_bug         : no
>> > f00f_bug        : no
>> > coma_bug        : no
>> > fpu             : yes
>> > fpu_exception   : yes
>> > cpuid level     : 10
>> > wp              : yes
>> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
>> mca
>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
>> cx16
>> > xtpr dca lahf_lm
>> > bogomips        : 4666.82
>> > clflush size    : 64
>> > power management:
>> >
>> >
>> >
>> > uname -a
>> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009 i686
>> > GNU/Linux
>> >
>> >
>> >
>> > Of course, i've tuned the machine up
>> >
>> > ulimit -c unlimited
>> > ulimit -d unlimited
>> > ulimit -f unlimited
>> > ulimit -i unlimited
>> > ulimit -n 999999
>> > ulimit -q unlimited
>> > ulimit -u unlimited
>> > ulimit -v unlimited
>> > ulimit -x unlimited
>> > ulimit -s 240
>> > ulimit -l unlimited
>> > ulimit -a
>> >
>> >
>> > Started FS with minimum modules but still 40 CPS seems to be the limit.
>> >
>> >
>> > So, is there any way to improve performance?
>> >
>> >
>> > Tihomir.
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>> > <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > <http://www.freeswitch.org>http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  <FreeSWITCH-users at lists.freeswitch.org>
>> FreeSWITCH-users at lists.freeswitch.org
>>  <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:<http://lists.freeswitch.org/mailman/options/freeswitch-users>
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  <http://www.freeswitch.org>http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From gmaruzz at celliax.org  Tue Aug 25 06:47:37 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 15:47:37 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com> 
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com> 
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com> 
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
Message-ID: <7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>

is a heavely multithreaded software, it benefits from number of CPUs
(or cores), RAM, and heavy duty kernel features (found in 64bit
kernels)

put all accesses on ramdisk, leave out the modules you don't use...

experiment, test, and find the best for your specific application/workload

test not only with sipp, but with real load too (often they're very different)

-gm


On Tue, Aug 25, 2009 at 3:42 PM, Tihomir Culjaga<tculjaga at gmail.com> wrote:
> thanks for the feedback... this is something im going to do tomorrow...
>
> what about other things?
>
>
> On Tue, Aug 25, 2009 at 3:39 PM, Jay Binks <jaybinks at gmail.com> wrote:
>>
>> Everytime someone asks this , ?the resounding answer is use a 64bit os..
>> No question
>> Jay
>>
>>
>>
>> On 25/08/2009, at 23:19, Tihomir Culjaga <tculjaga at gmail.com> wrote:
>>
>> Hey Giovanni,
>>
>> thanks for the tip... indeed the db files were heavily used regardless if
>> i started freeswitch with nosql option (freeswitch -nosql)... FS was not
>> writing anything into that files ... instead it was just accessing it....
>> This behaviour leads to a waste of 40% CPU time... waiting for other
>> processes (mainly disk access) to finish!!!
>>
>> I moved freeswitch/db/ to a ramdisk and the performance got a boost to 140
>> CPS with a CPU load of 80%. I was keeping the machine for a while (20 - 30
>> minutes) on that rate when i sow CPU suddenly went to 100% and FS becoming
>> irresponsive :).
>>
>>
>> What can be wrong?
>> What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
>> cross?
>> What fine tuning do we need in order to asure a long high load run?
>>
>>
>>
>> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
>> sense to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
>> will FS perofomr drastically better 20%+ ?
>>
>>
>> Tihomir.
>>
>>
>> On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
>> <gmaruzz at celliax.org> wrote:
>>>
>>> Maybe your load comes from disk access?
>>>
>>> Try putting the sql and log directories on a ramdisk.
>>>
>>> OTH,
>>>
>>> -giovanni
>>>
>>> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com>
>>> wrote:
>>> > Hello,
>>> >
>>> > i'm trying to use freeswitch as a redirecting server meaning FS has to
>>> > receive an INVITE and according to some rules it will redirect calls to
>>> > other destinations.
>>> >
>>> >
>>> > CALLING_USER??????????????? FREESWITCH??????????????????????? SOMEWHERE
>>> >
>>> > INVITE ------------------------------->
>>> > ?????????? <------------------------------ 100 Trying
>>> > ?????????? <------------------------------ 302 Moved Temporary
>>> > ACK ?? ------------------------------->
>>> >
>>> > INVITE--------------------------------------------------------------------------------->
>>> >
>>> >
>>> >
>>> > Well, wverything works well except i have perfromance issues .... on my
>>> > HW
>>> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved Temporary).
>>> > When
>>> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS i
>>> > see
>>> > "calls" being build up in FS and the delay begining to grow.
>>> >
>>> > When i observe the machine, load average is almost nothing (load
>>> > average:
>>> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
>>> > taking
>>> > most load... all others are just sitting there with 1-5 % CPU time.
>>> > This looks to me as FS handles 302 messages in a single thread?!?!
>>> >
>>> >
>>> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>>> >
>>> > top - 10:41:37 up 167 days, 20:42,? 3 users,? load average: 1.41, 0.61,
>>> > 0.60
>>> > Tasks:? 83 total,?? 2 running,? 81 sleeping,?? 0 stopped,?? 0 zombie
>>> > Cpu(s): 25.3%us,? 1.5%sy,? 0.0%ni, 30.3%id, 42.7%wa,? 0.0%hi,? 0.2%si,
>>> > 0.0%st
>>> > Mem:?? 2074520k total,?? 571244k used,? 1503276k free,?? 259604k
>>> > buffers
>>> > Swap:? 2650684k total,???? 3020k used,? 2647664k free,?? 153868k cached
>>> >
>>> > ? PID USER????? PR? NI? VIRT? RES? SHR S %CPU %MEM??? TIME+
>>> > COMMAND
>>> > ?4814 root????? 20?? 0 34188? 20m 3780 S?? 38? 1.0?? 3:10.29
>>> > freeswitch
>>> > ?4800 root????? 20?? 0 34188? 20m 3780 S??? 6? 1.0?? 0:08.26
>>> > freeswitch
>>> > ?4798 root????? 20?? 0 34188? 20m 3780 R??? 5? 1.0?? 0:24.46
>>> > freeswitch
>>> > ?4787 root????? 20?? 0 34188? 20m 3780 S??? 2? 1.0?? 0:11.24
>>> > freeswitch
>>> > ?4794 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.42
>>> > freeswitch
>>> > ?4803 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.74
>>> > freeswitch
>>> > ?4788 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.96
>>> > freeswitch
>>> > ?4804 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.64
>>> > freeswitch
>>> > ?4807 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.68
>>> > freeswitch
>>> > ?4811 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.50 freeswitch
>>> >
>>> >
>>> >
>>> > cat /proc/cpuinfo
>>> > processor?????? : 0
>>> > vendor_id?????? : GenuineIntel
>>> > cpu family????? : 6
>>> > model?????????? : 15
>>> > model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
>>> > stepping??????? : 6
>>> > cpu MHz???????? : 2333.560
>>> > cache size????? : 4096 KB
>>> > physical id???? : 0
>>> > siblings??????? : 2
>>> > core id???????? : 0
>>> > cpu cores?????? : 2
>>> > apicid????????? : 0
>>> > initial apicid? : 0
>>> > fdiv_bug??????? : no
>>> > hlt_bug???????? : no
>>> > f00f_bug??????? : no
>>> > coma_bug??????? : no
>>> > fpu???????????? : yes
>>> > fpu_exception?? : yes
>>> > cpuid level???? : 10
>>> > wp????????????? : yes
>>> > flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
>>> > mca
>>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
>>> > cx16
>>> > xtpr dca lahf_lm
>>> > bogomips??????? : 4670.78
>>> > clflush size??? : 64
>>> > power management:
>>> >
>>> > processor?????? : 1
>>> > vendor_id?????? : GenuineIntel
>>> > cpu family????? : 6
>>> > model?????????? : 15
>>> > model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
>>> > stepping??????? : 6
>>> > cpu MHz???????? : 2333.560
>>> > cache size????? : 4096 KB
>>> > physical id???? : 0
>>> > siblings??????? : 2
>>> > core id???????? : 1
>>> > cpu cores?????? : 2
>>> > apicid????????? : 1
>>> > initial apicid? : 1
>>> > fdiv_bug??????? : no
>>> > hlt_bug???????? : no
>>> > f00f_bug??????? : no
>>> > coma_bug??????? : no
>>> > fpu???????????? : yes
>>> > fpu_exception?? : yes
>>> > cpuid level???? : 10
>>> > wp????????????? : yes
>>> > flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
>>> > mca
>>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2 ssse3
>>> > cx16
>>> > xtpr dca lahf_lm
>>> > bogomips??????? : 4666.82
>>> > clflush size??? : 64
>>> > power management:
>>> >
>>> >
>>> >
>>> > uname -a
>>> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009
>>> > i686
>>> > GNU/Linux
>>> >
>>> >
>>> >
>>> > Of course, i've tuned the machine up
>>> >
>>> > ulimit -c unlimited
>>> > ulimit -d unlimited
>>> > ulimit -f unlimited
>>> > ulimit -i unlimited
>>> > ulimit -n 999999
>>> > ulimit -q unlimited
>>> > ulimit -u unlimited
>>> > ulimit -v unlimited
>>> > ulimit -x unlimited
>>> > ulimit -s 240
>>> > ulimit -l unlimited
>>> > ulimit -a
>>> >
>>> >
>>> > Started FS with minimum modules but still 40 CPS seems to be the limit.
>>> >
>>> >
>>> > So, is there any way to improve performance?
>>> >
>>> >
>>> > Tihomir.
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From csa at nowthor.com  Tue Aug 25 07:06:25 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Tue, 25 Aug 2009 10:06:25 -0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
Message-ID: <4A93EFE1.6010007@nowthor.com>

Max,

I would like to see something similar too. For example, it would be 
wonderful if one could specify multiple gateways to try like this or 
something similar:

<action application="bridge">
    <!-- set some kind of *bridge-specific* parameter/variable -->
    <gateway order=1 data="sofia/gateway/gw-1/1$1"/>
    <gateway order=1 data="sofia/gateway/gw-2/1$1"/>
    <gateway order=2 data="sofia/gateway/gw-3/1$1">
        <!-- set some kind of *gw-specific* parameter/variable -->
    </gateway>
    <gateway order=3 data="sofia/gateway/gw-4/1$1"/>
</action>

One would be able to avoid the "[]" and "{}" hacks and combine 
sequential and simultaneous trying of gateways.

What do the developers think of this?

Carlos

Max Ivanov wrote:
> Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
> sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
> ,
> the whole sting is tooo long, is there any shorter way to write same thing? Like
> "sofia/gateway/panas*/1000" will try all gateways matching the pattern.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From rs at runsolutions.com  Tue Aug 25 02:06:37 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Tue, 25 Aug 2009 11:06:37 +0200
Subject: [Freeswitch-users] using sipp to test call transfers
Message-ID: <84D97940-68D0-402F-A7E2-B5ADD287DC73@runsolutions.com>

Hi List,

i have some scripts to test our lab:

I have scripts to create sipp instances which act as individual  
agents, which log on, take a random amount of calls, log off, wait a  
bit, log on again, etc.
I have scripts to generate call traffic for our queues to saturate them

But what I have difficulties is how to create a good scenario xml file  
for call transfers? Has anybody here a sample for me to sink my teeth  
in?

thanks and best regards

-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares




From aep.lists at it46.se  Mon Aug 24 23:54:24 2009
From: aep.lists at it46.se (Alberto Escudero-Pascual (lists))
Date: Tue, 25 Aug 2009 08:54:24 +0200
Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
In-Reply-To: <191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
	<191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
Message-ID: <6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>

Dear all,

In one of the applications I am writing I need to convert a recorded wav
to mp3.

After using

session.recordFile() and obtaining a foo.wav file, I am calling

session.execute("system",lmLameCmd);

to invoke lame for the conversion.

The system command looks like this:
lmLameCmd = "/usr/local/freeswitch/bin/lame -V2 foo.wav foo.mp3 -S";


Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
appears, if I use version 3.97 (older version), it does!.

If I execute the conversion from the command line, i get the mp3 with both
3.97 and 3.98.2

In fact, i am considering doing the conversions as background job, but I
am very curious to hear if this behavior has a pseudo-scientific
explanation

/aep

-- 
Stopping junk mailers is good for the environment





From ederwander at gmail.com  Tue Aug 25 06:06:40 2009
From: ederwander at gmail.com (Eder Souza)
Date: Tue, 25 Aug 2009 10:06:40 -0300
Subject: [Freeswitch-users] MFC-R2 support for FreeSWITCH
In-Reply-To: <4A930D6B.2040202@devel-it.com.br>
References: <c4d05cbe0908211529t712fbd14me158e1d756fcdf68@mail.gmail.com>
	<4A930D6B.2040202@devel-it.com.br>
Message-ID: <622bedea0908250606w57d4eba8q3045eea272bf012f@mail.gmail.com>

Very Nice my friends i'm from Brazil i'm crazy for the tests ...


Eng Eder de Souza

On Mon, Aug 24, 2009 at 7:00 PM, Rodrigo P. Telles <
telles-listas at devel-it.com.br> wrote:

> It is very nice to hear that, great work!
> Thanks to support FS.
>
> Telles
>
>
> Moises Silva wrote:
>
> So, I finally took some days to put up OpenR2 working with OpenZAP, which
> means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has
> support for. Including Mexico, Brazil, Argentina and others. The stack has
> been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most
> countries that users may be interested in, support for new variants will be
> added on-demand only (in any case users can always tweak the advanced
> configuration file to create their own variants as a last resort).
> I created a wiki page to illustrate the basic setup:
> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2
>
> Now is time for testing. I just did minimal testing on my development
> environment, no serious testing, and I know that some stuff is not working
> at this point (I had some issues with variable length DNIS and ANI) which
> should be fixed soon.
>
> If anyone around happens to have an R2 link and wants to test R2 support in
> OpenZAP, I can give them a hand with the configuration and any issues you
> may find. You can find me on IRC at #freeswitch, #freeswitch-dev and
> #openzap as moy.
>
> --
> Moises Silva
> Software Developer
> Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R
> 9T3 Canada
> t. 1 905 474 1990 x 128 | e. moy at sangoma.com
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From rs at runsolutions.com  Tue Aug 25 06:35:13 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Tue, 25 Aug 2009 15:35:13 +0200
Subject: [Freeswitch-users] Freeswitch & DAHDI Kernel drivers / Hardware
	questions
Message-ID: <DD5C7209-7041-4D21-869D-2F3BABEADC78@runsolutions.com>

Hello,

i was reading through the openzap wiki page and searched the wiki, the  
only thing I found about dahdi is a note from January that freeswitch  
does not work with dahdi right now.

Is this current information? Can't I use dahdi kernel drivers for  
freeswitch? As the last update to the zaptel drivers is now nearly a  
year old (1.4.12) and development has, AFAIK, ceased and new stuff  
only gehts to dahdi, correct me please if I am wrong.

Second question,

is it possible to use this hardware:

00:0d.0 Network controller: Cologne Chip Designs GmbH ISDN network  
controller [HFC-PCI] (rev 02)
	Subsystem: Cologne Chip Designs GmbH ISDN Board
	Flags: bus master, medium devsel, latency 16, IRQ 10
	I/O ports at e000 [disabled] [size=8]
	Memory at ee001000 (32-bit, non-prefetchable) [size=256]
	Capabilities: [40] Power Management version 1
	Kernel modules: hisax

with openZap? I read in the wiki (openZap.conf_Examples) something  
about HFC 4 BRI, but i do not know if this is the same as the above  
mentioned card.


I have access to some pcmcia fxo, and pcmcia isdn cards, could they be  
used? I have as well an soekris board with two pcmcia slots, could i  
use this with an fxo pcmcia card to actually drive a small home - pbx  
with freeswitch?

I did not find answers for this questions in the wiki :-)

best regards


-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-
Email: rs at runsolutions.com
tel: 625 40 32 08

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

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From brian at freeswitch.org  Tue Aug 25 07:16:04 2009
From: brian at freeswitch.org (Brian West)
Date: Tue, 25 Aug 2009 09:16:04 -0500
Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
In-Reply-To: <6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
	<191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
	<6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>
Message-ID: <5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org>

Try running it at the CLI and see if you see any errors.  Also please  
do not hijack threads.  The original thread "[Freeswitch-users] XML- 
RPC on different ip than 0.0.0.0" which was hijacked by clicking  
reply, changing the subject and clicking send.  Please in the future  
do not do that as it clutters up the threading and could get your  
query lost in the noise.

Thanks,
Brian

On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote:

> Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
> appears, if I use version 3.97 (older version), it does!.




From MPeace at edcogroupinc.com  Tue Aug 25 07:20:41 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 09:20:41 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>

I fired up Wireshark on each side and I can see the SIP register request coming from the laptop, the Freeswitch server replies with a Destination Unreachable (Port unreachable) message.

I rebooted the Server and now I get a "Registration error; 405 Method not allowed" on the softphone and the Wiresharp capture shows: "Status 405 Method Not Allowerd (0 bindings).

It does this even after I stop and restart the Freeswitch service.

Shouldn't all the ports be open if the firewall and SELunix are disabled?

Mike Peace
Network Analyst
EDCO, The Document People
Direct 417-447-3367

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 5:32 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


On Mon, Aug 24, 2009 at 2:49 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

It does the same thing, does anything get set during the install that would remember or cache the old network settings? I can access anything from the FS server on any of several networks and vice-versa but the SIP will not register, again no firewalls are upon any of the test hosts.

Doesn't make sense to me.


Time to bust out tcpdump and/or wireshark to make 100% certain you know what's happening with all those SIP packets.

Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:34 PM

To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.





On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

"Registration error: 408-Request Timeout"

Sorry for the the typo - 408 = timeout. (480 = temp unavail)

Try stopping iptables in Linux and try again. Sounds like something is interfering with your packets getting from here to there...
Try:

/etc/init.d/iptables stop

And then see if your packets can move again.

-MC



Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:18 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.





On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

I haven't changed any of the conf files.



What happens when you try to register? Do you get 480? (timeout) Or something else?
-MC



________________________________



EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/



Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From moises.silva at gmail.com  Tue Aug 25 07:24:50 2009
From: moises.silva at gmail.com (Moises Silva)
Date: Tue, 25 Aug 2009 10:24:50 -0400
Subject: [Freeswitch-users] Freeswitch & DAHDI Kernel drivers / Hardware
	questions
In-Reply-To: <DD5C7209-7041-4D21-869D-2F3BABEADC78@runsolutions.com>
References: <DD5C7209-7041-4D21-869D-2F3BABEADC78@runsolutions.com>
Message-ID: <c4d05cbe0908250724j407636e0o33f0df4bbfbb461@mail.gmail.com>

On Tue, Aug 25, 2009 at 9:35 AM, Raimund Sacherer <rs at runsolutions.com>wrote:

> Hello,
> i was reading through the openzap wiki page and searched the wiki, the only
> thing I found about dahdi is a note from January that freeswitch does not
> work with dahdi right now.
>
>
You did not search well enough.
http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 , read note 5 there.


> Is this current information? Can't I use dahdi kernel drivers for
> freeswitch? As the last update to the zaptel drivers is now nearly a year
> old (1.4.12) and development has, AFAIK, ceased and new stuff only gehts to
> dahdi, correct me please if I am wrong.
>
both dahdi and zaptel work for FreeSWITCH and no headers are needed, we have
our own.


>
> 00:0d.0 Network controller: Cologne Chip Designs GmbH ISDN network
> controller [HFC-PCI] (rev 02)
> Subsystem: Cologne Chip Designs GmbH ISDN Board
> Flags: bus master, medium devsel, latency 16, IRQ 10
> I/O ports at e000 [disabled] [size=8]
> Memory at ee001000 (32-bit, non-prefetchable) [size=256]
> Capabilities: [40] Power Management version 1
> Kernel modules: hisax
>
> with openZap? I read in the wiki (openZap.conf_Examples) something about
> HFC 4 BRI, but i do not know if this is the same as the above mentioned
> card.
>
>
> I have access to some pcmcia fxo, and pcmcia isdn cards, could they be
> used? I have as well an soekris board with two pcmcia slots, could i use
> this with an fxo pcmcia card to actually drive a small home - pbx with
> freeswitch?
>

I don't have experience with such hardware, cannot tell, may be someone else
around here may know.

-- 
Moises Silva
Software Developer
Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3
Canada
t. 1 905 474 1990 x 128 | e. moy at sangoma.com
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From rs at runsolutions.com  Tue Aug 25 07:35:05 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Tue, 25 Aug 2009 16:35:05 +0200
Subject: [Freeswitch-users] Freeswitch & DAHDI Kernel drivers / Hardware
	questions
In-Reply-To: <c4d05cbe0908250724j407636e0o33f0df4bbfbb461@mail.gmail.com>
References: <DD5C7209-7041-4D21-869D-2F3BABEADC78@runsolutions.com>
	<c4d05cbe0908250724j407636e0o33f0df4bbfbb461@mail.gmail.com>
Message-ID: <7CEF2F0A-0735-41E8-ABA5-FBCB2A0AC0DF@runsolutions.com>


On Aug 25, 2009, at 4:24 PM, Moises Silva wrote:

> On Tue, Aug 25, 2009 at 9:35 AM, Raimund Sacherer  
> <rs at runsolutions.com> wrote:
> Hello,
>
> i was reading through the openzap wiki page and searched the wiki,  
> the only thing I found about dahdi is a note from January that  
> freeswitch does not work with dahdi right now.
>
>
> You did not search well enough. http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 
>  , read note 5 there.
Thank you, i did not see any link to OpenZAP_OpenR2, but this page  
explains a lot for me!
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From pjintheusa at gmail.com  Tue Aug 25 07:35:45 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Tue, 25 Aug 2009 10:35:45 -0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <4A93EFE1.6010007@nowthor.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
	<4A93EFE1.6010007@nowthor.com>
Message-ID: <367751820908250735t35cd96f6kfcf46c2c39e8d4d@mail.gmail.com>

Take a look at http://jira.freeswitch.org/browse/FSCORE-422. This a
feature request I submitted. This problem it solves is different - but
the solution is the same. Perhaps you add your take to the comments
there.

On Tue, Aug 25, 2009 at 10:06 AM, Carlos S. Antunes<csa at nowthor.com> wrote:
> Max,
>
> I would like to see something similar too. For example, it would be
> wonderful if one could specify multiple gateways to try like this or
> something similar:
>
> <action application="bridge">
> ? ?<!-- set some kind of *bridge-specific* parameter/variable -->
> ? ?<gateway order=1 data="sofia/gateway/gw-1/1$1"/>
> ? ?<gateway order=1 data="sofia/gateway/gw-2/1$1"/>
> ? ?<gateway order=2 data="sofia/gateway/gw-3/1$1">
> ? ? ? ?<!-- set some kind of *gw-specific* parameter/variable -->
> ? ?</gateway>
> ? ?<gateway order=3 data="sofia/gateway/gw-4/1$1"/>
> </action>
>
> One would be able to avoid the "[]" and "{}" hacks and combine
> sequential and simultaneous trying of gateways.
>
> What do the developers think of this?
>
> Carlos
>
> Max Ivanov wrote:
>> Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
>> sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
>> ,
>> the whole sting is tooo long, is there any shorter way to write same thing? Like
>> "sofia/gateway/panas*/1000" will try all gateways matching the pattern.
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From anthony.minessale at gmail.com  Tue Aug 25 07:45:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Aug 2009 09:45:04 -0500
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <4A93EFE1.6010007@nowthor.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
	<4A93EFE1.6010007@nowthor.com>
Message-ID: <191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>

This suggestion violates the scope boundaries.

gateways are specific concept to mod_sofia so a <gateway> tag in <action>
(part of agnostic xml dialplan)
does not flow properly.

you can also use combinations of continue_on_fail and hangup_after bridge so
you can
just put each bridge statement in it's own action.


On Tue, Aug 25, 2009 at 9:06 AM, Carlos S. Antunes <csa at nowthor.com> wrote:

> Max,
>
> I would like to see something similar too. For example, it would be
> wonderful if one could specify multiple gateways to try like this or
> something similar:
>
> <action application="bridge">
>    <!-- set some kind of *bridge-specific* parameter/variable -->
>    <gateway order=1 data="sofia/gateway/gw-1/1$1"/>
>    <gateway order=1 data="sofia/gateway/gw-2/1$1"/>
>    <gateway order=2 data="sofia/gateway/gw-3/1$1">
>        <!-- set some kind of *gw-specific* parameter/variable -->
>    </gateway>
>    <gateway order=3 data="sofia/gateway/gw-4/1$1"/>
> </action>
>
> One would be able to avoid the "[]" and "{}" hacks and combine
> sequential and simultaneous trying of gateways.
>
> What do the developers think of this?
>
> Carlos
>
> Max Ivanov wrote:
> > Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
> >
> sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
> > ,
> > the whole sting is tooo long, is there any shorter way to write same
> thing? Like
> > "sofia/gateway/panas*/1000" will try all gateways matching the pattern.
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Tue Aug 25 08:00:27 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 11:00:27 -0400
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
	<15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>
	<15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com>
	<65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com>
Message-ID: <84597815-246D-45BB-9C9D-755C2C74D4C4@jerris.com>

I beleive this is following the right rfc rules for dialog matching.   
If it is not, please open up a bug on jira.freeswitch.org with  
references of what exactly is not right.

Mike

On Aug 25, 2009, at 2:51 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Hello Takeshi,
>
> Thanks for your hint... it worked out... so to be precise:
>
> VIA header of both INVITE and ACK messages MUST be identical  
> (IP:PORT + branch)... and you are right... it might not be according  
> to SIP specification. Anyhow, i get FS understand my ACK message.
>
>
> Finally, here is what i used and I'm getting some poor results ..  
> but this is another topic :)
>
>
> Thanks for your help.
> Tihomir.
>
>
> sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err  
> -r 1 -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
>
>
> <?xml version="1.0" encoding="ISO-8859-1" ?>
> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>
>
> <scenario name="Basic Sipstone UAC">
>   <send retrans="500" start_rtd="1" start_rtd="2">
>
>     <![CDATA[
>
>       INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>       Max-Forwards: 70
>       Contact: <sip:[field1]@[local_ip]>
>       From: [field1] <sip:[field1]@[local_ip]:[local_port]>;tag= 
> [call_number]
>       To: [service] <sip:[service]@[remote_ip]:[remote_port]>
>       Call-ID: [call_id]
>       CSeq: 1 INVITE
>       Content-Type: application/sdp
>       Content-Length: [len]
>
>       v=0
>       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>       s=-
>       c=IN IP[media_ip_type] [media_ip]
>       t=0 0
>       m=audio [media_port] RTP/AVP 0
>       a=rtpmap:0 PCMU/8000
>
>     ]]>
>   </send>
>
>   <recv response="100"
>         optional="true" rtd="1">
>   </recv>
>
>
>   <recv response="302" rtd="2">
>   </recv>
>
>   <send>
>     <![CDATA[
>
>       ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch= 
> [branch-3]
>       From: [field1] <sip:[field1]@1[local_ip]:[local_port]>;tag= 
> [call_number]
>       To: [service] <sip:[service]@[remote_ip]:[remote_port]> 
> [peer_tag_param]
>       Call-ID: [call_id]
>       CSeq: 1 ACK
>       Max-Forwards: 70
>       Content-Length: 0
>
>     ]]>
>   </send>
>
>   <!-- definition of the response time repartition table (unit is  
> ms)   -->
>   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
>
>   <!-- definition of the call length repartition table (unit is  
> ms)     -->
>   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
>
> </scenario>
>
>
>
> On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi at gmail.com 
> > wrote:
> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com 
> > wrote:
> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir  
> Culjaga<tculjaga at gmail.com> wrote:
> >>
> >> sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf  
> test.txt -m 1 -l
> >> 4000
> >> scenario file:      uac_redirect.xml
> >> FS dialplan:       public.xml
> >> SIP trace:          trace.log
> >
> > The Via definition in your SIPp scenario differs between the  
> INVITE and the ACK:
> >
> > INVITE:
> > Via: SIP/2.0/[transport] [local_ip];branch=[branch]
> >
> > ACK:
> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
> >
> >
> > In the INVITE, you are not adding the [local_port] as you do in  
> the ACK.
> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept  
> the ACK.
> > So it seems FS is not checking just the ACK's branch against the
> > INVITE's; it seems it is checking the whole Via header.
> > I don't know if this is in accordance to SIP specs.
> > Another thing, about the way you are calling SIPp: do no use "-sn  
> uac"
> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
> > means "use the internal (embedded) scenario named xxx". So this
> > conflicts with the other parameter "-sf" which specifies an external
> > profile.
>
> I mean, an external scenario (file).
>
>  It seems this doesn't cause any problem (probably because in
> > the sipp startup, -sf overrides -sn), but it is misleading.
> >
> > regards,
> > takeshi
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From sdjernes at gmail.com  Tue Aug 25 07:30:49 2009
From: sdjernes at gmail.com (Shawn L. Djernes)
Date: Tue, 25 Aug 2009 09:30:49 -0500
Subject: [Freeswitch-users] Group Call
Message-ID: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>

Hello,
I am trying to get group calling to work.  AKA: dial an extension (7300 in
this case) and it ring all phones at once.  We are moving a server from
Asterisk to FS

Here is the conf/directory/default.xml section:

        <group name="sdjernes">
                <users>
                        <user id="7301" type="pointer"/>
                        <user id="7302" type="pointer"/>
                        <user id="7303" type="pointer"/>
                        <user id="7304" type="pointer"/>
                </users>
        </group>

Here is the conf/dialplan/default/7300_group.xml extension statement:

     <extension name="Group 7300">
       <condition field="destination_number" expression="(7300)">
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" data="continue_on_fail=true"/>
         <action application="set" data="call_timeout=15"/>
         <action application="bridge"
data="{ignore_early_media=true}${group(call:sdjernes+A@${domain_name})}"/>
<!--         <action application="transfer" data="7301 XML default"/> -->
         <action application="hangup"/>
       </condition>
     </extension>

The log output shows the following:

2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3289 Channel
sofia/internal/7305 at ewr.
djernes.net entering state [received][100]
2009-08-25 14:27:45.282619 [DEBUG] switch_core_state_machine.c:398
(sofia/intern
al/7305 at ewr.djernes.net) Running State Change CS_NEW
2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3296 Remote SDP:
v=0
o=root 628695665 628695665 IN IP4 98.188.201.109
s=call
c=IN IP4 98.188.201.109
t=0 0
m=audio 61012 RTP/AVP 0 8 9 99 3 18 4 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:9 g722/8000
a=rtpmap:99 g726-32/8000
a=rtpmap:3 gsm/8000
a=rtpmap:18 g729/8000
a=rtpmap:4 g723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20

2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[CELT:114:48000:10]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[CELT:114:32000:10]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[SPEEX:98:8000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[SPEEX:98:8000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[SPEEX:98:8000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[G7221:115:32000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[G7221:107:16000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[G722:9:8000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
[pcmu:0
:8000:20]/[PCMU:0:8000:20]
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:2029 Set Codec
sofia/internal/73
05 at ewr.djernes.net PCMU/8000 20 ms 160 samples
2009-08-25 14:27:45.282619 [DEBUG] switch_core_state_machine.c:404
(sofia/intern
al/7305 at ewr.djernes.net) State NEW
2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload
to 10
1
2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3455
(sofia/internal/7305 at ewr.djernes
.net) State Change CS_NEW -> CS_INIT
2009-08-25 14:27:45.282619 [DEBUG] switch_core_session.c:932 Send signal
sofia/i
nternal/7305 at ewr.djernes.net [BREAK]
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
(sofia/intern
al/7305 at ewr.djernes.net) Running State Change CS_INIT
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:481
(sofia/intern
al/7305 at ewr.djernes.net) State INIT
2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:83
sofia/internal/7305 at ewr.djerne
s.net SOFIA INIT
2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:111
(sofia/internal/7305 at ewr.djer
nes.net) State Change CS_INIT -> CS_ROUTING
2009-08-25 14:27:45.287037 [DEBUG] switch_core_session.c:932 Send signal
sofia/i
nternal/7305 at ewr.djernes.net [BREAK]
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:481
(sofia/intern
al/7305 at ewr.djernes.net) State INIT going to sleep
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
(sofia/intern
al/7305 at ewr.djernes.net) Running State Change CS_ROUTING
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:484
(sofia/intern
al/7305 at ewr.djernes.net) State ROUTING
2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:130
sofia/internal/7305 at ewr.djern
es.net SOFIA ROUTING
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:78
sofia/internal
/7305 at ewr.djernes.net Standard ROUTING
2009-08-25 14:27:45.287037 [INFO] mod_dialplan_xml.c:315 Processing Sdjernes
TOI
 Phone->7300 in context default
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unloop]
continue
=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unloop]
${unroll_loo
ps}(true) =~ /^true$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unloop]
${sip_looped
_call}() =~ /^true$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->tod_example]
con
tinue=true
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [tod_example]
${strft
ime(%w)}(2) =~ /^([1-5])$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [tod_example]
${strft
ime(%H%M)}(1427) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Action set(open=true)
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->global-intercept
] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[global-intercept] de
stination_number(7300) =~ /^886$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->group-intercept]
 continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [group-intercept]
des
tination_number(7300) =~ /^\*8$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->intercept-ext] c
ontinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [intercept-ext]
desti
nation_number(7300) =~ /^\*\*(\d+)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->redial]
continue
=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [redial]
destination_
number(7300) =~ /^870$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->global]
continue
=true
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [global]
${call_debug
}(false) =~ /^true$/ break=never
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [global]
${sip_has_cr
ypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never
Dialplan: sofia/internal/7305 at ewr.djernes.net Absolute Condition [global]
Dialplan: sofia/internal/7305 at ewr.djernes.net Action
hash(insert/${domain_name}-
spymap/${caller_id_number}/${uuid})
Dialplan: sofia/internal/7305 at ewr.djernes.net Action
hash(insert/${domain_name}-
last_dial/${caller_id_number}/${destination_number})
Dialplan: sofia/internal/7305 at ewr.djernes.net Action
hash(insert/${domain_name}-
last_dial/global/${uuid})
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->snom-demo-2]
con
tinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [snom-demo-2]
destina
tion_number(7300) =~ /^9001$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->snom-demo-1]
con
tinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [snom-demo-1]
destina
tion_number(7300) =~ /^9000$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->eavesdrop]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [eavesdrop]
destinati
on_number(7300) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->eavesdrop]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [eavesdrop]
destinati
on_number(7300) =~ /^779$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->call_return]
con
tinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call_return]
destina
tion_number(7300) =~ /^\*69$|^869$|^lcr$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->del-group]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [del-group]
destinati
on_number(7300) =~ /^80(\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->add-group]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [add-group]
destinati
on_number(7300) =~ /^81(\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->call-group-simo]
 continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call-group-simo]
des
tination_number(7300) =~ /^82(\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->call-group-order
] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[call-group-order] de
stination_number(7300) =~ /^83(\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->extension-interc
om] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[extension-intercom]
destination_number(7300) =~ /^8(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->Local_Extension]
 continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [Local_Extension]
des
tination_number(7300) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->group_dial_sales
] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[group_dial_sales] de
stination_number(7300) =~ /^2000$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->group_dial_suppo
rt] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[group_dial_support]
destination_number(7300) =~ /^2001$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->group_dial_billi
ng] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[group_dial_billing]
destination_number(7300) =~ /^2002$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->operator]
contin
ue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [operator]
destinatio
n_number(7300) =~ /^(operator|0)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->vmain]
continue=
false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [vmain]
destination_n
umber(7300) =~ /^vmain$|^4000$|^\*98$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->sip_uri]
continu
e=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [sip_uri]
destination
_number(7300) =~ /^sip:(.*)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->nb_conferences]
continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [nb_conferences]
dest
ination_number(7300) =~ /^(30\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->wb_conferences]
continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [wb_conferences]
dest
ination_number(7300) =~ /^(31\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->uwb_conferences]
 continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [uwb_conferences]
des
tination_number(7300) =~ /^(32\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->cdquality_confer
ences] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[cdquality_conference
s] destination_number(7300) =~ /^(33\d{2})$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->freeswitch_publi
c_conf_via_sip] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[freeswitch_public_co
nf_via_sip] destination_number(7300) =~ /^9(888|1616|3232)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->mad_boss_interco
m] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[mad_boss_intercom] d
estination_number(7300) =~ /^0911$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->mad_boss_interco
m] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[mad_boss_intercom] d
estination_number(7300) =~ /^0912$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->mad_boss]
contin
ue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [mad_boss]
destinatio
n_number(7300) =~ /^0913$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->ivr_demo]
contin
ue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ivr_demo]
destinatio
n_number(7300) =~ /^5000$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->dynamic_conferen
ce] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[dynamic_conference]
destination_number(7300) =~ /^5001$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->rtp_multicast_pa
ge] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[rtp_multicast_page]
destination_number(7300) =~ /^pagegroup$|^7243$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
continue=f
alse
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
destination_nu
mber(7300) =~ /^5900$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
continue
=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
destination_
number(7300) =~ /^5901$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
continue=f
alse
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [park]
source(mod_sof
ia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
destination_nu
mber(7300) =~ /park\+(\d+)/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
continue
=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unpark]
source(mod_s
ofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
destination_
number(7300) =~ /^parking$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
continue=f
alse
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [park]
source(mod_sof
ia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
destination_nu
mber(7300) =~ /callpark/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
continue
=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unpark]
source(mod_s
ofia) =~ /mod_sofia/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
destination_
number(7300) =~ /pickup/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->wait]
continue=f
alse
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [wait]
destination_nu
mber(7300) =~ /^wait$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->fax_receive]
con
tinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [fax_receive]
destina
tion_number(7300) =~ /^9978$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->fax_transmit] co
ntinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [fax_transmit]
destin
ation_number(7300) =~ /^9979$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->ringback_180] co
ntinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ringback_180]
destin
ation_number(7300) =~ /^9980$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->ringback_183_uk_
ring] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[ringback_183_uk_ring
] destination_number(7300) =~ /^9981$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->ringback_183_mus
ic_ring] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[ringback_183_music_r
ing] destination_number(7300) =~ /^9982$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->ringback_post_an
swer_uk_ring] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[ringback_post_answer
_uk_ring] destination_number(7300) =~ /^9983$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->ringback_post_an
swer_music] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[ringback_post_answer
_music] destination_number(7300) =~ /^9984$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->ClueCon]
continu
e=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ClueCon]
destination
_number(7300) =~ /^9991$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->show_info]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [show_info]
destinati
on_number(7300) =~ /^9992$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->video_record] co
ntinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [video_record]
destin
ation_number(7300) =~ /^9993$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->video_playback]
continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [video_playback]
dest
ination_number(7300) =~ /^9994$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->delay_echo]
cont
inue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [delay_echo]
destinat
ion_number(7300) =~ /^9995$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->echo]
continue=f
alse
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [echo]
destination_nu
mber(7300) =~ /^9996$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->milliwatt]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [milliwatt]
destinati
on_number(7300) =~ /^9997$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->tone_stream]
con
tinue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [tone_stream]
destina
tion_number(7300) =~ /^9998$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
[default->zrtp_enrollement
] continue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
[zrtp_enrollement] de
stination_number(7300) =~ /^9787$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->hold_music]
cont
inue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [hold_music]
destinat
ion_number(7300) =~ /^9999$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->Rednote]
continu
e=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [Rednote]
destination
_number(7300) =~ /^0220(\d+)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->carmickle]
conti
nue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [carmickle]
destinati
on_number(7300) =~ /^0352(\d+)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->call_jason]
cont
inue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call_jason]
destinat
ion_number(7300) =~ /^059(\d+)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->TOI]
continue=fa
lse
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [TOI]
destination_num
ber(7300) =~ /^08740(\d+)$/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->Group 7300]
cont
inue=false
Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [Group 7300]
destinat
ion_number(7300) =~ /(7300)/ break=on-false
Dialplan: sofia/internal/7305 at ewr.djernes.net Action
set(hangup_after_bridge=tru
e)
Dialplan: sofia/internal/7305 at ewr.djernes.net Action
set(continue_on_fail=true)
Dialplan: sofia/internal/7305 at ewr.djernes.net Action set(call_timeout=15)
Dialplan: sofia/internal/7305 at ewr.djernes.net Action
bridge({ignore_early_media=
true}${group_call(sdjernes+A@${domain_name})})
Dialplan: sofia/internal/7305 at ewr.djernes.net Action hangup()
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:114
(sofia/intern
al/7305 at ewr.djernes.net) State Change CS_ROUTING -> CS_EXECUTE
2009-08-25 14:27:45.287037 [DEBUG] switch_core_session.c:932 Send signal
sofia/i
nternal/7305 at ewr.djernes.net [BREAK]
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:484
(sofia/intern
al/7305 at ewr.djernes.net) State ROUTING going to sleep
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
(sofia/intern
al/7305 at ewr.djernes.net) Running State Change CS_EXECUTE
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:491
(sofia/intern
al/7305 at ewr.djernes.net) State EXECUTE
2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:173
sofia/internal/7305 at ewr.djern
es.net SOFIA EXECUTE
2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:151
sofia/interna
l/7305 at ewr.djernes.net Standard EXECUTE
EXECUTE sofia/internal/7305 at ewr.djernes.net set(open=true)
2009-08-25 14:27:45.287037 [DEBUG] mod_dptools.c:748
sofia/internal/7305 at ewr.dje
rnes.net SET [open]=[true]
EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-s
pymap/7305/733dbed2-9183-11de-95f0-07acd7038fc1)
EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-l
ast_dial/7305/7300)
EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-l
ast_dial/global/733dbed2-9183-11de-95f0-07acd7038fc1)
EXECUTE sofia/internal/7305 at ewr.djernes.net set(hangup_after_bridge=true)
2009-08-25 14:27:45.291899 [DEBUG] mod_dptools.c:748
sofia/internal/7305 at ewr.dje
rnes.net SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/7305 at ewr.djernes.net set(continue_on_fail=true)
2009-08-25 14:27:45.291899 [DEBUG] mod_dptools.c:748
sofia/internal/7305 at ewr.dje
rnes.net SET [continue_on_fail]=[true]
EXECUTE sofia/internal/7305 at ewr.djernes.net set(call_timeout=15)
2009-08-25 14:27:45.294755 [DEBUG] mod_dptools.c:748
sofia/internal/7305 at ewr.dje
rnes.net SET [call_timeout]=[15]
EXECUTE sofia/internal/7305 at ewr.djernes.netbridge({ignore_early_media=true})
2009-08-25 14:27:45.318579 [WARNING] switch_ivr_originate.c:1001 No
origination
URL specified!
2009-08-25 14:27:45.318579 [DEBUG] switch_ivr_originate.c:2138 Originate
Resulte
d in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
2009-08-25 14:27:45.318579 [INFO] mod_dptools.c:2093 Originate Failed.
 Cause: D
ESTINATION_OUT_OF_ORDER
2009-08-25 14:27:45.318579 [DEBUG] mod_dptools.c:2120 Continue on fail
[true]:
Cause: DESTINATION_OUT_OF_ORDER
EXECUTE sofia/internal/7305 at ewr.djernes.net hangup()
2009-08-25 14:27:45.318579 [NOTICE] mod_dptools.c:633 Hangup
sofia/internal/7305
@ewr.djernes.net [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-25 14:27:45.318579 [DEBUG] switch_channel.c:1683 Send signal
sofia/inter
nal/7305 at ewr.djernes.net [KILL]
2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:932 Send signal
sofia/i
nternal/7305 at ewr.djernes.net [BREAK]
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:491
(sofia/intern
al/7305 at ewr.djernes.net) State EXECUTE going to sleep
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:398
(sofia/intern
al/7305 at ewr.djernes.net) Running State Change CS_HANGUP
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:434
(sofia/intern
al/7305 at ewr.djernes.net) State HANGUP
2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:338 Channel
sofia/internal/7305 at e
wr.djernes.net hanging up, cause: NORMAL_CLEARING
2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:417 Responding to INVITE
with: 48
0
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:46
sofia/internal
/7305 at ewr.djernes.net Standard HANGUP, cause: NORMAL_CLEARING
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:434
(sofia/intern
al/7305 at ewr.djernes.net) State HANGUP going to sleep
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:476
(sofia/intern
al/7305 at ewr.djernes.net) State Change CS_HANGUP -> CS_REPORTING
2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:932 Send signal
sofia/i
nternal/7305 at ewr.djernes.net [BREAK]
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:398
(sofia/intern
al/7305 at ewr.djernes.net) Running State Change CS_REPORTING
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:612
(sofia/intern
al/7305 at ewr.djernes.net) State REPORTING
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:53
sofia/internal
/7305 at ewr.djernes.net Standard REPORTING, cause: NORMAL_CLEARING
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:612
(sofia/intern
al/7305 at ewr.djernes.net) State REPORTING going to sleep
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:411
(sofia/intern
al/7305 at ewr.djernes.net) State Change CS_REPORTING -> CS_DESTROY
2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:1068 Session 117
(sofia
/internal/7305 at ewr.djernes.net) Locked, Waiting on external entities
2009-08-25 14:27:45.318579 [NOTICE] switch_core_session.c:1086 Session 117
(sofi
a/internal/7305 at ewr.djernes.net) Ended
2009-08-25 14:27:45.318579 [NOTICE] switch_core_session.c:1088 Close Channel
sof
ia/internal/7305 at ewr.djernes.net [CS_DESTROY]
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:564
(sofia/intern
al/7305 at ewr.djernes.net) State DESTROY
2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:255
sofia/internal/7305 at ewr.djern
es.net SOFIA DESTROY
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:60
sofia/internal
/7305 at ewr.djernes.net Standard DESTROY
2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:564
(sofia/intern
al/7305 at ewr.djernes.net) State DESTROY going to sleep


Any Help would be much appreciated.
-- 
Shawn L. Djernes
SD Consulting
shawn at djernes.org | sdjernes at gmail.com
MSN: wizardwlf at hotmail.com
402.345.7734 | 402.350.6973 Cell
-------------- next part --------------
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From mike at jerris.com  Tue Aug 25 08:29:26 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 11:29:26 -0400
Subject: [Freeswitch-users] reload user data
In-Reply-To: <27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>
References: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>
	<C6B90B49.5D5AC%krice@freeswitch.org>
	<27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>
Message-ID: <9FE7AA03-C09B-4DB3-A15A-CF99D2052726@jerris.com>

You can also do xml config hooks in perl and some of the other  
embedded languages.

Mike

On Aug 25, 2009, at 5:22 AM, Juan Backson wrote:

> Hi Ken,
>
> xml_curl is a great idea.  Is there anyway to not having to setup  
> another HTTP server?  For instance, can I have freeswitch to call an  
> api or call a lua or php or c script that will return the xml  
> response?  That way, I don't need to maintain yet another service.
>
> Thanks,
> JB
>
> On Tue, Aug 25, 2009 at 4:30 PM, Ken Rice <krice at freeswitch.org>  
> wrote:
> You just need to reloadxml you don?t have to restart the whole  
> thing. You can also use xml_curl to feed the users from a database  
> see my contrib directory (contrib/swk) for some example scripts and  
> db code
>
>
> From: Juan Backson <juanbackson at gmail.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 25 Aug 2009 16:21:58 +0800
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] reload user data
>
>
> Hello,
>
> I would like to dynamically add user to freeswitch.  If I add a new  
> file to the directory dir, is there anyway to have freeswitch to  
> read the new user xml file without having to restart freeswitch?
>
> Other than using flat file, is there anyway to add user to  
> freeswitch user api command?
>
> Thanks,
> JB

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From mike at jerris.com  Tue Aug 25 08:31:49 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 11:31:49 -0400
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
Message-ID: <2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>

Actually in this case, we are bound to one thread in sofia.

Mike

On Aug 25, 2009, at 9:47 AM, Giovanni Maruzzelli wrote:

> is a heavely multithreaded software, it benefits from number of CPUs
> (or cores), RAM, and heavy duty kernel features (found in 64bit
> kernels)
>
> put all accesses on ramdisk, leave out the modules you don't use...
>
> experiment, test, and find the best for your specific application/ 
> workload
>
> test not only with sipp, but with real load too (often they're very  
> different)
>
> -gm
>




From msc at freeswitch.org  Tue Aug 25 08:55:29 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 08:55:29 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
Message-ID: <87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>

On Tue, Aug 25, 2009 at 7:20 AM, Mike Peace <MPeace at edcogroupinc.com> wrote:

>  I fired up Wireshark on each side and I can see the SIP register request
> coming from the laptop, the Freeswitch server replies with a Destination
> Unreachable (Port unreachable) message.
>
>
>
> I rebooted the Server and now I get a ?Registration error; 405 Method not
> allowed? on the softphone and the Wiresharp capture shows: ?Status 405
> Method Not Allowerd (0 bindings).
>
>
>
Hmm... can you fire up FS and then do "sofia status" and tell us if the
internal profile looks okay? Also, do "sofia status profile internal" and
capture the output. Something fishy is going on...
-MC

> It does this even after I stop and restart the Freeswitch service.
>
>
>
> Shouldn?t all the ports be open if the firewall and SELunix are disabled?
>
>
>
> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 5:32 PM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:49 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> It does the same thing, does anything get set during the install that would
> remember or cache the old network settings? I can access anything from the
> FS server on any of several networks and vice-versa but the SIP will not
> register, again no firewalls are upon any of the test hosts.
>
> Doesn?t make sense to me.
>
>
>
> Time to bust out tcpdump and/or wireshark to make 100% certain you know
> what's happening with all those SIP packets.
>
>   *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:34 PM
>
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> ?Registration error: 408-Request Timeout?
>
> Sorry for the the typo - 408 = timeout. (480 = temp unavail)
>
> Try stopping iptables in Linux and try again. Sounds like something is
> interfering with your packets getting from here to there...
> Try:
>
> /etc/init.d/iptables stop
>
> And then see if your packets can move again.
>
> -MC
>
>
>
> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:18 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> I haven?t changed any of the conf files.
>
>
>
> What happens when you try to register? Do you get 480? (timeout) Or
> something else?
> -MC
>
>
>  ------------------------------
>
>
>
> *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>
>
> Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>  ------------------------------
>
>
>
> *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>
>
> Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
>
>  *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>  Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Tue Aug 25 08:59:28 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 08:59:28 -0700
Subject: [Freeswitch-users] using sipp to test call transfers
In-Reply-To: <84D97940-68D0-402F-A7E2-B5ADD287DC73@runsolutions.com>
References: <84D97940-68D0-402F-A7E2-B5ADD287DC73@runsolutions.com>
Message-ID: <87f2f3b90908250859w5f5c8f25qd1ec1a83c9a706fd@mail.gmail.com>

On Tue, Aug 25, 2009 at 2:06 AM, Raimund Sacherer <rs at runsolutions.com>wrote:

> Hi List,
>
> i have some scripts to test our lab:
>
> I have scripts to create sipp instances which act as individual
> agents, which log on, take a random amount of calls, log off, wait a
> bit, log on again, etc.
> I have scripts to generate call traffic for our queues to saturate them
>
> But what I have difficulties is how to create a good scenario xml file
> for call transfers? Has anybody here a sample for me to sink my teeth
> in?


Start with this...
http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations

-MC


>
>
> thanks and best regards
>
> --
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Aug 25 09:05:14 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 09:05:14 -0700
Subject: [Freeswitch-users] Freeswitch & DAHDI Kernel drivers / Hardware
	questions
In-Reply-To: <7CEF2F0A-0735-41E8-ABA5-FBCB2A0AC0DF@runsolutions.com>
References: <DD5C7209-7041-4D21-869D-2F3BABEADC78@runsolutions.com>
	<c4d05cbe0908250724j407636e0o33f0df4bbfbb461@mail.gmail.com>
	<7CEF2F0A-0735-41E8-ABA5-FBCB2A0AC0DF@runsolutions.com>
Message-ID: <87f2f3b90908250905y4516550ag11b92d0642179f04@mail.gmail.com>

On Tue, Aug 25, 2009 at 7:35 AM, Raimund Sacherer <rs at runsolutions.com>wrote:

>
> On Aug 25, 2009, at 4:24 PM, Moises Silva wrote:
>
> On Tue, Aug 25, 2009 at 9:35 AM, Raimund Sacherer <rs at runsolutions.com>wrote:
>
>> Hello,
>> i was reading through the openzap wiki page and searched the wiki, the
>> only thing I found about dahdi is a note from January that freeswitch does
>> not work with dahdi right now.
>>
>>
> You did not search well enough.
> http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 , read note 5 there.
>
> Thank you, i did not see any link to OpenZAP_OpenR2, but this page explains
> a lot for me!
>

FYI,
I added a link to the R2 page on the main OpenZAP page.
-MC
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From msc at freeswitch.org  Tue Aug 25 09:11:51 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 09:11:51 -0700
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
Message-ID: <87f2f3b90908250911m31ef0914t136f7511b59ecb79@mail.gmail.com>

> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
> sense to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
> will FS perofomr drastically better 20%+ ?
>

If you really want to get on the same page as the developers then get the
64bit CentOS 5.3 loaded on your machine. Also, we've seen reports of really
weird things happening when people run 32bit OS on 64bit hardware. That's a
big no-no. :)

Let us know how it goes.
-MC
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From msc at freeswitch.org  Tue Aug 25 09:14:15 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 09:14:15 -0700
Subject: [Freeswitch-users] reload user data
In-Reply-To: <9FE7AA03-C09B-4DB3-A15A-CF99D2052726@jerris.com>
References: <27c25bc40908250121w461d1cb3v1b671cd72d5c6a8e@mail.gmail.com>
	<C6B90B49.5D5AC%krice@freeswitch.org>
	<27c25bc40908250222x2eed1c4r79af7853088cd6b3@mail.gmail.com>
	<9FE7AA03-C09B-4DB3-A15A-CF99D2052726@jerris.com>
Message-ID: <87f2f3b90908250914r4a864f25o677e1735d1856ff4@mail.gmail.com>

On Tue, Aug 25, 2009 at 8:29 AM, Michael Jerris <mike at jerris.com> wrote:

> You can also do xml config hooks in perl and some of the other embedded
> languages.
>

FYI, here's a page to go you started:
http://wiki.freeswitch.org/wiki/Mod_perl_and_Generating_XML

-MC
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From tculjaga at gmail.com  Tue Aug 25 09:34:42 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 18:34:42 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
Message-ID: <65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>

Exactly... the scenario i use seems operating on a single thread... why is
that ? can it be changed?

T.

On Tue, Aug 25, 2009 at 5:31 PM, Michael Jerris <mike at jerris.com> wrote:

> Actually in this case, we are bound to one thread in sofia.
>
> Mike
>
> On Aug 25, 2009, at 9:47 AM, Giovanni Maruzzelli wrote:
>
> > is a heavely multithreaded software, it benefits from number of CPUs
> > (or cores), RAM, and heavy duty kernel features (found in 64bit
> > kernels)
> >
> > put all accesses on ramdisk, leave out the modules you don't use...
> >
> > experiment, test, and find the best for your specific application/
> > workload
> >
> > test not only with sipp, but with real load too (often they're very
> > different)
> >
> > -gm
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Tue Aug 25 09:35:36 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 18:35:36 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <87f2f3b90908250911m31ef0914t136f7511b59ecb79@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<87f2f3b90908250911m31ef0914t136f7511b59ecb79@mail.gmail.com>
Message-ID: <65d96fc80908250935j4be3574ala817f9823204b462@mail.gmail.com>

well :) ... this is something we are going to change tomorrow.... of course
will let you posted.

T.

On Tue, Aug 25, 2009 at 6:11 PM, Michael Collins <msc at freeswitch.org> wrote:

>
> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
>> sense to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
>> will FS perofomr drastically better 20%+ ?
>>
>
> If you really want to get on the same page as the developers then get the
> 64bit CentOS 5.3 loaded on your machine. Also, we've seen reports of really
> weird things happening when people run 32bit OS on 64bit hardware. That's a
> big no-no. :)
>
> Let us know how it goes.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Tue Aug 25 09:41:29 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Tue, 25 Aug 2009 13:41:29 -0300
Subject: [Freeswitch-users] Group Call
In-Reply-To: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
Message-ID: <ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>

According to example dialplan you hould do this:

<action application="bridge" data="{ignore_early_media=true}${group(call:$1@
${domain_name})}"/>

According to wiki:

group,[insert|delete|call]:<group name>:<url>,group [insert|delete|call


Only difference I see here is the "+A" on your bridge statement.


jmesquita


On Tue, Aug 25, 2009 at 11:30 AM, Shawn L. Djernes <sdjernes at gmail.com>wrote:

> Hello,
> I am trying to get group calling to work.  AKA: dial an extension (7300 in
> this case) and it ring all phones at once.  We are moving a server from
> Asterisk to FS
>
> Here is the conf/directory/default.xml section:
>
>         <group name="sdjernes">
>                 <users>
>                         <user id="7301" type="pointer"/>
>                         <user id="7302" type="pointer"/>
>                         <user id="7303" type="pointer"/>
>                         <user id="7304" type="pointer"/>
>                 </users>
>         </group>
>
> Here is the conf/dialplan/default/7300_group.xml extension statement:
>
>      <extension name="Group 7300">
>        <condition field="destination_number" expression="(7300)">
>          <action application="set" data="hangup_after_bridge=true"/>
>          <action application="set" data="continue_on_fail=true"/>
>          <action application="set" data="call_timeout=15"/>
>          <action application="bridge"
> data="{ignore_early_media=true}${group(call:sdjernes+A@${domain_name})}"/>
> <!--         <action application="transfer" data="7301 XML default"/> -->
>          <action application="hangup"/>
>        </condition>
>      </extension>
>
> The log output shows the following:
>
> 2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3289 Channel
> sofia/internal/7305 at ewr.
> djernes.net entering state [received][100]
> 2009-08-25 14:27:45.282619 [DEBUG] switch_core_state_machine.c:398
> (sofia/intern
> al/7305 at ewr.djernes.net) Running State Change CS_NEW
> 2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3296 Remote SDP:
> v=0
> o=root 628695665 628695665 IN IP4 98.188.201.109
> s=call
> c=IN IP4 98.188.201.109
> t=0 0
> m=audio 61012 RTP/AVP 0 8 9 99 3 18 4 101
> a=rtpmap:0 pcmu/8000
> a=rtpmap:8 pcma/8000
> a=rtpmap:9 g722/8000
> a=rtpmap:99 g726-32/8000
> a=rtpmap:3 gsm/8000
> a=rtpmap:18 g729/8000
> a=rtpmap:4 g723/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
>
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[CELT:114:48000:10]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[CELT:114:32000:10]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[SPEEX:98:8000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[SPEEX:98:8000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[SPEEX:98:8000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[G7221:115:32000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[G7221:107:16000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[G722:9:8000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
> [pcmu:0
> :8000:20]/[PCMU:0:8000:20]
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:2029 Set Codec
> sofia/internal/73
> 05 at ewr.djernes.net PCMU/8000 20 ms 160 samples
> 2009-08-25 14:27:45.282619 [DEBUG] switch_core_state_machine.c:404
> (sofia/intern
> al/7305 at ewr.djernes.net) State NEW
> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload
> to 10
> 1
> 2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3455
> (sofia/internal/7305 at ewr.djernes
> .net) State Change CS_NEW -> CS_INIT
> 2009-08-25 14:27:45.282619 [DEBUG] switch_core_session.c:932 Send signal
> sofia/i
> nternal/7305 at ewr.djernes.net [BREAK]
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
> (sofia/intern
> al/7305 at ewr.djernes.net) Running State Change CS_INIT
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:481
> (sofia/intern
> al/7305 at ewr.djernes.net) State INIT
> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:83
> sofia/internal/7305 at ewr.djerne
> s.net SOFIA INIT
> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:111
> (sofia/internal/7305 at ewr.djer
> nes.net) State Change CS_INIT -> CS_ROUTING
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_session.c:932 Send signal
> sofia/i
> nternal/7305 at ewr.djernes.net [BREAK]
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:481
> (sofia/intern
> al/7305 at ewr.djernes.net) State INIT going to sleep
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
> (sofia/intern
> al/7305 at ewr.djernes.net) Running State Change CS_ROUTING
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:484
> (sofia/intern
> al/7305 at ewr.djernes.net) State ROUTING
> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:130
> sofia/internal/7305 at ewr.djern
> es.net SOFIA ROUTING
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:78
> sofia/internal
> /7305 at ewr.djernes.net Standard ROUTING
> 2009-08-25 14:27:45.287037 [INFO] mod_dialplan_xml.c:315 Processing
> Sdjernes TOI
>  Phone->7300 in context default
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unloop]
> continue
> =false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unloop]
> ${unroll_loo
> ps}(true) =~ /^true$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unloop]
> ${sip_looped
> _call}() =~ /^true$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->tod_example] con
> tinue=true
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [tod_example]
> ${strft
> ime(%w)}(2) =~ /^([1-5])$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [tod_example]
> ${strft
> ime(%H%M)}(1427) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action set(open=true)
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->global-intercept
> ] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [global-intercept] de
> stination_number(7300) =~ /^886$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->group-intercept]
>  continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [group-intercept] des
> tination_number(7300) =~ /^\*8$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->intercept-ext] c
> ontinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [intercept-ext]
> desti
> nation_number(7300) =~ /^\*\*(\d+)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->redial]
> continue
> =false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [redial]
> destination_
> number(7300) =~ /^870$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->global]
> continue
> =true
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [global]
> ${call_debug
> }(false) =~ /^true$/ break=never
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [global]
> ${sip_has_cr
> ypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
> break=never
> Dialplan: sofia/internal/7305 at ewr.djernes.net Absolute Condition [global]
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
> hash(insert/${domain_name}-
> spymap/${caller_id_number}/${uuid})
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
> hash(insert/${domain_name}-
> last_dial/${caller_id_number}/${destination_number})
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
> hash(insert/${domain_name}-
> last_dial/global/${uuid})
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->snom-demo-2] con
> tinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [snom-demo-2]
> destina
> tion_number(7300) =~ /^9001$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->snom-demo-1] con
> tinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [snom-demo-1]
> destina
> tion_number(7300) =~ /^9000$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->eavesdrop]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [eavesdrop]
> destinati
> on_number(7300) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->eavesdrop]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [eavesdrop]
> destinati
> on_number(7300) =~ /^779$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->call_return] con
> tinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call_return]
> destina
> tion_number(7300) =~ /^\*69$|^869$|^lcr$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->del-group]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [del-group]
> destinati
> on_number(7300) =~ /^80(\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->add-group]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [add-group]
> destinati
> on_number(7300) =~ /^81(\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->call-group-simo]
>  continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [call-group-simo] des
> tination_number(7300) =~ /^82(\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->call-group-order
> ] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [call-group-order] de
> stination_number(7300) =~ /^83(\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->extension-interc
> om] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [extension-intercom]
> destination_number(7300) =~ /^8(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->Local_Extension]
>  continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [Local_Extension] des
> tination_number(7300) =~ /^(10[01][0-9])$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->group_dial_sales
> ] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [group_dial_sales] de
> stination_number(7300) =~ /^2000$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->group_dial_suppo
> rt] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [group_dial_support]
> destination_number(7300) =~ /^2001$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->group_dial_billi
> ng] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [group_dial_billing]
> destination_number(7300) =~ /^2002$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->operator]
> contin
> ue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [operator]
> destinatio
> n_number(7300) =~ /^(operator|0)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->vmain]
> continue=
> false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [vmain]
> destination_n
> umber(7300) =~ /^vmain$|^4000$|^\*98$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->sip_uri]
> continu
> e=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [sip_uri]
> destination
> _number(7300) =~ /^sip:(.*)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->nb_conferences]
> continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [nb_conferences] dest
> ination_number(7300) =~ /^(30\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->wb_conferences]
> continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [wb_conferences] dest
> ination_number(7300) =~ /^(31\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->uwb_conferences]
>  continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [uwb_conferences] des
> tination_number(7300) =~ /^(32\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->cdquality_confer
> ences] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [cdquality_conference
> s] destination_number(7300) =~ /^(33\d{2})$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->freeswitch_publi
> c_conf_via_sip] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [freeswitch_public_co
> nf_via_sip] destination_number(7300) =~ /^9(888|1616|3232)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->mad_boss_interco
> m] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [mad_boss_intercom] d
> estination_number(7300) =~ /^0911$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->mad_boss_interco
> m] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [mad_boss_intercom] d
> estination_number(7300) =~ /^0912$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->mad_boss]
> contin
> ue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [mad_boss]
> destinatio
> n_number(7300) =~ /^0913$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->ivr_demo]
> contin
> ue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ivr_demo]
> destinatio
> n_number(7300) =~ /^5000$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->dynamic_conferen
> ce] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [dynamic_conference]
> destination_number(7300) =~ /^5001$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->rtp_multicast_pa
> ge] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [rtp_multicast_page]
> destination_number(7300) =~ /^pagegroup$|^7243$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
> continue=f
> alse
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
> destination_nu
> mber(7300) =~ /^5900$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
> continue
> =false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
> destination_
> number(7300) =~ /^5901$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
> continue=f
> alse
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [park]
> source(mod_sof
> ia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
> destination_nu
> mber(7300) =~ /park\+(\d+)/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
> continue
> =false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unpark]
> source(mod_s
> ofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
> destination_
> number(7300) =~ /^parking$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
> continue=f
> alse
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [park]
> source(mod_sof
> ia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
> destination_nu
> mber(7300) =~ /callpark/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
> continue
> =false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unpark]
> source(mod_s
> ofia) =~ /mod_sofia/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
> destination_
> number(7300) =~ /pickup/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->wait]
> continue=f
> alse
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [wait]
> destination_nu
> mber(7300) =~ /^wait$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->fax_receive] con
> tinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [fax_receive]
> destina
> tion_number(7300) =~ /^9978$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->fax_transmit] co
> ntinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [fax_transmit]
> destin
> ation_number(7300) =~ /^9979$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->ringback_180] co
> ntinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ringback_180]
> destin
> ation_number(7300) =~ /^9980$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->ringback_183_uk_
> ring] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [ringback_183_uk_ring
> ] destination_number(7300) =~ /^9981$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->ringback_183_mus
> ic_ring] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [ringback_183_music_r
> ing] destination_number(7300) =~ /^9982$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->ringback_post_an
> swer_uk_ring] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [ringback_post_answer
> _uk_ring] destination_number(7300) =~ /^9983$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->ringback_post_an
> swer_music] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [ringback_post_answer
> _music] destination_number(7300) =~ /^9984$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->ClueCon]
> continu
> e=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ClueCon]
> destination
> _number(7300) =~ /^9991$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->show_info]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [show_info]
> destinati
> on_number(7300) =~ /^9992$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->video_record] co
> ntinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [video_record]
> destin
> ation_number(7300) =~ /^9993$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->video_playback]
> continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [video_playback] dest
> ination_number(7300) =~ /^9994$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->delay_echo] cont
> inue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [delay_echo]
> destinat
> ion_number(7300) =~ /^9995$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->echo]
> continue=f
> alse
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [echo]
> destination_nu
> mber(7300) =~ /^9996$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->milliwatt]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [milliwatt]
> destinati
> on_number(7300) =~ /^9997$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->tone_stream] con
> tinue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [tone_stream]
> destina
> tion_number(7300) =~ /^9998$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->zrtp_enrollement
> ] continue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
> [zrtp_enrollement] de
> stination_number(7300) =~ /^9787$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->hold_music] cont
> inue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [hold_music]
> destinat
> ion_number(7300) =~ /^9999$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->Rednote]
> continu
> e=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [Rednote]
> destination
> _number(7300) =~ /^0220(\d+)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->carmickle]
> conti
> nue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [carmickle]
> destinati
> on_number(7300) =~ /^0352(\d+)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
> [default->call_jason] cont
> inue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call_jason]
> destinat
> ion_number(7300) =~ /^059(\d+)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->TOI]
> continue=fa
> lse
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [TOI]
> destination_num
> ber(7300) =~ /^08740(\d+)$/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->Group
> 7300] cont
> inue=false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [Group 7300]
> destinat
> ion_number(7300) =~ /(7300)/ break=on-false
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
> set(hangup_after_bridge=tru
> e)
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
> set(continue_on_fail=true)
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action set(call_timeout=15)
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
> bridge({ignore_early_media=
> true}${group_call(sdjernes+A@${domain_name})})
> Dialplan: sofia/internal/7305 at ewr.djernes.net Action hangup()
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:114
> (sofia/intern
> al/7305 at ewr.djernes.net) State Change CS_ROUTING -> CS_EXECUTE
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_session.c:932 Send signal
> sofia/i
> nternal/7305 at ewr.djernes.net [BREAK]
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:484
> (sofia/intern
> al/7305 at ewr.djernes.net) State ROUTING going to sleep
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
> (sofia/intern
> al/7305 at ewr.djernes.net) Running State Change CS_EXECUTE
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:491
> (sofia/intern
> al/7305 at ewr.djernes.net) State EXECUTE
> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:173
> sofia/internal/7305 at ewr.djern
> es.net SOFIA EXECUTE
> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:151
> sofia/interna
> l/7305 at ewr.djernes.net Standard EXECUTE
> EXECUTE sofia/internal/7305 at ewr.djernes.net set(open=true)
> 2009-08-25 14:27:45.287037 [DEBUG] mod_dptools.c:748
> sofia/internal/7305 at ewr.dje
> rnes.net SET [open]=[true]
> EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-s
> pymap/7305/733dbed2-9183-11de-95f0-07acd7038fc1)
> EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-l
> ast_dial/7305/7300)
> EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-l
> ast_dial/global/733dbed2-9183-11de-95f0-07acd7038fc1)
> EXECUTE sofia/internal/7305 at ewr.djernes.net set(hangup_after_bridge=true)
> 2009-08-25 14:27:45.291899 [DEBUG] mod_dptools.c:748
> sofia/internal/7305 at ewr.dje
> rnes.net SET [hangup_after_bridge]=[true]
> EXECUTE sofia/internal/7305 at ewr.djernes.net set(continue_on_fail=true)
> 2009-08-25 14:27:45.291899 [DEBUG] mod_dptools.c:748
> sofia/internal/7305 at ewr.dje
> rnes.net SET [continue_on_fail]=[true]
> EXECUTE sofia/internal/7305 at ewr.djernes.net set(call_timeout=15)
> 2009-08-25 14:27:45.294755 [DEBUG] mod_dptools.c:748
> sofia/internal/7305 at ewr.dje
> rnes.net SET [call_timeout]=[15]
> EXECUTE sofia/internal/7305 at ewr.djernes.netbridge({ignore_early_media=true})
> 2009-08-25 14:27:45.318579 [WARNING] switch_ivr_originate.c:1001 No
> origination
> URL specified!
> 2009-08-25 14:27:45.318579 [DEBUG] switch_ivr_originate.c:2138 Originate
> Resulte
> d in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
> 2009-08-25 14:27:45.318579 [INFO] mod_dptools.c:2093 Originate Failed.
>  Cause: D
> ESTINATION_OUT_OF_ORDER
> 2009-08-25 14:27:45.318579 [DEBUG] mod_dptools.c:2120 Continue on fail
> [true]:
> Cause: DESTINATION_OUT_OF_ORDER
> EXECUTE sofia/internal/7305 at ewr.djernes.net hangup()
> 2009-08-25 14:27:45.318579 [NOTICE] mod_dptools.c:633 Hangup
> sofia/internal/7305
> @ewr.djernes.net [CS_EXECUTE] [NORMAL_CLEARING]
> 2009-08-25 14:27:45.318579 [DEBUG] switch_channel.c:1683 Send signal
> sofia/inter
> nal/7305 at ewr.djernes.net [KILL]
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:932 Send signal
> sofia/i
> nternal/7305 at ewr.djernes.net [BREAK]
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:491
> (sofia/intern
> al/7305 at ewr.djernes.net) State EXECUTE going to sleep
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:398
> (sofia/intern
> al/7305 at ewr.djernes.net) Running State Change CS_HANGUP
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:434
> (sofia/intern
> al/7305 at ewr.djernes.net) State HANGUP
> 2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:338 Channel
> sofia/internal/7305 at e
> wr.djernes.net hanging up, cause: NORMAL_CLEARING
> 2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:417 Responding to INVITE
> with: 48
> 0
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:46
> sofia/internal
> /7305 at ewr.djernes.net Standard HANGUP, cause: NORMAL_CLEARING
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:434
> (sofia/intern
> al/7305 at ewr.djernes.net) State HANGUP going to sleep
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:476
> (sofia/intern
> al/7305 at ewr.djernes.net) State Change CS_HANGUP -> CS_REPORTING
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:932 Send signal
> sofia/i
> nternal/7305 at ewr.djernes.net [BREAK]
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:398
> (sofia/intern
> al/7305 at ewr.djernes.net) Running State Change CS_REPORTING
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:612
> (sofia/intern
> al/7305 at ewr.djernes.net) State REPORTING
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:53
> sofia/internal
> /7305 at ewr.djernes.net Standard REPORTING, cause: NORMAL_CLEARING
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:612
> (sofia/intern
> al/7305 at ewr.djernes.net) State REPORTING going to sleep
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:411
> (sofia/intern
> al/7305 at ewr.djernes.net) State Change CS_REPORTING -> CS_DESTROY
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:1068 Session 117
> (sofia
> /internal/7305 at ewr.djernes.net) Locked, Waiting on external entities
> 2009-08-25 14:27:45.318579 [NOTICE] switch_core_session.c:1086 Session 117
> (sofi
> a/internal/7305 at ewr.djernes.net) Ended
> 2009-08-25 14:27:45.318579 [NOTICE] switch_core_session.c:1088 Close
> Channel sof
> ia/internal/7305 at ewr.djernes.net [CS_DESTROY]
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:564
> (sofia/intern
> al/7305 at ewr.djernes.net) State DESTROY
> 2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:255
> sofia/internal/7305 at ewr.djern
> es.net SOFIA DESTROY
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:60
> sofia/internal
> /7305 at ewr.djernes.net Standard DESTROY
> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:564
> (sofia/intern
> al/7305 at ewr.djernes.net) State DESTROY going to sleep
>
>
> Any Help would be much appreciated.
> --
> Shawn L. Djernes
> SD Consulting
> shawn at djernes.org | sdjernes at gmail.com
> MSN: wizardwlf at hotmail.com
> 402.345.7734 | 402.350.6973 Cell
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From MPeace at edcogroupinc.com  Tue Aug 25 10:04:50 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 12:04:50 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>

I get a bash: sofia: command not found. Is there something I need to add to my config to use these commands?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 25, 2009 10:55 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


On Tue, Aug 25, 2009 at 7:20 AM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

I fired up Wireshark on each side and I can see the SIP register request coming from the laptop, the Freeswitch server replies with a Destination Unreachable (Port unreachable) message.



I rebooted the Server and now I get a "Registration error; 405 Method not allowed" on the softphone and the Wiresharp capture shows: "Status 405 Method Not Allowerd (0 bindings).


Hmm... can you fire up FS and then do "sofia status" and tell us if the internal profile looks okay? Also, do "sofia status profile internal" and capture the output. Something fishy is going on...
-MC

It does this even after I stop and restart the Freeswitch service.



Shouldn't all the ports be open if the firewall and SELunix are disabled?



Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 5:32 PM

To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.





On Mon, Aug 24, 2009 at 2:49 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

It does the same thing, does anything get set during the install that would remember or cache the old network settings? I can access anything from the FS server on any of several networks and vice-versa but the SIP will not register, again no firewalls are upon any of the test hosts.

Doesn't make sense to me.



Time to bust out tcpdump and/or wireshark to make 100% certain you know what's happening with all those SIP packets.

Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:34 PM

To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.





On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

"Registration error: 408-Request Timeout"

Sorry for the the typo - 408 = timeout. (480 = temp unavail)

Try stopping iptables in Linux and try again. Sounds like something is interfering with your packets getting from here to there...
Try:

/etc/init.d/iptables stop

And then see if your packets can move again.

-MC



Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367



From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] On Behalf Of Michael Collins
Sent: Monday, August 24, 2009 4:18 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.





On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com<mailto:MPeace at edcogroupinc.com>> wrote:

I haven't changed any of the conf files.



What happens when you try to register? Do you get 480? (timeout) Or something else?
-MC



________________________________



EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/



Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org





________________________________



EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/



Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org



________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From anthony.minessale at gmail.com  Tue Aug 25 10:05:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Aug 2009 12:05:14 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
	<15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>
	<15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com>
	<65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com>
Message-ID: <191c3a030908251005v6657943l5cdd7645446fcbc5@mail.gmail.com>

I wish I had a nickel for every guy struggling with sipp load testing vs
real world traffic.


On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Hello Takeshi,
>
> Thanks for your hint... it worked out... so to be precise:
>
> VIA header of both INVITE and ACK messages MUST be identical (IP:PORT +
> branch)... and you are right... it might not be according to SIP
> specification. Anyhow, i get FS understand my ACK message.
>
>
> Finally, here is what i used and I'm getting some poor results .. but this
> is another topic :)
>
>
> Thanks for your help.
> Tihomir.
>
>
> sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1
> -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
>
>
> <?xml version="1.0" encoding="ISO-8859-1" ?>
> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>
>
> <scenario name="Basic Sipstone UAC">
>   <send retrans="500" start_rtd="1" start_rtd="2">
>
>     <![CDATA[
>
>       INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>       Max-Forwards: 70
>       Contact: <sip:[field1]@[local_ip]>
>       From: [field1]
> <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
>       To: [service] <sip:[service]@[remote_ip]:[remote_port]>
>       Call-ID: [call_id]
>       CSeq: 1 INVITE
>       Content-Type: application/sdp
>       Content-Length: [len]
>
>       v=0
>       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>       s=-
>       c=IN IP[media_ip_type] [media_ip]
>       t=0 0
>       m=audio [media_port] RTP/AVP 0
>       a=rtpmap:0 PCMU/8000
>
>     ]]>
>   </send>
>
>   <recv response="100"
>         optional="true" rtd="1">
>   </recv>
>
>
>   <recv response="302" rtd="2">
>   </recv>
>
>   <send>
>     <![CDATA[
>
>       ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>       From: [field1]
> <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]
>       To: [service]
> <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
>       Call-ID: [call_id]
>       CSeq: 1 ACK
>       Max-Forwards: 70
>       Content-Length: 0
>
>     ]]>
>   </send>
>
>   <!-- definition of the response time repartition table (unit is ms)   -->
>   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
>
>   <!-- definition of the call length repartition table (unit is ms)     -->
>   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
>
> </scenario>
>
>
>
> On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>
>> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com>
>> wrote:
>> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com>
>> wrote:
>> >>
>> >> sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
>> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m
>> 1 -l
>> >> 4000
>> >> scenario file:      uac_redirect.xml
>> >> FS dialplan:       public.xml
>> >> SIP trace:          trace.log
>> >
>> > The Via definition in your SIPp scenario differs between the INVITE and
>> the ACK:
>> >
>> > INVITE:
>> > Via: SIP/2.0/[transport] [local_ip];branch=[branch]
>> >
>> > ACK:
>> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>> >
>> >
>> > In the INVITE, you are not adding the [local_port] as you do in the ACK.
>> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept the
>> ACK.
>> > So it seems FS is not checking just the ACK's branch against the
>> > INVITE's; it seems it is checking the whole Via header.
>> > I don't know if this is in accordance to SIP specs.
>> > Another thing, about the way you are calling SIPp: do no use "-sn uac"
>> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
>> > means "use the internal (embedded) scenario named xxx". So this
>> > conflicts with the other parameter "-sf" which specifies an external
>> > profile.
>>
>> I mean, an external scenario (file).
>>
>>  It seems this doesn't cause any problem (probably because in
>> > the sipp startup, -sf overrides -sn), but it is misleading.
>> >
>> > regards,
>> > takeshi
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Tue Aug 25 10:10:54 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Aug 2009 12:10:54 -0500
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
Message-ID: <191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>

You also should put your extension first in your dialplan ahead of the
default extensions which match every call and do a lot of db access for
record keeping etc.

The single thread in sofia is part of their concurrency model.
The single thread acts as a scheduler and indicates to us that an invite is
requested
we react to that and spawn a new thread for that call as soon as  the data
is received.

like I said above you are probably hitting all the default stuff to save
that last dialed number etc
that exists in the default config.

sigh, we go through this every new guy doing load testing.


On Tue, Aug 25, 2009 at 11:34 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> Exactly... the scenario i use seems operating on a single thread... why is
> that ? can it be changed?
>
> T.
>
>
> On Tue, Aug 25, 2009 at 5:31 PM, Michael Jerris <mike at jerris.com> wrote:
>
>> Actually in this case, we are bound to one thread in sofia.
>>
>> Mike
>>
>> On Aug 25, 2009, at 9:47 AM, Giovanni Maruzzelli wrote:
>>
>> > is a heavely multithreaded software, it benefits from number of CPUs
>> > (or cores), RAM, and heavy duty kernel features (found in 64bit
>> > kernels)
>> >
>> > put all accesses on ramdisk, leave out the modules you don't use...
>> >
>> > experiment, test, and find the best for your specific application/
>> > workload
>> >
>> > test not only with sipp, but with real load too (often they're very
>> > different)
>> >
>> > -gm
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From mike at jerris.com  Tue Aug 25 10:13:05 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 13:13:05 -0400
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
Message-ID: <278E22F4-46BE-4FB0-8FD1-52874B900C91@jerris.com>

This requires invasive changes in the sofia-sip stack to get thread- 
pooling working again.  I am sure they would accept patches if you can  
provide some that fully address any issues that may come up from  
adding this such as race conditions.

Mike

On Aug 25, 2009, at 12:34 PM, Tihomir Culjaga wrote:

> Exactly... the scenario i use seems operating on a single thread...  
> why is that ? can it be changed?
>
> T.
>
> On Tue, Aug 25, 2009 at 5:31 PM, Michael Jerris <mike at jerris.com>  
> wrote:
> Actually in this case, we are bound to one thread in sofia.
>
> Mike
>
> On Aug 25, 2009, at 9:47 AM, Giovanni Maruzzelli wrote:
>
> > is a heavely multithreaded software, it benefits from number of CPUs
> > (or cores), RAM, and heavy duty kernel features (found in 64bit
> > kernels)
> >
> > put all accesses on ramdisk, leave out the modules you don't use...
> >
> > experiment, test, and find the best for your specific application/
> > workload
> >
> > test not only with sipp, but with real load too (often they're very
> > different)
> >
> > -gm
> >
>

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From sdjernes at gmail.com  Tue Aug 25 10:15:49 2009
From: sdjernes at gmail.com (Shawn L. Djernes)
Date: Tue, 25 Aug 2009 12:15:49 -0500
Subject: [Freeswitch-users] Group Call
In-Reply-To: <ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
Message-ID: <5ff3e3a70908251015ue88b8b9p7a5e091d15a6b5ae@mail.gmail.com>

Woops,
That line was an attempt to try something from an older diaplan example.
 Copied and pasted from a file on my computer not the server.  The actual
line is:

<action application="bridge"
data="{ignore_early_media=true}${group_call(sdjernes+A@${domain_name})}"/>

The +A is to ring all phones in the group at the same time.



2009/8/25 Jo?o Mesquita <jmesquita at freeswitch.org>

> According to example dialplan you hould do this:
>
> <action application="bridge"
> data="{ignore_early_media=true}${group(call:$1@${domain_name})}"/>
>
> According to wiki:
>
> group,[insert|delete|call]:<group name>:<url>,group [insert|delete|call
>
>
> Only difference I see here is the "+A" on your bridge statement.
>
>
> jmesquita
>
>
> On Tue, Aug 25, 2009 at 11:30 AM, Shawn L. Djernes <sdjernes at gmail.com>wrote:
>
>> Hello,
>> I am trying to get group calling to work.  AKA: dial an extension (7300 in
>> this case) and it ring all phones at once.  We are moving a server from
>> Asterisk to FS
>>
>> Here is the conf/directory/default.xml section:
>>
>>         <group name="sdjernes">
>>                 <users>
>>                         <user id="7301" type="pointer"/>
>>                         <user id="7302" type="pointer"/>
>>                         <user id="7303" type="pointer"/>
>>                         <user id="7304" type="pointer"/>
>>                 </users>
>>         </group>
>>
>> Here is the conf/dialplan/default/7300_group.xml extension statement:
>>
>>      <extension name="Group 7300">
>>        <condition field="destination_number" expression="(7300)">
>>          <action application="set" data="hangup_after_bridge=true"/>
>>          <action application="set" data="continue_on_fail=true"/>
>>          <action application="set" data="call_timeout=15"/>
>>          <action application="bridge"
>> data="{ignore_early_media=true}${group(call:sdjernes+A@
>> ${domain_name})}"/>
>> <!--         <action application="transfer" data="7301 XML default"/> -->
>>          <action application="hangup"/>
>>        </condition>
>>      </extension>
>>
>> The log output shows the following:
>>
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3289 Channel
>> sofia/internal/7305 at ewr.
>> djernes.net entering state [received][100]
>> 2009-08-25 14:27:45.282619 [DEBUG] switch_core_state_machine.c:398
>> (sofia/intern
>> al/7305 at ewr.djernes.net) Running State Change CS_NEW
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3296 Remote SDP:
>> v=0
>> o=root 628695665 628695665 IN IP4 98.188.201.109
>> s=call
>> c=IN IP4 98.188.201.109
>> t=0 0
>> m=audio 61012 RTP/AVP 0 8 9 99 3 18 4 101
>> a=rtpmap:0 pcmu/8000
>> a=rtpmap:8 pcma/8000
>> a=rtpmap:9 g722/8000
>> a=rtpmap:99 g726-32/8000
>> a=rtpmap:3 gsm/8000
>> a=rtpmap:18 g729/8000
>> a=rtpmap:4 g723/8000
>> a=rtpmap:101 telephone-event/8000
>> a=fmtp:101 0-16
>> a=ptime:20
>>
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[CELT:114:48000:10]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[CELT:114:32000:10]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[SPEEX:98:8000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[SPEEX:98:8000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[SPEEX:98:8000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[G7221:115:32000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[G7221:107:16000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[G722:9:8000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3071 Audio Codec Compare
>> [pcmu:0
>> :8000:20]/[PCMU:0:8000:20]
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:2029 Set Codec
>> sofia/internal/73
>> 05 at ewr.djernes.net PCMU/8000 20 ms 160 samples
>> 2009-08-25 14:27:45.282619 [DEBUG] switch_core_state_machine.c:404
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State NEW
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload
>> to 10
>> 1
>> 2009-08-25 14:27:45.282619 [DEBUG] sofia.c:3455
>> (sofia/internal/7305 at ewr.djernes
>> .net) State Change CS_NEW -> CS_INIT
>> 2009-08-25 14:27:45.282619 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/i
>> nternal/7305 at ewr.djernes.net [BREAK]
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
>> (sofia/intern
>> al/7305 at ewr.djernes.net) Running State Change CS_INIT
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:481
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State INIT
>> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:83
>> sofia/internal/7305 at ewr.djerne
>> s.net SOFIA INIT
>> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:111
>> (sofia/internal/7305 at ewr.djer
>> nes.net) State Change CS_INIT -> CS_ROUTING
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/i
>> nternal/7305 at ewr.djernes.net [BREAK]
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:481
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State INIT going to sleep
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
>> (sofia/intern
>> al/7305 at ewr.djernes.net) Running State Change CS_ROUTING
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:484
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State ROUTING
>> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:130
>> sofia/internal/7305 at ewr.djern
>> es.net SOFIA ROUTING
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:78
>> sofia/internal
>> /7305 at ewr.djernes.net Standard ROUTING
>> 2009-08-25 14:27:45.287037 [INFO] mod_dialplan_xml.c:315 Processing
>> Sdjernes TOI
>>  Phone->7300 in context default
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unloop]
>> continue
>> =false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unloop]
>> ${unroll_loo
>> ps}(true) =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unloop]
>> ${sip_looped
>> _call}() =~ /^true$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->tod_example] con
>> tinue=true
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [tod_example]
>> ${strft
>> ime(%w)}(2) =~ /^([1-5])$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [tod_example]
>> ${strft
>> ime(%H%M)}(1427) =~ /^((09|1[0-7])[0-5][0-9]|1800)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action set(open=true)
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->global-intercept
>> ] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [global-intercept] de
>> stination_number(7300) =~ /^886$/ break=on-false
>>  Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->group-intercept]
>>  continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [group-intercept] des
>> tination_number(7300) =~ /^\*8$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->intercept-ext] c
>> ontinue=false
>>  Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [intercept-ext] desti
>> nation_number(7300) =~ /^\*\*(\d+)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->redial]
>> continue
>> =false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [redial]
>> destination_
>> number(7300) =~ /^870$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->global]
>> continue
>> =true
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [global]
>> ${call_debug
>> }(false) =~ /^true$/ break=never
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [global]
>> ${sip_has_cr
>> ypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/
>> break=never
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Absolute Condition [global]
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
>> hash(insert/${domain_name}-
>> spymap/${caller_id_number}/${uuid})
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
>> hash(insert/${domain_name}-
>> last_dial/${caller_id_number}/${destination_number})
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
>> hash(insert/${domain_name}-
>> last_dial/global/${uuid})
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->snom-demo-2] con
>> tinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [snom-demo-2]
>> destina
>> tion_number(7300) =~ /^9001$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->snom-demo-1] con
>> tinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [snom-demo-1]
>> destina
>> tion_number(7300) =~ /^9000$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->eavesdrop] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [eavesdrop]
>> destinati
>> on_number(7300) =~ /^88(.*)$|^\*0(.*)$/ break=on-false
>>  Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->eavesdrop] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [eavesdrop]
>> destinati
>> on_number(7300) =~ /^779$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->call_return] con
>> tinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call_return]
>> destina
>> tion_number(7300) =~ /^\*69$|^869$|^lcr$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->del-group] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [del-group]
>> destinati
>> on_number(7300) =~ /^80(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->add-group] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [add-group]
>> destinati
>> on_number(7300) =~ /^81(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->call-group-simo]
>>  continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [call-group-simo] des
>> tination_number(7300) =~ /^82(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->call-group-order
>> ] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [call-group-order] de
>> stination_number(7300) =~ /^83(\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->extension-interc
>> om] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [extension-intercom]
>> destination_number(7300) =~ /^8(10[01][0-9])$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->Local_Extension]
>>  continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [Local_Extension] des
>> tination_number(7300) =~ /^(10[01][0-9])$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->group_dial_sales
>> ] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [group_dial_sales] de
>> stination_number(7300) =~ /^2000$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->group_dial_suppo
>> rt] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [group_dial_support]
>> destination_number(7300) =~ /^2001$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->group_dial_billi
>> ng] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [group_dial_billing]
>> destination_number(7300) =~ /^2002$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->operator]
>> contin
>> ue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [operator]
>> destinatio
>> n_number(7300) =~ /^(operator|0)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->vmain]
>> continue=
>> false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [vmain]
>> destination_n
>> umber(7300) =~ /^vmain$|^4000$|^\*98$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->sip_uri]
>> continu
>> e=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [sip_uri]
>> destination
>> _number(7300) =~ /^sip:(.*)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->nb_conferences]
>> continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [nb_conferences] dest
>> ination_number(7300) =~ /^(30\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->wb_conferences]
>> continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [wb_conferences] dest
>> ination_number(7300) =~ /^(31\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->uwb_conferences]
>>  continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [uwb_conferences] des
>> tination_number(7300) =~ /^(32\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->cdquality_confer
>> ences] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [cdquality_conference
>> s] destination_number(7300) =~ /^(33\d{2})$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->freeswitch_publi
>> c_conf_via_sip] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [freeswitch_public_co
>> nf_via_sip] destination_number(7300) =~ /^9(888|1616|3232)$/
>> break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->mad_boss_interco
>> m] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [mad_boss_intercom] d
>> estination_number(7300) =~ /^0911$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->mad_boss_interco
>> m] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [mad_boss_intercom] d
>> estination_number(7300) =~ /^0912$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->mad_boss]
>> contin
>> ue=false
>>  Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [mad_boss]
>> destinatio
>> n_number(7300) =~ /^0913$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->ivr_demo]
>> contin
>> ue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ivr_demo]
>> destinatio
>> n_number(7300) =~ /^5000$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->dynamic_conferen
>> ce] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [dynamic_conference]
>> destination_number(7300) =~ /^5001$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->rtp_multicast_pa
>> ge] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [rtp_multicast_page]
>> destination_number(7300) =~ /^pagegroup$|^7243$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
>> continue=f
>> alse
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
>> destination_nu
>> mber(7300) =~ /^5900$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
>> continue
>> =false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
>> destination_
>> number(7300) =~ /^5901$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
>> continue=f
>> alse
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [park]
>> source(mod_sof
>> ia) =~ /mod_sofia/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
>> destination_nu
>> mber(7300) =~ /park\+(\d+)/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
>> continue
>> =false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unpark]
>> source(mod_s
>> ofia) =~ /mod_sofia/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
>> destination_
>> number(7300) =~ /^parking$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->park]
>> continue=f
>> alse
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [park]
>> source(mod_sof
>> ia) =~ /mod_sofia/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [park]
>> destination_nu
>> mber(7300) =~ /callpark/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->unpark]
>> continue
>> =false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [unpark]
>> source(mod_s
>> ofia) =~ /mod_sofia/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [unpark]
>> destination_
>> number(7300) =~ /pickup/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->wait]
>> continue=f
>> alse
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [wait]
>> destination_nu
>> mber(7300) =~ /^wait$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->fax_receive] con
>> tinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [fax_receive]
>> destina
>> tion_number(7300) =~ /^9978$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->fax_transmit] co
>> ntinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [fax_transmit]
>> destin
>> ation_number(7300) =~ /^9979$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->ringback_180] co
>> ntinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ringback_180]
>> destin
>> ation_number(7300) =~ /^9980$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->ringback_183_uk_
>> ring] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [ringback_183_uk_ring
>> ] destination_number(7300) =~ /^9981$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->ringback_183_mus
>> ic_ring] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [ringback_183_music_r
>> ing] destination_number(7300) =~ /^9982$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->ringback_post_an
>> swer_uk_ring] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [ringback_post_answer
>> _uk_ring] destination_number(7300) =~ /^9983$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->ringback_post_an
>> swer_music] continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [ringback_post_answer
>> _music] destination_number(7300) =~ /^9984$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->ClueCon]
>> continu
>> e=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [ClueCon]
>> destination
>> _number(7300) =~ /^9991$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->show_info] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [show_info]
>> destinati
>> on_number(7300) =~ /^9992$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->video_record] co
>> ntinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [video_record]
>> destin
>> ation_number(7300) =~ /^9993$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->video_playback]
>> continue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [video_playback] dest
>> ination_number(7300) =~ /^9994$/ break=on-false
>>  Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->delay_echo] cont
>> inue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [delay_echo]
>> destinat
>> ion_number(7300) =~ /^9995$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->echo]
>> continue=f
>> alse
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [echo]
>> destination_nu
>> mber(7300) =~ /^9996$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->milliwatt] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [milliwatt]
>> destinati
>> on_number(7300) =~ /^9997$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->tone_stream] con
>> tinue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [tone_stream]
>> destina
>> tion_number(7300) =~ /^9998$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->zrtp_enrollement
>> ] continue=false
>>  Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL)
>> [zrtp_enrollement] de
>> stination_number(7300) =~ /^9787$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->hold_music] cont
>> inue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [hold_music]
>> destinat
>> ion_number(7300) =~ /^9999$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->Rednote]
>> continu
>> e=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [Rednote]
>> destination
>> _number(7300) =~ /^0220(\d+)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->carmickle] conti
>> nue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [carmickle]
>> destinati
>> on_number(7300) =~ /^0352(\d+)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing
>> [default->call_jason] cont
>> inue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [call_jason]
>> destinat
>> ion_number(7300) =~ /^059(\d+)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->TOI]
>> continue=fa
>> lse
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (FAIL) [TOI]
>> destination_num
>> ber(7300) =~ /^08740(\d+)$/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net parsing [default->Group
>> 7300] cont
>> inue=false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Regex (PASS) [Group 7300]
>> destinat
>> ion_number(7300) =~ /(7300)/ break=on-false
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
>> set(hangup_after_bridge=tru
>> e)
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
>> set(continue_on_fail=true)
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action set(call_timeout=15)
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action
>> bridge({ignore_early_media=
>> true}${group_call(sdjernes+A@${domain_name})})
>> Dialplan: sofia/internal/7305 at ewr.djernes.net Action hangup()
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:114
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State Change CS_ROUTING -> CS_EXECUTE
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/i
>> nternal/7305 at ewr.djernes.net [BREAK]
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:484
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State ROUTING going to sleep
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:398
>> (sofia/intern
>> al/7305 at ewr.djernes.net) Running State Change CS_EXECUTE
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:491
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State EXECUTE
>> 2009-08-25 14:27:45.287037 [DEBUG] mod_sofia.c:173
>> sofia/internal/7305 at ewr.djern
>> es.net SOFIA EXECUTE
>> 2009-08-25 14:27:45.287037 [DEBUG] switch_core_state_machine.c:151
>> sofia/interna
>> l/7305 at ewr.djernes.net Standard EXECUTE
>> EXECUTE sofia/internal/7305 at ewr.djernes.net set(open=true)
>> 2009-08-25 14:27:45.287037 [DEBUG] mod_dptools.c:748
>> sofia/internal/7305 at ewr.dje
>> rnes.net SET [open]=[true]
>> EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-s
>> pymap/7305/733dbed2-9183-11de-95f0-07acd7038fc1)
>> EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-l
>> ast_dial/7305/7300)
>> EXECUTE sofia/internal/7305 at ewr.djernes.nethash(insert/freeswitch.djernes.net-l
>> ast_dial/global/733dbed2-9183-11de-95f0-07acd7038fc1)
>> EXECUTE sofia/internal/7305 at ewr.djernes.net set(hangup_after_bridge=true)
>> 2009-08-25 14:27:45.291899 [DEBUG] mod_dptools.c:748
>> sofia/internal/7305 at ewr.dje
>> rnes.net SET [hangup_after_bridge]=[true]
>> EXECUTE sofia/internal/7305 at ewr.djernes.net set(continue_on_fail=true)
>> 2009-08-25 14:27:45.291899 [DEBUG] mod_dptools.c:748
>> sofia/internal/7305 at ewr.dje
>> rnes.net SET [continue_on_fail]=[true]
>> EXECUTE sofia/internal/7305 at ewr.djernes.net set(call_timeout=15)
>> 2009-08-25 14:27:45.294755 [DEBUG] mod_dptools.c:748
>> sofia/internal/7305 at ewr.dje
>> rnes.net SET [call_timeout]=[15]
>> EXECUTE sofia/internal/7305 at ewr.djernes.netbridge({ignore_early_media=true})
>> 2009-08-25 14:27:45.318579 [WARNING] switch_ivr_originate.c:1001 No
>> origination
>> URL specified!
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_ivr_originate.c:2138 Originate
>> Resulte
>> d in Error Cause: 27 [DESTINATION_OUT_OF_ORDER]
>> 2009-08-25 14:27:45.318579 [INFO] mod_dptools.c:2093 Originate Failed.
>>  Cause: D
>> ESTINATION_OUT_OF_ORDER
>> 2009-08-25 14:27:45.318579 [DEBUG] mod_dptools.c:2120 Continue on fail
>> [true]:
>> Cause: DESTINATION_OUT_OF_ORDER
>> EXECUTE sofia/internal/7305 at ewr.djernes.net hangup()
>> 2009-08-25 14:27:45.318579 [NOTICE] mod_dptools.c:633 Hangup
>> sofia/internal/7305
>> @ewr.djernes.net [CS_EXECUTE] [NORMAL_CLEARING]
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_channel.c:1683 Send signal
>> sofia/inter
>> nal/7305 at ewr.djernes.net [KILL]
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/i
>> nternal/7305 at ewr.djernes.net [BREAK]
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:491
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State EXECUTE going to sleep
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:398
>> (sofia/intern
>> al/7305 at ewr.djernes.net) Running State Change CS_HANGUP
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:434
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State HANGUP
>> 2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:338 Channel
>> sofia/internal/7305 at e
>> wr.djernes.net hanging up, cause: NORMAL_CLEARING
>> 2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:417 Responding to INVITE
>> with: 48
>> 0
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:46
>> sofia/internal
>> /7305 at ewr.djernes.net Standard HANGUP, cause: NORMAL_CLEARING
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:434
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State HANGUP going to sleep
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:476
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State Change CS_HANGUP -> CS_REPORTING
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:932 Send signal
>> sofia/i
>> nternal/7305 at ewr.djernes.net [BREAK]
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:398
>> (sofia/intern
>> al/7305 at ewr.djernes.net) Running State Change CS_REPORTING
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:612
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State REPORTING
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:53
>> sofia/internal
>> /7305 at ewr.djernes.net Standard REPORTING, cause: NORMAL_CLEARING
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:612
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State REPORTING going to sleep
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:411
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State Change CS_REPORTING -> CS_DESTROY
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_session.c:1068 Session 117
>> (sofia
>> /internal/7305 at ewr.djernes.net) Locked, Waiting on external entities
>> 2009-08-25 14:27:45.318579 [NOTICE] switch_core_session.c:1086 Session 117
>> (sofi
>> a/internal/7305 at ewr.djernes.net) Ended
>> 2009-08-25 14:27:45.318579 [NOTICE] switch_core_session.c:1088 Close
>> Channel sof
>> ia/internal/7305 at ewr.djernes.net [CS_DESTROY]
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:564
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State DESTROY
>> 2009-08-25 14:27:45.318579 [DEBUG] mod_sofia.c:255
>> sofia/internal/7305 at ewr.djern
>> es.net SOFIA DESTROY
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:60
>> sofia/internal
>> /7305 at ewr.djernes.net Standard DESTROY
>> 2009-08-25 14:27:45.318579 [DEBUG] switch_core_state_machine.c:564
>> (sofia/intern
>> al/7305 at ewr.djernes.net) State DESTROY going to sleep
>>
>>
>> Any Help would be much appreciated.
>> --
>> Shawn L. Djernes
>> SD Consulting
>> shawn at djernes.org | sdjernes at gmail.com
>> MSN: wizardwlf at hotmail.com
>> 402.345.7734 | 402.350.6973 Cell
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Shawn L. Djernes
SD Consulting
shawn at djernes.org | sdjernes at gmail.com
MSN: wizardwlf at hotmail.com
402.345.7734 | 402.350.6973 Cell
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From tculjaga at gmail.com  Tue Aug 25 10:31:13 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 19:31:13 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
	<191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
Message-ID: <65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>

Of course i removed everytihng from teh dialplan except my extension :)

when exactly do you react and bring up a new thread ? ... is it on INVITE or
on 1st 1xx response ?

i beleive i can have several lets call it SIP interfaces ... on different
ports 5060, 5070, 5080 ... every interface will have it's own sip profile.

does it mean i will have one thread per profile?

T.


On Tue, Aug 25, 2009 at 7:10 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> You also should put your extension first in your dialplan ahead of the
> default extensions which match every call and do a lot of db access for
> record keeping etc.
>
> The single thread in sofia is part of their concurrency model.
> The single thread acts as a scheduler and indicates to us that an invite is
> requested
> we react to that and spawn a new thread for that call as soon as  the data
> is received.
>
> like I said above you are probably hitting all the default stuff to save
> that last dialed number etc
> that exists in the default config.
>
> sigh, we go through this every new guy doing load testing.
>
>
>
> On Tue, Aug 25, 2009 at 11:34 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> Exactly... the scenario i use seems operating on a single thread... why is
>> that ? can it be changed?
>>
>> T.
>>
>>
>> On Tue, Aug 25, 2009 at 5:31 PM, Michael Jerris <mike at jerris.com> wrote:
>>
>>> Actually in this case, we are bound to one thread in sofia.
>>>
>>> Mike
>>>
>>> On Aug 25, 2009, at 9:47 AM, Giovanni Maruzzelli wrote:
>>>
>>> > is a heavely multithreaded software, it benefits from number of CPUs
>>> > (or cores), RAM, and heavy duty kernel features (found in 64bit
>>> > kernels)
>>> >
>>> > put all accesses on ramdisk, leave out the modules you don't use...
>>> >
>>> > experiment, test, and find the best for your specific application/
>>> > workload
>>> >
>>> > test not only with sipp, but with real load too (often they're very
>>> > different)
>>> >
>>> > -gm
>>> >
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Tue Aug 25 10:39:22 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 13:39:22 -0400
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
Message-ID: <34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>

if you want to run that at your prompt instead of at the fs_cli you  
can do this:

function sofia() { fs_cli -x "$(echo "sofia $@")"; }

(thanks ray for the bash foo)

Mike

On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:

> I get a bash: sofia: command not found. Is there something I need to  
> add to my config to use these commands?
>
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
> Michael Collins
> Sent: Tuesday, August 25, 2009 10:55 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> Hmm... can you fire up FS and then do "sofia status" and tell us if  
> the internal profile looks okay? Also, do "sofia status profile  
> internal" and capture the output. Something fishy is going on...
> -MC
> Mike Peace
>
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>

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From msc at freeswitch.org  Tue Aug 25 10:40:36 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 10:40:36 -0700
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
	<191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
	<65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>
Message-ID: <87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>

On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> Of course i removed everytihng from teh dialplan except my extension :)
>
> when exactly do you react and bring up a new thread ? ... is it on INVITE
> or on 1st 1xx response ?
>
> i beleive i can have several lets call it SIP interfaces ... on different
> ports 5060, 5070, 5080 ... every interface will have it's own sip profile.
>
> does it mean i will have one thread per profile?

Yes.
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From tculjaga at gmail.com  Tue Aug 25 10:44:07 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 19:44:07 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
Message-ID: <65d96fc80908251044u9dfa12bi717cccd4e91f1560@mail.gmail.com>

Hi Giovanny,

thanks for your help,
everything that heavyly accesses the disk is on ramdisk now...
hopefully will get some real traffic pretty soon...


On Tue, Aug 25, 2009 at 3:47 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> is a heavely multithreaded software, it benefits from number of CPUs
> (or cores), RAM, and heavy duty kernel features (found in 64bit
> kernels)
>
> put all accesses on ramdisk, leave out the modules you don't use...
>
> experiment, test, and find the best for your specific application/workload
>
> test not only with sipp, but with real load too (often they're very
> different)
>
> -gm
>
>
> On Tue, Aug 25, 2009 at 3:42 PM, Tihomir Culjaga<tculjaga at gmail.com>
> wrote:
> > thanks for the feedback... this is something im going to do tomorrow...
> >
> > what about other things?
> >
> >
> > On Tue, Aug 25, 2009 at 3:39 PM, Jay Binks <jaybinks at gmail.com> wrote:
> >>
> >> Everytime someone asks this ,  the resounding answer is use a 64bit os..
> >> No question
> >> Jay
> >>
> >>
> >>
> >> On 25/08/2009, at 23:19, Tihomir Culjaga <tculjaga at gmail.com> wrote:
> >>
> >> Hey Giovanni,
> >>
> >> thanks for the tip... indeed the db files were heavily used regardless
> if
> >> i started freeswitch with nosql option (freeswitch -nosql)... FS was not
> >> writing anything into that files ... instead it was just accessing
> it....
> >> This behaviour leads to a waste of 40% CPU time... waiting for other
> >> processes (mainly disk access) to finish!!!
> >>
> >> I moved freeswitch/db/ to a ramdisk and the performance got a boost to
> 140
> >> CPS with a CPU load of 80%. I was keeping the machine for a while (20 -
> 30
> >> minutes) on that rate when i sow CPU suddenly went to 100% and FS
> becoming
> >> irresponsive :).
> >>
> >>
> >> What can be wrong?
> >> What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
> >> cross?
> >> What fine tuning do we need in order to asure a long high load run?
> >>
> >>
> >>
> >> Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
> >> sense to move my OS to 64 bit? ... will FS gain more preformance ?... I
> mean
> >> will FS perofomr drastically better 20%+ ?
> >>
> >>
> >> Tihomir.
> >>
> >>
> >> On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
> >> <gmaruzz at celliax.org> wrote:
> >>>
> >>> Maybe your load comes from disk access?
> >>>
> >>> Try putting the sql and log directories on a ramdisk.
> >>>
> >>> OTH,
> >>>
> >>> -giovanni
> >>>
> >>> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com>
> >>> wrote:
> >>> > Hello,
> >>> >
> >>> > i'm trying to use freeswitch as a redirecting server meaning FS has
> to
> >>> > receive an INVITE and according to some rules it will redirect calls
> to
> >>> > other destinations.
> >>> >
> >>> >
> >>> > CALLING_USER                FREESWITCH
> SOMEWHERE
> >>> >
> >>> > INVITE ------------------------------->
> >>> >            <------------------------------ 100 Trying
> >>> >            <------------------------------ 302 Moved Temporary
> >>> > ACK    ------------------------------->
> >>> >
> >>> >
> INVITE--------------------------------------------------------------------------------->
> >>> >
> >>> >
> >>> >
> >>> > Well, wverything works well except i have perfromance issues .... on
> my
> >>> > HW
> >>> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
> Temporary).
> >>> > When
> >>> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS
> i
> >>> > see
> >>> > "calls" being build up in FS and the delay begining to grow.
> >>> >
> >>> > When i observe the machine, load average is almost nothing (load
> >>> > average:
> >>> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
> >>> > taking
> >>> > most load... all others are just sitting there with 1-5 % CPU time.
> >>> > This looks to me as FS handles 302 messages in a single thread?!?!
> >>> >
> >>> >
> >>> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
> >>> >
> >>> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41,
> 0.61,
> >>> > 0.60
> >>> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
> >>> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,
> 0.2%si,
> >>> > 0.0%st
> >>> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k
> >>> > buffers
> >>> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k
> cached
> >>> >
> >>> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
> >>> > COMMAND
> >>> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
> >>> > freeswitch
> >>> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
> >>> > freeswitch
> >>> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
> >>> > freeswitch
> >>> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
> >>> > freeswitch
> >>> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
> >>> > freeswitch
> >>> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
> >>> > freeswitch
> >>> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
> >>> > freeswitch
> >>> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
> >>> > freeswitch
> >>> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
> >>> > freeswitch
> >>> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50
> freeswitch
> >>> >
> >>> >
> >>> >
> >>> > cat /proc/cpuinfo
> >>> > processor       : 0
> >>> > vendor_id       : GenuineIntel
> >>> > cpu family      : 6
> >>> > model           : 15
> >>> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> >>> > stepping        : 6
> >>> > cpu MHz         : 2333.560
> >>> > cache size      : 4096 KB
> >>> > physical id     : 0
> >>> > siblings        : 2
> >>> > core id         : 0
> >>> > cpu cores       : 2
> >>> > apicid          : 0
> >>> > initial apicid  : 0
> >>> > fdiv_bug        : no
> >>> > hlt_bug         : no
> >>> > f00f_bug        : no
> >>> > coma_bug        : no
> >>> > fpu             : yes
> >>> > fpu_exception   : yes
> >>> > cpuid level     : 10
> >>> > wp              : yes
> >>> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
> pge
> >>> > mca
> >>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> >>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
> ssse3
> >>> > cx16
> >>> > xtpr dca lahf_lm
> >>> > bogomips        : 4670.78
> >>> > clflush size    : 64
> >>> > power management:
> >>> >
> >>> > processor       : 1
> >>> > vendor_id       : GenuineIntel
> >>> > cpu family      : 6
> >>> > model           : 15
> >>> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> >>> > stepping        : 6
> >>> > cpu MHz         : 2333.560
> >>> > cache size      : 4096 KB
> >>> > physical id     : 0
> >>> > siblings        : 2
> >>> > core id         : 1
> >>> > cpu cores       : 2
> >>> > apicid          : 1
> >>> > initial apicid  : 1
> >>> > fdiv_bug        : no
> >>> > hlt_bug         : no
> >>> > f00f_bug        : no
> >>> > coma_bug        : no
> >>> > fpu             : yes
> >>> > fpu_exception   : yes
> >>> > cpuid level     : 10
> >>> > wp              : yes
> >>> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
> pge
> >>> > mca
> >>> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> >>> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
> ssse3
> >>> > cx16
> >>> > xtpr dca lahf_lm
> >>> > bogomips        : 4666.82
> >>> > clflush size    : 64
> >>> > power management:
> >>> >
> >>> >
> >>> >
> >>> > uname -a
> >>> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009
> >>> > i686
> >>> > GNU/Linux
> >>> >
> >>> >
> >>> >
> >>> > Of course, i've tuned the machine up
> >>> >
> >>> > ulimit -c unlimited
> >>> > ulimit -d unlimited
> >>> > ulimit -f unlimited
> >>> > ulimit -i unlimited
> >>> > ulimit -n 999999
> >>> > ulimit -q unlimited
> >>> > ulimit -u unlimited
> >>> > ulimit -v unlimited
> >>> > ulimit -x unlimited
> >>> > ulimit -s 240
> >>> > ulimit -l unlimited
> >>> > ulimit -a
> >>> >
> >>> >
> >>> > Started FS with minimum modules but still 40 CPS seems to be the
> limit.
> >>> >
> >>> >
> >>> > So, is there any way to improve performance?
> >>> >
> >>> >
> >>> > Tihomir.
> >>> >
> >>> >
> >>> >
> >>> >
> >>> >
> >>> >
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Tue Aug 25 10:44:50 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 10:44:50 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
Message-ID: <87f2f3b90908251044p4f2ccf6cif79f5b86956708bc@mail.gmail.com>

On Tue, Aug 25, 2009 at 10:04 AM, Mike Peace <MPeace at edcogroupinc.com>wrote:

>  I get a bash: sofia: command not found. Is there something I need to add
> to my config to use these commands?
>
Hehe, sorry. The "sofia" command is for use at the FreeSWITCH command line.
If you have FS running as a daemon (that is, you launched it with the -nc
option) then you'll need to use the fs_cli program to connect to your
running FS daemon. On a typical install it would be something like this:

/usr/local/freeswitch/bin/fs_cli

Once you're there, then issue the two commands:
sofia status
sofia status profile internal

Have fun!
-MC

BTW, you should really get a copy of the September issue of Linux-Pro
magazine because it has a nice article that is a gentle introduction to
using FreeSWITCH. If you're in Western Europe it's already on the shelves.
In the US it should be out any day. In Australia/NZ at the end of the month.
Hope you don't mind the shameless plug.

>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Tuesday, August 25, 2009 10:55 AM
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Tue, Aug 25, 2009 at 7:20 AM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> I fired up Wireshark on each side and I can see the SIP register request
> coming from the laptop, the Freeswitch server replies with a Destination
> Unreachable (Port unreachable) message.
>
>
>
> I rebooted the Server and now I get a ?Registration error; 405 Method not
> allowed? on the softphone and the Wiresharp capture shows: ?Status 405
> Method Not Allowerd (0 bindings).
>
>
>
> Hmm... can you fire up FS and then do "sofia status" and tell us if the
> internal profile looks okay? Also, do "sofia status profile internal" and
> capture the output. Something fishy is going on...
> -MC
>
>  It does this even after I stop and restart the Freeswitch service.
>
>
>
> Shouldn?t all the ports be open if the firewall and SELunix are disabled?
>
>
>
> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 5:32 PM
>
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:49 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> It does the same thing, does anything get set during the install that would
> remember or cache the old network settings? I can access anything from the
> FS server on any of several networks and vice-versa but the SIP will not
> register, again no firewalls are upon any of the test hosts.
>
> Doesn?t make sense to me.
>
>
>
> Time to bust out tcpdump and/or wireshark to make 100% certain you know
> what's happening with all those SIP packets.
>
>   *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:34 PM
>
>
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:26 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> ?Registration error: 408-Request Timeout?
>
> Sorry for the the typo - 408 = timeout. (480 = temp unavail)
>
> Try stopping iptables in Linux and try again. Sounds like something is
> interfering with your packets getting from here to there...
> Try:
>
> /etc/init.d/iptables stop
>
> And then see if your packets can move again.
>
> -MC
>
>
>
> *Mike Peace*
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael
> Collins
> *Sent:* Monday, August 24, 2009 4:18 PM
> *To:* freeswitch-users at lists.freeswitch.org
> *Subject:* Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
>
>
>
>
> On Mon, Aug 24, 2009 at 2:01 PM, Mike Peace <MPeace at edcogroupinc.com>
> wrote:
>
> I haven?t changed any of the conf files.
>
>
>
> What happens when you try to register? Do you get 480? (timeout) Or
> something else?
> -MC
>
>
>  ------------------------------
>
>
>
> *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>
>
> Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>  ------------------------------
>
>
>
> *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>
>
> Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
>  ------------------------------
>
>
>
> *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>
>
> Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> ------------------------------
>
>  *EDCO Group, Inc.*
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
>  Confidentiality Notice: The information contained in this e-mail message
> (including any attachments) may contain confidential and privileged
> information, and is for the sole use of the intended recipient(s). If you
> are not the intended recipient, any unauthorized review, use, or disclosure
> or distribution is strictly prohibited. If you have received this message in
> error, please notify the sender by replying to this e-mail message or by
> telephone at (800) 999-3456 and permanently destroy all copies of the
> original message (including any attachments), along with any reply, and
> delete them from your system.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From MPeace at edcogroupinc.com  Tue Aug 25 10:53:00 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 12:53:00 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>

Ok, figured out from your post to load the FS-CLI in order to run the Sophia status Michael mentioned.

Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error] when running ./fs_cli from a terminal window in /usr/local/freeswitch/bin where fs_cli is located.

Mike
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, August 25, 2009 12:39 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

if you want to run that at your prompt instead of at the fs_cli you can do this:

function sofia() { fs_cli -x "$(echo "sofia $@")"; }

(thanks ray for the bash foo)

Mike

On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:


I get a bash: sofia: command not found. Is there something I need to add to my config to use these commands?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 25, 2009 10:55 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

Hmm... can you fire up FS and then do "sofia status" and tell us if the internal profile looks okay? Also, do "sofia status profile internal" and capture the output. Something fishy is going on...
-MC

Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From tculjaga at gmail.com  Tue Aug 25 10:53:54 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 19:53:54 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
	<191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
	<65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>
	<87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>
Message-ID: <65d96fc80908251053s53f45f62vc58656f24b5ddc6a@mail.gmail.com>

nice, this is one way to go...

btw: when do you exactly bring up a new thread ?

T.

On Tue, Aug 25, 2009 at 7:40 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> Of course i removed everytihng from teh dialplan except my extension :)
>>
>> when exactly do you react and bring up a new thread ? ... is it on INVITE
>> or on 1st 1xx response ?
>>
>> i beleive i can have several lets call it SIP interfaces ... on different
>> ports 5060, 5070, 5080 ... every interface will have it's own sip profile.
>>
>> does it mean i will have one thread per profile?
>
> Yes.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mrene_lists at avgs.ca  Tue Aug 25 10:54:36 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 25 Aug 2009 13:54:36 -0400
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
Message-ID: <D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>

It means freeswitch is not running. You need to start it with /usr/ 
local/freeswitch/bin/freeswitch -nc

@MikeJ: Nice bash scripting :D

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 25-Aug-09, at 1:53 PM, Mike Peace wrote:

> Ok, figured out from your post to load the FS-CLI in order to run  
> the Sophia status Michael mentioned.
>
> Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting  
> [Socket Connection Error] when running ./fs_cli from a terminal  
> window in /usr/local/freeswitch/bin where fs_cli is located.
>
> Mike
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Jerris
> Sent: Tuesday, August 25, 2009 12:39 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> if you want to run that at your prompt instead of at the fs_cli you  
> can do this:
>
> function sofia() { fs_cli -x "$(echo "sofia $@")"; }
>
> (thanks ray for the bash foo)
>
> Mike
>
> On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:
>
>
> I get a bash: sofia: command not found. Is there something I need to  
> add to my config to use these commands?
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Tuesday, August 25, 2009 10:55 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> Hmm... can you fire up FS and then do "sofia status" and tell us if  
> the internal profile looks okay? Also, do "sofia status profile  
> internal" and capture the output. Something fishy is going on...
> -MC
> Mike Peace
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
>
> EDCO Group, Inc.
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
> Confidentiality Notice: The information contained in this e-mail  
> message (including any attachments) may contain confidential and  
> privileged information, and is for the sole use of the intended  
> recipient(s). If you are not the intended recipient, any  
> unauthorized review, use, or disclosure or distribution is strictly  
> prohibited. If you have received this message in error, please  
> notify the sender by replying to this e-mail message or by telephone  
> at (800) 999-3456 and permanently destroy all copies of the original  
> message (including any attachments), along with any reply, and  
> delete them from your system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From mike at jerris.com  Tue Aug 25 10:55:39 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 13:55:39 -0400
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
	<191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
	<65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>
	<87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>
Message-ID: <40411BDB-A19E-42CD-8F2B-70D45A490AFC@jerris.com>


On Aug 25, 2009, at 1:40 PM, Michael Collins wrote:

>
>
> On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga  
> <tculjaga at gmail.com> wrote:
> Of course i removed everytihng from teh dialplan except my  
> extension :)
>
> when exactly do you react and bring up a new thread ? ... is it on  
> INVITE or on 1st 1xx response ?

Before it hits the dialplan.

>
> i beleive i can have several lets call it SIP interfaces ... on  
> different ports 5060, 5070, 5080 ... every interface will have it's  
> own sip profile.
>
> does it mean i will have one thread per profile?
> Yes.

And no. There are still parts of the stack that are in one thread.

Mike
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From mrene_lists at avgs.ca  Tue Aug 25 10:57:50 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 25 Aug 2009 13:57:50 -0400
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908251053s53f45f62vc58656f24b5ddc6a@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
	<191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
	<65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>
	<87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>
	<65d96fc80908251053s53f45f62vc58656f24b5ddc6a@mail.gmail.com>
Message-ID: <B53FBFB4-F319-4DFE-9AEA-BBC949E014C1@avgs.ca>

mod_sofia will take care of spawning the session thread once it  
authenticated the call and loaded all the variables related to the  
call such as the caller profile (callerid, destination number, etc).

If you want to check the source, this is done in  
sofia_handle_sip_i_invite() (sofia.c:5261) with  
switch_core_session_thread_launch().

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 25-Aug-09, at 1:53 PM, Tihomir Culjaga wrote:

> nice, this is one way to go...
>
> btw: when do you exactly bring up a new thread ?
>
> T.
>
> On Tue, Aug 25, 2009 at 7:40 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga  
> <tculjaga at gmail.com> wrote:
> Of course i removed everytihng from teh dialplan except my  
> extension :)
>
> when exactly do you react and bring up a new thread ? ... is it on  
> INVITE or on 1st 1xx response ?
>
> i beleive i can have several lets call it SIP interfaces ... on  
> different ports 5060, 5070, 5080 ... every interface will have it's  
> own sip profile.
>
> does it mean i will have one thread per profile?
> Yes.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tculjaga at gmail.com  Tue Aug 25 10:59:14 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 25 Aug 2009 19:59:14 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <40411BDB-A19E-42CD-8F2B-70D45A490AFC@jerris.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<BE993306-5E78-42B6-8CE8-05DB9C5ED514@gmail.com>
	<65d96fc80908250642s197de5demfb2b6f057b8221e5@mail.gmail.com>
	<7b197bef0908250647i32bfc491i24f3112264426dec@mail.gmail.com>
	<2CC6705A-D8D3-4559-B0CC-409785384857@jerris.com>
	<65d96fc80908250934p49e1b36eu2cac0a4e8db34832@mail.gmail.com>
	<191c3a030908251010n5f4d30c0u961ef67942785535@mail.gmail.com>
	<65d96fc80908251031y3fd4ca84tc0a91f9de5535dd@mail.gmail.com>
	<87f2f3b90908251040t3bcabb5dv54ad15f041e39369@mail.gmail.com>
	<40411BDB-A19E-42CD-8F2B-70D45A490AFC@jerris.com>
Message-ID: <65d96fc80908251059x5ec061dfm60ae8fe9b45aec43@mail.gmail.com>

clear... thanks!

On Tue, Aug 25, 2009 at 7:55 PM, Michael Jerris <mike at jerris.com> wrote:

>
> On Aug 25, 2009, at 1:40 PM, Michael Collins wrote:
>
>
>
> On Tue, Aug 25, 2009 at 10:31 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> Of course i removed everytihng from teh dialplan except my extension :)
>>
>> when exactly do you react and bring up a new thread ? ... is it on INVITE
>> or on 1st 1xx response ?
>>
>
> Before it hits the dialplan.
>
>
>> i beleive i can have several lets call it SIP interfaces ... on different
>> ports 5060, 5070, 5080 ... every interface will have it's own sip profile.
>>
>> does it mean i will have one thread per profile?
>
> Yes.
>
>
> And no. There are still parts of the stack that are in one thread.
>
> Mike
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From carlos.talbot at gmail.com  Tue Aug 25 11:06:56 2009
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Tue, 25 Aug 2009 13:06:56 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
Message-ID: <5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>

Giovanni,
you mean like this message?

"Unable to determine location for device. Voicemail password set via FreePBX
will not be valid."

This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36

Let's keep in ming FreePBX v3 is a developer release and as such many
features are in flux and might not work. That being said there some features
in the Windows build that still do not work. The biggest one right now is
the lack of the php ESL library for Windows which affects the voicemail app.
I'm trying to get this to compile but it's been difficult.

I do include the .svn files with the FreePBX install so you can freely
install TortoiseSVN and update FreePBX at your leisure.

With regards to the sip_profiles, did you create a trunk group and trunk?

regards,

Carlos

On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Windows installer does not work for me.
>
> I've reinstalled various times, same results.
>
> I can correctly create a number, but when I try to create a device for
> that number, it tells me that cannot locate the device, and the
> password for vicemail will be invalid.
>
> After that, it begins to give the php error page, it cannot find the
> start < tag in directory/default.xml
>
> Also for me there are no sofia profiles...
>
> So, I cannot start to test it (eg: I would like to add mod_skypiax
> support to it).
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
>
> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
> Guidi<raffaele.p.guidi at gmail.com> wrote:
> > This is what I was asking! :D When the installer finished it started the
> > whole thing and everything got loaded fine, but when I restarted my
> system
> > it didn't (and did not anymore). Well, I will try to install everything
> from
> > scratch again and see...
> >
> > On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org> wrote:
> >>
> >> If you installed FreePBX then it would be that softwares job to manage
> >> the sofia profiles... wouldn't it?
> >>
> >> /b
> >>
> >> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
> >>
> >> > Actually I did that and it worked fine. I had the problem the SECOND
> >> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
> >> > sip_profiles were not
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jerry.richards at teotech.com  Tue Aug 25 11:12:39 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 25 Aug 2009 11:12:39 -0700
Subject: [Freeswitch-users] Scalability
Message-ID: <B43123F543104F3E894E15B06D705571@greyhawk.tonecommander.com>

Hello All,

Does anyone know what the capacity of a stand-alone Freeswitch, in terms of
how many users?

Also, when that number is exceeded, how can Freeswitch server be distributed
to accommodate a larger installation?

Best Regards,
Jerry 




From mdm at openaccessinc.com  Tue Aug 25 11:20:16 2009
From: mdm at openaccessinc.com (Michael Di Martino)
Date: Tue, 25 Aug 2009 13:20:16 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer -
	great	but I have a little problem
In-Reply-To: <5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
Message-ID: <5EA32FE3006E2740884F5A09AD98A5A718CDCB415A@34093-MBX-C03.mex07a.mlsrvr.com>

Is FreePBX V3 based on Freeswitch?

Michael DiMartino | Director of IT | Open Access, Inc.
115 Bi County Blvd | Farmingdale, NY 11735
631.227.1034| 631.694.6730 FAX |631.988.6060 MOBILE
www.openaccessinc.com

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Carlos Talbot
Sent: Tuesday, August 25, 2009 2:07 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem

Giovanni,

you mean like this message?

"Unable to determine location for device. Voicemail password set via FreePBX will not be valid."

This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36

Let's keep in ming FreePBX v3 is a developer release and as such many features are in flux and might not work. That being said there some features in the Windows build that still do not work. The biggest one right now is the lack of the php ESL library for Windows which affects the voicemail app. I'm trying to get this to compile but it's been difficult.

I do include the .svn files with the FreePBX install so you can freely install TortoiseSVN and update FreePBX at your leisure.

With regards to the sip_profiles, did you create a trunk group and trunk?

regards,

Carlos
On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <gmaruzz at celliax.org<mailto:gmaruzz at celliax.org>> wrote:
Windows installer does not work for me.

I've reinstalled various times, same results.

I can correctly create a number, but when I try to create a device for
that number, it tells me that cannot locate the device, and the
password for vicemail will be invalid.

After that, it begins to give the php error page, it cannot find the
start < tag in directory/default.xml

Also for me there are no sofia profiles...

So, I cannot start to test it (eg: I would like to add mod_skypiax
support to it).


Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
Guidi<raffaele.p.guidi at gmail.com<mailto:raffaele.p.guidi at gmail.com>> wrote:
> This is what I was asking! :D When the installer finished it started the
> whole thing and everything got loaded fine, but when I restarted my system
> it didn't (and did not anymore). Well, I will try to install everything from
> scratch again and see...
>
> On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
>>
>> If you installed FreePBX then it would be that softwares job to manage
>> the sofia profiles... wouldn't it?
>>
>> /b
>>
>> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>>
>> > Actually I did that and it worked fine. I had the problem the SECOND
>> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
>> > sip_profiles were not
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From gmaruzz at celliax.org  Tue Aug 25 11:28:34 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 20:28:34 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com> 
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com> 
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com> 
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com> 
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com> 
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
Message-ID: <7b197bef0908251128y39de0bc8tcf123ee6343bfb33@mail.gmail.com>

Carlos,

you're very kind, as always.

I'm aware that this is a dev preview, and I'm interested just in that,
to begin getting acquainted with the framework (and adding support to
the endpoints/trunk I take care of).

I probably have not got the logic right :-) (I tried both Windows
Installer and Linux ISO)

I started fpbx with fs running, it works. I create a number, then I
create a device and I connect it to that number. It works.

If I restart FS, do not works anymore, complaining no sofia profiles.

>From the front page of FPBX is not clear you *must* create a trunk/trunk group.

I was thinking trunks were for outgoing calls, or for receiving from external.

I was just testing internal phones, trying an IVR, so I was thinking
trunks were not needed.

Can you explain to me?

Thanks again,

-giovanni

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 8:06 PM, Carlos Talbot<carlos.talbot at gmail.com> wrote:
> Giovanni,
> you mean like this message?
> "Unable to determine location for device. Voicemail password set via FreePBX
> will not be valid."
> This is a known FreePBX issue.?http://www.freepbx.org/v3/ticket/36
> Let's keep in ming FreePBX v3 is a developer release and as such many
> features are in flux and might not work.?That being said there some features
> in the Windows build that still do not work. The biggest one right now is
> the lack of the php ESL library for Windows which affects the voicemail app.
> I'm trying to get this to compile but it's been difficult.
> I do include the .svn files with the FreePBX install so you can freely
> install TortoiseSVN and update FreePBX at your leisure.
> With regards to the sip_profiles, did you create a trunk group and trunk?
> regards,
> Carlos
>
> On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> Windows installer does not work for me.
>>
>> I've reinstalled various times, same results.
>>
>> I can correctly create a number, but when I try to create a device for
>> that number, it tells me that cannot locate the device, and the
>> password for vicemail will be invalid.
>>
>> After that, it begins to give the php error page, it cannot find the
>> start < tag in directory/default.xml
>>
>> Also for me there are no sofia profiles...
>>
>> So, I cannot start to test it (eg: I would like to add mod_skypiax
>> support to it).
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>
>>
>>
>> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
>> Guidi<raffaele.p.guidi at gmail.com> wrote:
>> > This is what I was asking! :D When the installer finished it started the
>> > whole thing and everything got loaded fine, but when I restarted my
>> > system
>> > it didn't (and did not anymore). Well, I will try to install everything
>> > from
>> > scratch again and see...
>> >
>> > On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org> wrote:
>> >>
>> >> If you installed FreePBX then it would be that softwares job to manage
>> >> the sofia profiles... wouldn't it?
>> >>
>> >> /b
>> >>
>> >> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>> >>
>> >> > Actually I did that and it worked fine. I had the problem the SECOND
>> >> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
>> >> > sip_profiles were not
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From gmaruzz at celliax.org  Tue Aug 25 11:31:09 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 20:31:09 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <5EA32FE3006E2740884F5A09AD98A5A718CDCB415A@34093-MBX-C03.mex07a.mlsrvr.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com> 
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com> 
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com> 
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com> 
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com> 
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com> 
	<5EA32FE3006E2740884F5A09AD98A5A718CDCB415A@34093-MBX-C03.mex07a.mlsrvr.com>
Message-ID: <7b197bef0908251131r63d4f3a2t9d02724adcba7ea0@mail.gmail.com>

yes, and will support Asterisk and other engines in the future.

On Tue, Aug 25, 2009 at 8:20 PM, Michael Di
Martino<mdm at openaccessinc.com> wrote:
> Is FreePBX V3 based on Freeswitch?
>
>
>
> Michael DiMartino?| Director of IT | Open Access, Inc.
> 115 Bi County Blvd | Farmingdale, NY 11735
> 631.227.1034| 631.694.6730 FAX |631.988.6060?MOBILE
>
> www.openaccessinc.com
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Carlos
> Talbot
> Sent: Tuesday, August 25, 2009 2:07 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
> but I have a little problem
>
>
>
> Giovanni,
>
>
>
> you mean like this message?
>
>
>
> "Unable to determine location for device. Voicemail password set via FreePBX
> will not be valid."
>
>
>
> This is a known FreePBX issue.?http://www.freepbx.org/v3/ticket/36
>
>
>
> Let's keep in ming FreePBX v3 is a developer release and as such many
> features are in flux and might not work.?That being said there some features
> in the Windows build that still do not work. The biggest one right now is
> the lack of the php ESL library for Windows which affects the voicemail app.
> I'm trying to get this to compile but it's been difficult.
>
>
>
> I do include the .svn files with the FreePBX install so you can freely
> install TortoiseSVN and update FreePBX at your leisure.
>
>
>
> With regards to the sip_profiles, did you create a trunk group and trunk?
>
>
>
> regards,
>
>
>
> Carlos
>
> On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>
> Windows installer does not work for me.
>
> I've reinstalled various times, same results.
>
> I can correctly create a number, but when I try to create a device for
> that number, it tells me that cannot locate the device, and the
> password for vicemail will be invalid.
>
> After that, it begins to give the php error page, it cannot find the
> start < tag in directory/default.xml
>
> Also for me there are no sofia profiles...
>
> So, I cannot start to test it (eg: I would like to add mod_skypiax
> support to it).
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
>
> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
>
> Guidi<raffaele.p.guidi at gmail.com> wrote:
>
>> This is what I was asking! :D When the installer finished it started the
>> whole thing and everything got loaded fine, but when I restarted my system
>> it didn't (and did not anymore). Well, I will try to install everything
>> from
>> scratch again and see...
>>
>> On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org> wrote:
>>>
>>> If you installed FreePBX then it would be that softwares job to manage
>>> the sofia profiles... wouldn't it?
>>>
>>> /b
>>>
>>> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>>>
>>> > Actually I did that and it worked fine. I had the problem the SECOND
>>> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
>>> > sip_profiles were not
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From mike at jerris.com  Tue Aug 25 11:33:44 2009
From: mike at jerris.com (Michael Jerris)
Date: Tue, 25 Aug 2009 14:33:44 -0400
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer -
	great	but I have a little problem
In-Reply-To: <5EA32FE3006E2740884F5A09AD98A5A718CDCB415A@34093-MBX-C03.mex07a.mlsrvr.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
	<5EA32FE3006E2740884F5A09AD98A5A718CDCB415A@34093-MBX-C03.mex07a.mlsrvr.com>
Message-ID: <C46DBBA8-88CA-4B3D-935F-0E0A3D0F7382@jerris.com>

It supports FreeSWITCH and soon other engines as well.  More  
information is at:

http://www.freepbx.org/freepbx-v3

Mike

On Aug 25, 2009, at 2:20 PM, Michael Di Martino wrote:

> Is FreePBX V3 based on Freeswitch?
>
> Michael DiMartino | Director of IT | Open Access, Inc.
> 115 Bi County Blvd | Farmingdale, NY 11735
> 631.227.1034| 631.694.6730 FAX |631.988.6060 MOBILE
> www.openaccessinc.com
>
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From jerry.richards at teotech.com  Tue Aug 25 11:35:16 2009
From: jerry.richards at teotech.com (Jerry Richards)
Date: Tue, 25 Aug 2009 11:35:16 -0700
Subject: [Freeswitch-users] RTP Packet Routing
Message-ID: <0003D215849D415591E531947383F082@greyhawk.tonecommander.com>

Hello All,

I noticed Freeswitch becomes the "middle-man", handling RTP traffic for an
active call.  How do I configure it so it allows the two SIP endpoints to
send RTP packet to each other directly?

Best Regards,
Jerry




From msc at freeswitch.org  Tue Aug 25 11:42:53 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 11:42:53 -0700
Subject: [Freeswitch-users] RTP Packet Routing
In-Reply-To: <0003D215849D415591E531947383F082@greyhawk.tonecommander.com>
References: <0003D215849D415591E531947383F082@greyhawk.tonecommander.com>
Message-ID: <87f2f3b90908251142w772a4a5ai47f1b100908da373@mail.gmail.com>

On Tue, Aug 25, 2009 at 11:35 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

> Hello All,
>
> I noticed Freeswitch becomes the "middle-man", handling RTP traffic for an
> active call.  How do I configure it so it allows the two SIP endpoints to
> send RTP packet to each other directly?
>

Check out bypass media mode:
http://wiki.freeswitch.org/wiki/Bypass_Media

-MC
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From msc at freeswitch.org  Tue Aug 25 11:47:35 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 11:47:35 -0700
Subject: [Freeswitch-users] Scalability
In-Reply-To: <B43123F543104F3E894E15B06D705571@greyhawk.tonecommander.com>
References: <B43123F543104F3E894E15B06D705571@greyhawk.tonecommander.com>
Message-ID: <87f2f3b90908251147h48b80dex384ff30912a902f6@mail.gmail.com>

On Tue, Aug 25, 2009 at 11:12 AM, Jerry Richards <jerry.richards at teotech.com
> wrote:

> Hello All,
>
> Does anyone know what the capacity of a stand-alone Freeswitch, in terms of
> how many users?
>

You mean concurrent registered users? I suppose the answer is "a lot"! :)

>
> Also, when that number is exceeded, how can Freeswitch server be
> distributed
> to accommodate a larger installation?


FreeSWITCH supports a number of ways of spreading things around. Check this
out:
http://wiki.freeswitch.org/wiki/Mod_xml_curl

-MC
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From krice at freeswitch.org  Tue Aug 25 11:53:22 2009
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 25 Aug 2009 13:53:22 -0500
Subject: [Freeswitch-users] Scalability
In-Reply-To: <B43123F543104F3E894E15B06D705571@greyhawk.tonecommander.com>
Message-ID: <C6B99D52.5D691%krice@freeswitch.org>

Deployable scalability varies based on a number of things... Number of users
registering, how often they register, concurrent call volume, call rate
(calls/second) etc...  Defining that a little better may illicit a better
response... But generally FS can scale into the 1000s of concurrent calls at
that 100s of calls/sec with 100s of registered users... The however as with
any software how well it actually scales for you will be dependant on the
hardware you deploy it on and its actual configuration...


> From: Jerry Richards <jerry.richards at teotech.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Tue, 25 Aug 2009 11:12:39 -0700
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: [Freeswitch-users] Scalability
> 
> Hello All,
> 
> Does anyone know what the capacity of a stand-alone Freeswitch, in terms of
> how many users?
> 
> Also, when that number is exceeded, how can Freeswitch server be distributed
> to accommodate a larger installation?
> 
> Best Regards,
> Jerry 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From krice at freeswitch.org  Tue Aug 25 11:54:27 2009
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 25 Aug 2009 13:54:27 -0500
Subject: [Freeswitch-users] RTP Packet Routing
In-Reply-To: <87f2f3b90908251142w772a4a5ai47f1b100908da373@mail.gmail.com>
Message-ID: <C6B99D93.5D692%krice@freeswitch.org>

Just remember that bypass_media is a special mode and not everything will
work when a call is in that mode... Don?t expect to do anything that
remotely relies on media being in the FS box...



From: Michael Collins <msc at freeswitch.org>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Tue, 25 Aug 2009 11:42:53 -0700
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] RTP Packet Routing



On Tue, Aug 25, 2009 at 11:35 AM, Jerry Richards
<jerry.richards at teotech.com> wrote:
> Hello All,
> 
> I noticed Freeswitch becomes the "middle-man", handling RTP traffic for an
> active call. ?How do I configure it so it allows the two SIP endpoints to
> send RTP packet to each other directly?

Check out bypass media mode:
http://wiki.freeswitch.org/wiki/Bypass_Media

-MC



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From MPeace at edcogroupinc.com  Tue Aug 25 11:58:07 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 13:58:07 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>

It was started from the services configuration window first and I still get the error. I set the service not to load rebooted then ran it from the prompt as you suggested. It stated "3585 Backgrounding" I then ran ps -eaf from the prompt and that PID didn't show up. Then when I attempted to run the fs_cli I received the same error as before  [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error]


From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 12:55 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

It means freeswitch is not running. You need to start it with /usr/local/freeswitch/bin/freeswitch -nc

@MikeJ: Nice bash scripting :D

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>




On 25-Aug-09, at 1:53 PM, Mike Peace wrote:


Ok, figured out from your post to load the FS-CLI in order to run the Sophia status Michael mentioned.

Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error] when running ./fs_cli from a terminal window in /usr/local/freeswitch/bin where fs_cli is located.

Mike
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, August 25, 2009 12:39 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

if you want to run that at your prompt instead of at the fs_cli you can do this:

function sofia() { fs_cli -x "$(echo "sofia $@")"; }

(thanks ray for the bash foo)

Mike

On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:



I get a bash: sofia: command not found. Is there something I need to add to my config to use these commands?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 25, 2009 10:55 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

Hmm... can you fire up FS and then do "sofia status" and tell us if the internal profile looks okay? Also, do "sofia status profile internal" and capture the output. Something fishy is going on...
-MC

Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367


________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From mrene_lists at avgs.ca  Tue Aug 25 12:01:43 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 25 Aug 2009 15:01:43 -0400
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
Message-ID: <FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>

You can start it without the -nc switch and it'll stay active on the  
current terminal so you can look for any errors.

Just do:

/usr/local/freeswitch/bin/freeswitch


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca



On 25-Aug-09, at 2:58 PM, Mike Peace wrote:

> It was started from the services configuration window first and I  
> still get the error. I set the service not to load rebooted then ran  
> it from the prompt as you suggested. It stated ?3585 Backgrounding?  
> I then ran ps ?eaf from the prompt and that PID didn?t show up. Then  
> when I attempted to run the fs_cli I received the same error as  
> before  [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting  
> [Socket Connection Error]
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Mathieu Rene
> Sent: Tuesday, August 25, 2009 12:55 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> It means freeswitch is not running. You need to start it with /usr/ 
> local/freeswitch/bin/freeswitch -nc
>
> @MikeJ: Nice bash scripting :D
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 25-Aug-09, at 1:53 PM, Mike Peace wrote:
>
>
> Ok, figured out from your post to load the FS-CLI in order to run  
> the Sophia status Michael mentioned.
>
> Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting  
> [Socket Connection Error] when running ./fs_cli from a terminal  
> window in /usr/local/freeswitch/bin where fs_cli is located.
>
> Mike
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Jerris
> Sent: Tuesday, August 25, 2009 12:39 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> if you want to run that at your prompt instead of at the fs_cli you  
> can do this:
>
> function sofia() { fs_cli -x "$(echo "sofia $@")"; }
>
> (thanks ray for the bash foo)
>
> Mike
>
> On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:
>
>
>
> I get a bash: sofia: command not found. Is there something I need to  
> add to my config to use these commands?
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Tuesday, August 25, 2009 10:55 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> Hmm... can you fire up FS and then do "sofia status" and tell us if  
> the internal profile looks okay? Also, do "sofia status profile  
> internal" and capture the output. Something fishy is going on...
> -MC
> Mike Peace
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
>
> EDCO Group, Inc.
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
> Confidentiality Notice: The information contained in this e-mail  
> message (including any attachments) may contain confidential and  
> privileged information, and is for the sole use of the intended  
> recipient(s). If you are not the intended recipient, any  
> unauthorized review, use, or disclosure or distribution is strictly  
> prohibited. If you have received this message in error, please  
> notify the sender by replying to this e-mail message or by telephone  
> at (800) 999-3456 and permanently destroy all copies of the original  
> message (including any attachments), along with any reply, and  
> delete them from your system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> EDCO Group, Inc.
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
> Confidentiality Notice: The information contained in this e-mail  
> message (including any attachments) may contain confidential and  
> privileged information, and is for the sole use of the intended  
> recipient(s). If you are not the intended recipient, any  
> unauthorized review, use, or disclosure or distribution is strictly  
> prohibited. If you have received this message in error, please  
> notify the sender by replying to this e-mail message or by telephone  
> at (800) 999-3456 and permanently destroy all copies of the original  
> message (including any attachments), along with any reply, and  
> delete them from your system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From carlos.talbot at gmail.com  Tue Aug 25 12:02:02 2009
From: carlos.talbot at gmail.com (Carlos Talbot)
Date: Tue, 25 Aug 2009 14:02:02 -0500
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <7b197bef0908251128y39de0bc8tcf123ee6343bfb33@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
	<7b197bef0908251128y39de0bc8tcf123ee6343bfb33@mail.gmail.com>
Message-ID: <5800526b0908251202o730d2d18nb6948bad3766b705@mail.gmail.com>

This would be a question for Darren and the FreePBX group. :) I guess it
does not help that the *User Documentation* link on this page is currently
empty: http://www.freepbx.org/v3/wiki/

If you note the message during FreePBX initialization *all* files in the
sip_profiles directory are removed (including internal*.xml). This causes
'sofia status' to come back empty.


Incompatible ConfigurationWARNING: THE FOLLOWING FILES WILL BE DELETED!

   - D:/FreeSWITCH/conf/sip_profiles/external.xml
   - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
   - D:/FreeSWITCH/conf/sip_profiles/internal.xml


regards,

Carlos

On Tue, Aug 25, 2009 at 1:28 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
 wrote:

>
> >From the front page of FPBX is not clear you *must* create a trunk/trunk
> group.
>
> I was thinking trunks were for outgoing calls, or for receiving from
> external.
>
> I was just testing internal phones, trying an IVR, so I was thinking
> trunks were not needed.
>
> Can you explain to me?


>
> Thanks again,
>
> -giovanni
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
>
> On Tue, Aug 25, 2009 at 8:06 PM, Carlos Talbot<carlos.talbot at gmail.com>
> wrote:
> > Giovanni,
> > you mean like this message?
> > "Unable to determine location for device. Voicemail password set via
> FreePBX
> > will not be valid."
> > This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36
> > Let's keep in ming FreePBX v3 is a developer release and as such many
> > features are in flux and might not work. That being said there some
> features
> > in the Windows build that still do not work. The biggest one right now is
> > the lack of the php ESL library for Windows which affects the voicemail
> app.
> > I'm trying to get this to compile but it's been difficult.
> > I do include the .svn files with the FreePBX install so you can freely
> > install TortoiseSVN and update FreePBX at your leisure.
> > With regards to the sip_profiles, did you create a trunk group and trunk?
> > regards,
> > Carlos
> >
> > On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <
> gmaruzz at celliax.org>
> > wrote:
> >>
> >> Windows installer does not work for me.
> >>
> >> I've reinstalled various times, same results.
> >>
> >> I can correctly create a number, but when I try to create a device for
> >> that number, it tells me that cannot locate the device, and the
> >> password for vicemail will be invalid.
> >>
> >> After that, it begins to give the php error page, it cannot find the
> >> start < tag in directory/default.xml
> >>
> >> Also for me there are no sofia profiles...
> >>
> >> So, I cannot start to test it (eg: I would like to add mod_skypiax
> >> support to it).
> >>
> >>
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >>
> >>
> >>
> >> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
> >> Guidi<raffaele.p.guidi at gmail.com> wrote:
> >> > This is what I was asking! :D When the installer finished it started
> the
> >> > whole thing and everything got loaded fine, but when I restarted my
> >> > system
> >> > it didn't (and did not anymore). Well, I will try to install
> everything
> >> > from
> >> > scratch again and see...
> >> >
> >> > On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org>
> wrote:
> >> >>
> >> >> If you installed FreePBX then it would be that softwares job to
> manage
> >> >> the sofia profiles... wouldn't it?
> >> >>
> >> >> /b
> >> >>
> >> >> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
> >> >>
> >> >> > Actually I did that and it worked fine. I had the problem the
> SECOND
> >> >> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
> >> >> > sip_profiles were not
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From MPeace at edcogroupinc.com  Tue Aug 25 12:41:23 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 14:41:23 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
	<FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A688B@ESNEXS2.edcogroup.net>


OK,

Received a different error: Cannot Initialize [[error near line 294]: unclosed <!--]

?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 2:02 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

You can start it without the -nc switch and it'll stay active on the current terminal so you can look for any errors.

Just do:

/usr/local/freeswitch/bin/freeswitch


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>



On 25-Aug-09, at 2:58 PM, Mike Peace wrote:


It was started from the services configuration window first and I still get the error. I set the service not to load rebooted then ran it from the prompt as you suggested. It stated "3585 Backgrounding" I then ran ps -eaf from the prompt and that PID didn't show up. Then when I attempted to run the fs_cli I received the same error as before  [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error]


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 12:55 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

It means freeswitch is not running. You need to start it with /usr/local/freeswitch/bin/freeswitch -nc

@MikeJ: Nice bash scripting :D

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>




On 25-Aug-09, at 1:53 PM, Mike Peace wrote:



Ok, figured out from your post to load the FS-CLI in order to run the Sophia status Michael mentioned.

Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error] when running ./fs_cli from a terminal window in /usr/local/freeswitch/bin where fs_cli is located.

Mike
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, August 25, 2009 12:39 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

if you want to run that at your prompt instead of at the fs_cli you can do this:

function sofia() { fs_cli -x "$(echo "sofia $@")"; }

(thanks ray for the bash foo)

Mike

On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:




I get a bash: sofia: command not found. Is there something I need to add to my config to use these commands?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 25, 2009 10:55 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

Hmm... can you fire up FS and then do "sofia status" and tell us if the internal profile looks okay? Also, do "sofia status profile internal" and capture the output. Something fishy is going on...
-MC

Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367


________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

_______________________________________________
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________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

_______________________________________________
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=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From mrene_lists at avgs.ca  Tue Aug 25 12:45:01 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Tue, 25 Aug 2009 15:45:01 -0400
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A688B@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
	<FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A688B@ESNEXS2.edcogroup.net>
Message-ID: <9C4244D1-EC2C-4B3C-BD5E-16643BA5D0D4@avgs.ca>

It means you have an error in your XML configuration. You can open / 
usr/local/freeswitch/log/freeswitch.xml.fsxml to figure out where it  
is (but do not modify this file, as it is generated from all the other  
xml files).

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 25-Aug-09, at 3:41 PM, Mike Peace wrote:

>
> OK,
>
> Received a different error: Cannot Initialize [[error near line  
> 294]: unclosed <!--]
>
> ?
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Mathieu Rene
> Sent: Tuesday, August 25, 2009 2:02 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> You can start it without the -nc switch and it'll stay active on the  
> current terminal so you can look for any errors.
>
> Just do:
>
> /usr/local/freeswitch/bin/freeswitch
>
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
> On 25-Aug-09, at 2:58 PM, Mike Peace wrote:
>
>
> It was started from the services configuration window first and I  
> still get the error. I set the service not to load rebooted then ran  
> it from the prompt as you suggested. It stated ?3585 Backgrounding?  
> I then ran ps ?eaf from the prompt and that PID didn?t show up. Then  
> when I attempted to run the fs_cli I received the same error as  
> before  [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting  
> [Socket Connection Error]
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Mathieu Rene
> Sent: Tuesday, August 25, 2009 12:55 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> It means freeswitch is not running. You need to start it with /usr/ 
> local/freeswitch/bin/freeswitch -nc
>
> @MikeJ: Nice bash scripting :D
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 25-Aug-09, at 1:53 PM, Mike Peace wrote:
>
>
>
> Ok, figured out from your post to load the FS-CLI in order to run  
> the Sophia status Michael mentioned.
>
> Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting  
> [Socket Connection Error] when running ./fs_cli from a terminal  
> window in /usr/local/freeswitch/bin where fs_cli is located.
>
> Mike
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Jerris
> Sent: Tuesday, August 25, 2009 12:39 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> if you want to run that at your prompt instead of at the fs_cli you  
> can do this:
>
> function sofia() { fs_cli -x "$(echo "sofia $@")"; }
>
> (thanks ray for the bash foo)
>
> Mike
>
> On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:
>
>
>
>
> I get a bash: sofia: command not found. Is there something I need to  
> add to my config to use these commands?
>
> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] On Behalf Of Michael Collins
> Sent: Tuesday, August 25, 2009 10:55 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.
>
> Hmm... can you fire up FS and then do "sofia status" and tell us if  
> the internal profile looks okay? Also, do "sofia status profile  
> internal" and capture the output. Something fishy is going on...
> -MC
> Mike Peace
>
> Network Analyst
>
> EDCO, The Document People
>
> Direct 417-447-3367
>
>
>
>
> EDCO Group, Inc.
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
> Confidentiality Notice: The information contained in this e-mail  
> message (including any attachments) may contain confidential and  
> privileged information, and is for the sole use of the intended  
> recipient(s). If you are not the intended recipient, any  
> unauthorized review, use, or disclosure or distribution is strictly  
> prohibited. If you have received this message in error, please  
> notify the sender by replying to this e-mail message or by telephone  
> at (800) 999-3456 and permanently destroy all copies of the original  
> message (including any attachments), along with any reply, and  
> delete them from your system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> EDCO Group, Inc.
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
> Confidentiality Notice: The information contained in this e-mail  
> message (including any attachments) may contain confidential and  
> privileged information, and is for the sole use of the intended  
> recipient(s). If you are not the intended recipient, any  
> unauthorized review, use, or disclosure or distribution is strictly  
> prohibited. If you have received this message in error, please  
> notify the sender by replying to this e-mail message or by telephone  
> at (800) 999-3456 and permanently destroy all copies of the original  
> message (including any attachments), along with any reply, and  
> delete them from your system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> EDCO Group, Inc.
> Phone: (800) 999-3456
> Fax: (800) 999-3551
> Web: http://www.edcogroupinc.com/
>
> Confidentiality Notice: The information contained in this e-mail  
> message (including any attachments) may contain confidential and  
> privileged information, and is for the sole use of the intended  
> recipient(s). If you are not the intended recipient, any  
> unauthorized review, use, or disclosure or distribution is strictly  
> prohibited. If you have received this message in error, please  
> notify the sender by replying to this e-mail message or by telephone  
> at (800) 999-3456 and permanently destroy all copies of the original  
> message (including any attachments), along with any reply, and  
> delete them from your system.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From bjbrashier at gmail.com  Tue Aug 25 13:23:53 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Tue, 25 Aug 2009 14:23:53 -0600
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <191c3a030908251005v6657943l5cdd7645446fcbc5@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com> 
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com> 
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com> 
	<15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com> 
	<15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com> 
	<65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com> 
	<191c3a030908251005v6657943l5cdd7645446fcbc5@mail.gmail.com>
Message-ID: <7bcfdd290908251323o65654ba3w663900ec693b81cb@mail.gmail.com>

Well, you'd have another nickel from over here, then.
If I can get this working before I'm tasked with something else I'll write
up something more on the wiki about "Freeswitch and SIPp", but I'm not sure
I'll get that chance.

BB

On Tue, Aug 25, 2009 at 11:05 AM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> I wish I had a nickel for every guy struggling with sipp load testing vs
> real world traffic.
>
>
>
> On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> Hello Takeshi,
>>
>> Thanks for your hint... it worked out... so to be precise:
>>
>> VIA header of both INVITE and ACK messages MUST be identical (IP:PORT +
>> branch)... and you are right... it might not be according to SIP
>> specification. Anyhow, i get FS understand my ACK message.
>>
>>
>> Finally, here is what i used and I'm getting some poor results .. but this
>> is another topic :)
>>
>>
>> Thanks for your help.
>> Tihomir.
>>
>>
>> sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1
>> -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
>>
>>
>> <?xml version="1.0" encoding="ISO-8859-1" ?>
>> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>>
>>
>> <scenario name="Basic Sipstone UAC">
>>   <send retrans="500" start_rtd="1" start_rtd="2">
>>
>>     <![CDATA[
>>
>>       INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>       Max-Forwards: 70
>>       Contact: <sip:[field1]@[local_ip]>
>>       From: [field1]
>> <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
>>       To: [service] <sip:[service]@[remote_ip]:[remote_port]>
>>       Call-ID: [call_id]
>>       CSeq: 1 INVITE
>>       Content-Type: application/sdp
>>       Content-Length: [len]
>>
>>       v=0
>>       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>>       s=-
>>       c=IN IP[media_ip_type] [media_ip]
>>       t=0 0
>>       m=audio [media_port] RTP/AVP 0
>>       a=rtpmap:0 PCMU/8000
>>
>>     ]]>
>>   </send>
>>
>>   <recv response="100"
>>         optional="true" rtd="1">
>>   </recv>
>>
>>
>>   <recv response="302" rtd="2">
>>   </recv>
>>
>>   <send>
>>     <![CDATA[
>>
>>       ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>>       From: [field1]
>> <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]
>>       To: [service]
>> <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
>>       Call-ID: [call_id]
>>       CSeq: 1 ACK
>>       Max-Forwards: 70
>>       Content-Length: 0
>>
>>     ]]>
>>   </send>
>>
>>   <!-- definition of the response time repartition table (unit is ms)
>> -->
>>   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
>>
>>   <!-- definition of the call length repartition table (unit is ms)
>> -->
>>   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
>>
>> </scenario>
>>
>>
>>
>> On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>>
>>> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com>
>>> wrote:
>>> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com>
>>> wrote:
>>> >>
>>> >> sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
>>> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt -m
>>> 1 -l
>>> >> 4000
>>> >> scenario file:      uac_redirect.xml
>>> >> FS dialplan:       public.xml
>>> >> SIP trace:          trace.log
>>> >
>>> > The Via definition in your SIPp scenario differs between the INVITE and
>>> the ACK:
>>> >
>>> > INVITE:
>>> > Via: SIP/2.0/[transport] [local_ip];branch=[branch]
>>> >
>>> > ACK:
>>> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>>> >
>>> >
>>> > In the INVITE, you are not adding the [local_port] as you do in the
>>> ACK.
>>> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept the
>>> ACK.
>>> > So it seems FS is not checking just the ACK's branch against the
>>> > INVITE's; it seems it is checking the whole Via header.
>>> > I don't know if this is in accordance to SIP specs.
>>> > Another thing, about the way you are calling SIPp: do no use "-sn uac"
>>> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
>>> > means "use the internal (embedded) scenario named xxx". So this
>>> > conflicts with the other parameter "-sf" which specifies an external
>>> > profile.
>>>
>>> I mean, an external scenario (file).
>>>
>>>  It seems this doesn't cause any problem (probably because in
>>> > the sipp startup, -sf overrides -sn), but it is misleading.
>>> >
>>> > regards,
>>> > takeshi
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Tue Aug 25 13:39:21 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Aug 2009 15:39:21 -0500
Subject: [Freeswitch-users] SIPp issues - seems FS doesn't understand
	ACK message
In-Reply-To: <7bcfdd290908251323o65654ba3w663900ec693b81cb@mail.gmail.com>
References: <65d96fc80908241513ja15d071h5d40e4a11e3cb9a@mail.gmail.com>
	<191c3a030908241520q1da54ee1r172ebb9fb25f4085@mail.gmail.com>
	<65d96fc80908241531p5cc89983h2c842ff85e1907a8@mail.gmail.com>
	<15b9404e0908241852s6c0630ebp1b85bff664621f8f@mail.gmail.com>
	<15b9404e0908241857r15bce88rc04066092bd8173e@mail.gmail.com>
	<65d96fc80908242351p26804e2agb85fd7fc07f4a73c@mail.gmail.com>
	<191c3a030908251005v6657943l5cdd7645446fcbc5@mail.gmail.com>
	<7bcfdd290908251323o65654ba3w663900ec693b81cb@mail.gmail.com>
Message-ID: <191c3a030908251339n57370a92nb6c404903e6f0353@mail.gmail.com>

here's the one i use for making a call waiting x seconds and hanging up
http://www.freeswitch.org/eg/load_test/dft_cap.xml

This requires that the sipp terminate all the calls.

careful with sipp, it's like a roach motel, you can get stuck trying to make
it work and never get it to produce real-life situations.



On Tue, Aug 25, 2009 at 3:23 PM, Bradley Brashier <bjbrashier at gmail.com>wrote:

> Well, you'd have another nickel from over here, then.
> If I can get this working before I'm tasked with something else I'll write
> up something more on the wiki about "Freeswitch and SIPp", but I'm not sure
> I'll get that chance.
>
> BB
>
> On Tue, Aug 25, 2009 at 11:05 AM, Anthony Minessale <
> anthony.minessale at gmail.com> wrote:
>
>> I wish I had a nickel for every guy struggling with sipp load testing vs
>> real world traffic.
>>
>>
>>
>> On Tue, Aug 25, 2009 at 1:51 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> Hello Takeshi,
>>>
>>> Thanks for your hint... it worked out... so to be precise:
>>>
>>> VIA header of both INVITE and ACK messages MUST be identical (IP:PORT +
>>> branch)... and you are right... it might not be according to SIP
>>> specification. Anyhow, i get FS understand my ACK message.
>>>
>>>
>>> Finally, here is what i used and I'm getting some poor results .. but
>>> this is another topic :)
>>>
>>>
>>> Thanks for your help.
>>> Tihomir.
>>>
>>>
>>> sipp 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1
>>> -rp 100 -trace_msg -inf test.txt -m 20000 -l 4000
>>>
>>>
>>> <?xml version="1.0" encoding="ISO-8859-1" ?>
>>> <!DOCTYPE scenario SYSTEM "sipp.dtd">
>>>
>>>
>>> <scenario name="Basic Sipstone UAC">
>>>   <send retrans="500" start_rtd="1" start_rtd="2">
>>>
>>>     <![CDATA[
>>>
>>>       INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>>>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
>>>       Max-Forwards: 70
>>>       Contact: <sip:[field1]@[local_ip]>
>>>       From: [field1]
>>> <sip:[field1]@[local_ip]:[local_port]>;tag=[call_number]
>>>       To: [service] <sip:[service]@[remote_ip]:[remote_port]>
>>>       Call-ID: [call_id]
>>>       CSeq: 1 INVITE
>>>       Content-Type: application/sdp
>>>       Content-Length: [len]
>>>
>>>       v=0
>>>       o=user1 53655765 2353687637 IN IP[local_ip_type] [local_ip]
>>>       s=-
>>>       c=IN IP[media_ip_type] [media_ip]
>>>       t=0 0
>>>       m=audio [media_port] RTP/AVP 0
>>>       a=rtpmap:0 PCMU/8000
>>>
>>>     ]]>
>>>   </send>
>>>
>>>   <recv response="100"
>>>         optional="true" rtd="1">
>>>   </recv>
>>>
>>>
>>>   <recv response="302" rtd="2">
>>>   </recv>
>>>
>>>   <send>
>>>     <![CDATA[
>>>
>>>       ACK sip:[service]@[remote_ip]:[remote_port] SIP/2.0
>>>       Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>>>       From: [field1]
>>> <sip:[field1]@1[local_ip]:[local_port]>;tag=[call_number]
>>>       To: [service]
>>> <sip:[service]@[remote_ip]:[remote_port]>[peer_tag_param]
>>>       Call-ID: [call_id]
>>>       CSeq: 1 ACK
>>>       Max-Forwards: 70
>>>       Content-Length: 0
>>>
>>>     ]]>
>>>   </send>
>>>
>>>   <!-- definition of the response time repartition table (unit is ms)
>>> -->
>>>   <ResponseTimeRepartition value="10, 20, 30, 40, 50, 100, 150, 200"/>
>>>
>>>   <!-- definition of the call length repartition table (unit is ms)
>>> -->
>>>   <CallLengthRepartition value="10, 50, 100, 500, 1000, 5000, 10000"/>
>>>
>>> </scenario>
>>>
>>>
>>>
>>> On Tue, Aug 25, 2009 at 3:57 AM, mayamatakeshi <mayamatakeshi at gmail.com>wrote:
>>>
>>>> On Tue, Aug 25, 2009 at 10:52 AM, mayamatakeshi<mayamatakeshi at gmail.com>
>>>> wrote:
>>>> > On Tue, Aug 25, 2009 at 7:31 AM, Tihomir Culjaga<tculjaga at gmail.com>
>>>> wrote:
>>>> >>
>>>> >> sipp_cmd:         sipp -sn uac 10.4.4.251 -sf uac_redirect.xml -s
>>>> >> 30003016094191500 -trace_err -r 1 -rp 1000 -trace_msg -inf test.txt
>>>> -m 1 -l
>>>> >> 4000
>>>> >> scenario file:      uac_redirect.xml
>>>> >> FS dialplan:       public.xml
>>>> >> SIP trace:          trace.log
>>>> >
>>>> > The Via definition in your SIPp scenario differs between the INVITE
>>>> and the ACK:
>>>> >
>>>> > INVITE:
>>>> > Via: SIP/2.0/[transport] [local_ip];branch=[branch]
>>>> >
>>>> > ACK:
>>>> > Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch-3]
>>>> >
>>>> >
>>>> > In the INVITE, you are not adding the [local_port] as you do in the
>>>> ACK.
>>>> > Just adding the [local_port] in the INVITE makes FreeSWITCH accept the
>>>> ACK.
>>>> > So it seems FS is not checking just the ACK's branch against the
>>>> > INVITE's; it seems it is checking the whole Via header.
>>>> > I don't know if this is in accordance to SIP specs.
>>>> > Another thing, about the way you are calling SIPp: do no use "-sn uac"
>>>> > and "-sf uac_redirect.xml" at the same time. The parameter "-sn xxx"
>>>> > means "use the internal (embedded) scenario named xxx". So this
>>>> > conflicts with the other parameter "-sf" which specifies an external
>>>> > profile.
>>>>
>>>> I mean, an external scenario (file).
>>>>
>>>>  It seems this doesn't cause any problem (probably because in
>>>> > the sipp startup, -sf overrides -sn), but it is misleading.
>>>> >
>>>> > regards,
>>>> > takeshi
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From aep.lists at it46.se  Tue Aug 25 13:46:50 2009
From: aep.lists at it46.se (Alberto Escudero-Pascual (lists))
Date: Tue, 25 Aug 2009 22:46:50 +0200
Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
In-Reply-To: <5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
	<191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
	<6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>
	<5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org>
Message-ID: <45b2e91d7aa77afcf6430a14dde4b886.squirrel@correo.nodo50.org>

Hi Brian,

>From the CLI>

freeswitch at open46> system /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
/tmp/foo.mp3 -S
2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing command:
/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S
API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
/tmp/foo.mp3 -S)] output:
+OK

open46:/tmp# ls
foo.wav


and running the command from the command line:


open46:/tmp#  /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3
-Sopen46:/tmp# ls
foo.mp3  foo.wav


If I do the same with lame397

freeswitch at open46> system /usr/local/freeswitch/bin/lame397 -V2
/tmp/foo.wav /tmp/foo.mp3 -S
2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing command:
/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S
API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav
/tmp/foo.mp3 -S)] output:
+OK

open46:/tmp# ls
foo.mp3  foo.wav


Highly paranormal! Sorry for hijacking the previous thread.

/aep

-- 
Stopping junk mailers is good for the environment

> Try running it at the CLI and see if you see any errors.  Also please
> do not hijack threads.  The original thread "[Freeswitch-users] XML-
> RPC on different ip than 0.0.0.0" which was hijacked by clicking
> reply, changing the subject and clicking send.  Please in the future
> do not do that as it clutters up the threading and could get your
> query lost in the noise.
>
> Thanks,
> Brian
>
> On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote:
>
>> Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
>> appears, if I use version 3.97 (older version), it does!.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>





From anthony.minessale at gmail.com  Tue Aug 25 13:54:15 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Tue, 25 Aug 2009 15:54:15 -0500
Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
In-Reply-To: <45b2e91d7aa77afcf6430a14dde4b886.squirrel@correo.nodo50.org>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
	<191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
	<6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>
	<5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org>
	<45b2e91d7aa77afcf6430a14dde4b886.squirrel@correo.nodo50.org>
Message-ID: <191c3a030908251354l25206d0ch19836c8996f43e84@mail.gmail.com>

maybe it's writing some err to stderr that is being suppressed somehow

On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists) <
aep.lists at it46.se> wrote:

> Hi Brian,
>
> >From the CLI>
>
> freeswitch at open46> system /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
> /tmp/foo.mp3 -S
> 2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing command:
> /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S
> API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
> /tmp/foo.mp3 -S)] output:
> +OK
>
> open46:/tmp# ls
> foo.wav
>
>
> and running the command from the command line:
>
>
> open46:/tmp#  /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3
> -Sopen46:/tmp# ls
> foo.mp3  foo.wav
>
>
> If I do the same with lame397
>
> freeswitch at open46> system /usr/local/freeswitch/bin/lame397 -V2
> /tmp/foo.wav /tmp/foo.mp3 -S
> 2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing command:
> /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S
> API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav
> /tmp/foo.mp3 -S)] output:
> +OK
>
> open46:/tmp# ls
> foo.mp3  foo.wav
>
>
> Highly paranormal! Sorry for hijacking the previous thread.
>
> /aep
>
> --
> Stopping junk mailers is good for the environment
>
> > Try running it at the CLI and see if you see any errors.  Also please
> > do not hijack threads.  The original thread "[Freeswitch-users] XML-
> > RPC on different ip than 0.0.0.0" which was hijacked by clicking
> > reply, changing the subject and clicking send.  Please in the future
> > do not do that as it clutters up the threading and could get your
> > query lost in the noise.
> >
> > Thanks,
> > Brian
> >
> > On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote:
> >
> >> Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
> >> appears, if I use version 3.97 (older version), it does!.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From d at d-man.org  Tue Aug 25 13:59:03 2009
From: d at d-man.org (Darren Schreiber)
Date: Tue, 25 Aug 2009 13:59:03 -0700
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer -
	great	but I have a little problem
In-Reply-To: <5800526b0908251202o730d2d18nb6948bad3766b705@mail.gmail.com>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
	<7b197bef0908251128y39de0bc8tcf123ee6343bfb33@mail.gmail.com>
	<5800526b0908251202o730d2d18nb6948bad3766b705@mail.gmail.com>
Message-ID: <8A034A3098ED3C4990F7D9DE40F5585F1762B73424@EXVMBX020-3.exch020.serverdata.net>

Hi there...

So a few things on this.

1) We have a module that's still being worked on called Sip Interface that allows you to configure Sip Profiles in FreeSWITCH. Unfortunately we don't have the ability to easily import your existing SIP profiles, and by NOT displaying them in the UI your stock config conflicts with those profiles. In other words, internal.xml and external.xml define sip profiles on ports 5060 and 5080 that the GUI is unaware. So to simplify things, we just delete those files and expect you to recreate them via the UI on install. We'll probably make this a little more obvious in the near future, but for now, that's what we do.

2) The trunk creation system in the ISO does give the impression that trunk groups & trunks are for external. This is a design flaw I have already fixed in trunk (referenced above). We have now split this module into two modules - one for configuring Sip Interface/Sip Profiles (which are for defining your IPs & ports to use for sending/receiving calls and authentication settings) and the other for defining gateways ('trunks") which are generally used for making outbound calls.

So in other words, we're aware of the issues you are detailing. We will be finalizing the fix for this issue hopefully on Sunday and will rebuild the ISO by Monday.

Hope that helps.

- Darren




________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Carlos Talbot
Sent: Tuesday, August 25, 2009 12:02 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great but I have a little problem


This would be a question for Darren and the FreePBX group. :) I guess it does not help that the User Documentation link on this page is currently empty: http://www.freepbx.org/v3/wiki/

If you note the message during FreePBX initialization *all* files in the sip_profiles directory are removed (including internal*.xml). This causes 'sofia status' to come back empty.


Incompatible Configuration
WARNING: THE FOLLOWING FILES WILL BE DELETED!


 *   D:/FreeSWITCH/conf/sip_profiles/external.xml
 *   D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
 *   D:/FreeSWITCH/conf/sip_profiles/internal.xml



regards,

Carlos

On Tue, Aug 25, 2009 at 1:28 PM, Giovanni Maruzzelli <gmaruzz at celliax.org<mailto:gmaruzz at celliax.org>> wrote:

>From the front page of FPBX is not clear you *must* create a trunk/trunk group.

I was thinking trunks were for outgoing calls, or for receiving from external.

I was just testing internal phones, trying an IVR, so I was thinking
trunks were not needed.

Can you explain to me?


Thanks again,

-giovanni

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Tue, Aug 25, 2009 at 8:06 PM, Carlos Talbot<carlos.talbot at gmail.com<mailto:carlos.talbot at gmail.com>> wrote:
> Giovanni,
> you mean like this message?
> "Unable to determine location for device. Voicemail password set via FreePBX
> will not be valid."
> This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36
> Let's keep in ming FreePBX v3 is a developer release and as such many
> features are in flux and might not work. That being said there some features
> in the Windows build that still do not work. The biggest one right now is
> the lack of the php ESL library for Windows which affects the voicemail app.
> I'm trying to get this to compile but it's been difficult.
> I do include the .svn files with the FreePBX install so you can freely
> install TortoiseSVN and update FreePBX at your leisure.
> With regards to the sip_profiles, did you create a trunk group and trunk?
> regards,
> Carlos
>
> On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <gmaruzz at celliax.org<mailto:gmaruzz at celliax.org>>
> wrote:
>>
>> Windows installer does not work for me.
>>
>> I've reinstalled various times, same results.
>>
>> I can correctly create a number, but when I try to create a device for
>> that number, it tells me that cannot locate the device, and the
>> password for vicemail will be invalid.
>>
>> After that, it begins to give the php error page, it cannot find the
>> start < tag in directory/default.xml
>>
>> Also for me there are no sofia profiles...
>>
>> So, I cannot start to test it (eg: I would like to add mod_skypiax
>> support to it).
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>
>>
>>
>> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
>> Guidi<raffaele.p.guidi at gmail.com<mailto:raffaele.p.guidi at gmail.com>> wrote:
>> > This is what I was asking! :D When the installer finished it started the
>> > whole thing and everything got loaded fine, but when I restarted my
>> > system
>> > it didn't (and did not anymore). Well, I will try to install everything
>> > from
>> > scratch again and see...
>> >
>> > On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:
>> >>
>> >> If you installed FreePBX then it would be that softwares job to manage
>> >> the sofia profiles... wouldn't it?
>> >>
>> >> /b
>> >>
>> >> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>> >>
>> >> > Actually I did that and it worked fine. I had the problem the SECOND
>> >> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
>> >> > sip_profiles were not
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From MPeace at edcogroupinc.com  Tue Aug 25 13:59:48 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 15:59:48 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <9C4244D1-EC2C-4B3C-BD5E-16643BA5D0D4@avgs.ca>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<8C0ED0F1-0842-4E22-8C7D-46E22862BB48@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A621A63D3@ESNEXS2.edcogroup.net>
	<87f2f3b90908241418ub6fa71fo56084eba3dc8d2cb@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6407@ESNEXS2.edcogroup.net>
	<87f2f3b90908241434k3aad88c6h96222c49f4965561@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6445@ESNEXS2.edcogroup.net>
	<87f2f3b90908241531h229b178ayf4ce08a6069347de@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A65B5@ESNEXS2.edcogroup.net>
	<87f2f3b90908250855s1bb7e01asf5f08fa0501ecb69@mail.gmail.com>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
	<FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A688B@ESNEXS2.edcogroup.net>
	<9C4244D1-EC2C-4B3C-BD5E-16643BA5D0D4@avgs.ca>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A621A6921@ESNEXS2.edcogroup.net>

That was it! I had been try to setup the conference java script example and I made an error in the conference-conf.xml. Thanks a lot to everyone!

Can somebody point me in the right direction to get my new T-1 taking to the Sangoma A101?

I have it installed and setup the Wanpipe driver off their site.

When I issue a wanrouter status it shows connected, an ifconfig shows RX and TX packets incrementing on the w1g1 interface.

This is the spec on the phone line:

LD T1 with ISDN, ESF, B8ZS, DTMF, ASCENDING HUNTING, PSEUDO ANI 417-862-4351, REAL TIME ANI, COS US 50 CA,
AND TO ADD 10 8XX #S TO POINT AT THE DS1 WITH NO DNIS.

Thanks again!

Mike







From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 2:45 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

It means you have an error in your XML configuration. You can open /usr/local/freeswitch/log/freeswitch.xml.fsxml to figure out where it is (but do not modify this file, as it is generated from all the other xml files).

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>




On 25-Aug-09, at 3:41 PM, Mike Peace wrote:



OK,

Received a different error: Cannot Initialize [[error near line 294]: unclosed <!--]

?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 2:02 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

You can start it without the -nc switch and it'll stay active on the current terminal so you can look for any errors.

Just do:

/usr/local/freeswitch/bin/freeswitch


Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>



On 25-Aug-09, at 2:58 PM, Mike Peace wrote:



It was started from the services configuration window first and I still get the error. I set the service not to load rebooted then ran it from the prompt as you suggested. It stated "3585 Backgrounding" I then ran ps -eaf from the prompt and that PID didn't show up. Then when I attempted to run the fs_cli I received the same error as before  [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error]


From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mathieu Rene
Sent: Tuesday, August 25, 2009 12:55 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

It means freeswitch is not running. You need to start it with /usr/local/freeswitch/bin/freeswitch -nc

@MikeJ: Nice bash scripting :D

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca<mailto:mrene at avgs.ca>




On 25-Aug-09, at 1:53 PM, Mike Peace wrote:




Ok, figured out from your post to load the FS-CLI in order to run the Sophia status Michael mentioned.

Now I get: [ERROR] libs/esl/fs_cli.c.639 main() Error Connecting [Socket Connection Error] when running ./fs_cli from a terminal window in /usr/local/freeswitch/bin where fs_cli is located.

Mike
From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Jerris
Sent: Tuesday, August 25, 2009 12:39 PM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

if you want to run that at your prompt instead of at the fs_cli you can do this:

function sofia() { fs_cli -x "$(echo "sofia $@")"; }

(thanks ray for the bash foo)

Mike

On Aug 25, 2009, at 1:04 PM, Mike Peace wrote:





I get a bash: sofia: command not found. Is there something I need to add to my config to use these commands?

From: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 25, 2009 10:55 AM
To: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

Hmm... can you fire up FS and then do "sofia status" and tell us if the internal profile looks okay? Also, do "sofia status profile internal" and capture the output. Something fishy is going on...
-MC

Mike Peace

Network Analyst

EDCO, The Document People

Direct 417-447-3367


________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

_______________________________________________
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________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


________________________________

EDCO Group, Inc.
Phone: (800) 999-3456
Fax: (800) 999-3551
Web: http://www.edcogroupinc.com/

Confidentiality Notice: The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s). If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited. If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org


=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From gmaruzz at celliax.org  Tue Aug 25 14:06:04 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Tue, 25 Aug 2009 23:06:04 +0200
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
	but I have a little problem
In-Reply-To: <8A034A3098ED3C4990F7D9DE40F5585F1762B73424@EXVMBX020-3.exch020.serverdata.net>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com> 
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com> 
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com> 
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com> 
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com> 
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com> 
	<7b197bef0908251128y39de0bc8tcf123ee6343bfb33@mail.gmail.com> 
	<5800526b0908251202o730d2d18nb6948bad3766b705@mail.gmail.com> 
	<8A034A3098ED3C4990F7D9DE40F5585F1762B73424@EXVMBX020-3.exch020.serverdata.net>
Message-ID: <7b197bef0908251406ub380301p1ea74cc40c2ae358@mail.gmail.com>

Darren,

thanks! it helps a lot.

If we svn up in /var/www/freepbx we got the trunk?


On Tue, Aug 25, 2009 at 10:59 PM, Darren Schreiber<d at d-man.org> wrote:
> Hi there...
>
> So a few things on this.
>
> 1) We have a module that's still being worked on called Sip Interface that
> allows you to configure Sip Profiles in FreeSWITCH. Unfortunately we don't
> have the ability to easily import your existing SIP profiles, and by NOT
> displaying them in the UI your stock config conflicts with those profiles.
> In other words, internal.xml and external.xml define sip profiles on ports
> 5060 and 5080 that the GUI is unaware. So to simplify things, we just delete
> those files and expect you to recreate them via the UI on install. We'll
> probably make this a little more obvious in the near future, but for now,
> that's what we do.
>
> 2) The trunk creation system in the ISO does give the impression that trunk
> groups & trunks are for external. This is a design flaw I have already fixed
> in trunk (referenced above). We have now split this module into two modules
> - one for configuring Sip Interface/Sip Profiles (which are for defining
> your IPs & ports to use for sending/receiving calls and authentication
> settings) and the other for defining gateways ('trunks") which are generally
> used for making outbound calls.
>
> So in other words, we're aware of the issues you are detailing. We will be
> finalizing the fix for this issue hopefully on Sunday and will rebuild the
> ISO by Monday.
>
> Hope that helps.
>
> - Darren
>
>
>
> ________________________________
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Carlos
> Talbot
> Sent: Tuesday, August 25, 2009 12:02 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great
> but I have a little problem
>
>
> This would be a question for Darren and the FreePBX group. :) I guess it
> does not help that the?User Documentation?link on this page is currently
> empty:?http://www.freepbx.org/v3/wiki/
> If you note the message during FreePBX initialization *all* files in the
> sip_profiles directory are removed (including internal*.xml). This causes
> 'sofia status' to come back empty.
>
> Incompatible Configuration
>
> WARNING: THE FOLLOWING FILES WILL BE DELETED!
>
> D:/FreeSWITCH/conf/sip_profiles/external.xml
> D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
> D:/FreeSWITCH/conf/sip_profiles/internal.xml
>
>
> regards,
> Carlos
>
> On Tue, Aug 25, 2009 at 1:28 PM, Giovanni
> Maruzzelli?<gmaruzz at celliax.org>?wrote:
>>
>> >From the front page of FPBX is not clear you *must* create a trunk/trunk
>> group.
>>
>> I was thinking trunks were for outgoing calls, or for receiving from
>> external.
>>
>> I was just testing internal phones, trying an IVR, so I was thinking
>> trunks were not needed.
>>
>> Can you explain to me?
>>
>> Thanks again,
>>
>> -giovanni
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>
>>
>>
>> On Tue, Aug 25, 2009 at 8:06 PM, Carlos Talbot<carlos.talbot at gmail.com>
>> wrote:
>> > Giovanni,
>> > you mean like this message?
>> > "Unable to determine location for device. Voicemail password set via
>> > FreePBX
>> > will not be valid."
>> > This is a known FreePBX issue.?http://www.freepbx.org/v3/ticket/36
>> > Let's keep in ming FreePBX v3 is a developer release and as such many
>> > features are in flux and might not work.?That being said there some
>> > features
>> > in the Windows build that still do not work. The biggest one right now
>> > is
>> > the lack of the php ESL library for Windows which affects the voicemail
>> > app.
>> > I'm trying to get this to compile but it's been difficult.
>> > I do include the .svn files with the FreePBX install so you can freely
>> > install TortoiseSVN and update FreePBX at your leisure.
>> > With regards to the sip_profiles, did you create a trunk group and
>> > trunk?
>> > regards,
>> > Carlos
>> >
>> > On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli
>> > <gmaruzz at celliax.org>
>> > wrote:
>> >>
>> >> Windows installer does not work for me.
>> >>
>> >> I've reinstalled various times, same results.
>> >>
>> >> I can correctly create a number, but when I try to create a device for
>> >> that number, it tells me that cannot locate the device, and the
>> >> password for vicemail will be invalid.
>> >>
>> >> After that, it begins to give the php error page, it cannot find the
>> >> start < tag in directory/default.xml
>> >>
>> >> Also for me there are no sofia profiles...
>> >>
>> >> So, I cannot start to test it (eg: I would like to add mod_skypiax
>> >> support to it).
>> >>
>> >>
>> >> Sincerely,
>> >>
>> >> Giovanni Maruzzelli
>> >> Cell : +39-347-2665618
>> >>
>> >>
>> >>
>> >>
>> >> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
>> >> Guidi<raffaele.p.guidi at gmail.com> wrote:
>> >> > This is what I was asking! :D When the installer finished it started
>> >> > the
>> >> > whole thing and everything got loaded fine, but when I restarted my
>> >> > system
>> >> > it didn't (and did not anymore). Well, I will try to install
>> >> > everything
>> >> > from
>> >> > scratch again and see...
>> >> >
>> >> > On Mon, Aug 24, 2009 at 20:30, Brian West <brian at freeswitch.org>
>> >> > wrote:
>> >> >>
>> >> >> If you installed FreePBX then it would be that softwares job to
>> >> >> manage
>> >> >> the sofia profiles... wouldn't it?
>> >> >>
>> >> >> /b
>> >> >>
>> >> >> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
>> >> >>
>> >> >> > Actually I did that and it worked fine. I had the problem the
>> >> >> > SECOND
>> >> >> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded but
>> >> >> > sip_profiles were not
>> >> >>
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Tue Aug 25 14:59:56 2009
From: msc at freeswitch.org (Michael Collins)
Date: Tue, 25 Aug 2009 14:59:56 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A621A6921@ESNEXS2.edcogroup.net>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
	<FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A688B@ESNEXS2.edcogroup.net>
	<9C4244D1-EC2C-4B3C-BD5E-16643BA5D0D4@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A6921@ESNEXS2.edcogroup.net>
Message-ID: <87f2f3b90908251459p29f033c6g7cf2ed234f4b4479@mail.gmail.com>

>
> Can somebody point me in the right direction to get my new T-1 taking to
> the Sangoma A101?
>
>
>
Are you using PRI or just a 24 channel T1, aka CAS or RBS? Check out this
page for some handy information:
http://wiki.freeswitch.org/wiki/OpenZAP

-MC

P.S. - If you join #openzap and/or #freeswitch on irc.freenode.net you can
usually get real-time help with most of these questions.
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From MPeace at edcogroupinc.com  Tue Aug 25 17:14:40 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Tue, 25 Aug 2009 19:14:40 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <87f2f3b90908251459p29f033c6g7cf2ed234f4b4479@mail.gmail.com>
References: <69D652F49932C34199968DFB8AEAABA33A621A63B8@ESNEXS2.edcogroup.net>
	<69D652F49932C34199968DFB8AEAABA33A621A6767@ESNEXS2.edcogroup.net>
	<34791B59-CBE6-4F57-B4AB-D7398994FB3E@jerris.com>
	<69D652F49932C34199968DFB8AEAABA33A621A67C0@ESNEXS2.edcogroup.net>
	<D7EBC34D-D57C-49F2-976B-3CB4179D2EA1@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A683B@ESNEXS2.edcogroup.net>
	<FD9F3CD8-579B-4A59-B0AF-ED13654C5BBA@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A688B@ESNEXS2.edcogroup.net>
	<9C4244D1-EC2C-4B3C-BD5E-16643BA5D0D4@avgs.ca>
	<69D652F49932C34199968DFB8AEAABA33A621A6921@ESNEXS2.edcogroup.net>
	<87f2f3b90908251459p29f033c6g7cf2ed234f4b4479@mail.gmail.com>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A622D21B4@ESNEXS2.edcogroup.net>

I have looked at that but I am confused on which files need to be edited. Since I have already installed in Wanpipe mode with the Sangoma card I skipped straight to the Wanpipe section. It mentions setting the [span wanpie PRI_1] etc in the openzap.conf then further down it mentions editing the autoload_configs/openzap.conf.xml with a completely different format of config.

Are these the same file? If not where does the first one mentioned reside?

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins
Sent: Tuesday, August 25, 2009 5:00 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


Can somebody point me in the right direction to get my new T-1 taking to the Sangoma A101?


Are you using PRI or just a 24 channel T1, aka CAS or RBS? Check out this page for some handy information:
http://wiki.freeswitch.org/wiki/OpenZAP

-MC

P.S. - If you join #openzap and/or #freeswitch on irc.freenode.net<http://irc.freenode.net> you can usually get real-time help with most of these questions.

=-=-=-=-=-=-=-=-=-=-=-=-=-=-=
EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.
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From mitul at enterux.com  Tue Aug 25 18:53:45 2009
From: mitul at enterux.com (Mitul Limbani)
Date: Wed, 26 Aug 2009 07:23:45 +0530
Subject: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer -
	great	but I have a little problem
In-Reply-To: <8A034A3098ED3C4990F7D9DE40F5585F1762B73424@EXVMBX020-3.exch020.serverdata.net>
References: <b0a8792f0908160243q116471ejea388c56de6c1320@mail.gmail.com>
	<5800526b0908160704t51500392v8a4b633fc6f49a2d@mail.gmail.com>
	<b0a8792f0908241124k492084f9n58142a988dab2b70@mail.gmail.com>
	<22A50622-E391-4C9F-AF8A-FD0122660204@freeswitch.org>
	<b0a8792f0908241506r27380bc8sd95994fe9eb4452b@mail.gmail.com>
	<7b197bef0908241956g46784079g78d548e443f847d4@mail.gmail.com>
	<5800526b0908251106k2b295c66wa8b49dcf171acc2@mail.gmail.com>
	<7b197bef0908251128y39de0bc8tcf123ee6343bfb33@mail.gmail.com>
	<5800526b0908251202o730d2d18nb6948bad3766b705@mail.gmail.com>
	<8A034A3098ED3C4990F7D9DE40F5585F1762B73424@EXVMBX020-3.exch020.serverdata.net>
Message-ID: <596902CF-DC31-4E73-81CB-AA41B4CADE12@enterux.com>

Darren,

I shall look forward to check version 3 of FreePBX for FS.
I hope it not trixswitch anyone aware of status on trixswitch (not  
that I m a big fan of Trixbox, but quite a while back I had seen  
trixswitch iso so wondering if anything is progresing on that end.

Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com

On 26-Aug-2009, at 2:29 AM, Darren Schreiber <d at d-man.org> wrote:

> Hi there...
>
> So a few things on this.
>
> 1) We have a module that's still being worked on called Sip  
> Interface that allows you to configure Sip Profiles in FreeSWITCH.  
> Unfortunately we don't have the ability to easily import your  
> existing SIP profiles, and by NOT displaying them in the UI your  
> stock config conflicts with those profiles. In other words,  
> internal.xml and external.xml define sip profiles on ports 5060 and  
> 5080 that the GUI is unaware. So to simplify things, we just delete  
> those files and expect you to recreate them via the UI on install.  
> We'll probably make this a little more obvious in the near future,  
> but for now, that's what we do.
>
> 2) The trunk creation system in the ISO does give the impression  
> that trunk groups & trunks are for external. This is a design flaw I  
> have already fixed in trunk (referenced above). We have now split  
> this module into two modules - one for configuring Sip Interface/Sip  
> Profiles (which are for defining your IPs & ports to use for sending/ 
> receiving calls and authentication settings) and the other for  
> defining gateways ('trunks") which are generally used for making  
> outbound calls.
>
> So in other words, we're aware of the issues you are detailing. We  
> will be finalizing the fix for this issue hopefully on Sunday and  
> will rebuild the ISO by Monday.
>
> Hope that helps.
>
> - Darren
>
>
>
>
> From: freeswitch-users-bounces at lists.freeswitch.org  
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of  
> Carlos Talbot
> Sent: Tuesday, August 25, 2009 12:02 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer -  
> great but I have a little problem
>
>
> This would be a question for Darren and the FreePBX group. :) I  
> guess it does not help that the User Documentation link on this page  
> is currently empty: http://www.freepbx.org/v3/wiki/
>
> If you note the message during FreePBX initialization *all* files in  
> the sip_profiles directory are removed (including internal*.xml).  
> This causes 'sofia status' to come back empty.
>
>
> Incompatible Configuration
> WARNING: THE FOLLOWING FILES WILL BE DELETED!
>
>
> D:/FreeSWITCH/conf/sip_profiles/external.xml
> D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml
> D:/FreeSWITCH/conf/sip_profiles/internal.xml
>
> regards,
>
> Carlos
>
> On Tue, Aug 25, 2009 at 1:28 PM, Giovanni Maruzzelli <gmaruzz at celliax.org 
> > wrote:
>
> >From the front page of FPBX is not clear you *must* create a trunk/ 
> trunk group.
>
> I was thinking trunks were for outgoing calls, or for receiving from  
> external.
>
> I was just testing internal phones, trying an IVR, so I was thinking
> trunks were not needed.
>
> Can you explain to me?
>
>
> Thanks again,
>
> -giovanni
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
>
> On Tue, Aug 25, 2009 at 8:06 PM, Carlos  
> Talbot<carlos.talbot at gmail.com> wrote:
> > Giovanni,
> > you mean like this message?
> > "Unable to determine location for device. Voicemail password set  
> via FreePBX
> > will not be valid."
> > This is a known FreePBX issue. http://www.freepbx.org/v3/ticket/36
> > Let's keep in ming FreePBX v3 is a developer release and as such  
> many
> > features are in flux and might not work. That being said there  
> some features
> > in the Windows build that still do not work. The biggest one right  
> now is
> > the lack of the php ESL library for Windows which affects the  
> voicemail app.
> > I'm trying to get this to compile but it's been difficult.
> > I do include the .svn files with the FreePBX install so you can  
> freely
> > install TortoiseSVN and update FreePBX at your leisure.
> > With regards to the sip_profiles, did you create a trunk group and  
> trunk?
> > regards,
> > Carlos
> >
> > On Mon, Aug 24, 2009 at 9:56 PM, Giovanni Maruzzelli <gmaruzz at celliax.org 
> >
> > wrote:
> >>
> >> Windows installer does not work for me.
> >>
> >> I've reinstalled various times, same results.
> >>
> >> I can correctly create a number, but when I try to create a  
> device for
> >> that number, it tells me that cannot locate the device, and the
> >> password for vicemail will be invalid.
> >>
> >> After that, it begins to give the php error page, it cannot find  
> the
> >> start < tag in directory/default.xml
> >>
> >> Also for me there are no sofia profiles...
> >>
> >> So, I cannot start to test it (eg: I would like to add mod_skypiax
> >> support to it).
> >>
> >>
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >>
> >>
> >>
> >> On Tue, Aug 25, 2009 at 12:06 AM, Raffaele P.
> >> Guidi<raffaele.p.guidi at gmail.com> wrote:
> >> > This is what I was asking! :D When the installer finished it  
> started the
> >> > whole thing and everything got loaded fine, but when I  
> restarted my
> >> > system
> >> > it didn't (and did not anymore). Well, I will try to install  
> everything
> >> > from
> >> > scratch again and see...
> >> >
> >> > On Mon, Aug 24, 2009 at 20:30, Brian West  
> <brian at freeswitch.org> wrote:
> >> >>
> >> >> If you installed FreePBX then it would be that softwares job  
> to manage
> >> >> the sofia profiles... wouldn't it?
> >> >>
> >> >> /b
> >> >>
> >> >> On Aug 24, 2009, at 1:24 PM, Raffaele P. Guidi wrote:
> >> >>
> >> >> > Actually I did that and it worked fine. I had the problem  
> the SECOND
> >> >> > time I run FS and freepbx. And (@Brian) mod_sofia was loaded  
> but
> >> >> > sip_profiles were not
> >> >>
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
> freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From csa at nowthor.com  Tue Aug 25 19:50:08 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Tue, 25 Aug 2009 22:50:08 -0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>	<4A93EFE1.6010007@nowthor.com>
	<191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>
Message-ID: <4A94A2E0.6040306@nowthor.com>

Anthony,

Yes, you are right, I was thinking strictly in terms of SIP gateways. I 
guess that instead on the tag "gateway", one could use "channel"? For 
example:

<action application="bridge">
   <!-- set some kind of *bridge-specific* variable -->
   <channel order=1 data="sofia/gateway/gw-1/1$1"/>
   <channel order=1 data="sofia/gateway/gw-2/1$1"/>
   <channel order=2 data="sofia/gateway/gw-3/1$1">
       <!-- set some kind of *channel-specific* variable -->
   </channel >
   <channel order=3 data="sofia/gateway/gw-4/1$1"/>
</action>

Would this be acceptable to you?

Regarding the combinations of continue_on_fail and hangup_after bridge, 
I'll give that a try although I suspect it will result in less 
structured and harder to understand markup...

Thanks!

Carlos

Anthony Minessale wrote:
> This suggestion violates the scope boundaries.
>
> gateways are specific concept to mod_sofia so a <gateway> tag in 
> <action> (part of agnostic xml dialplan)
> does not flow properly.
>
> you can also use combinations of continue_on_fail and hangup_after 
> bridge so you can
> just put each bridge statement in it's own action.
>
>
> On Tue, Aug 25, 2009 at 9:06 AM, Carlos S. Antunes <csa at nowthor.com 
> <mailto:csa at nowthor.com>> wrote:
>
>     Max,
>
>     I would like to see something similar too. For example, it would be
>     wonderful if one could specify multiple gateways to try like this or
>     something similar:
>
>     <action application="bridge">
>        <!-- set some kind of *bridge-specific* parameter/variable -->
>        <gateway order=1 data="sofia/gateway/gw-1/1$1"/>
>        <gateway order=1 data="sofia/gateway/gw-2/1$1"/>
>        <gateway order=2 data="sofia/gateway/gw-3/1$1">
>            <!-- set some kind of *gw-specific* parameter/variable -->
>        </gateway>
>        <gateway order=3 data="sofia/gateway/gw-4/1$1"/>
>     </action>
>
>     One would be able to avoid the "[]" and "{}" hacks and combine
>     sequential and simultaneous trying of gateways.
>
>     What do the developers think of this?
>
>     Carlos
>
>     Max Ivanov wrote:
>     > Nowdays I 'm forced to put multiple "|" to find first free
>     gateway, ie
>     >
>     sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
>     > ,
>     > the whole sting is tooo long, is there any shorter way to write
>     same thing? Like
>     > "sofia/gateway/panas*/1000" will try all gateways matching the
>     pattern.
>     >
>     > _______________________________________________
>     > FreeSWITCH-users mailing list
>     > FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     >
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     > http://www.freeswitch.org
>     >
>     >
>
>     _______________________________________________
>     FreeSWITCH-users mailing list
>     FreeSWITCH-users at lists.freeswitch.org
>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>     http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com 
> <mailto:MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com 
> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org 
> <mailto:sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888 
> <http://iax:guest at conference.freeswitch.org/888>
> googletalk:conf+888 at conference.freeswitch.org 
> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
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From gshfreesw at gmail.com  Tue Aug 25 19:51:15 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Tue, 25 Aug 2009 22:51:15 -0400
Subject: [Freeswitch-users] caller id on origination does not work
Message-ID: <5070fcbd0908251951j5a7163b1se346abcdf0664c94@mail.gmail.com>

I am setting the caller id like this in my ESL script:

@con.sendRecv("api originate
{origination_caller_id_number=15103245678}sofia/gateway/junctionnetworks/1#{@number}
&park()")

And the caller id comes out as all zeroes.  The sip trace shows the "from"
as shown in the sofia status command.  This is output of sofia gateway
status and my gateway is under
/usr/local/freeswitch/conf/sip_profiles/external/junctionnetworks.xml .
Also, adding the gateway under /conf/directory/default did not work as the
wiki suugests. What I need to do for the caller_id to come through?


sofia status gateway junctionnetworks
=================================================================================================
Name        junctionnetworks
Scheme      Digest
Realm       jnctn.net
Username    username
Password    yes
>From        <sip:username at jnctn.net <sip%3Ausername at jnctn.net>
;transport=udp>
Contact     <sip:gw+junctionnetworks at 24.87.80.12:5080;transport=udp>
Exten       899xxxxxxx
To          sip:username at sip.jnctn.net <sip%3Ausername at sip.jnctn.net>
Proxy       sip:sip.jnctn.net
Context     public
Expires     3600
Freq        3600
Ping        0
PingFreq    0
State       REGED
Status      UP
CallsIN     0
CallsOUT    1
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From krice at freeswitch.org  Tue Aug 25 20:00:30 2009
From: krice at freeswitch.org (Ken Rice)
Date: Tue, 25 Aug 2009 22:00:30 -0500
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <4A94A2E0.6040306@nowthor.com>
Message-ID: <C6BA0F7E.5D757%krice@freeswitch.org>

That still wouldn?t work... An action has 2 parameters application and
data... And deeper then that and you have to start re-arranging all sorts of
things...

Continue_on_fail and hangup_after_bridge like tony pointed out are what you
want if you don?t want to use the | delimiting ... I use these all the time
with gateway counts > 10 just stacking additional actions for each bridge
line



From: "Carlos S. Antunes" <csa at nowthor.com>
Organization: Nowthor Corporation
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Tue, 25 Aug 2009 22:50:08 -0400
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] how to avoid many "|" in bridge application?

Anthony,

Yes, you are right, I was thinking strictly in terms of SIP gateways. I
guess that instead on the tag "gateway", one could use "channel"? For
example:

<action application="bridge">
   <!-- set some kind of *bridge-specific* variable -->
   <channel order=1 data="sofia/gateway/gw-1/1$1"/>
   <channel order=1 data="sofia/gateway/gw-2/1$1"/>
   <channel order=2 data="sofia/gateway/gw-3/1$1">
       <!-- set some kind of *channel-specific* variable -->
   </channel >
   <channel order=3 data="sofia/gateway/gw-4/1$1"/>
</action>

Would this be acceptable to you?

Regarding the combinations of continue_on_fail and hangup_after bridge, I'll
give that a try although I suspect it will result in less structured and
harder to understand markup...

Thanks!

Carlos

Anthony Minessale wrote:
> This suggestion violates the scope boundaries.
>  
> gateways are specific concept to mod_sofia so a <gateway> tag in <action>
> (part of agnostic xml dialplan)
> does not flow properly.
>  
> you can also use combinations of continue_on_fail and hangup_after bridge so
> you can
> just put each bridge statement in it's own action.
>  
>  
>  
> On Tue, Aug 25, 2009 at 9:06 AM, Carlos S. Antunes <csa at nowthor.com> wrote:
>  
>> Max,
>>  
>> I would like to see something similar too. For example, it would be
>> wonderful if one could specify multiple gateways to try like this or
>> something similar:
>>  
>> <action application="bridge">
>>    <!-- set some kind of *bridge-specific* parameter/variable -->
>>    <gateway order=1 data="sofia/gateway/gw-1/1$1"/>
>>    <gateway order=1 data="sofia/gateway/gw-2/1$1"/>
>>    <gateway order=2 data="sofia/gateway/gw-3/1$1">
>>        <!-- set some kind of *gw-specific* parameter/variable -->
>>    </gateway>
>>    <gateway order=3 data="sofia/gateway/gw-4/1$1"/>
>> </action>
>>  
>> One would be able to avoid the "[]" and "{}" hacks and combine
>> sequential and simultaneous trying of gateways.
>>  
>> What do the developers think of this?
>>  
>> Carlos
>>   
>>  
>> 
>> Max Ivanov wrote:
>>> > Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
>>> > 
>>> sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas1
>>> 13/1000
>>> > ,
>>> > the whole sting is tooo long, is there any shorter way to write same
>>> thing? Like
>>> > "sofia/gateway/panas*/1000" will try all gateways matching the pattern.
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>  
>> _______________________________________________
>> FreeSWITCH-users mailing list
>>  FreeSWITCH-users at lists.freeswitch.org
>>  http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>  http://www.freeswitch.org
>>  
>>  
>>  
>>  
>>  
>>  
>>  
>> -- 
>> Anthony Minessale II
>>  
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>  
>> AIM: anthm
>>  MSN:anthony_minessale at hotmail.com
>> <mailto:MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> <mailto:PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net <http://irc.freenode.net> #freeswitch
>>  
>> FreeSWITCH Developer Conference
>>  sip:888 at conference.freeswitch.org
>> <mailto:sip%3A888 at conference.freeswitch.org>
>>  iax:guest at conference.freeswitch.org/888
>> <http://iax:guest at conference.freeswitch.org/888>
>>  googletalk:conf+888 at conference.freeswitch.org
>> <mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>  
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>   
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org

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From sprice at gmail.com  Tue Aug 25 20:07:33 2009
From: sprice at gmail.com (SP)
Date: Tue, 25 Aug 2009 22:07:33 -0500
Subject: [Freeswitch-users] caller id on origination does not work
In-Reply-To: <5070fcbd0908251951j5a7163b1se346abcdf0664c94@mail.gmail.com>
References: <5070fcbd0908251951j5a7163b1se346abcdf0664c94@mail.gmail.com>
Message-ID: <7e2ac3270908252007g5dfb9390w806267550726b99a@mail.gmail.com>

You can try playing with this in your gateway profile
  <param name="caller-id-in-from" value="true"/>

Not sure what it'll do to your registration, give it a try

On Tue, Aug 25, 2009 at 21:51, Shameem Shiek <gshfreesw at gmail.com> wrote:

> I am setting the caller id like this in my ESL script:
>
> @con.sendRecv("api originate
> {origination_caller_id_number=15103245678}sofia/gateway/junctionnetworks/1#{@number}
> &park()")
>
> And the caller id comes out as all zeroes.  The sip trace shows the "from"
> as shown in the sofia status command.  This is output of sofia gateway
> status and my gateway is under
> /usr/local/freeswitch/conf/sip_profiles/external/junctionnetworks.xml .
> Also, adding the gateway under /conf/directory/default did not work as the
> wiki suugests. What I need to do for the caller_id to come through?
>
>
> sofia status gateway junctionnetworks
>
> =================================================================================================
> Name        junctionnetworks
> Scheme      Digest
> Realm       jnctn.net
> Username    username
> Password    yes
> From        <sip:username at jnctn.net <sip%3Ausername at jnctn.net>
> ;transport=udp>
> Contact     <sip:gw+junctionnetworks at 24.87.80.12:5080;transport=udp>
> Exten       899xxxxxxx
> To          sip:username at sip.jnctn.net <sip%3Ausername at sip.jnctn.net>
> Proxy       sip:sip.jnctn.net
> Context     public
> Expires     3600
> Freq        3600
> Ping        0
> PingFreq    0
> State       REGED
> Status      UP
> CallsIN     0
> CallsOUT    1
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Shannon
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From ivanov.maxim at gmail.com  Tue Aug 25 22:44:48 2009
From: ivanov.maxim at gmail.com (Max Ivanov)
Date: Wed, 26 Aug 2009 09:44:48 +0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <C6BA0F7E.5D757%krice@freeswitch.org>
References: <4A94A2E0.6040306@nowthor.com>
	<C6BA0F7E.5D757%krice@freeswitch.org>
Message-ID: <fbfa7b0c0908252244x65d19808r967a23332de6a0a0@mail.gmail.com>

>
> Continue_on_fail and hangup_after_bridge like tony pointed out are what you
> want if you don?t want to use the | delimiting ... I use these all the time
> with gateway counts > 10 just stacking additional actions for each bridge
> line

Let's imagine that I need to call 1000,1001,1002 via  gw1,gw2,gw3 by
choosing first available. How dialplan would look like?

continue_on_fail=True
hangup_after_bridge=True
bridge gw1/1000
bridge gw1/1001
bridge gw1/1002
bridge gw2/1000
bridge gw2/1001
bridge gw2/1002
bridge gw3/1000
bridge gw3/1001
bridge gw3/1002

Is it easy to understand? From my point of view it's not. Compare to this:
continue_on_fail=True
hangup_after_bridge=True
bridge ${get_avail_gw(gw1,gw2,gw3)}/1000
bridge ${get_avail_gw(gw1,gw2,gw3)}/1001
bridge ${get_avail_gw(gw1,gw2,gw3)}/1002



From mike at yes.net.ua  Wed Aug 26 00:51:27 2009
From: mike at yes.net.ua (Mike Tkachuk)
Date: Wed, 26 Aug 2009 10:51:27 +0300
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <fbfa7b0c0908252244x65d19808r967a23332de6a0a0@mail.gmail.com>
References: <4A94A2E0.6040306@nowthor.com>
	<C6BA0F7E.5D757%krice@freeswitch.org>
	<fbfa7b0c0908252244x65d19808r967a23332de6a0a0@mail.gmail.com>
Message-ID: <1236433376.20090826105127@yes.net.ua>

Hello Max,

 Don't improve that what's working well. ;)

Wednesday, August 26, 2009 8:44:48 AM, you wrote:

>>
>> Continue_on_fail and hangup_after_bridge like tony pointed out are what you
>> want if you don?t want to use the | delimiting ... I use these all the time
>> with gateway counts > 10 just stacking additional actions for each bridge
>> line

MI> Let's imagine that I need to call 1000,1001,1002 via  gw1,gw2,gw3 by
MI> choosing first available. How dialplan would look like?

MI> continue_on_fail=True
MI> hangup_after_bridge=True
MI> bridge gw1/1000
MI> bridge gw1/1001
MI> bridge gw1/1002
MI> bridge gw2/1000
MI> bridge gw2/1001
MI> bridge gw2/1002
MI> bridge gw3/1000
MI> bridge gw3/1001
MI> bridge gw3/1002

MI> Is it easy to understand? From my point of view it's not. Compare to this:
MI> continue_on_fail=True
MI> hangup_after_bridge=True
MI> bridge ${get_avail_gw(gw1,gw2,gw3)}/1000
MI> bridge ${get_avail_gw(gw1,gw2,gw3)}/1001
MI> bridge ${get_avail_gw(gw1,gw2,gw3)}/1002



--
Mike Tkachuk




From msc at freeswitch.org  Wed Aug 26 01:01:34 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 26 Aug 2009 01:01:34 -0700
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
Message-ID: <5DF610F9-6EB5-43CC-BB4F-3513C78F608F@freeswitch.org>


> I have looked at that but I am confused on which files need to be  
> edited. Since I have already installed in Wanpipe mode with the  
> Sangoma card I skipped straight to the Wanpipe section. It mentions  
> setting the [span wanpie PRI_1] etc in the openzap.conf then further  
> down it mentions editing the autoload_configs/openzap.conf.xml with  
> a completely different format of config.
>
>
>
> Are these the same file? If not where does the first one mentioned  
> reside?
>
>
These are two different files. The openzap.conf file is in the  
freeswitch/conf directory. It is for configuring OZ to talk to the  
Sangoma card. The openzap.conf.xml file is for configuring the actual  
openzap module in FreeSWITCH.

The specs you listed on the T1 you have indicate that it's a PRI.  
However, you need to know which protocol, like NI2 or DMS. Have you  
ever done a PRI turn up? Who is the carrier?

-MC
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From aep.lists at it46.se  Wed Aug 26 01:04:00 2009
From: aep.lists at it46.se (Alberto Escudero-Pascual (lists))
Date: Wed, 26 Aug 2009 10:04:00 +0200
Subject: [Freeswitch-users] Internal Profile sends RTP to external IP when
 using two network interfaces
Message-ID: <46b96649f81b0bfcb111928a55450211.squirrel@correo.nodo50.org>

Hi,

I have a FS box with two physical network interfaces. The internal
interface is hosting several internal phones. I have binded the internal
profile SIP/RTP/IP to the private interface. Phones registered correctly
but with User: 1000 at external.ip.address in sofia status profile internal

When I place a call between two internal phones, RTP traffic is send to
the external IP address of the FS box instead of the internal.

The SIP/SDP messages send from FS carry the external IP instead of the
internal.

The result is that no RTP media arrives to any of the phones.

/aep


-- 
Stopping junk mailers is good for the environment






From tculjaga at gmail.com  Wed Aug 26 01:08:57 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 26 Aug 2009 10:08:57 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com>
Message-ID: <65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com>

Hi Giovanny,

regarding ubuntu, did you mean 8.04 server or desktop ?


On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Definitely go for 64 bit OS.
>
> If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
> used both for development and for heavy duty production.
>
> Also Ubuntu 8.04 is good.
>
> Other versions/distros are less used by the community.
>
> Adding RAM and CPUs helps to scale up.
>
> -gm
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
> Cell : +39-347-2665618
>
>
>
>
> On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com>
> wrote:
> > Hey Giovanni,
> >
> > thanks for the tip... indeed the db files were heavily used regardless if
> i
> > started freeswitch with nosql option (freeswitch -nosql)... FS was not
> > writing anything into that files ... instead it was just accessing it....
> > This behaviour leads to a waste of 40% CPU time... waiting for other
> > processes (mainly disk access) to finish!!!
> >
> > I moved freeswitch/db/ to a ramdisk and the performance got a boost to
> 140
> > CPS with a CPU load of 80%. I was keeping the machine for a while (20 -
> 30
> > minutes) on that rate when i sow CPU suddenly went to 100% and FS
> becoming
> > irresponsive :).
> >
> >
> > What can be wrong?
> > What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
> > cross?
> > What fine tuning do we need in order to asure a long high load run?
> >
> >
> >
> > Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
> sense
> > to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
> will
> > FS perofomr drastically better 20%+ ?
> >
> >
> > Tihomir.
> >
> >
> > On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli <
> gmaruzz at celliax.org>
> > wrote:
> >>
> >> Maybe your load comes from disk access?
> >>
> >> Try putting the sql and log directories on a ramdisk.
> >>
> >> OTH,
> >>
> >> -giovanni
> >>
> >> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com>
> >> wrote:
> >> > Hello,
> >> >
> >> > i'm trying to use freeswitch as a redirecting server meaning FS has to
> >> > receive an INVITE and according to some rules it will redirect calls
> to
> >> > other destinations.
> >> >
> >> >
> >> > CALLING_USER                FREESWITCH
> SOMEWHERE
> >> >
> >> > INVITE ------------------------------->
> >> >            <------------------------------ 100 Trying
> >> >            <------------------------------ 302 Moved Temporary
> >> > ACK    ------------------------------->
> >> >
> >> >
> INVITE--------------------------------------------------------------------------------->
> >> >
> >> >
> >> >
> >> > Well, wverything works well except i have perfromance issues .... on
> my
> >> > HW
> >> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
> Temporary).
> >> > When
> >> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS
> i
> >> > see
> >> > "calls" being build up in FS and the delay begining to grow.
> >> >
> >> > When i observe the machine, load average is almost nothing (load
> >> > average:
> >> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
> >> > taking
> >> > most load... all others are just sitting there with 1-5 % CPU time.
> >> > This looks to me as FS handles 302 messages in a single thread?!?!
> >> >
> >> >
> >> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
> >> >
> >> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41,
> 0.61,
> >> > 0.60
> >> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0 zombie
> >> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,  0.2%si,
> >> > 0.0%st
> >> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k
> buffers
> >> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k
> cached
> >> >
> >> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
> >> > COMMAND
> >> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
> >> > freeswitch
> >> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
> >> > freeswitch
> >> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
> >> > freeswitch
> >> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
> >> > freeswitch
> >> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
> >> > freeswitch
> >> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
> >> > freeswitch
> >> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
> >> > freeswitch
> >> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
> >> > freeswitch
> >> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
> >> > freeswitch
> >> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50
> freeswitch
> >> >
> >> >
> >> >
> >> > cat /proc/cpuinfo
> >> > processor       : 0
> >> > vendor_id       : GenuineIntel
> >> > cpu family      : 6
> >> > model           : 15
> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> >> > stepping        : 6
> >> > cpu MHz         : 2333.560
> >> > cache size      : 4096 KB
> >> > physical id     : 0
> >> > siblings        : 2
> >> > core id         : 0
> >> > cpu cores       : 2
> >> > apicid          : 0
> >> > initial apicid  : 0
> >> > fdiv_bug        : no
> >> > hlt_bug         : no
> >> > f00f_bug        : no
> >> > coma_bug        : no
> >> > fpu             : yes
> >> > fpu_exception   : yes
> >> > cpuid level     : 10
> >> > wp              : yes
> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
> >> > mca
> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
> ssse3
> >> > cx16
> >> > xtpr dca lahf_lm
> >> > bogomips        : 4670.78
> >> > clflush size    : 64
> >> > power management:
> >> >
> >> > processor       : 1
> >> > vendor_id       : GenuineIntel
> >> > cpu family      : 6
> >> > model           : 15
> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> >> > stepping        : 6
> >> > cpu MHz         : 2333.560
> >> > cache size      : 4096 KB
> >> > physical id     : 0
> >> > siblings        : 2
> >> > core id         : 1
> >> > cpu cores       : 2
> >> > apicid          : 1
> >> > initial apicid  : 1
> >> > fdiv_bug        : no
> >> > hlt_bug         : no
> >> > f00f_bug        : no
> >> > coma_bug        : no
> >> > fpu             : yes
> >> > fpu_exception   : yes
> >> > cpuid level     : 10
> >> > wp              : yes
> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr pge
> >> > mca
> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
> ssse3
> >> > cx16
> >> > xtpr dca lahf_lm
> >> > bogomips        : 4666.82
> >> > clflush size    : 64
> >> > power management:
> >> >
> >> >
> >> >
> >> > uname -a
> >> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009
> i686
> >> > GNU/Linux
> >> >
> >> >
> >> >
> >> > Of course, i've tuned the machine up
> >> >
> >> > ulimit -c unlimited
> >> > ulimit -d unlimited
> >> > ulimit -f unlimited
> >> > ulimit -i unlimited
> >> > ulimit -n 999999
> >> > ulimit -q unlimited
> >> > ulimit -u unlimited
> >> > ulimit -v unlimited
> >> > ulimit -x unlimited
> >> > ulimit -s 240
> >> > ulimit -l unlimited
> >> > ulimit -a
> >> >
> >> >
> >> > Started FS with minimum modules but still 40 CPS seems to be the
> limit.
> >> >
> >> >
> >> > So, is there any way to improve performance?
> >> >
> >> >
> >> > Tihomir.
> >> >
> >> >
> >> >
> >> >
> >> >
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From gmaruzz at celliax.org  Wed Aug 26 01:25:19 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 26 Aug 2009 10:25:19 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com> 
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com> 
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com> 
	<7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com> 
	<65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com>
Message-ID: <7b197bef0908260125m56c3d681l26cb18a5e34aaa33@mail.gmail.com>

netbook remix


joking! Server 64bit :-)

-gm



On Wed, Aug 26, 2009 at 10:08 AM, Tihomir Culjaga<tculjaga at gmail.com> wrote:
> Hi Giovanny,
>
> regarding ubuntu, did you mean 8.04 server or desktop ?
>
>
> On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> Definitely go for 64 bit OS.
>>
>> If you want to be safe and sure, ?go for CentOS 5.2 64bit. Is the one
>> used both for development and for heavy duty production.
>>
>> Also Ubuntu 8.04 is good.
>>
>> Other versions/distros are less used by the community.
>>
>> Adding RAM and CPUs helps to scale up.
>>
>> -gm
>>
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>> Cell : +39-347-2665618
>>
>>
>>
>>
>> On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com>
>> wrote:
>> > Hey Giovanni,
>> >
>> > thanks for the tip... indeed the db files were heavily used regardless
>> > if i
>> > started freeswitch with nosql option (freeswitch -nosql)... FS was not
>> > writing anything into that files ... instead it was just accessing
>> > it....
>> > This behaviour leads to a waste of 40% CPU time... waiting for other
>> > processes (mainly disk access) to finish!!!
>> >
>> > I moved freeswitch/db/ to a ramdisk and the performance got a boost to
>> > 140
>> > CPS with a CPU load of 80%. I was keeping the machine for a while (20 -
>> > 30
>> > minutes) on that rate when i sow CPU suddenly went to 100% and FS
>> > becoming
>> > irresponsive :).
>> >
>> >
>> > What can be wrong?
>> > What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
>> > cross?
>> > What fine tuning do we need in order to asure a long high load run?
>> >
>> >
>> >
>> > Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
>> > sense
>> > to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
>> > will
>> > FS perofomr drastically better 20%+ ?
>> >
>> >
>> > Tihomir.
>> >
>> >
>> > On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
>> > <gmaruzz at celliax.org>
>> > wrote:
>> >>
>> >> Maybe your load comes from disk access?
>> >>
>> >> Try putting the sql and log directories on a ramdisk.
>> >>
>> >> OTH,
>> >>
>> >> -giovanni
>> >>
>> >> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com>
>> >> wrote:
>> >> > Hello,
>> >> >
>> >> > i'm trying to use freeswitch as a redirecting server meaning FS has
>> >> > to
>> >> > receive an INVITE and according to some rules it will redirect calls
>> >> > to
>> >> > other destinations.
>> >> >
>> >> >
>> >> > CALLING_USER??????????????? FREESWITCH
>> >> > SOMEWHERE
>> >> >
>> >> > INVITE ------------------------------->
>> >> > ?????????? <------------------------------ 100 Trying
>> >> > ?????????? <------------------------------ 302 Moved Temporary
>> >> > ACK ?? ------------------------------->
>> >> >
>> >> >
>> >> > INVITE--------------------------------------------------------------------------------->
>> >> >
>> >> >
>> >> >
>> >> > Well, wverything works well except i have perfromance issues .... on
>> >> > my
>> >> > HW
>> >> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
>> >> > Temporary).
>> >> > When
>> >> > i increase the rate, FS starts delaying 302 response. Right at 50 CPS
>> >> > i
>> >> > see
>> >> > "calls" being build up in FS and the delay begining to grow.
>> >> >
>> >> > When i observe the machine, load average is almost nothing (load
>> >> > average:
>> >> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
>> >> > taking
>> >> > most load... all others are just sitting there with 1-5 % CPU time.
>> >> > This looks to me as FS handles 302 messages in a single thread?!?!
>> >> >
>> >> >
>> >> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>> >> >
>> >> > top - 10:41:37 up 167 days, 20:42,? 3 users,? load average: 1.41,
>> >> > 0.61,
>> >> > 0.60
>> >> > Tasks:? 83 total,?? 2 running,? 81 sleeping,?? 0 stopped,?? 0 zombie
>> >> > Cpu(s): 25.3%us,? 1.5%sy,? 0.0%ni, 30.3%id, 42.7%wa,? 0.0%hi,
>> >> > 0.2%si,
>> >> > 0.0%st
>> >> > Mem:?? 2074520k total,?? 571244k used,? 1503276k free,?? 259604k
>> >> > buffers
>> >> > Swap:? 2650684k total,???? 3020k used,? 2647664k free,?? 153868k
>> >> > cached
>> >> >
>> >> > ? PID USER????? PR? NI? VIRT? RES? SHR S %CPU %MEM??? TIME+
>> >> > COMMAND
>> >> > ?4814 root????? 20?? 0 34188? 20m 3780 S?? 38? 1.0?? 3:10.29
>> >> > freeswitch
>> >> > ?4800 root????? 20?? 0 34188? 20m 3780 S??? 6? 1.0?? 0:08.26
>> >> > freeswitch
>> >> > ?4798 root????? 20?? 0 34188? 20m 3780 R??? 5? 1.0?? 0:24.46
>> >> > freeswitch
>> >> > ?4787 root????? 20?? 0 34188? 20m 3780 S??? 2? 1.0?? 0:11.24
>> >> > freeswitch
>> >> > ?4794 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.42
>> >> > freeswitch
>> >> > ?4803 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:11.74
>> >> > freeswitch
>> >> > ?4788 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.96
>> >> > freeswitch
>> >> > ?4804 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.64
>> >> > freeswitch
>> >> > ?4807 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:01.68
>> >> > freeswitch
>> >> > ?4811 root????? 20?? 0 34188? 20m 3780 S??? 1? 1.0?? 0:02.50
>> >> > freeswitch
>> >> >
>> >> >
>> >> >
>> >> > cat /proc/cpuinfo
>> >> > processor?????? : 0
>> >> > vendor_id?????? : GenuineIntel
>> >> > cpu family????? : 6
>> >> > model?????????? : 15
>> >> > model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
>> >> > stepping??????? : 6
>> >> > cpu MHz???????? : 2333.560
>> >> > cache size????? : 4096 KB
>> >> > physical id???? : 0
>> >> > siblings??????? : 2
>> >> > core id???????? : 0
>> >> > cpu cores?????? : 2
>> >> > apicid????????? : 0
>> >> > initial apicid? : 0
>> >> > fdiv_bug??????? : no
>> >> > hlt_bug???????? : no
>> >> > f00f_bug??????? : no
>> >> > coma_bug??????? : no
>> >> > fpu???????????? : yes
>> >> > fpu_exception?? : yes
>> >> > cpuid level???? : 10
>> >> > wp????????????? : yes
>> >> > flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
>> >> > pge
>> >> > mca
>> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
>> >> > ssse3
>> >> > cx16
>> >> > xtpr dca lahf_lm
>> >> > bogomips??????? : 4670.78
>> >> > clflush size??? : 64
>> >> > power management:
>> >> >
>> >> > processor?????? : 1
>> >> > vendor_id?????? : GenuineIntel
>> >> > cpu family????? : 6
>> >> > model?????????? : 15
>> >> > model name????? : Intel(R) Xeon(R) CPU??????????? 5140? @ 2.33GHz
>> >> > stepping??????? : 6
>> >> > cpu MHz???????? : 2333.560
>> >> > cache size????? : 4096 KB
>> >> > physical id???? : 0
>> >> > siblings??????? : 2
>> >> > core id???????? : 1
>> >> > cpu cores?????? : 2
>> >> > apicid????????? : 1
>> >> > initial apicid? : 1
>> >> > fdiv_bug??????? : no
>> >> > hlt_bug???????? : no
>> >> > f00f_bug??????? : no
>> >> > coma_bug??????? : no
>> >> > fpu???????????? : yes
>> >> > fpu_exception?? : yes
>> >> > cpuid level???? : 10
>> >> > wp????????????? : yes
>> >> > flags?????????? : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
>> >> > pge
>> >> > mca
>> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
>> >> > ssse3
>> >> > cx16
>> >> > xtpr dca lahf_lm
>> >> > bogomips??????? : 4666.82
>> >> > clflush size??? : 64
>> >> > power management:
>> >> >
>> >> >
>> >> >
>> >> > uname -a
>> >> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009
>> >> > i686
>> >> > GNU/Linux
>> >> >
>> >> >
>> >> >
>> >> > Of course, i've tuned the machine up
>> >> >
>> >> > ulimit -c unlimited
>> >> > ulimit -d unlimited
>> >> > ulimit -f unlimited
>> >> > ulimit -i unlimited
>> >> > ulimit -n 999999
>> >> > ulimit -q unlimited
>> >> > ulimit -u unlimited
>> >> > ulimit -v unlimited
>> >> > ulimit -x unlimited
>> >> > ulimit -s 240
>> >> > ulimit -l unlimited
>> >> > ulimit -a
>> >> >
>> >> >
>> >> > Started FS with minimum modules but still 40 CPS seems to be the
>> >> > limit.
>> >> >
>> >> >
>> >> > So, is there any way to improve performance?
>> >> >
>> >> >
>> >> > Tihomir.
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From aep.lists at it46.se  Wed Aug 26 01:28:36 2009
From: aep.lists at it46.se (Alberto Escudero-Pascual (lists))
Date: Wed, 26 Aug 2009 10:28:36 +0200
Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
In-Reply-To: <191c3a030908251354l25206d0ch19836c8996f43e84@mail.gmail.com>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
	<191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
	<6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>
	<5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org>
	<45b2e91d7aa77afcf6430a14dde4b886.squirrel@correo.nodo50.org>
	<191c3a030908251354l25206d0ch19836c8996f43e84@mail.gmail.com>
Message-ID: <a41cc07ca7c2a0f3c51ceb9b37ea6bc6.squirrel@correo.nodo50.org>

I ran strace from freeswitch and from the command line. lame segfaults
when run from system FS.

The only obvious different i see is in the execve() /* XX vars */ apart
from the final Segfault

From
execve("/usr/local/freeswitch/bin/lame",
["/usr/local/freeswitch/bin/lame", "/tmp/foo.wav", "/tmp/foo.mp3", "-S"],
[/* 16 vars */]) = 0


>From FS
execve("/usr/local/freeswitch/bin/lame",
["/usr/local/freeswitch/bin/lame", "/tmp/foo.wav",
"/tmp/fooooooooooooooo.mp3", "-S"], [/* 14 vars */]) = 0

I am attaching the full straces in case they are of any help. Not sure if
this deserves a jira

/aep
-- 
Stopping junk mailers is good for the environment

> maybe it's writing some err to stderr that is being suppressed somehow
>
> On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists) <
> aep.lists at it46.se> wrote:
>
>> Hi Brian,
>>
>> >From the CLI>
>>
>> freeswitch at open46> system /usr/local/freeswitch/bin/lame -V2
>> /tmp/foo.wav
>> /tmp/foo.mp3 -S
>> 2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing
>> command:
>> /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S
>> API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
>> /tmp/foo.mp3 -S)] output:
>> +OK
>>
>> open46:/tmp# ls
>> foo.wav
>>
>>
>> and running the command from the command line:
>>
>>
>> open46:/tmp#  /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
>> /tmp/foo.mp3
>> -Sopen46:/tmp# ls
>> foo.mp3  foo.wav
>>
>>
>> If I do the same with lame397
>>
>> freeswitch at open46> system /usr/local/freeswitch/bin/lame397 -V2
>> /tmp/foo.wav /tmp/foo.mp3 -S
>> 2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing
>> command:
>> /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S
>> API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav
>> /tmp/foo.mp3 -S)] output:
>> +OK
>>
>> open46:/tmp# ls
>> foo.mp3  foo.wav
>>
>>
>> Highly paranormal! Sorry for hijacking the previous thread.
>>
>> /aep
>>
>> --
>> Stopping junk mailers is good for the environment
>>
>> > Try running it at the CLI and see if you see any errors.  Also please
>> > do not hijack threads.  The original thread "[Freeswitch-users] XML-
>> > RPC on different ip than 0.0.0.0" which was hijacked by clicking
>> > reply, changing the subject and clicking send.  Please in the future
>> > do not do that as it clutters up the threading and could get your
>> > query lost in the noise.
>> >
>> > Thanks,
>> > Brian
>> >
>> > On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists) wrote:
>> >
>> >> Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
>> >> appears, if I use version 3.97 (older version), it does!.
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Wed Aug 26 01:29:07 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 26 Aug 2009 10:29:07 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <7b197bef0908260125m56c3d681l26cb18a5e34aaa33@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com>
	<65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com>
	<7b197bef0908260125m56c3d681l26cb18a5e34aaa33@mail.gmail.com>
Message-ID: <65d96fc80908260129xebbf460raffaa38c72c3281a@mail.gmail.com>

intanto e il centos che si sta installando :)

grazie.

T.

On Wed, Aug 26, 2009 at 10:25 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> netbook remix
>
>
> joking! Server 64bit :-)
>
> -gm
>
>
>
> On Wed, Aug 26, 2009 at 10:08 AM, Tihomir Culjaga<tculjaga at gmail.com>
> wrote:
> > Hi Giovanny,
> >
> > regarding ubuntu, did you mean 8.04 server or desktop ?
> >
> >
> > On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli <
> gmaruzz at celliax.org>
> > wrote:
> >>
> >> Definitely go for 64 bit OS.
> >>
> >> If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
> >> used both for development and for heavy duty production.
> >>
> >> Also Ubuntu 8.04 is good.
> >>
> >> Other versions/distros are less used by the community.
> >>
> >> Adding RAM and CPUs helps to scale up.
> >>
> >> -gm
> >>
> >>
> >>
> >> Sincerely,
> >>
> >> Giovanni Maruzzelli
> >> Cell : +39-347-2665618
> >>
> >>
> >>
> >>
> >> On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com>
> >> wrote:
> >> > Hey Giovanni,
> >> >
> >> > thanks for the tip... indeed the db files were heavily used regardless
> >> > if i
> >> > started freeswitch with nosql option (freeswitch -nosql)... FS was not
> >> > writing anything into that files ... instead it was just accessing
> >> > it....
> >> > This behaviour leads to a waste of 40% CPU time... waiting for other
> >> > processes (mainly disk access) to finish!!!
> >> >
> >> > I moved freeswitch/db/ to a ramdisk and the performance got a boost to
> >> > 140
> >> > CPS with a CPU load of 80%. I was keeping the machine for a while (20
> -
> >> > 30
> >> > minutes) on that rate when i sow CPU suddenly went to 100% and FS
> >> > becoming
> >> > irresponsive :).
> >> >
> >> >
> >> > What can be wrong?
> >> > What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should not
> >> > cross?
> >> > What fine tuning do we need in order to asure a long high load run?
> >> >
> >> >
> >> >
> >> > Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it have
> >> > sense
> >> > to move my OS to 64 bit? ... will FS gain more preformance ?... I mean
> >> > will
> >> > FS perofomr drastically better 20%+ ?
> >> >
> >> >
> >> > Tihomir.
> >> >
> >> >
> >> > On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
> >> > <gmaruzz at celliax.org>
> >> > wrote:
> >> >>
> >> >> Maybe your load comes from disk access?
> >> >>
> >> >> Try putting the sql and log directories on a ramdisk.
> >> >>
> >> >> OTH,
> >> >>
> >> >> -giovanni
> >> >>
> >> >> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<tculjaga at gmail.com
> >
> >> >> wrote:
> >> >> > Hello,
> >> >> >
> >> >> > i'm trying to use freeswitch as a redirecting server meaning FS has
> >> >> > to
> >> >> > receive an INVITE and according to some rules it will redirect
> calls
> >> >> > to
> >> >> > other destinations.
> >> >> >
> >> >> >
> >> >> > CALLING_USER                FREESWITCH
> >> >> > SOMEWHERE
> >> >> >
> >> >> > INVITE ------------------------------->
> >> >> >            <------------------------------ 100 Trying
> >> >> >            <------------------------------ 302 Moved Temporary
> >> >> > ACK    ------------------------------->
> >> >> >
> >> >> >
> >> >> >
> INVITE--------------------------------------------------------------------------------->
> >> >> >
> >> >> >
> >> >> >
> >> >> > Well, wverything works well except i have perfromance issues ....
> on
> >> >> > my
> >> >> > HW
> >> >> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
> >> >> > Temporary).
> >> >> > When
> >> >> > i increase the rate, FS starts delaying 302 response. Right at 50
> CPS
> >> >> > i
> >> >> > see
> >> >> > "calls" being build up in FS and the delay begining to grow.
> >> >> >
> >> >> > When i observe the machine, load average is almost nothing (load
> >> >> > average:
> >> >> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one thread
> >> >> > taking
> >> >> > most load... all others are just sitting there with 1-5 % CPU time.
> >> >> > This looks to me as FS handles 302 messages in a single thread?!?!
> >> >> >
> >> >> >
> >> >> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
> >> >> >
> >> >> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41,
> >> >> > 0.61,
> >> >> > 0.60
> >> >> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0
> zombie
> >> >> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,
> >> >> > 0.2%si,
> >> >> > 0.0%st
> >> >> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k
> >> >> > buffers
> >> >> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k
> >> >> > cached
> >> >> >
> >> >> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
> >> >> > COMMAND
> >> >> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
> >> >> > freeswitch
> >> >> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
> >> >> > freeswitch
> >> >> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
> >> >> > freeswitch
> >> >> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
> >> >> > freeswitch
> >> >> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
> >> >> > freeswitch
> >> >> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
> >> >> > freeswitch
> >> >> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
> >> >> > freeswitch
> >> >> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
> >> >> > freeswitch
> >> >> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
> >> >> > freeswitch
> >> >> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50
> >> >> > freeswitch
> >> >> >
> >> >> >
> >> >> >
> >> >> > cat /proc/cpuinfo
> >> >> > processor       : 0
> >> >> > vendor_id       : GenuineIntel
> >> >> > cpu family      : 6
> >> >> > model           : 15
> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> >> >> > stepping        : 6
> >> >> > cpu MHz         : 2333.560
> >> >> > cache size      : 4096 KB
> >> >> > physical id     : 0
> >> >> > siblings        : 2
> >> >> > core id         : 0
> >> >> > cpu cores       : 2
> >> >> > apicid          : 0
> >> >> > initial apicid  : 0
> >> >> > fdiv_bug        : no
> >> >> > hlt_bug         : no
> >> >> > f00f_bug        : no
> >> >> > coma_bug        : no
> >> >> > fpu             : yes
> >> >> > fpu_exception   : yes
> >> >> > cpuid level     : 10
> >> >> > wp              : yes
> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
> >> >> > pge
> >> >> > mca
> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
> >> >> > ssse3
> >> >> > cx16
> >> >> > xtpr dca lahf_lm
> >> >> > bogomips        : 4670.78
> >> >> > clflush size    : 64
> >> >> > power management:
> >> >> >
> >> >> > processor       : 1
> >> >> > vendor_id       : GenuineIntel
> >> >> > cpu family      : 6
> >> >> > model           : 15
> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
> >> >> > stepping        : 6
> >> >> > cpu MHz         : 2333.560
> >> >> > cache size      : 4096 KB
> >> >> > physical id     : 0
> >> >> > siblings        : 2
> >> >> > core id         : 1
> >> >> > cpu cores       : 2
> >> >> > apicid          : 1
> >> >> > initial apicid  : 1
> >> >> > fdiv_bug        : no
> >> >> > hlt_bug         : no
> >> >> > f00f_bug        : no
> >> >> > coma_bug        : no
> >> >> > fpu             : yes
> >> >> > fpu_exception   : yes
> >> >> > cpuid level     : 10
> >> >> > wp              : yes
> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
> >> >> > pge
> >> >> > mca
> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
> >> >> > ssse3
> >> >> > cx16
> >> >> > xtpr dca lahf_lm
> >> >> > bogomips        : 4666.82
> >> >> > clflush size    : 64
> >> >> > power management:
> >> >> >
> >> >> >
> >> >> >
> >> >> > uname -a
> >> >> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC 2009
> >> >> > i686
> >> >> > GNU/Linux
> >> >> >
> >> >> >
> >> >> >
> >> >> > Of course, i've tuned the machine up
> >> >> >
> >> >> > ulimit -c unlimited
> >> >> > ulimit -d unlimited
> >> >> > ulimit -f unlimited
> >> >> > ulimit -i unlimited
> >> >> > ulimit -n 999999
> >> >> > ulimit -q unlimited
> >> >> > ulimit -u unlimited
> >> >> > ulimit -v unlimited
> >> >> > ulimit -x unlimited
> >> >> > ulimit -s 240
> >> >> > ulimit -l unlimited
> >> >> > ulimit -a
> >> >> >
> >> >> >
> >> >> > Started FS with minimum modules but still 40 CPS seems to be the
> >> >> > limit.
> >> >> >
> >> >> >
> >> >> > So, is there any way to improve performance?
> >> >> >
> >> >> >
> >> >> > Tihomir.
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Wed Aug 26 01:44:42 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 26 Aug 2009 04:44:42 -0400
Subject: [Freeswitch-users] wav2mp3 conversion inside of spidermonkey
In-Reply-To: <a41cc07ca7c2a0f3c51ceb9b37ea6bc6.squirrel@correo.nodo50.org>
References: <4A92789C.5040900@gmx.net>
	<191c3a030908240951p570fca48v4476f6c87d1fcae2@mail.gmail.com>
	<4A93035B.5010907@gmx.net>
	<191c3a030908241441k46edc055tfae7d3e207d2cae0@mail.gmail.com>
	<6965e489a2bc4b6c38395623a4f2ffa8.squirrel@correo.nodo50.org>
	<5F8DA85A-5492-4828-B1C5-21ADBF5F42FA@freeswitch.org>
	<45b2e91d7aa77afcf6430a14dde4b886.squirrel@correo.nodo50.org>
	<191c3a030908251354l25206d0ch19836c8996f43e84@mail.gmail.com>
	<a41cc07ca7c2a0f3c51ceb9b37ea6bc6.squirrel@correo.nodo50.org>
Message-ID: <5896508D-3BA9-4B9A-A0DF-E8BE1EA69CB4@jerris.com>

Running out of stack space?  The stack space we run freeswitch in is  
fairly small.  Programs launched from the freeswitch process inherit  
this.

Mike

On Aug 26, 2009, at 4:28 AM, Alberto Escudero-Pascual (lists) wrote:

> I ran strace from freeswitch and from the command line. lame segfaults
> when run from system FS.
>
> The only obvious different i see is in the execve() /* XX vars */  
> apart
> from the final Segfault
>
> From
> execve("/usr/local/freeswitch/bin/lame",
> ["/usr/local/freeswitch/bin/lame", "/tmp/foo.wav", "/tmp/foo.mp3", "- 
> S"],
> [/* 16 vars */]) = 0
>
>
>> From FS
> execve("/usr/local/freeswitch/bin/lame",
> ["/usr/local/freeswitch/bin/lame", "/tmp/foo.wav",
> "/tmp/fooooooooooooooo.mp3", "-S"], [/* 14 vars */]) = 0
>
> I am attaching the full straces in case they are of any help. Not  
> sure if
> this deserves a jira
>
> /aep
> -- 
> Stopping junk mailers is good for the environment
>
>> maybe it's writing some err to stderr that is being suppressed  
>> somehow
>>
>> On Tue, Aug 25, 2009 at 3:46 PM, Alberto Escudero-Pascual (lists) <
>> aep.lists at it46.se> wrote:
>>
>>> Hi Brian,
>>>
>>>> From the CLI>
>>>
>>> freeswitch at open46> system /usr/local/freeswitch/bin/lame -V2
>>> /tmp/foo.wav
>>> /tmp/foo.mp3 -S
>>> 2009-08-25 22:41:51.556484 [NOTICE] mod_commands.c:3386 Executing
>>> command:
>>> /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav /tmp/foo.mp3 -S
>>> API CALL [system(/usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
>>> /tmp/foo.mp3 -S)] output:
>>> +OK
>>>
>>> open46:/tmp# ls
>>> foo.wav
>>>
>>>
>>> and running the command from the command line:
>>>
>>>
>>> open46:/tmp#  /usr/local/freeswitch/bin/lame -V2 /tmp/foo.wav
>>> /tmp/foo.mp3
>>> -Sopen46:/tmp# ls
>>> foo.mp3  foo.wav
>>>
>>>
>>> If I do the same with lame397
>>>
>>> freeswitch at open46> system /usr/local/freeswitch/bin/lame397 -V2
>>> /tmp/foo.wav /tmp/foo.mp3 -S
>>> 2009-08-25 22:44:32.743998 [NOTICE] mod_commands.c:3386 Executing
>>> command:
>>> /usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav /tmp/foo.mp3 -S
>>> API CALL [system(/usr/local/freeswitch/bin/lame397 -V2 /tmp/foo.wav
>>> /tmp/foo.mp3 -S)] output:
>>> +OK
>>>
>>> open46:/tmp# ls
>>> foo.mp3  foo.wav
>>>
>>>
>>> Highly paranormal! Sorry for hijacking the previous thread.
>>>
>>> /aep
>>>
>>> --
>>> Stopping junk mailers is good for the environment
>>>
>>>> Try running it at the CLI and see if you see any errors.  Also  
>>>> please
>>>> do not hijack threads.  The original thread "[Freeswitch-users]  
>>>> XML-
>>>> RPC on different ip than 0.0.0.0" which was hijacked by clicking
>>>> reply, changing the subject and clicking send.  Please in the  
>>>> future
>>>> do not do that as it clutters up the threading and could get your
>>>> query lost in the noise.
>>>>
>>>> Thanks,
>>>> Brian
>>>>
>>>> On Aug 25, 2009, at 1:54 AM, Alberto Escudero-Pascual (lists)  
>>>> wrote:
>>>>
>>>>> Here it comes the mystery. I am use lame 3.98.2 the mp3 file never
>>>>> appears, if I use version 3.97 (older version), it does!.
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN 
>> %3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL 
>> %3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip 
>> %3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf 
>> %2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
> <lame_strace.txt>_______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From raffaele.p.guidi at gmail.com  Wed Aug 26 03:12:31 2009
From: raffaele.p.guidi at gmail.com (Raffaele P. Guidi)
Date: Wed, 26 Aug 2009 12:12:31 +0200
Subject: [Freeswitch-users] caller id on origination does not work
In-Reply-To: <5070fcbd0908251951j5a7163b1se346abcdf0664c94@mail.gmail.com>
References: <5070fcbd0908251951j5a7163b1se346abcdf0664c94@mail.gmail.com>
Message-ID: <b0a8792f0908260312v6a84d66bh3d30e582b347e6ab@mail.gmail.com>

I had the same problem with the Sipek2 Softphone (based on the pjsip stack)
and I solved using the sip_contact_user variable like this:

api originate
{sip_contact_user=15103245678}sofia/gateway/junctionnetworks/1#{@number}
&park()


Hope this helps.
Ciao,
   Raffaele

On Wed, Aug 26, 2009 at 04:51, Shameem Shiek <gshfreesw at gmail.com> wrote:

> I am setting the caller id like this in my ESL script:
>
> @con.sendRecv("api originate
> {origination_caller_id_number=15103245678}sofia/gateway/junctionnetworks/1#{@number}
> &park()")
>
> And the caller id comes out as all zeroes.  The sip trace shows the "from"
> as shown in the sofia status command.  This is output of sofia gateway
> status and my gateway is under
> /usr/local/freeswitch/conf/sip_profiles/external/junctionnetworks.xml .
> Also, adding the gateway under /conf/directory/default did not work as the
> wiki suugests. What I need to do for the caller_id to come through?
>
>
> sofia status gateway junctionnetworks
>
> =================================================================================================
> Name        junctionnetworks
> Scheme      Digest
> Realm       jnctn.net
> Username    username
> Password    yes
> From        <sip:username at jnctn.net <sip%3Ausername at jnctn.net>
> ;transport=udp>
> Contact     <sip:gw+junctionnetworks at 24.87.80.12:5080;transport=udp>
> Exten       899xxxxxxx
> To          sip:username at sip.jnctn.net <sip%3Ausername at sip.jnctn.net>
> Proxy       sip:sip.jnctn.net
> Context     public
> Expires     3600
> Freq        3600
> Ping        0
> PingFreq    0
> State       REGED
> Status      UP
> CallsIN     0
> CallsOUT    1
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From odermann at googlemail.com  Wed Aug 26 05:59:33 2009
From: odermann at googlemail.com (Dennis)
Date: Wed, 26 Aug 2009 14:59:33 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
Message-ID: <5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>

> You have a NAT issue.

as i wrote, we are quite sure, that this is a nat problem. but we have
no idea how we could fix this. there must be a reason, why the hangup
always comes after 120 seconds.


>    <!--RTP port range -->
>    <param name="rtp-start-port" value="10000"/>
>    <param name="rtp-end-port" value="32767"/>

we tested that, but this does not change anything (nothing becomes
worse or better).


> param name="session-timeout" value="120"

this line is commented in our profile and we believe, that this has
nothing to do with our problem.


to tell a little bit more about our problem:

under http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios#Scenario_2
you can find the scenario, which is exactly like ours. we played with
all the mentioned settings, but it does not work.

to mention one thing: if we connect with xlite (softphone), we do not
have any problems. but we have snom-voip-phones and with them we have
the problems.

the big question is: wo or what might cause/trigger the hangup? is it
freeswitch or something else? we have a firewall (IPCop) - might there
be a setting, which needs to be set, to avoid theses problems?


kind regards
dennis



From gmaruzz at celliax.org  Wed Aug 26 06:08:38 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 26 Aug 2009 15:08:38 +0200
Subject: [Freeswitch-users] HOW-TO: being on "trunk" of FreePBX,
	starting from the ISO
Message-ID: <7b197bef0908260608l2c3f454cn8c2b8a12bc3d30ce@mail.gmail.com>

Instructions for being on "trunk" of FreePBX, starting from the ISO
(cut and paste to the ssh console after ISO install):

/etc/init.d/httpd stop
cd /var/www/html
mv freepbx freepbx-originale
svn co http://www.freepbx.org/v3/svn/trunk/ freepbx
chown -R apache.apache freepbx
ln -s freepbx  freepbx-v3
cd freepbx-v3/
ln -s freepbx  freepbx-v3
/etc/init.d/httpd start

then browse to:

http://192.168.0.12/freepbx-v3/index.php/installer

it will work! :-)


Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From odermann at googlemail.com  Wed Aug 26 06:18:34 2009
From: odermann at googlemail.com (Dennis)
Date: Wed, 26 Aug 2009 15:18:34 +0200
Subject: [Freeswitch-users] Question about presence
Message-ID: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>

hi,

we have set manage-presence = true to see, who is talking on the phone
and who is free. everyone here has his own snom-voip phone with 12
led-lights.

incoming calls are handled in groups. this means, if someone is
ringing, all voip-phones of one group are ringing and one can see all
led-lights flashing. if someone answers the phonecall, the led of this
person (p1) is on on everybodys phone, so everybody can see, that the
person is talking.
now, if another phonecall comes in, all led-lights are flashing -
including the led of the person who is talking. if another person (p2)
answers the second phonecall, the led of p2 is on, but the led of p1
is off, although p1 is still talking.

is there something we can do about this?


thanks & kind regards
dennis



From brian at freeswitch.org  Wed Aug 26 06:54:40 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 Aug 2009 08:54:40 -0500
Subject: [Freeswitch-users] Internal Profile sends RTP to external IP
	when using two network interfaces
In-Reply-To: <46b96649f81b0bfcb111928a55450211.squirrel@correo.nodo50.org>
References: <46b96649f81b0bfcb111928a55450211.squirrel@correo.nodo50.org>
Message-ID: <3694FC90-3FD1-4F36-8CA9-A3C18EBDEF34@freeswitch.org>

First off you can't bind one profile to two interfaces you have to  
launch two sofia profiles, one for each IP.  Secondly if you're doing  
things like this you'll have to refer to the in tree internal.xml.   
Third can you outline the network topology a little bit more?  Is nat  
involved?

/b

On Aug 26, 2009, at 3:04 AM, Alberto Escudero-Pascual (lists) wrote:

> Hi,
>
> I have a FS box with two physical network interfaces. The internal
> interface is hosting several internal phones. I have binded the  
> internal
> profile SIP/RTP/IP to the private interface. Phones registered  
> correctly
> but with User: 1000 at external.ip.address in sofia status profile  
> internal
>
> When I place a call between two internal phones, RTP traffic is send  
> to
> the external IP address of the FS box instead of the internal.
>
> The SIP/SDP messages send from FS carry the external IP instead of the
> internal.
>
> The result is that no RTP media arrives to any of the phones.
>
> /aep

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From kadantsev.d at gmail.com  Wed Aug 26 04:53:45 2009
From: kadantsev.d at gmail.com (Dmitry Kadantsev)
Date: Wed, 26 Aug 2009 13:53:45 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908260129xebbf460raffaa38c72c3281a@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com> 
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com> 
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com> 
	<7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com> 
	<65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com> 
	<7b197bef0908260125m56c3d681l26cb18a5e34aaa33@mail.gmail.com> 
	<65d96fc80908260129xebbf460raffaa38c72c3281a@mail.gmail.com>
Message-ID: <681a20520908260453q57877b1eg8f0949061cd129ee@mail.gmail.com>

Hi all,

is there same situation with FS for Windows? I mean 64bit is more preferable
than 32bit, isn't it?

Any performance test on Win 32/64 were done?

--
Best regards,
Dmitry Kadantsev


On Wed, Aug 26, 2009 at 10:29 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> intanto e il centos che si sta installando :)
>
> grazie.
>
> T.
>
>
> On Wed, Aug 26, 2009 at 10:25 AM, Giovanni Maruzzelli <gmaruzz at celliax.org
> > wrote:
>
>> netbook remix
>>
>>
>> joking! Server 64bit :-)
>>
>> -gm
>>
>>
>>
>> On Wed, Aug 26, 2009 at 10:08 AM, Tihomir Culjaga<tculjaga at gmail.com>
>> wrote:
>> > Hi Giovanny,
>> >
>> > regarding ubuntu, did you mean 8.04 server or desktop ?
>> >
>> >
>> > On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli <
>> gmaruzz at celliax.org>
>> > wrote:
>> >>
>> >> Definitely go for 64 bit OS.
>> >>
>> >> If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
>> >> used both for development and for heavy duty production.
>> >>
>> >> Also Ubuntu 8.04 is good.
>> >>
>> >> Other versions/distros are less used by the community.
>> >>
>> >> Adding RAM and CPUs helps to scale up.
>> >>
>> >> -gm
>> >>
>> >>
>> >>
>> >> Sincerely,
>> >>
>> >> Giovanni Maruzzelli
>> >> Cell : +39-347-2665618
>> >>
>> >>
>> >>
>> >>
>> >> On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com>
>> >> wrote:
>> >> > Hey Giovanni,
>> >> >
>> >> > thanks for the tip... indeed the db files were heavily used
>> regardless
>> >> > if i
>> >> > started freeswitch with nosql option (freeswitch -nosql)... FS was
>> not
>> >> > writing anything into that files ... instead it was just accessing
>> >> > it....
>> >> > This behaviour leads to a waste of 40% CPU time... waiting for other
>> >> > processes (mainly disk access) to finish!!!
>> >> >
>> >> > I moved freeswitch/db/ to a ramdisk and the performance got a boost
>> to
>> >> > 140
>> >> > CPS with a CPU load of 80%. I was keeping the machine for a while (20
>> -
>> >> > 30
>> >> > minutes) on that rate when i sow CPU suddenly went to 100% and FS
>> >> > becoming
>> >> > irresponsive :).
>> >> >
>> >> >
>> >> > What can be wrong?
>> >> > What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should
>> not
>> >> > cross?
>> >> > What fine tuning do we need in order to asure a long high load run?
>> >> >
>> >> >
>> >> >
>> >> > Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it
>> have
>> >> > sense
>> >> > to move my OS to 64 bit? ... will FS gain more preformance ?... I
>> mean
>> >> > will
>> >> > FS perofomr drastically better 20%+ ?
>> >> >
>> >> >
>> >> > Tihomir.
>> >> >
>> >> >
>> >> > On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
>> >> > <gmaruzz at celliax.org>
>> >> > wrote:
>> >> >>
>> >> >> Maybe your load comes from disk access?
>> >> >>
>> >> >> Try putting the sql and log directories on a ramdisk.
>> >> >>
>> >> >> OTH,
>> >> >>
>> >> >> -giovanni
>> >> >>
>> >> >> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<
>> tculjaga at gmail.com>
>> >> >> wrote:
>> >> >> > Hello,
>> >> >> >
>> >> >> > i'm trying to use freeswitch as a redirecting server meaning FS
>> has
>> >> >> > to
>> >> >> > receive an INVITE and according to some rules it will redirect
>> calls
>> >> >> > to
>> >> >> > other destinations.
>> >> >> >
>> >> >> >
>> >> >> > CALLING_USER                FREESWITCH
>> >> >> > SOMEWHERE
>> >> >> >
>> >> >> > INVITE ------------------------------->
>> >> >> >            <------------------------------ 100 Trying
>> >> >> >            <------------------------------ 302 Moved Temporary
>> >> >> > ACK    ------------------------------->
>> >> >> >
>> >> >> >
>> >> >> >
>> INVITE--------------------------------------------------------------------------------->
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > Well, wverything works well except i have perfromance issues ....
>> on
>> >> >> > my
>> >> >> > HW
>> >> >> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
>> >> >> > Temporary).
>> >> >> > When
>> >> >> > i increase the rate, FS starts delaying 302 response. Right at 50
>> CPS
>> >> >> > i
>> >> >> > see
>> >> >> > "calls" being build up in FS and the delay begining to grow.
>> >> >> >
>> >> >> > When i observe the machine, load average is almost nothing (load
>> >> >> > average:
>> >> >> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one
>> thread
>> >> >> > taking
>> >> >> > most load... all others are just sitting there with 1-5 % CPU
>> time.
>> >> >> > This looks to me as FS handles 302 messages in a single thread?!?!
>> >> >> >
>> >> >> >
>> >> >> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>> >> >> >
>> >> >> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41,
>> >> >> > 0.61,
>> >> >> > 0.60
>> >> >> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0
>> zombie
>> >> >> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,
>> >> >> > 0.2%si,
>> >> >> > 0.0%st
>> >> >> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k
>> >> >> > buffers
>> >> >> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k
>> >> >> > cached
>> >> >> >
>> >> >> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
>> >> >> > COMMAND
>> >> >> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
>> >> >> > freeswitch
>> >> >> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
>> >> >> > freeswitch
>> >> >> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
>> >> >> > freeswitch
>> >> >> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
>> >> >> > freeswitch
>> >> >> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
>> >> >> > freeswitch
>> >> >> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
>> >> >> > freeswitch
>> >> >> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
>> >> >> > freeswitch
>> >> >> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
>> >> >> > freeswitch
>> >> >> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
>> >> >> > freeswitch
>> >> >> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50
>> >> >> > freeswitch
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > cat /proc/cpuinfo
>> >> >> > processor       : 0
>> >> >> > vendor_id       : GenuineIntel
>> >> >> > cpu family      : 6
>> >> >> > model           : 15
>> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
>> >> >> > stepping        : 6
>> >> >> > cpu MHz         : 2333.560
>> >> >> > cache size      : 4096 KB
>> >> >> > physical id     : 0
>> >> >> > siblings        : 2
>> >> >> > core id         : 0
>> >> >> > cpu cores       : 2
>> >> >> > apicid          : 0
>> >> >> > initial apicid  : 0
>> >> >> > fdiv_bug        : no
>> >> >> > hlt_bug         : no
>> >> >> > f00f_bug        : no
>> >> >> > coma_bug        : no
>> >> >> > fpu             : yes
>> >> >> > fpu_exception   : yes
>> >> >> > cpuid level     : 10
>> >> >> > wp              : yes
>> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
>> >> >> > pge
>> >> >> > mca
>> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
>> >> >> > ssse3
>> >> >> > cx16
>> >> >> > xtpr dca lahf_lm
>> >> >> > bogomips        : 4670.78
>> >> >> > clflush size    : 64
>> >> >> > power management:
>> >> >> >
>> >> >> > processor       : 1
>> >> >> > vendor_id       : GenuineIntel
>> >> >> > cpu family      : 6
>> >> >> > model           : 15
>> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
>> >> >> > stepping        : 6
>> >> >> > cpu MHz         : 2333.560
>> >> >> > cache size      : 4096 KB
>> >> >> > physical id     : 0
>> >> >> > siblings        : 2
>> >> >> > core id         : 1
>> >> >> > cpu cores       : 2
>> >> >> > apicid          : 1
>> >> >> > initial apicid  : 1
>> >> >> > fdiv_bug        : no
>> >> >> > hlt_bug         : no
>> >> >> > f00f_bug        : no
>> >> >> > coma_bug        : no
>> >> >> > fpu             : yes
>> >> >> > fpu_exception   : yes
>> >> >> > cpuid level     : 10
>> >> >> > wp              : yes
>> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep mtrr
>> >> >> > pge
>> >> >> > mca
>> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
>> >> >> > ssse3
>> >> >> > cx16
>> >> >> > xtpr dca lahf_lm
>> >> >> > bogomips        : 4666.82
>> >> >> > clflush size    : 64
>> >> >> > power management:
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > uname -a
>> >> >> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC
>> 2009
>> >> >> > i686
>> >> >> > GNU/Linux
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > Of course, i've tuned the machine up
>> >> >> >
>> >> >> > ulimit -c unlimited
>> >> >> > ulimit -d unlimited
>> >> >> > ulimit -f unlimited
>> >> >> > ulimit -i unlimited
>> >> >> > ulimit -n 999999
>> >> >> > ulimit -q unlimited
>> >> >> > ulimit -u unlimited
>> >> >> > ulimit -v unlimited
>> >> >> > ulimit -x unlimited
>> >> >> > ulimit -s 240
>> >> >> > ulimit -l unlimited
>> >> >> > ulimit -a
>> >> >> >
>> >> >> >
>> >> >> > Started FS with minimum modules but still 40 CPS seems to be the
>> >> >> > limit.
>> >> >> >
>> >> >> >
>> >> >> > So, is there any way to improve performance?
>> >> >> >
>> >> >> >
>> >> >> > Tihomir.
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From se02005-mec at yahoo.com  Wed Aug 26 06:25:51 2009
From: se02005-mec at yahoo.com (Merle J. Ebbert)
Date: Wed, 26 Aug 2009 06:25:51 -0700
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
Message-ID: <4A9537DF.8070609@yahoo.com>


Hi,

I'm trying to avoid taking up a lot of peoples valuable time.

SIP & FS have  brought some ideas for some commercial products but I 
need to know where to start.

Having once written a proprietary DOS & helped with writing a RTOS, I 
consider myself capable
of learning.  I (we) just need to know where to start to come up to 
speed rapidly.

Is there a FreeSWITCH tutorial available?
Should someone new start with Asterisk and then possibly move to FS?

Thanks,
Merle




From harry at vangberg.name  Wed Aug 26 06:54:41 2009
From: harry at vangberg.name (Harry Vangberg)
Date: Wed, 26 Aug 2009 15:54:41 +0200
Subject: [Freeswitch-users] bind_meta_app: "already broadcasting..broadcast
	aborted" if bound via event socket
Message-ID: <74d41a3d0908260654x52bf9d94g124ceaa27f6df6bb@mail.gmail.com>

Hello.

After pestering the IRC channel for a few days with useless
information, I think I've finally made a breakthrough, but I have no
idea why it works this way.

Basically, what I want to achieve is: A calls in and is bridged to B
(1234). If B presses *1, he should be taken out, and A be bridged to C
(8888) instead. Pretty simple.

With the following dialplan it works very nicely:

    <extension name="ivr">
      <condition field="destination_number" expression="^(\d+)$">
        <action application="set" data="bypass_media=false"/>
        <action application="answer"/>
        <action application="bind_meta_app" data="1 b a
bridge::sofia/gateway/secretgw/8888"/>
        <action application="bridge" data="sofia/gateway/secretgw/1234" />
      </condition>
    </extension>

On the other hand, if I try to replicate the above via an outbound
event socket it fails with a: "already broadcasting...broadcast
aborted". In this case my dialplan XML looks like this:

    <extension name="ff-ivr">
      <condition field="destination_number" expression="^(.*)$">
        <action application="set" data="bypass_media=false"/>
        <action application="socket" data="127.0.0.1:8084 full"/>
      </condition>
    </extension>

And this is the event socket communication (in reality I use a Ruby
framework, do a bunch of database calls etc., but for the sake of
clarity I'm mimicking it with a netcat session):

--------------------------------
$ nc -v -l 127.0.0.1 8084
connect

Event-Name: CHANNEL_DATA
Core-UUID: ba78636a-9241-11de-ac94-b736da546252
FreeSWITCH-Hostname: ip-10-226-231-225
FreeSWITCH-IPv4: 10.226.231.225
FreeSWITCH-IPv6: %3A%3A1
Event-Date-Local: 2009-08-26%2013%3A43%3A20
Event-Date-GMT: Wed,%2026%20Aug%202009%2013%3A43%3A20%20GMT
Event-Date-Timestamp: 1251294200244673
Event-Calling-File: mod_event_socket.c
Event-Calling-Function: parse_command
Event-Calling-Line-Number: 1482
Channel-Username: hemmeligt
Channel-Dialplan: XML
Channel-Caller-ID-Name: hemmeligt
Channel-Caller-ID-Number: hemmeligt
Channel-Network-Addr: 129.142.224.250
Channel-Destination-Number: 77344541
Channel-Unique-ID: 681102bc-9246-11de-ac94-b736da546252
Channel-Source: mod_sofia
Channel-Context: public
Channel-Channel-Name: sofia/external/hemmeligt%40129.142.224.250
Channel-Profile-Index: 1
Channel-Profile-Created-Time: 1251294195778388
Channel-Channel-Created-Time: 1251294195778388
Channel-Channel-Answered-Time: 0
Channel-Channel-Progress-Time: 0
Channel-Channel-Progress-Media-Time: 0
Channel-Channel-Hangup-Time: 0
Channel-Channel-Transfer-Time: 0
Channel-Screen-Bit: false
Channel-Privacy-Hide-Name: true
Channel-Privacy-Hide-Number: true
Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/external/hemmeligt%40129.142.224.250
Unique-ID: 681102bc-9246-11de-ac94-b736da546252
Call-Direction: inbound
Presence-Call-Direction: inbound
Answer-State: ringing
Channel-Read-Codec-Name: PCMA
Channel-Read-Codec-Rate: 8000
Channel-Write-Codec-Name: PCMA
Channel-Write-Codec-Rate: 8000
Caller-Username: hemmeligt
Caller-Dialplan: XML
Caller-Caller-ID-Name: hemmeligt
Caller-Caller-ID-Number: hemmeligt
Caller-Network-Addr: 129.142.224.250
Caller-Destination-Number: 77344541
Caller-Unique-ID: 681102bc-9246-11de-ac94-b736da546252
Caller-Source: mod_sofia
Caller-Context: public
Caller-Channel-Name: sofia/external/hemmeligt%40129.142.224.250
Caller-Profile-Index: 1
Caller-Profile-Created-Time: 1251294195778388
Caller-Channel-Created-Time: 1251294195778388
Caller-Channel-Answered-Time: 0
Caller-Channel-Progress-Time: 0
Caller-Channel-Progress-Media-Time: 0
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: false
Caller-Privacy-Hide-Name: true
Caller-Privacy-Hide-Number: true
variable_sip_received_ip: 129.142.224.250
variable_sip_received_port: 5060
variable_sip_via_protocol: udp
variable_sip_from_user: hemmeligt
variable_sip_from_uri: hemmeligt%40129.142.224.250
variable_sip_from_host: 129.142.224.250
variable_sip_from_user_stripped: hemmeligt
variable_sip_from_tag: as19641342
variable_sofia_profile_name: external
variable_sip_Remote-Party-ID:
%22hemmeligt%22%20%3Csip%3Ahemmeligt%40129.142.224.250%3E%3Bprivacy%3Dfull%3Bscreen%3Dno
variable_sip_cid_type: rpid
variable_sip_req_user: 77344541
variable_sip_req_port: 5080
variable_sip_req_uri: 77344541%4079.125.42.248%3A5080
variable_sip_req_host: 79.125.42.248
variable_sip_to_user: 77344541
variable_sip_to_port: 5080
variable_sip_to_uri: 77344541%4079.125.42.248%3A5080
variable_sip_to_host: 79.125.42.248
variable_sip_contact_user: hemmeligt
variable_sip_contact_uri: hemmeligt%40129.142.224.250
variable_sip_contact_host: 129.142.224.250
variable_channel_name: sofia/external/hemmeligt%40129.142.224.250
variable_sip_call_id: 1aa24dab25cf03f07933f7136822ceff%40129.142.224.250
variable_sip_user_agent: Condor_gw1
variable_sip_via_host: 129.142.224.250
variable_sip_via_port: 5060
variable_sip_via_rport: 5060
variable_max_forwards: 70
variable_switch_r_sdp:
v%3D0%0D%0Ao%3Droot%2018982%2018982%20IN%20IP4%20129.142.224.250%0D%0As%3Dsession%0D%0Ac%3DIN%20IP4%20129.142.224.250%0D%0At%3D0%200%0D%0Am%3Daudio%2015430%20RTP/AVP%208%200%203%2097%20101%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A97%20iLBC/8000%0D%0Aa%3Dfmtp%3A97%20mode%3D30%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0D%0Aa%3Dptime%3A20%0D%0A
variable_remote_media_ip: 129.142.224.250
variable_remote_media_port: 15430
variable_read_codec: PCMA
variable_read_rate: 8000
variable_write_codec: PCMA
variable_write_rate: 8000
variable_endpoint_disposition: RECEIVED
variable_bypass_media: false
variable_current_application_data: 127.0.0.1%3A8084%20full
variable_current_application: socket
variable_socket_host: 127.0.0.1
Content-Type: command/reply
Reply-Text: %2BOK%0A
Socket-Mode: static
Control: full

sendmsg
call-command: execute
execute-app-name: answer

Content-Type: command/reply
Reply-Text: +OK

sendmsg
call-command: execute
execute-app-name: bind_meta_app
execute-app-arg: 1 b a bridge::sofia/gateway/secretgw/8888

Content-Type: command/reply
Reply-Text: +OK

sendmsg
call-command: execute
execute-app-name: bridge
execute-app-arg: sofia/gateway/secretgw/1234

Content-Type: command/reply
Reply-Text: +OK
--------------------------------

And on the FreeSWITCH side:

2009-08-26 13:43:15.778388 [NOTICE] switch_channel.c:602 New Channel
sofia/external/hemmeligt at 129.142.224.250
[681102bc-9246-11de-ac94-b736da546252]
2009-08-26 13:43:15.778388 [DEBUG] sofia.c:3302 Channel
sofia/external/hemmeligt at 129.142.224.250 entering state
[received][100]
2009-08-26 13:43:15.778388 [DEBUG] sofia.c:3309 Remote SDP:
v=0
o=root 18982 18982 IN IP4 129.142.224.250
s=session
c=IN IP4 129.142.224.250
t=0 0
m=audio 15430 RTP/AVP 8 0 3 97 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

2009-08-26 13:43:15.778388 [DEBUG] sofia_glue.c:3132 Audio Codec
Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2009-08-26 13:43:15.778388 [DEBUG] sofia_glue.c:2090 Set Codec
sofia/external/hemmeligt at 129.142.224.250 PCMA/8000 20 ms 160 samples
2009-08-26 13:43:15.778388 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
payload to 101
2009-08-26 13:43:15.778388 [DEBUG] sofia.c:3468
(sofia/external/hemmeligt at 129.142.224.250) State Change CS_NEW ->
CS_INIT
2009-08-26 13:43:15.778388 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:398
(sofia/external/hemmeligt at 129.142.224.250) Running State Change
CS_INIT
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:481
(sofia/external/hemmeligt at 129.142.224.250) State INIT
2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:83
sofia/external/hemmeligt at 129.142.224.250 SOFIA INIT
2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:111
(sofia/external/hemmeligt at 129.142.224.250) State Change CS_INIT ->
CS_ROUTING
2009-08-26 13:43:15.778388 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:481
(sofia/external/hemmeligt at 129.142.224.250) State INIT going to sleep
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:398
(sofia/external/hemmeligt at 129.142.224.250) Running State Change
CS_ROUTING
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:484
(sofia/external/hemmeligt at 129.142.224.250) State ROUTING
2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:130
sofia/external/hemmeligt at 129.142.224.250 SOFIA ROUTING
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:78
sofia/external/hemmeligt at 129.142.224.250 Standard ROUTING
2009-08-26 13:43:15.778388 [INFO] mod_dialplan_xml.c:315 Processing
hemmeligt->77344541 in context public
Dialplan: sofia/external/hemmeligt at 129.142.224.250 parsing
[public->ff-ivr] continue=false
Dialplan: sofia/external/hemmeligt at 129.142.224.250 Regex (PASS)
[ff-ivr] destination_number(77344541) =~ /^(.*)$/ break=on-false
Dialplan: sofia/external/hemmeligt at 129.142.224.250 Action
set(bypass_media=false)
Dialplan: sofia/external/hemmeligt at 129.142.224.250 Action
socket(127.0.0.1:8084 full)
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:114
(sofia/external/hemmeligt at 129.142.224.250) State Change CS_ROUTING ->
CS_EXECUTE
2009-08-26 13:43:15.778388 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:484
(sofia/external/hemmeligt at 129.142.224.250) State ROUTING going to
sleep
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:398
(sofia/external/hemmeligt at 129.142.224.250) Running State Change
CS_EXECUTE
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:491
(sofia/external/hemmeligt at 129.142.224.250) State EXECUTE
2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:173
sofia/external/hemmeligt at 129.142.224.250 SOFIA EXECUTE
2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:151
sofia/external/hemmeligt at 129.142.224.250 Standard EXECUTE
EXECUTE sofia/external/hemmeligt at 129.142.224.250 set(bypass_media=false)
2009-08-26 13:43:15.778388 [DEBUG] mod_dptools.c:748
sofia/external/hemmeligt at 129.142.224.250 SET [bypass_media]=[false]
EXECUTE sofia/external/hemmeligt at 129.142.224.250 socket(127.0.0.1:8084 full)
2009-08-26 13:43:21.701447 [DEBUG] switch_ivr.c:540
sofia/external/hemmeligt at 129.142.224.250 Command Execute answer()
EXECUTE sofia/external/hemmeligt at 129.142.224.250 answer()
2009-08-26 13:43:21.701447 [DEBUG] mod_dptools.c:649
sofia/external/hemmeligt at 129.142.224.250 receive message [ANSWER]
2009-08-26 13:43:21.701447 [DEBUG] sofia_glue.c:2324 AUDIO RTP
[sofia/external/hemmeligt at 129.142.224.250] 10.226.231.225 port 19016
-> 129.142.224.250 port 15430 codec: 8 ms: 20
2009-08-26 13:43:21.701447 [DEBUG] switch_rtp.c:1138 Starting timer
[soft] 160 bytes per 20ms
2009-08-26 13:43:21.702447 [DEBUG] mod_sofia.c:536 Local SDP
sofia/external/hemmeligt at 129.142.224.250:
v=0
o=FreeSWITCH 1251275185 1251275186 IN IP4 79.125.42.248
s=FreeSWITCH
c=IN IP4 79.125.42.248
t=0 0
m=audio 19016 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-08-26 13:43:21.702447 [DEBUG] switch_core_session.c:630 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:21.702447 [NOTICE] mod_dptools.c:649 Channel
[sofia/external/hemmeligt at 129.142.224.250] has been answered
2009-08-26 13:43:21.702447 [DEBUG] switch_channel.c:182
sofia/external/hemmeligt at 129.142.224.250 receive message [AUDIO_SYNC]
2009-08-26 13:43:21.702447 [DEBUG] sofia.c:3302 Channel
sofia/external/hemmeligt at 129.142.224.250 entering state
[completed][200]
2009-08-26 13:43:21.782616 [DEBUG] sofia.c:3302 Channel
sofia/external/hemmeligt at 129.142.224.250 entering state [ready][200]
2009-08-26 13:43:26.388737 [DEBUG] switch_ivr.c:540
sofia/external/hemmeligt at 129.142.224.250 Command Execute
bind_meta_app(1 b a bridge::sofia/gateway/secretgw/8888)
EXECUTE sofia/external/hemmeligt at 129.142.224.250 bind_meta_app(1 b a
bridge::sofia/gateway/secretgw/8888)
2009-08-26 13:43:26.388737 [INFO] switch_ivr_async.c:1795 Bound B-Leg:
1 bridge::sofia/gateway/secretgw/8888
2009-08-26 13:43:29.684236 [DEBUG] switch_ivr.c:540
sofia/external/hemmeligt at 129.142.224.250 Command Execute
bridge(sofia/gateway/secretgw/1234)
EXECUTE sofia/external/hemmeligt at 129.142.224.250
bridge(sofia/gateway/secretgw/1234)
2009-08-26 13:43:29.684236 [NOTICE] switch_channel.c:602 New Channel
sofia/external/1234 [705ad5d8-9246-11de-ac94-b736da546252]
2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:2809
(sofia/external/1234) State Change CS_NEW -> CS_INIT
2009-08-26 13:43:29.684236 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:398
(sofia/external/1234) Running State Change CS_INIT
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:481
(sofia/external/1234) State INIT
2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:83 sofia/external/1234 SOFIA INIT
2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:111
(sofia/external/1234) State Change CS_INIT -> CS_ROUTING
2009-08-26 13:43:29.684236 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:29.684236 [DEBUG] sofia.c:3302 Channel
sofia/external/1234 entering state [calling][0]
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:481
(sofia/external/1234) State INIT going to sleep
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:398
(sofia/external/1234) Running State Change CS_ROUTING
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:484
(sofia/external/1234) State ROUTING
2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:130 sofia/external/1234
SOFIA ROUTING
2009-08-26 13:43:29.684236 [DEBUG] switch_ivr_originate.c:63
(sofia/external/1234) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-08-26 13:43:29.684236 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:484
(sofia/external/1234) State ROUTING going to sleep
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:398
(sofia/external/1234) Running State Change CS_CONSUME_MEDIA
2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:503
(sofia/external/1234) State CONSUME_MEDIA
2009-08-26 13:43:29.731221 [DEBUG] sofia.c:3302 Channel
sofia/external/1234 entering state [calling][0]
2009-08-26 13:43:30.660221 [DEBUG] sofia.c:3302 Channel
sofia/external/1234 entering state [proceeding][183]
2009-08-26 13:43:30.660221 [DEBUG] sofia.c:3309 Remote SDP:
v=0
o=root 18982 18982 IN IP4 129.142.224.250
s=session
c=IN IP4 129.142.224.250
t=0 0
m=audio 12480 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:3132 Audio Codec
Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:2090 Set Codec
sofia/external/1234 PCMA/8000 20 ms 160 samples
2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
payload to 101
2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:2324 AUDIO RTP
[sofia/external/1234] 10.226.231.225 port 25540 -> 129.142.224.250
port 12480 codec: 8 ms: 20
2009-08-26 13:43:30.660221 [DEBUG] switch_rtp.c:1138 Starting timer
[soft] 160 bytes per 20ms
2009-08-26 13:43:30.661084 [NOTICE] sofia_glue.c:2759 Pre-Answer
sofia/external/1234!
2009-08-26 13:43:30.661084 [DEBUG] switch_channel.c:1778 Send signal
sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_originate.c:2061
Originate Resulted in Success: [sofia/external/1234]
2009-08-26 13:43:30.665170 [DEBUG] switch_channel.c:182
sofia/external/1234 receive message [AUDIO_SYNC]
2009-08-26 13:43:30.665170 [DEBUG] switch_channel.c:182
sofia/external/hemmeligt at 129.142.224.250 receive message [AUDIO_SYNC]
2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_bridge.c:889
sofia/external/1234 receive message [BRIDGE]
2009-08-26 13:43:30.665170 [DEBUG] switch_core_session.c:630 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_bridge.c:896
sofia/external/hemmeligt at 129.142.224.250 receive message [BRIDGE]
2009-08-26 13:43:30.665170 [DEBUG] switch_core_session.c:630 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_bridge.c:940
(sofia/external/1234) State Change CS_CONSUME_MEDIA ->
CS_EXCHANGE_MEDIA
2009-08-26 13:43:30.665170 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:30.705074 [DEBUG] switch_core_state_machine.c:503
(sofia/external/1234) State CONSUME_MEDIA going to sleep
2009-08-26 13:43:30.705074 [DEBUG] switch_core_state_machine.c:398
(sofia/external/1234) Running State Change CS_EXCHANGE_MEDIA
2009-08-26 13:43:30.705074 [DEBUG] switch_core_state_machine.c:494
(sofia/external/1234) State EXCHANGE_MEDIA
2009-08-26 13:43:30.705074 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK
2009-08-26 13:43:36.642173 [DEBUG] sofia.c:3302 Channel
sofia/external/1234 entering state [ready][200]
2009-08-26 13:43:36.642173 [DEBUG] switch_channel.c:1891 Send signal
sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:36.642173 [NOTICE] sofia.c:3752 Channel
[sofia/external/1234] has been answered
2009-08-26 13:43:36.642173 [DEBUG] switch_channel.c:182
sofia/external/1234 receive message [AUDIO_SYNC]
2009-08-26 13:43:40.309616 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF *:2000
2009-08-26 13:43:40.970513 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF 1:2000
2009-08-26 13:43:40.970513 [DEBUG] switch_ivr_async.c:1711
sofia/external/hemmeligt at 129.142.224.250 Processing meta digit '1'
[bridge::sofia/gateway/secretgw/8888]
2009-08-26 13:43:40.970513 [WARNING] switch_ivr_async.c:2310 Channel
[sofia/external/hemmeligt at 129.142.224.250][bridge::sofia/gateway/secretgw/8888]
already broadcasting...broadcast aborted
2009-08-26 13:43:46.272706 [NOTICE] sofia.c:327 Hangup
sofia/external/1234 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
2009-08-26 13:43:46.272706 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/1234 [KILL]
2009-08-26 13:43:46.272706 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:371
sofia/external/1234 ending bridge by request from write function
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:426
sofia/external/hemmeligt at 129.142.224.250 receive message [UNBRIDGE]
2009-08-26 13:43:46.284704 [DEBUG] switch_core_session.c:630 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:452 BRIDGE
THREAD DONE [sofia/external/hemmeligt at 129.142.224.250]
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:454 Send signal
sofia/external/1234 [BREAK]
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:377
sofia/external/1234 ending bridge by request from read function
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:452 BRIDGE
THREAD DONE [sofia/external/1234]
2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:454 Send signal
sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:494
(sofia/external/1234) State EXCHANGE_MEDIA going to sleep
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:398
(sofia/external/1234) Running State Change CS_HANGUP
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:434
(sofia/external/1234) State HANGUP
2009-08-26 13:43:46.284704 [DEBUG] mod_sofia.c:338 Channel
sofia/external/1234 hanging up, cause: NORMAL_CLEARING
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:46
sofia/external/1234 Standard HANGUP, cause: NORMAL_CLEARING
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:434
(sofia/external/1234) State HANGUP going to sleep
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:476
(sofia/external/1234) State Change CS_HANGUP -> CS_REPORTING
2009-08-26 13:43:46.284704 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/1234 [BREAK]
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:398
(sofia/external/1234) Running State Change CS_REPORTING
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:612
(sofia/external/1234) State REPORTING
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:53
sofia/external/1234 Standard REPORTING, cause: NORMAL_CLEARING
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:612
(sofia/external/1234) State REPORTING going to sleep
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:411
(sofia/external/1234) State Change CS_REPORTING -> CS_DESTROY
2009-08-26 13:43:46.284704 [DEBUG] switch_core_session.c:1068 Session
5 (sofia/external/1234) Locked, Waiting on external entities
2009-08-26 13:43:46.284704 [NOTICE] switch_core_session.c:1086 Session
5 (sofia/external/1234) Ended
2009-08-26 13:43:46.284704 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/external/1234 [CS_DESTROY]
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:564
(sofia/external/1234) State DESTROY
2009-08-26 13:43:46.284704 [DEBUG] mod_sofia.c:255 sofia/external/1234
SOFIA DESTROY
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:60
sofia/external/1234 Standard DESTROY
2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:564
(sofia/external/1234) State DESTROY going to sleep
2009-08-26 13:44:46.746524 [NOTICE] sofia.c:327 Hangup
sofia/external/hemmeligt at 129.142.224.250 [CS_EXECUTE]
[NORMAL_CLEARING]
2009-08-26 13:44:46.746524 [DEBUG] switch_channel.c:1683 Send signal
sofia/external/hemmeligt at 129.142.224.250 [KILL]
2009-08-26 13:44:46.746524 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:491
(sofia/external/hemmeligt at 129.142.224.250) State EXECUTE going to
sleep
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:398
(sofia/external/hemmeligt at 129.142.224.250) Running State Change
CS_HANGUP
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:434
(sofia/external/hemmeligt at 129.142.224.250) State HANGUP
2009-08-26 13:44:46.747796 [DEBUG] mod_sofia.c:338 Channel
sofia/external/hemmeligt at 129.142.224.250 hanging up, cause:
NORMAL_CLEARING
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:46
sofia/external/hemmeligt at 129.142.224.250 Standard HANGUP, cause:
NORMAL_CLEARING
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:434
(sofia/external/hemmeligt at 129.142.224.250) State HANGUP going to sleep
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:476
(sofia/external/hemmeligt at 129.142.224.250) State Change CS_HANGUP ->
CS_REPORTING
2009-08-26 13:44:46.747796 [DEBUG] switch_core_session.c:932 Send
signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:398
(sofia/external/hemmeligt at 129.142.224.250) Running State Change
CS_REPORTING
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:612
(sofia/external/hemmeligt at 129.142.224.250) State REPORTING
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:53
sofia/external/hemmeligt at 129.142.224.250 Standard REPORTING, cause:
NORMAL_CLEARING
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:612
(sofia/external/hemmeligt at 129.142.224.250) State REPORTING going to
sleep
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:411
(sofia/external/hemmeligt at 129.142.224.250) State Change CS_REPORTING
-> CS_DESTROY
2009-08-26 13:44:46.747796 [DEBUG] switch_core_session.c:1068 Session
4 (sofia/external/hemmeligt at 129.142.224.250) Locked, Waiting on
external entities
2009-08-26 13:44:46.747796 [NOTICE] switch_core_session.c:1086 Session
4 (sofia/external/hemmeligt at 129.142.224.250) Ended
2009-08-26 13:44:46.747796 [NOTICE] switch_core_session.c:1088 Close
Channel sofia/external/hemmeligt at 129.142.224.250 [CS_DESTROY]
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:564
(sofia/external/hemmeligt at 129.142.224.250) State DESTROY
2009-08-26 13:44:46.747796 [DEBUG] mod_sofia.c:255
sofia/external/hemmeligt at 129.142.224.250 SOFIA DESTROY
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:60
sofia/external/hemmeligt at 129.142.224.250 Standard DESTROY
2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:564
(sofia/external/hemmeligt at 129.142.224.250) State DESTROY going to
sleep

Of which I mainly find these interesting:

EXECUTE sofia/external/hemmeligt at 129.142.224.250 bind_meta_app(1 b a
bridge::sofia/gateway/secretgw/8888)
2009-08-26 13:43:26.388737 [INFO] switch_ivr_async.c:1795 Bound B-Leg:
1 bridge::sofia/gateway/secretgw/8888
[...]
2009-08-26 13:43:40.309616 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF *:2000
2009-08-26 13:43:40.970513 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF 1:2000
2009-08-26 13:43:40.970513 [DEBUG] switch_ivr_async.c:1711
sofia/external/hemmeligt at 129.142.224.250 Processing meta digit '1'
[bridge::sofia/gateway/secretgw/8888]
2009-08-26 13:43:40.970513 [WARNING] switch_ivr_async.c:2310 Channel
[sofia/external/hemmeligt at 129.142.224.250][bridge::sofia/gateway/secretgw/8888]
already broadcasting...broadcast aborted

Can anybody explain what is going on here? It seems like the
bind_meta_app is bound perfectly via event socket, but when it comes
down, it doesn't work.

Best regards,
Harry



From MPeace at edcogroupinc.com  Wed Aug 26 06:58:02 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Wed, 26 Aug 2009 08:58:02 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <5DF610F9-6EB5-43CC-BB4F-3513C78F608F@freeswitch.org>
References: <5DF610F9-6EB5-43CC-BB4F-3513C78F608F@freeswitch.org>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A622D22BE@ESNEXS2.edcogroup.net>

I found and edited openzap.cong file, the openzap.conf.xml file is in the correct dir but the only example I can find is for a Euro E1 setup not a US T1. I

I have turned up data T1 circuits for WAN links before but not telco stuff. The provider is Qwest, I?ll have to ask about the NI2 , DMS question.

Mike

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins
Sent: Wednesday, August 26, 2009 3:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


I have looked at that but I am confused on which files need to be edited. Since I have already installed in Wanpipe mode with the Sangoma card I skipped straight to the Wanpipe section. It mentions setting the [span wanpie PRI_1] etc in the openzap.conf then further down it mentions editing the autoload_configs/openzap.conf.xml with a completely different format of config.

Are these the same file? If not where does the first one mentioned reside?

These are two different files. The openzap.conf file is in the freeswitch/conf directory. It is for configuring OZ to talk to the Sangoma card. The openzap.conf.xml file is for configuring the actual openzap module in FreeSWITCH.

The specs you listed on the T1 you have indicate that it's a PRI. However, you need to know which protocol, like NI2 or DMS. Have you ever done a PRI turn up? Who is the carrier?

-MC
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From gmaruzz at celliax.org  Wed Aug 26 07:01:44 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Wed, 26 Aug 2009 16:01:44 +0200
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
In-Reply-To: <4A9537DF.8070609@yahoo.com>
References: <4A9537DF.8070609@yahoo.com>
Message-ID: <7b197bef0908260701n60a671dct26394947e4db65c4@mail.gmail.com>

You will find all the informations, and then some, here:
http://wiki.freeswitch.org

Then, after reading and testing and experimenting, come to the IRC
channel for direct help with esoteric (or not so esoteric) problems.

You'll find a nice and friendly community.

-gm




Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618




On Wed, Aug 26, 2009 at 3:25 PM, Merle J. Ebbert<se02005-mec at yahoo.com> wrote:
>
> Hi,
>
> I'm trying to avoid taking up a lot of peoples valuable time.
>
> SIP & FS have ?brought some ideas for some commercial products but I
> need to know where to start.
>
> Having once written a proprietary DOS & helped with writing a RTOS, I
> consider myself capable
> of learning. ?I (we) just need to know where to start to come up to
> speed rapidly.
>
> Is there a FreeSWITCH tutorial available?
> Should someone new start with Asterisk and then possibly move to FS?
>
> Thanks,
> Merle
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From bjbrashier at gmail.com  Wed Aug 26 07:05:05 2009
From: bjbrashier at gmail.com (Bradley Brashier)
Date: Wed, 26 Aug 2009 08:05:05 -0600
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
In-Reply-To: <4A9537DF.8070609@yahoo.com>
References: <4A9537DF.8070609@yahoo.com>
Message-ID: <7bcfdd290908260705t644c42f0g52fd6c10b9bbaed1@mail.gmail.com>

Speaking as someone who went through this recently myself, my
suggestion is to learn by doing. Choose some simple-sounding task you
want to accomplish in FS, and try to carry it out. Use the wiki as a
reference and the mailing list if you can't figure out what you need
from that.

I don't suggest you start with Asterisk or anything else, as I'd guess
that would just confuse the issue. I came from a BIOS background, went
directly into Freeswitch, and am so far glad I did.

BB

On Wed, Aug 26, 2009 at 7:25 AM, Merle J. Ebbert<se02005-mec at yahoo.com> wrote:
>
> Hi,
>
> I'm trying to avoid taking up a lot of peoples valuable time.
>
> SIP & FS have ?brought some ideas for some commercial products but I
> need to know where to start.
>
> Having once written a proprietary DOS & helped with writing a RTOS, I
> consider myself capable
> of learning. ?I (we) just need to know where to start to come up to
> speed rapidly.
>
> Is there a FreeSWITCH tutorial available?
> Should someone new start with Asterisk and then possibly move to FS?
>
> Thanks,
> Merle
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From aep.lists at it46.se  Wed Aug 26 07:18:03 2009
From: aep.lists at it46.se (Alberto Escudero-Pascual (lists))
Date: Wed, 26 Aug 2009 16:18:03 +0200
Subject: [Freeswitch-users] Internal Profile sends RTP to external IP
 when using two network interfaces
In-Reply-To: <3694FC90-3FD1-4F36-8CA9-A3C18EBDEF34@freeswitch.org>
References: <46b96649f81b0bfcb111928a55450211.squirrel@correo.nodo50.org>
	<3694FC90-3FD1-4F36-8CA9-A3C18EBDEF34@freeswitch.org>
Message-ID: <7682ba3b2cbb1cf578dcc2226e7d4787.squirrel@correo.nodo50.org>



ASCII art follows

voip-phones..... 10.0.46.1 [ FS ] 216.82.231.69 ----> Internet


I have two profiles binded to two different IPs

internal is binded to 10.0.46.1 port 5060 /24 network
external is binded to 216.82.231.69 port 5080

My IP phones are in the 10.0.46.X range and they register @ 10.0.46.1 IP
address.

When I place a call between the two phones in the 10.0.46.0/24 network
the phones register but audio RTP is sent to 216.82.231.69 instead of
10.0.46.1 (FS)

In this scenario i have FS without any NAT configuration.

-- 
The way that i solved was to run both profiles (internal and external) in
the public IP (local_ipv4) and force the phones to register with
216.82.231.69 instead of 10.0.46.1

Thanks looong time!


-- 
Stopping junk mailers is good for the environment

> First off you can't bind one profile to two interfaces you have to
> launch two sofia profiles, one for each IP.  Secondly if you're doing
> things like this you'll have to refer to the in tree internal.xml.
> Third can you outline the network topology a little bit more?  Is nat
> involved?
>
> /b
>
> On Aug 26, 2009, at 3:04 AM, Alberto Escudero-Pascual (lists) wrote:
>
>> Hi,
>>
>> I have a FS box with two physical network interfaces. The internal
>> interface is hosting several internal phones. I have binded the
>> internal
>> profile SIP/RTP/IP to the private interface. Phones registered
>> correctly
>> but with User: 1000 at external.ip.address in sofia status profile
>> internal
>>
>> When I place a call between two internal phones, RTP traffic is send
>> to
>> the external IP address of the FS box instead of the internal.
>>
>> The SIP/SDP messages send from FS carry the external IP instead of the
>> internal.
>>
>> The result is that no RTP media arrives to any of the phones.
>>
>> /aep
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>





From brian at freeswitch.org  Wed Aug 26 07:25:02 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 Aug 2009 09:25:02 -0500
Subject: [Freeswitch-users] Internal Profile sends RTP to external IP
	when using two network interfaces
In-Reply-To: <7682ba3b2cbb1cf578dcc2226e7d4787.squirrel@correo.nodo50.org>
References: <46b96649f81b0bfcb111928a55450211.squirrel@correo.nodo50.org>
	<3694FC90-3FD1-4F36-8CA9-A3C18EBDEF34@freeswitch.org>
	<7682ba3b2cbb1cf578dcc2226e7d4787.squirrel@correo.nodo50.org>
Message-ID: <56E6F9F8-82FB-4A4D-9B7A-44D9BE2BDE43@freeswitch.org>

Well actually you do have issues you're going to have to specify the  
rtp-ip and sip-ip on BOTH profiles yourself because you're config is  
currently putting your external IP into the internal's settings.   
Please correct that and I'm sure it'll work then.

/b

On Aug 26, 2009, at 9:18 AM, Alberto Escudero-Pascual (lists) wrote:

>
>
> ASCII art follows
>
> voip-phones..... 10.0.46.1 [ FS ] 216.82.231.69 ----> Internet
>
>
> I have two profiles binded to two different IPs
>
> internal is binded to 10.0.46.1 port 5060 /24 network
> external is binded to 216.82.231.69 port 5080
>
> My IP phones are in the 10.0.46.X range and they register @  
> 10.0.46.1 IP
> address.
>
> When I place a call between the two phones in the 10.0.46.0/24 network
> the phones register but audio RTP is sent to 216.82.231.69 instead of
> 10.0.46.1 (FS)
>
> In this scenario i have FS without any NAT configuration.
>
> -- 
> The way that i solved was to run both profiles (internal and  
> external) in
> the public IP (local_ipv4) and force the phones to register with
> 216.82.231.69 instead of 10.0.46.1
>
> Thanks looong time!
>




From MPeace at edcogroupinc.com  Wed Aug 26 07:26:45 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Wed, 26 Aug 2009 09:26:45 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A622D22BE@ESNEXS2.edcogroup.net>
References: <5DF610F9-6EB5-43CC-BB4F-3513C78F608F@freeswitch.org>
	<69D652F49932C34199968DFB8AEAABA33A622D22BE@ESNEXS2.edcogroup.net>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A622D2307@ESNEXS2.edcogroup.net>

Ok it is NI2 according to Qwest,

Mike Peace
Network Analyst
EDCO, The Document People
Direct 417-447-3367

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike Peace
Sent: Wednesday, August 26, 2009 8:58 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

I found and edited openzap.cong file, the openzap.conf.xml file is in the correct dir but the only example I can find is for a Euro E1 setup not a US T1. I

I have turned up data T1 circuits for WAN links before but not telco stuff. The provider is Qwest, I?ll have to ask about the NI2 , DMS question.

Mike

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins
Sent: Wednesday, August 26, 2009 3:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


I have looked at that but I am confused on which files need to be edited. Since I have already installed in Wanpipe mode with the Sangoma card I skipped straight to the Wanpipe section. It mentions setting the [span wanpie PRI_1] etc in the openzap.conf then further down it mentions editing the autoload_configs/openzap.conf.xml with a completely different format of config.

Are these the same file? If not where does the first one mentioned reside?

These are two different files. The openzap.conf file is in the freeswitch/conf directory. It is for configuring OZ to talk to the Sangoma card. The openzap.conf.xml file is for configuring the actual openzap module in FreeSWITCH.

The specs you listed on the T1 you have indicate that it's a PRI. However, you need to know which protocol, like NI2 or DMS. Have you ever done a PRI turn up? Who is the carrier?

-MC
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From se02005-mec at yahoo.com  Wed Aug 26 07:45:11 2009
From: se02005-mec at yahoo.com (Merle J. Ebbert)
Date: Wed, 26 Aug 2009 07:45:11 -0700
Subject: [Freeswitch-users] I would like to send an IM. From a Windows
 PC. To a BlackBerry @vzw.blackberry.com
Message-ID: <4A954A77.7050508@yahoo.com>

An HTML attachment was scrubbed...
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From anatoliy at kounitskiy.com  Wed Aug 26 07:08:55 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Wed, 26 Aug 2009 17:08:55 +0300
Subject: [Freeswitch-users] Questions about att_xfer (freeswitch version 1.0
	trunk 14633M)
Message-ID: <4A9541F7.4070901@kounitskiy.com>

Hello everybody!
I have few questions about the att_xfer application. First, what i want 
to accomplish is: user A calls user B, after that user B makes attended 
transfer to user C.
In the dialplan i have:

<context name="vpbx">
  <extension name="local_number">
...
      <action application="bind_meta_app" data="1 b s 
execute_extension::dx XML features"/>
      <action application="bind_meta_app" data="2 b s 
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
      <action application="bind_meta_app" data="3 b s 
execute_extension::cf XML features"/>
      <action application="bind_meta_app" data="4 b s 
execute_extension::attented_xfer XML features"/>
 ....
    </condition>
  </extension>

So when user B answers the call, he sends *4 and the extensions for the 
attended transfer is started - the usual - plays message and read the 
input dtmf:

features.xml
...
    <extension name="attented_xfer">
      <condition field="${toll_allow}" expression="local"/>
      <condition field="destination_number" expression="^attented_xfer$">
        <action application="info"/>
        <action application="read" data="3 4 ivr/ivr-enter_ext.wav 
attxfer_callthis 30000 #"/>
        <action application="set" data="call_timeout=15"/>
        <action application="att_xfer" 
data="user/${attxfer_callthis}@${domain_name}"/>
      </condition>
    </extension>
...

To this problems everything is perfect. But here comes the questions, so 
if you can give some tips would be great.

1) when user B enters the extension number of C - the C's phone starts 
ringing in the tcpdump i can see that the phone is sending 180 ringing, 
BUT user B does not hear the ringing.
2) as mentioned in the 
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
quote: "If the other leg is a voicemail or doesn't answered you can 
hangup that leg by pressing dtmf # (fixed in r14438) "
It doesn't work. The option 0 is working even before C answering the 
phone - after he answers it's a threeway conference :) - i like this 
feature.

I'm using FreeSWITCH Version 1.0.trunk (14633M)

Also I tried to set call timeout to see if I can go back the user A, who 
is listening to MOH - no luck here.

Probably I'm missing something. Tried to look in the source of att_xfer 
to understand why the feature i want is not working - but it seems my 
C/C++ skills are not so good, as i want :( .

Thank you in advance,
Anatoliy Kounitskiy



From anatoliy at kounitskiy.com  Wed Aug 26 07:51:49 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Wed, 26 Aug 2009 17:51:49 +0300
Subject: [Freeswitch-users] Questions about att_xfer
Message-ID: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>

Hello everybody!
I have few questions about the att_xfer application. First, what i want
to accomplish is: user A calls user B, after that user B makes attended
transfer to user C.
In the dialplan i have:

<context name="vpbx">
  <extension name="local_number">
...
      <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
      <action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
      <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
      <action application="bind_meta_app" data="4 b s
execute_extension::attented_xfer XML features"/>
 ....
    </condition>
  </extension>

So when user B answers the call, he sends *4 and the extensions for the
attended transfer is started - the usual - plays message and read the
input dtmf:

features.xml
...
    <extension name="attented_xfer">
      <condition field="${toll_allow}" expression="local"/>
      <condition field="destination_number" expression="^attented_xfer$">
        <action application="info"/>
        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 30000 #"/>
        <action application="set" data="call_timeout=15"/>
        <action application="att_xfer"
data="user/${attxfer_callthis}@${domain_name}"/>
      </condition>
    </extension>
...

To this problems everything is perfect. But here comes the questions, so
if you can give some tips would be great.

1) when user B enters the extension number of C - the C's phone starts
ringing in the tcpdump i can see that the phone is sending 180 ringing,
BUT user B does not hear the ringing.
2) as mentioned in the
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
quote: "If the other leg is a voicemail or doesn't answered you can
hangup that leg by pressing dtmf # (fixed in r14438) "
It doesn't work. The option 0 is working even before C answering the
phone - after he answers it's a threeway conference :) - i like this
feature.

I'm using FreeSWITCH Version 1.0.trunk (14633M)

Also I tried to set call timeout to see if I can go back the user A, who
is listening to MOH - no luck here.

Probably I'm missing something. Tried to look in the source of att_xfer
to understand why the feature i want is not working - but it seems my
C/C++ skills are not so good, as i want :( .

Thank you in advance,
Anatoliy Kounitskiy



From delianspam at gmail.com  Wed Aug 26 07:59:08 2009
From: delianspam at gmail.com (delianSPAM)
Date: Wed, 26 Aug 2009 17:59:08 +0300
Subject: [Freeswitch-users] Timers/DTMFs During a Call
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>

Hello Everybody!

 

1.       Scenario.

I am writing an IVR in Python that gets a destination from the calling party
(party A) and then connects to the destination (party B).

When the call is CONNECTED, I want to:

-          Receive DTMFs

-          Have a timer that can call a certain function in my script. The
script will have to play a message to party A.

-          Have a timer that can call a certain function in my script. The
script will have to drop the call.

Please notice that I want to do the things after the two parties are
connected, and not after I send the Invite to party B.

 

2.       Problem.

I will be happy to receive help on:

-          Which methods should I look for to implement this.

 

3.       Details

Here is how I connect the call currently:

session.execute("bridge","sofia/internal/" + destination_number +
"@domain.com")

 

I have tried to create a timer callback function "my_method()" using:

ivr_timer =threading.Timer(30,my_method)

This never called the function "my_method()".

 

Maybe I am wrong in using threading.Timer and the "bridge" application?
Maybe I need to create a new thread and a new timer using the API of
freeswitch, plus to use the "session.setInputCallback", plus use a
conference rather than a bridge? Can you please provide any suggestions or
examples? 

 

Thank you!

Best Regards, Delian Tashev

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From mike at jerris.com  Wed Aug 26 08:18:38 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 26 Aug 2009 11:18:38 -0400
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
Message-ID: <0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>


On Aug 26, 2009, at 9:18 AM, Dennis wrote:

> hi,
>
> we have set manage-presence = true to see, who is talking on the phone
> and who is free. everyone here has his own snom-voip phone with 12
> led-lights.
>
> incoming calls are handled in groups. this means, if someone is
> ringing, all voip-phones of one group are ringing and one can see all
> led-lights flashing. if someone answers the phonecall, the led of this
> person (p1) is on on everybodys phone, so everybody can see, that the
> person is talking.
> now, if another phonecall comes in, all led-lights are flashing -
> including the led of the person who is talking. if another person (p2)
> answers the second phonecall, the led of p2 is on, but the led of p1
> is off, although p1 is still talking.
>
> is there something we can do about this?
>

Do about what?  Your description sounds fine, what is the problem

Mike




From anthony.minessale at gmail.com  Wed Aug 26 08:25:05 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 10:25:05 -0500
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <4A94A2E0.6040306@nowthor.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
	<4A93EFE1.6010007@nowthor.com>
	<191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>
	<4A94A2E0.6040306@nowthor.com>
Message-ID: <191c3a030908260825n73d6ffc2kbc87296197e72c90@mail.gmail.com>

Keep in mind I am not trying to just arrogantly shoot down your idea, I
enjoy the brainstorming.

Unfortunately it's still a problem because the core bridge/originate
mechanism does not parse XML it's only parses the
dial string markup.

The XML dialplan is completely abstract from the originate mechanism because
there can be many
dialplans not just XML.

In the end you must end up with one string of application and one string of
data.

I am very strict about maintaining scope boundaries between abstract
concepts.  I think it's the number 1 breakdown that
led to asterisk getting so messed up over the years and I want to avoid that
same plague.

I hope you understand and if you want to continue to ponder some more
functionality I welcome it.



On Tue, Aug 25, 2009 at 9:50 PM, Carlos S. Antunes <csa at nowthor.com> wrote:

>  Anthony,
>
> Yes, you are right, I was thinking strictly in terms of SIP gateways. I
> guess that instead on the tag "gateway", one could use "channel"? For
> example:
>
> <action application="bridge">
>    <!-- set some kind of *bridge-specific* variable -->
>    <channel order=1 data="sofia/gateway/gw-1/1$1"/>
>    <channel order=1 data="sofia/gateway/gw-2/1$1"/>
>    <channel order=2 data="sofia/gateway/gw-3/1$1">
>        <!-- set some kind of *channel-specific* variable -->
>    </channel >
>    <channel order=3 data="sofia/gateway/gw-4/1$1"/>
> </action>
>
> Would this be acceptable to you?
>
> Regarding the combinations of continue_on_fail and hangup_after bridge,
> I'll give that a try although I suspect it will result in less structured
> and harder to understand markup...
>
> Thanks!
>
> Carlos
>
> Anthony Minessale wrote:
>
> This suggestion violates the scope boundaries.
>
> gateways are specific concept to mod_sofia so a <gateway> tag in <action>
> (part of agnostic xml dialplan)
> does not flow properly.
>
> you can also use combinations of continue_on_fail and hangup_after bridge
> so you can
> just put each bridge statement in it's own action.
>
>
> On Tue, Aug 25, 2009 at 9:06 AM, Carlos S. Antunes <csa at nowthor.com>wrote:
>
>> Max,
>>
>> I would like to see something similar too. For example, it would be
>> wonderful if one could specify multiple gateways to try like this or
>> something similar:
>>
>> <action application="bridge">
>>    <!-- set some kind of *bridge-specific* parameter/variable -->
>>    <gateway order=1 data="sofia/gateway/gw-1/1$1"/>
>>    <gateway order=1 data="sofia/gateway/gw-2/1$1"/>
>>    <gateway order=2 data="sofia/gateway/gw-3/1$1">
>>        <!-- set some kind of *gw-specific* parameter/variable -->
>>    </gateway>
>>    <gateway order=3 data="sofia/gateway/gw-4/1$1"/>
>> </action>
>>
>> One would be able to avoid the "[]" and "{}" hacks and combine
>> sequential and simultaneous trying of gateways.
>>
>> What do the developers think of this?
>>
>> Carlos
>>
>> Max Ivanov wrote:
>> > Nowdays I 'm forced to put multiple "|" to find first free gateway, ie
>> >
>> sofia/gateway/panas111/1000|sofia/gateway/panas112/1000|sofia/gateway/panas113/1000
>> > ,
>> > the whole sting is tooo long, is there any shorter way to write same
>> thing? Like
>> > "sofia/gateway/panas*/1000" will try all gateways matching the pattern.
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> ------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From mitul at enterux.com  Wed Aug 26 08:31:09 2009
From: mitul at enterux.com (Mitul Limbani)
Date: Wed, 26 Aug 2009 11:31:09 -0400
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
Message-ID: <2918.1251300669@enterux.com>

  BODY { font-family:Arial, Helvetica, sans-serif;font-size:12px;
}Infact coming from Asterisk world is not too hard to figure out FS,
but yeah i wont recommend a newbie that route :)
 Although an Asterisk dude /dudette can check out the Rosetta_Stone
page on wiki
 Thanks & Regards, 
 Mitul Limbani, 
 Founder & CEO, 
 Enterux Solutions, 
 The Enterprise Linux Company (TM), 
 www.enterux.com 
 On Wed 26/08/09 19:35 , Bradley Brashier bjbrashier at gmail.com sent:
 Speaking as someone who went through this recently myself, my
 suggestion is to learn by doing. Choose some simple-sounding task
you
 want to accomplish in FS, and try to carry it out. Use the wiki as a
 reference and the mailing list if you can't figure out what you need
 from that.
 I don't suggest you start with Asterisk or anything else, as I'd
guess
 that would just confuse the issue. I came from a BIOS background,
went
 directly into Freeswitch, and am so far glad I did.
 BB
 On Wed, Aug 26, 2009 at 7:25 AM, Merle J. Ebbert wrote:
 >
 > Hi,
 >
 > I'm trying to avoid taking up a lot of peoples valuable time.
 >
 > SIP & FS have  brought some ideas for some commercial products but
I
 > need to know where to start.
 >
 > Having once written a proprietary DOS & helped with writing a
RTOS, I
 > consider myself capable
 > of learning.  I (we) just need to know where to start to come up
to
 > speed rapidly.
 >
 > Is there a FreeSWITCH tutorial available?
 > Should someone new start with Asterisk and then possibly move to
FS?
 >
 > Thanks,
 > Merle
 >
 >
 > _______________________________________________
 > FreeSWITCH-users mailing list
 > FreeSWITCH-users at lists.freeswitch.org [2]
 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > http://www.freeswitch.org
 >
 _______________________________________________
 FreeSWITCH-users mailing list
 FreeSWITCH-users at lists.freeswitch.org [3]
 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users

UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 http://www.freeswitch.org
-------------------------
Msg sent via Enterux Enterprise Email Server :
http://www.enterux.com/

Links:
------
[1] mailto:se02005-mec at yahoo.com
[2] mailto:FreeSWITCH-users at lists.freeswitch.org
[3] mailto:FreeSWITCH-users at lists.freeswitch.org
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From odermann at googlemail.com  Wed Aug 26 08:34:13 2009
From: odermann at googlemail.com (Dennis)
Date: Wed, 26 Aug 2009 17:34:13 +0200
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
Message-ID: <5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>

> Do about what? ?Your description sounds fine, what is the problem

2 people are talking and only ONE led is on? for me it sounds wrong.

the led of p1 AND p2 should be on! the led should indicate, who in the
company/group is busy/talking and who is available - i think that's
the joke about the leds.


kind regards
dennis



From tparikh at gmail.com  Wed Aug 26 08:38:53 2009
From: tparikh at gmail.com (Tapan Parikh)
Date: Wed, 26 Aug 2009 08:38:53 -0700
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
In-Reply-To: <2918.1251300669@enterux.com>
References: <2918.1251300669@enterux.com>
Message-ID: <1ecdcb6a0908260838r405dfcf5j44485101b3447505@mail.gmail.com>

I went through this a couple of weeks ago, and found that this
particular example is the most straightforward way to get things
working end-to-end.  I just chose to implement the Hello World Lua
script instead.

http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts


On Wed, Aug 26, 2009 at 8:31 AM, Mitul Limbani<mitul at enterux.com> wrote:
> Infact coming from Asterisk world is not too hard to figure out FS, but yeah
> i wont recommend a newbie that route :)
>
> Although an Asterisk dude /dudette can check out the Rosetta_Stone page on
> wiki
>
> Thanks & Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions,
> The Enterprise Linux Company (TM),
> www.enterux.com
>
>
> On Wed 26/08/09 19:35 , Bradley Brashier bjbrashier at gmail.com sent:
>
> Speaking as someone who went through this recently myself, my
> suggestion is to learn by doing. Choose some simple-sounding task you
> want to accomplish in FS, and try to carry it out. Use the wiki as a
> reference and the mailing list if you can't figure out what you need
> from that.
>
> I don't suggest you start with Asterisk or anything else, as I'd guess
> that would just confuse the issue. I came from a BIOS background, went
> directly into Freeswitch, and am so far glad I did.
>
> BB
>
> On Wed, Aug 26, 2009 at 7:25 AM, Merle J. Ebbert<se02005-mec at yahoo.com>
> wrote:
>>
>> Hi,
>>
>> I'm trying to avoid taking up a lot of peoples valuable time.
>>
>> SIP & FS have ?brought some ideas for some commercial products but I
>> need to know where to start.
>>
>> Having once written a proprietary DOS & helped with writing a RTOS, I
>> consider myself capable
>> of learning. ?I (we) just need to know where to start to come up to
>> speed rapidly.
>>
>> Is there a FreeSWITCH tutorial available?
>> Should someone new start with Asterisk and then possibly move to FS?
>>
>> Thanks,
>> Merle
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> ________________________________
> Msg sent via Enterux Enterprise Email Server : http://www.enterux.com/
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From brian at freeswitch.org  Wed Aug 26 08:41:40 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 Aug 2009 10:41:40 -0500
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
In-Reply-To: <1ecdcb6a0908260838r405dfcf5j44485101b3447505@mail.gmail.com>
References: <2918.1251300669@enterux.com>
	<1ecdcb6a0908260838r405dfcf5j44485101b3447505@mail.gmail.com>
Message-ID: <F7D32744-B3A0-4769-BC98-02F866A77607@freeswitch.org>

This example need to be updated as the grammar format was changed to  
JSGF which is a standard grammar format.

/b

On Aug 26, 2009, at 10:38 AM, Tapan Parikh wrote:

> I went through this a couple of weeks ago, and found that this
> particular example is the most straightforward way to get things
> working end-to-end.  I just chose to implement the Hello World Lua
> script instead.
>
> http://wiki.freeswitch.org/wiki/Examples_directory_lua_asr_tts

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From anthony.minessale at gmail.com  Wed Aug 26 08:42:05 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 10:42:05 -0500
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <fbfa7b0c0908252244x65d19808r967a23332de6a0a0@mail.gmail.com>
References: <4A94A2E0.6040306@nowthor.com>
	<C6BA0F7E.5D757%krice@freeswitch.org>
	<fbfa7b0c0908252244x65d19808r967a23332de6a0a0@mail.gmail.com>
Message-ID: <191c3a030908260842s573de3b8ncf1ea4b76ee86232@mail.gmail.com>

But how does get_avail_gw find out it's avail?
It would have to try making the call so you would call over it to find out
it's ok to call then pass that back up to bridge?

You can try to hide the complexity but you still have to stash it somewhere.

So if you don't like looking at it consider making a script or a C mod to
hide it.

Here's one cool way you can do with just regular dialplan.


    <extension name="gateways">
      <condition field="destination_number" expression="gateway_macro">
        <action application="set" data="continue_on_fail=true"/>
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="bridge"
data="sofia/gateway/gw1/${dialed_number}"/>
        <action application="bridge"
data="sofia/gateway/gw2/${dialed_number}"/>
        <action application="bridge"
data="sofia/gateway/gw3/${dialed_number}"/>
        <action application="bridge"
data="sofia/gateway/gw4/${dialed_number}"/>
        <action application="bridge"
data="sofia/gateway/gw5/${dialed_number}"/>
      </condition>
    </extension>

    <extension name="myext">
      <condition field="destination_number" expression="^(whatever)$">
        <action application="set" data="dialed_number=$1"/>
        <action application="transfer" data="gateway_macro"/>
      </condition>
    </extension>

change whatever to your favorite regex
so now all you have to do is set dialed_number and transfer to gateway_macro
(you could have many of these with different names)


One way or another all your real goal seems to be is hiding the complexity
so *shrug* here's and easy way with existing tools.



On Wed, Aug 26, 2009 at 12:44 AM, Max Ivanov <ivanov.maxim at gmail.com> wrote:

> >
> > Continue_on_fail and hangup_after_bridge like tony pointed out are what
> you
> > want if you don?t want to use the | delimiting ... I use these all the
> time
> > with gateway counts > 10 just stacking additional actions for each bridge
> > line
>
> Let's imagine that I need to call 1000,1001,1002 via  gw1,gw2,gw3 by
> choosing first available. How dialplan would look like?
>
> continue_on_fail=True
> hangup_after_bridge=True
> bridge gw1/1000
> bridge gw1/1001
> bridge gw1/1002
> bridge gw2/1000
> bridge gw2/1001
> bridge gw2/1002
> bridge gw3/1000
> bridge gw3/1001
> bridge gw3/1002
>
> Is it easy to understand? From my point of view it's not. Compare to this:
> continue_on_fail=True
> hangup_after_bridge=True
> bridge ${get_avail_gw(gw1,gw2,gw3)}/1000
> bridge ${get_avail_gw(gw1,gw2,gw3)}/1001
> bridge ${get_avail_gw(gw1,gw2,gw3)}/1002
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Wed Aug 26 08:44:40 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 10:44:40 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
Message-ID: <191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>

if you are talking to a device that pretends to do session timers but really
does not, the re-invite can cause a hangup.
We have seen this with sonus many times, do you have sonus involved in your
call per chance?


On Wed, Aug 26, 2009 at 7:59 AM, Dennis <odermann at googlemail.com> wrote:

> > You have a NAT issue.
>
> as i wrote, we are quite sure, that this is a nat problem. but we have
> no idea how we could fix this. there must be a reason, why the hangup
> always comes after 120 seconds.
>
>
> >    <!--RTP port range -->
> >    <param name="rtp-start-port" value="10000"/>
> >    <param name="rtp-end-port" value="32767"/>
>
> we tested that, but this does not change anything (nothing becomes
> worse or better).
>
>
> > param name="session-timeout" value="120"
>
> this line is commented in our profile and we believe, that this has
> nothing to do with our problem.
>
>
> to tell a little bit more about our problem:
>
> under
> http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios#Scenario_2
> you can find the scenario, which is exactly like ours. we played with
> all the mentioned settings, but it does not work.
>
> to mention one thing: if we connect with xlite (softphone), we do not
> have any problems. but we have snom-voip-phones and with them we have
> the problems.
>
> the big question is: wo or what might cause/trigger the hangup? is it
> freeswitch or something else? we have a firewall (IPCop) - might there
> be a setting, which needs to be set, to avoid theses problems?
>
>
> kind regards
> dennis
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From mike at jerris.com  Wed Aug 26 08:52:57 2009
From: mike at jerris.com (Michael Jerris)
Date: Wed, 26 Aug 2009 11:52:57 -0400
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
	<5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
Message-ID: <274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>


On Aug 26, 2009, at 11:34 AM, Dennis wrote:

>> Do about what?  Your description sounds fine, what is the problem
>
> 2 people are talking and only ONE led is on? for me it sounds wrong.
>
> the led of p1 AND p2 should be on! the led should indicate, who in the
> company/group is busy/talking and who is available - i think that's
> the joke about the leds.
>
ahh, I understand the issue now.  Please open a jira on jira.freeswitch.org 
  for this issue.

Mike




From odermann at googlemail.com  Wed Aug 26 09:10:06 2009
From: odermann at googlemail.com (Dennis)
Date: Wed, 26 Aug 2009 18:10:06 +0200
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
	<5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
	<274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>
Message-ID: <5e414ed0908260910h55122d77o5013ef1795d0c75@mail.gmail.com>

> ahh, I understand the issue now. ?Please open a jira on jira.freeswitch.org
> ?for this issue.

ok, we could not imagine, that this behavior is meant to be.

we will try to open a jira with this issue (never opened a jira before).


kind regards
dennis



From odermann at googlemail.com  Wed Aug 26 09:23:00 2009
From: odermann at googlemail.com (Dennis)
Date: Wed, 26 Aug 2009 18:23:00 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
Message-ID: <5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>

> if you are talking to a device that pretends to do session timers but really
> does not, the re-invite can cause a hangup.
> We have seen this with sonus many times, do you have sonus involved in your
> call per chance?

hi anthony,

this is what we found on the snom website about the session timer
settings: "If SIP Session Timer Support is enabled, this option
specifies the SIP session timer in seconds. For instance, a Re-INVITE
will be sent after 50% of its value has elapsed." our setting is 3600.

is session timer and re-invite helpful or can we disable it? behaps
disabling works?

is it possible, that snom does not support a REAL session timer?

sonus is not involved.


kind regards
dennis



From MPeace at edcogroupinc.com  Wed Aug 26 09:20:49 2009
From: MPeace at edcogroupinc.com (Mike Peace)
Date: Wed, 26 Aug 2009 11:20:49 -0500
Subject: [Freeswitch-users] Moved Freeswitch to new subnet.
In-Reply-To: <69D652F49932C34199968DFB8AEAABA33A622D22BE@ESNEXS2.edcogroup.net>
References: <5DF610F9-6EB5-43CC-BB4F-3513C78F608F@freeswitch.org>,
	<69D652F49932C34199968DFB8AEAABA33A622D22BE@ESNEXS2.edcogroup.net>
Message-ID: <69D652F49932C34199968DFB8AEAABA33A622F13DB@ESNEXS2.edcogroup.net>

OK I edited the openzap.conf and the open.conf.xml and added the gateway info from the Wiki for dilaing out through a gateway. Still no love but I also noticed this when starting Freeswitch:

2009-08-26 11:04:53 [CRIT] switch_loadable_module.c:839 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_openzap.so
**/usr/local/freeswitch/mod/mod_openzap.so: cannot open shared object file: No such file or directory**


Sure enough the openzap.so is not there, does the fact I used the "wanpipe mode" have anything to do with this ? I have searched the drive and it is not there. 
________________________________________
From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mike Peace [MPeace at edcogroupinc.com]
Sent: Wednesday, August 26, 2009 8:58 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.

I found and edited openzap.cong file, the openzap.conf.xml file is in the correct dir but the only example I can find is for a Euro E1 setup not a US T1. I

I have turned up data T1 circuits for WAN links before but not telco stuff. The provider is Qwest, I?ll have to ask about the NI2 , DMS question.

Mike

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael S Collins
Sent: Wednesday, August 26, 2009 3:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Moved Freeswitch to new subnet.


I have looked at that but I am confused on which files need to be edited. Since I have already installed in Wanpipe mode with the Sangoma card I skipped straight to the Wanpipe section. It mentions setting the [span wanpie PRI_1] etc in the openzap.conf then further down it mentions editing the autoload_configs/openzap.conf.xml with a completely different format of config.

Are these the same file? If not where does the first one mentioned reside?

These are two different files. The openzap.conf file is in the freeswitch/conf directory. It is for configuring OZ to talk to the Sangoma card. The openzap.conf.xml file is for configuring the actual openzap module in FreeSWITCH.

The specs you listed on the T1 you have indicate that it's a PRI. However, you need to know which protocol, like NI2 or DMS. Have you ever done a PRI turn up? Who is the carrier?

-MC
=-=-=-=-=-=-=-=-=-=-=-=-=-=-EDCO Group, Inc.
Phone: (800) 999-3456
Fax:   (800) 999-3551
Web:   http://www.edcogroupinc.com/

Confidentiality Notice:  The information contained in this e-mail message (including any attachments) may contain confidential and privileged information, and is for the sole use of the intended recipient(s).  If you are not the intended recipient, any unauthorized review, use, or disclosure or distribution is strictly prohibited.  If you have received this message in error, please notify the sender by replying to this e-mail message or by telephone at (800) 999-3456 and permanently destroy all copies of the original message (including any attachments), along with any reply, and delete them from your system.



From foxb at abv.bg  Wed Aug 26 09:37:16 2009
From: foxb at abv.bg (Hristo Benev)
Date: Wed, 26 Aug 2009 19:37:16 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <2089188486.120014.1251304644860.JavaMail.apache@mail22.abv.bg>

 Hello

I followed the tutorial
http://wiki.freeswitch.org/wiki/SBC_Setup

I have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not route

Where to look for problems?

Here is the debug:

2009-08-26 20:21:42.332154 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context default
Dialplan: sofia/internal/1001 at 10.10.10.10 parsing [default->LOOKUP_ROUTE] continue=false
Dialplan: sofia/internal/1001 at 10.10.10.10 Regex (PASS) [LOOKUP_ROUTE] destination_number(1000) =~ /(\d+)$/ break=on-false
Dialplan: sofia/internal/1001 at 10.10.10.10 Action set(hangup_after_bridge=true)
Dialplan: sofia/internal/1001 at 10.10.10.10 Action set(continue_on_fail=true)
Dialplan: sofia/internal/1001 at 10.10.10.10 Action export(sip_h_X-ROUTE=LOOKUP)
Dialplan: sofia/internal/1001 at 10.10.10.10 Action bridge(sofia/internal/${destination_number}@127.0.0.1:5062)
Dialplan: sofia/internal/1001 at 10.10.10.10 Action set(ROUTE_GW=${sip_redirect_contact_user_0})
Dialplan: sofia/internal/1001 at 10.10.10.10 Action set(AREA=${sip_redirect_contact_user_0})
Dialplan: sofia/internal/1001 at 10.10.10.10 Action transfer(${destination_number} XML ROUTING)
2009-08-26 20:21:42.334579 [DEBUG] switch_core_state_machine.c:114 (sofia/internal/1001 at 10.10.10.10) State Change CS_ROUTING -> CS_EXECUTE
2009-08-26 20:21:42.334579 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1001 at 10.10.10.10 [BREAK]
2009-08-26 20:21:42.334579 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/1001 at 10.10.10.10) State ROUTING going to sleep
2009-08-26 20:21:42.335292 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1001 at 10.10.10.10) Running State Change CS_EXECUTE
2009-08-26 20:21:42.335292 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1001 at 10.10.10.10) State EXECUTE
2009-08-26 20:21:42.335292 [DEBUG] mod_sofia.c:173 sofia/internal/1001 at 10.10.10.10 SOFIA EXECUTE
2009-08-26 20:21:42.335292 [DEBUG] switch_core_state_machine.c:151 sofia/internal/1001 at 10.10.10.10 Standard EXECUTE
EXECUTE sofia/internal/1001 at 10.10.10.10 set(hangup_after_bridge=true)
2009-08-26 20:21:42.336509 [DEBUG] mod_dptools.c:748 sofia/internal/1001 at 10.10.10.10 SET [hangup_after_bridge]=[true]
EXECUTE sofia/internal/1001 at 10.10.10.10 set(continue_on_fail=true)
2009-08-26 20:21:42.337353 [DEBUG] mod_dptools.c:748 sofia/internal/1001 at 10.10.10.10 SET [continue_on_fail]=[true]
EXECUTE sofia/internal/1001 at 10.10.10.10 export(sip_h_X-ROUTE=LOOKUP)
2009-08-26 20:21:42.338352 [DEBUG] mod_dptools.c:886 EXPORT [sip_h_X-ROUTE]=[LOOKUP]
EXECUTE sofia/internal/1001 at 10.10.10.10 bridge(sofia/internal/1000 at 127.0.0.1:5062)
2009-08-26 20:21:42.339309 [NOTICE] switch_channel.c:602 New Channel sofia/internal/1000 at 127.0.0.1:5062 [8a588e6a-925c-11de-85dd-15dc0a06983f]
2009-08-26 20:21:42.339309 [DEBUG] mod_sofia.c:2811 (sofia/internal/1000 at 127.0.0.1:5062) State Change CS_NEW -> CS_INIT
2009-08-26 20:21:42.340376 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1000 at 127.0.0.1:5062 [BREAK]
2009-08-26 20:21:42.341175 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1000 at 127.0.0.1:5062) Running State Change CS_INIT
2009-08-26 20:21:42.341175 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/1000 at 127.0.0.1:5062) State INIT
2009-08-26 20:21:42.341175 [DEBUG] mod_sofia.c:83 sofia/internal/1000 at 127.0.0.1:5062 SOFIA INIT
2009-08-26 20:21:42.344378 [DEBUG] mod_sofia.c:111 (sofia/internal/1000 at 127.0.0.1:5062) State Change CS_INIT -> CS_ROUTING
2009-08-26 20:21:42.344378 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1000 at 127.0.0.1:5062 [BREAK]
2009-08-26 20:21:42.344378 [DEBUG] switch_core_state_machine.c:481 (sofia/internal/1000 at 127.0.0.1:5062) State INIT going to sleep
2009-08-26 20:21:42.344378 [DEBUG] sofia.c:3289 Channel sofia/internal/1000 at 127.0.0.1:5062 entering state [calling][0]
2009-08-26 20:21:42.344378 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1000 at 127.0.0.1:5062) Running State Change CS_ROUTING
2009-08-26 20:21:42.345314 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/1000 at 127.0.0.1:5062) State ROUTING
2009-08-26 20:21:42.345314 [DEBUG] mod_sofia.c:130 sofia/internal/1000 at 127.0.0.1:5062 SOFIA ROUTING
2009-08-26 20:21:42.345314 [DEBUG] switch_ivr_originate.c:63 (sofia/internal/1000 at 127.0.0.1:5062) State Change CS_ROUTING -> CS_CONSUME_MEDIA
2009-08-26 20:21:42.345314 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1000 at 127.0.0.1:5062 [BREAK]
2009-08-26 20:21:42.345314 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/1000 at 127.0.0.1:5062) State ROUTING going to sleep
2009-08-26 20:21:42.346255 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1000 at 127.0.0.1:5062) Running State Change CS_CONSUME_MEDIA
2009-08-26 20:21:42.346255 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/1000 at 127.0.0.1:5062) State CONSUME_MEDIA
2009-08-26 20:21:42.349173 [DEBUG] sofia.c:3289 Channel sofia/internal/1000 at 127.0.0.1:5062 entering state [calling][0]
2009-08-26 20:21:42.350242 [DEBUG] sofia.c:3289 Channel sofia/internal/1000 at 127.0.0.1:5062 entering state [terminated][503]
2009-08-26 20:21:42.350242 [NOTICE] sofia.c:3849 Hangup sofia/internal/1000 at 127.0.0.1:5062 [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE]
2009-08-26 20:21:42.351172 [DEBUG] switch_ivr_originate.c:2138 Originate Resulted in Error Cause: 41 [NORMAL_TEMPORARY_FAILURE]
2009-08-26 20:21:42.351172 [INFO] mod_dptools.c:2093 Originate Failed.  Cause: NORMAL_TEMPORARY_FAILURE
2009-08-26 20:21:42.351172 [DEBUG] mod_dptools.c:2120 Continue on fail [true]:  Cause: NORMAL_TEMPORARY_FAILURE
EXECUTE sofia/internal/1001 at 10.10.10.10 set(ROUTE_GW=PEER-01)
2009-08-26 20:21:42.352234 [DEBUG] mod_dptools.c:748 sofia/internal/1001 at 10.10.10.10 SET [ROUTE_GW]=[PEER-01]
2009-08-26 20:21:42.352234 [DEBUG] switch_channel.c:1683 Send signal sofia/internal/1000 at 127.0.0.1:5062 [KILL]
2009-08-26 20:21:42.352234 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1000 at 127.0.0.1:5062 [BREAK]
EXECUTE sofia/internal/1001 at 10.10.10.10 set(AREA=PEER-01)
2009-08-26 20:21:42.353179 [DEBUG] mod_dptools.c:748 sofia/internal/1001 at 10.10.10.10 SET [AREA]=[PEER-01]
EXECUTE sofia/internal/1001 at 10.10.10.10 transfer(1000 XML ROUTING)
2009-08-26 20:21:42.355174 [DEBUG] switch_ivr.c:1343 (sofia/internal/1001 at 10.10.10.10) State Change CS_EXECUTE -> CS_ROUTING
2009-08-26 20:21:42.355174 [DEBUG] switch_core_session.c:932 Send signal sofia/internal/1001 at 10.10.10.10 [BREAK]
2009-08-26 20:21:42.355174 [DEBUG] switch_ivr.c:1347 sofia/internal/1001 at 10.10.10.10 receive message [TRANSFER]
2009-08-26 20:21:42.355174 [DEBUG] switch_core_session.c:630 Send signal sofia/internal/1001 at 10.10.10.10 [BREAK]
2009-08-26 20:21:42.355174 [NOTICE] switch_ivr.c:1349 Transfer sofia/internal/1001 at 10.10.10.10 to XML[1000 at ROUTING]
2009-08-26 20:21:42.356405 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1001 at 10.10.10.10) State EXECUTE going to sleep
2009-08-26 20:21:42.356405 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1001 at 10.10.10.10) Running State Change CS_ROUTING
2009-08-26 20:21:42.357172 [DEBUG] switch_core_state_machine.c:503 (sofia/internal/1000 at 127.0.0.1:5062) State CONSUME_MEDIA going to sleep
2009-08-26 20:21:42.357172 [DEBUG] switch_core_state_machine.c:398 (sofia/internal/1000 at 127.0.0.1:5062) Running State Change CS_HANGUP
2009-08-26 20:21:42.357172 [DEBUG] switch_core_state_machine.c:484 (sofia/internal/1001 at 10.10.10.10) State ROUTING
2009-08-26 20:21:42.357172 [DEBUG] mod_sofia.c:130 sofia/internal/1001 at 10.10.10.10 SOFIA ROUTING
2009-08-26 20:21:42.357172 [DEBUG] switch_core_state_machine.c:78 sofia/internal/1001 at 10.10.10.10 Standard ROUTING
2009-08-26 20:21:42.357172 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
Dialplan: sofia/internal/1001 at 10.10.10.10 parsing [ROUTING->PEER_01] continue=false
2009-08-26 20:21:42.358179 [DEBUG] switch_core_state_machine.c:434 (sofia/internal/1000 at 127.0.0.1:5062) State HANGUP
Dialplan: sofia/internal/1001 at 10.10.10.10 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
2009-08-26 20:21:42.358179 [INFO] switch_core_state_machine.c:136 No Route, Aborting
2009-08-26 20:21:42.358179 [DEBUG] mod_sofia.c:306 sofia/internal/1000 at 127.0.0.1:5062 Overriding SIP cause 503 with 503 from the other leg
2009-08-26 20:21:42.359202 [NOTICE] switch_core_state_machine.c:137 Hangup sofia/internal/1001 at 10.10.10.10 [CS_ROUTING] [NO_ROUTE_DESTINATION]




From brian at freeswitch.org  Wed Aug 26 09:47:37 2009
From: brian at freeswitch.org (Brian West)
Date: Wed, 26 Aug 2009 11:47:37 -0500
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
In-Reply-To: <2089188486.120014.1251304644860.JavaMail.apache@mail22.abv.bg>
References: <2089188486.120014.1251304644860.JavaMail.apache@mail22.abv.bg>
Message-ID: <94B8E08E-2537-4FA4-9047-8904975D7BFB@freeswitch.org>

We do not blindly follow 302's as that is a dangerous thing to do.   
You have to process all 302's in the dialplan.

Set this on your sofia profile <param name="manual-redirect"  
value="true"/>

You can set these variables sip_redirect_profile,  
sip_redirect_context, sip_redirect_dialplan,

When a redirect happens you get these variables - sip_redirect_contact_ 
%d, sip_redirected_to, sip_redirect_contact_user_%d,  
sip_redirect_contact_host_%d, sip_redirect_contact_params_%d,  
sip_redirect_dialstring_%d, sip_redirect_dialstring, sip_redirected_by

Then its up to you to process the redirect in your dialplan, If you  
don't set the sip_redirect_context then it'll default to redirected  
context and XML as the dialplan.

/b


On Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:

> Hello
>
> I followed the tutorial
> http://wiki.freeswitch.org/wiki/SBC_Setup
>
> I have following problem when I dial 1000 Kamalio reports 302, but  
> freeswitch does not route
>
> Where to look for problems?

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From anthony.minessale at gmail.com  Wed Aug 26 09:53:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 11:53:03 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
Message-ID: <191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>

Is the snom firmware up to the latest?
I believe session timers should work properly with snom?
you can try disabling it but make sure you have your firmware to latest.


On Wed, Aug 26, 2009 at 11:23 AM, Dennis <odermann at googlemail.com> wrote:

> > if you are talking to a device that pretends to do session timers but
> really
> > does not, the re-invite can cause a hangup.
> > We have seen this with sonus many times, do you have sonus involved in
> your
> > call per chance?
>
> hi anthony,
>
> this is what we found on the snom website about the session timer
> settings: "If SIP Session Timer Support is enabled, this option
> specifies the SIP session timer in seconds. For instance, a Re-INVITE
> will be sent after 50% of its value has elapsed." our setting is 3600.
>
> is session timer and re-invite helpful or can we disable it? behaps
> disabling works?
>
> is it possible, that snom does not support a REAL session timer?
>
> sonus is not involved.
>
>
> kind regards
> dennis
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From pbd at suspiria.net  Wed Aug 26 10:28:50 2009
From: pbd at suspiria.net (Public Dump)
Date: Wed, 26 Aug 2009 19:28:50 +0200
Subject: [Freeswitch-users] delay buildup in conference
Message-ID: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>

When running conferences with users dialed in from a PSTN gateway (SIP) and directly from remote SIP endpoints there is an ever longer buildup in delay, reaching up to multiple seconds. Is there any way to limit the delay ?



I am not 100% sure whether the delays is caused by the SIP jitter buffer of freeswitch or directly by the conference module.



Any advice?



Thanks

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From tculjaga at gmail.com  Wed Aug 26 10:42:19 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Wed, 26 Aug 2009 19:42:19 +0200
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <681a20520908260453q57877b1eg8f0949061cd129ee@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com>
	<65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com>
	<7b197bef0908260125m56c3d681l26cb18a5e34aaa33@mail.gmail.com>
	<65d96fc80908260129xebbf460raffaa38c72c3281a@mail.gmail.com>
	<681a20520908260453q57877b1eg8f0949061cd129ee@mail.gmail.com>
Message-ID: <65d96fc80908261042l42437ed9xc4fab87b757adc80@mail.gmail.com>

Guys you made a monster!!


so, i moved the machine to 64bit CentOS 5.3... recompiled the latest trunk
and did my tests again.

The old tests on 32bit debian 5 on the same hardware shown a CPS rate of 120
as 75 - 80% CPU.... and after some time on that 120 CPS rate the CPU goes to
100% without any chance FS recovers at all.
New tests on 64bit CentOS shown a monster.... 400 CPS rate at 75% CPU....
during the tests FS was really stable and responsive... i placed few calls
that went through as a charm :).


Also, i didn't optimize the machine at all ... as it is after CentOS
install!.... not even db files are on ramdisk.

What did it really happen? .. did you guys change something in the trunk
overnight or it is just moving to CentOS 64bit that boosted drastically?




Here are some details:

?nmon?12a??????[H for help]???Hostname=l01sipindir2?Refresh= 2secs
???19:17.48??????????????????????????????????????????????????????
? CPU
+-------------------------------------------------------------------------+
?
?100%-|
|
?
? 95%-|
|
?
? 90%-|
|
?
? 85%-|
|
?
? 80%-|  | w              s  w      www  w        s   sw    s    s
s                                                     ?
? 75%-|
|ssssssssssssssssssssswssssssssssssssssssssssssssssssssssssssssssssss
?
? 70%-|
+sssssssssssssssssssssssssssssssUssssssssUsssUssssssssssUssssssssssss
?
? 65%-|
|UUUUUsUsUUUUUUUUUUUUUsUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUsUUU
?
? 60%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 55%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 50%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 45%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 40%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUw
?
? 35%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUs
?
? 30%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 25%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 20%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 15%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
? 10%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
?  5%-|
|UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
?
?
+-------------------------------------------------------------------------+
?
? CPU Utilisation
??????????????????????????????????????????????????????????????????????????????????????????????????????????????????
?
+-------------------------------------------------+
?
?CPU  User%  Sys% Wait% Idle|0          |25         |50          |75
100|                                                    ?
? 1   1.0   0.5   0.0   98.5|
>          |                                                    ?
? 2   1.5   1.0   0.0   97.5|
>  |                                                    ?
?
+-------------------------------------------------+
?
?Avg  1.2   0.5   0.0   98.3|
>         |                                                    ?
?
+-------------------------------------------------+
?
? Disk I/O ?????(/proc/diskstats)????????all data is Kbytes per
second??????????????????????????????????????????????????????????????
?DiskName Busy  Read WriteKB|0          |25         |50          |75
100|                                                    ?
?iss/c0d0   0%    0.0    0.0|
>   |                                                    ?
?s/c0d0p1   0%    0.0
0.0|>
|                                                    ?
?s/c0d0p2   0%    0.0    0.0|
>  |                                                    ?
?dm-0       0%    0.0    0.0|
>  |                                                    ?
?dm-1       0%    0.0
0.0|>
|                                                    ?
????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????







------------------------------ Scenario Screen -------- [1-9]: Change Screen
--
  Call-rate(length)   Port   Total-time  Total-calls  Remote-host
  40.0(0 ms)/0.100s   5060     488.20 s       193555  10.4.4.252:5060(UDP)

  402 new calls during 1.002 s period    0 ms scheduler resolution
  3 calls (limit 4000)                   Peak was 53 calls, after 351 s
  0 Running, 13216 Paused, 670 Woken up
  0 dead call msg (discarded)            0 out-of-call msg
(discarded)
  3 open sockets

                                 Messages  Retrans   Timeout
Unexpected-Msg
      INVITE ---------->  B-RTD1 193553    0         0
         100 <----------  E-RTD1 193553    0         0         0
         302 <----------  E-RTD2 193552    0         0         0
         ACK ---------->         193552    0
------ [+|-|*|/]: Adjust rate ---- [q]: Soft exit ---- [p]: Pause traffic
-----

------------------------------ Scenario Screen -------- [1-9]: Change Screen
--
  Call-rate(length)   Port   Total-time  Total-calls  Remote-host
  40.0(0 ms)/0.100s   5060     489.10 s       193917  10.4.4.252:5060(UDP)

  362 new calls during 0.906 s period    1 ms scheduler resolution
  2 calls (limit 4000)                   Peak was 53 calls, after 351 s
  0 Running, 13215 Paused, 623 Woken up
  0 dead call msg (discarded)            0 out-of-call msg
(discarded)
  3 open sockets

                                 Messages  Retrans   Timeout
Unexpected-Msg
      INVITE ---------->  B-RTD1 193917    0         0
         100 <----------  E-RTD1 193917    0         0         0
         302 <----------  E-RTD2 193915    0         0         0
         ACK ---------->         193915    0
------------------------------ Test Terminated
--------------------------------


----------------------------- Statistics Screen ------- [1-9]: Change Screen
--
  Start Time             | 2009-08-26   19:09:34:575
1251306574.575684
  Last Reset Time        | 2009-08-26   19:17:42:779
1251307062.779468
  Current Time           | 2009-08-26   19:17:43:685
1251307063.685281
-------------------------+---------------------------+--------------------------
  Counter Name           | Periodic value            | Cumulative value
-------------------------+---------------------------+--------------------------
  Elapsed Time           | 00:00:00:905              |
00:08:09:109
  Call Rate              |  400.000 cps              |  396.470
cps
-------------------------+---------------------------+--------------------------
  Incoming call created  |        0                  |
0
  OutGoing call created  |      362                  |
193917
  Total Call created     |                           |
193917
  Current Call           |        2
|
-------------------------+---------------------------+--------------------------
  Successful call        |      363                  |
193915
  Failed call            |        0                  |
0
-------------------------+---------------------------+--------------------------
  Response Time 1        | 00:00:00:001              |
00:00:00:000
  Response Time 2        | 00:00:00:010              |
00:00:00:008
  Call Length            | 00:00:00:010              |
00:00:00:008
------------------------------ Test Terminated
--------------------------------



...i didn't beleive to SIPp and i went to FS console issuing status command
to conferm the results.


freeswitch at l01sipindir2.ot.hr> status
API CALL [status()] output:
UP 0 years, 0 days, 0 hours, 8 minutes, 13 seconds, 703 milliseconds, 971
microseconds
183382 session(s) since startup
1 session(s) 410/800
8000 session(s) max

freeswitch at l01sipindir2.ot.hr> status
API CALL [status()] output:
UP 0 years, 0 days, 0 hours, 8 minutes, 15 seconds, 109 milliseconds, 891
microseconds
183944 session(s) since startup
1 session(s) 401/800
8000 session(s) max

freeswitch at l01sipindir2.ot.hr> status
API CALL [status()] output:
UP 0 years, 0 days, 0 hours, 8 minutes, 16 seconds, 139 milliseconds, 412
microseconds
184356 session(s) since startup
2 session(s) 389/800
8000 session(s) max

freeswitch at l01sipindir2.ot.hr> status
API CALL [status()] output:
UP 0 years, 0 days, 0 hours, 8 minutes, 17 seconds, 62 milliseconds, 16
microseconds
184717 session(s) since startup
6 session(s) 410/800
8000 session(s) max

freeswitch at l01sipindir2.ot.hr> status
API CALL [status()] output:
UP 0 years, 0 days, 0 hours, 8 minutes, 35 seconds, 150 milliseconds, 253
microseconds
191959 session(s) since startup
1 session(s) 400/800
8000 session(s) max

freeswitch at l01sipindir2.ot.hr> status
API CALL [status()] output:
UP 0 years, 0 days, 0 hours, 8 minutes, 36 seconds, 892 milliseconds, 672
microseconds
192657 session(s) since startup
1 session(s) 393/800
8000 session(s) max



On Wed, Aug 26, 2009 at 1:53 PM, Dmitry Kadantsev <kadantsev.d at gmail.com>wrote:

> Hi all,
>
> is there same situation with FS for Windows? I mean 64bit is more
> preferable than 32bit, isn't it?
>
> Any performance test on Win 32/64 were done?
>
> --
> Best regards,
> Dmitry Kadantsev
>
>
>
> On Wed, Aug 26, 2009 at 10:29 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> intanto e il centos che si sta installando :)
>>
>> grazie.
>>
>> T.
>>
>>
>> On Wed, Aug 26, 2009 at 10:25 AM, Giovanni Maruzzelli <
>> gmaruzz at celliax.org> wrote:
>>
>>> netbook remix
>>>
>>>
>>> joking! Server 64bit :-)
>>>
>>> -gm
>>>
>>>
>>>
>>> On Wed, Aug 26, 2009 at 10:08 AM, Tihomir Culjaga<tculjaga at gmail.com>
>>> wrote:
>>> > Hi Giovanny,
>>> >
>>> > regarding ubuntu, did you mean 8.04 server or desktop ?
>>> >
>>> >
>>> > On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli <
>>> gmaruzz at celliax.org>
>>> > wrote:
>>> >>
>>> >> Definitely go for 64 bit OS.
>>> >>
>>> >> If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
>>> >> used both for development and for heavy duty production.
>>> >>
>>> >> Also Ubuntu 8.04 is good.
>>> >>
>>> >> Other versions/distros are less used by the community.
>>> >>
>>> >> Adding RAM and CPUs helps to scale up.
>>> >>
>>> >> -gm
>>> >>
>>> >>
>>> >>
>>> >> Sincerely,
>>> >>
>>> >> Giovanni Maruzzelli
>>> >> Cell : +39-347-2665618
>>> >>
>>> >>
>>> >>
>>> >>
>>> >> On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com>
>>> >> wrote:
>>> >> > Hey Giovanni,
>>> >> >
>>> >> > thanks for the tip... indeed the db files were heavily used
>>> regardless
>>> >> > if i
>>> >> > started freeswitch with nosql option (freeswitch -nosql)... FS was
>>> not
>>> >> > writing anything into that files ... instead it was just accessing
>>> >> > it....
>>> >> > This behaviour leads to a waste of 40% CPU time... waiting for other
>>> >> > processes (mainly disk access) to finish!!!
>>> >> >
>>> >> > I moved freeswitch/db/ to a ramdisk and the performance got a boost
>>> to
>>> >> > 140
>>> >> > CPS with a CPU load of 80%. I was keeping the machine for a while
>>> (20 -
>>> >> > 30
>>> >> > minutes) on that rate when i sow CPU suddenly went to 100% and FS
>>> >> > becoming
>>> >> > irresponsive :).
>>> >> >
>>> >> >
>>> >> > What can be wrong?
>>> >> > What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should
>>> not
>>> >> > cross?
>>> >> > What fine tuning do we need in order to asure a long high load run?
>>> >> >
>>> >> >
>>> >> >
>>> >> > Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it
>>> have
>>> >> > sense
>>> >> > to move my OS to 64 bit? ... will FS gain more preformance ?... I
>>> mean
>>> >> > will
>>> >> > FS perofomr drastically better 20%+ ?
>>> >> >
>>> >> >
>>> >> > Tihomir.
>>> >> >
>>> >> >
>>> >> > On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
>>> >> > <gmaruzz at celliax.org>
>>> >> > wrote:
>>> >> >>
>>> >> >> Maybe your load comes from disk access?
>>> >> >>
>>> >> >> Try putting the sql and log directories on a ramdisk.
>>> >> >>
>>> >> >> OTH,
>>> >> >>
>>> >> >> -giovanni
>>> >> >>
>>> >> >> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<
>>> tculjaga at gmail.com>
>>> >> >> wrote:
>>> >> >> > Hello,
>>> >> >> >
>>> >> >> > i'm trying to use freeswitch as a redirecting server meaning FS
>>> has
>>> >> >> > to
>>> >> >> > receive an INVITE and according to some rules it will redirect
>>> calls
>>> >> >> > to
>>> >> >> > other destinations.
>>> >> >> >
>>> >> >> >
>>> >> >> > CALLING_USER                FREESWITCH
>>> >> >> > SOMEWHERE
>>> >> >> >
>>> >> >> > INVITE ------------------------------->
>>> >> >> >            <------------------------------ 100 Trying
>>> >> >> >            <------------------------------ 302 Moved Temporary
>>> >> >> > ACK    ------------------------------->
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> INVITE--------------------------------------------------------------------------------->
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > Well, wverything works well except i have perfromance issues ....
>>> on
>>> >> >> > my
>>> >> >> > HW
>>> >> >> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
>>> >> >> > Temporary).
>>> >> >> > When
>>> >> >> > i increase the rate, FS starts delaying 302 response. Right at 50
>>> CPS
>>> >> >> > i
>>> >> >> > see
>>> >> >> > "calls" being build up in FS and the delay begining to grow.
>>> >> >> >
>>> >> >> > When i observe the machine, load average is almost nothing (load
>>> >> >> > average:
>>> >> >> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one
>>> thread
>>> >> >> > taking
>>> >> >> > most load... all others are just sitting there with 1-5 % CPU
>>> time.
>>> >> >> > This looks to me as FS handles 302 messages in a single
>>> thread?!?!
>>> >> >> >
>>> >> >> >
>>> >> >> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>>> >> >> >
>>> >> >> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average: 1.41,
>>> >> >> > 0.61,
>>> >> >> > 0.60
>>> >> >> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0
>>> zombie
>>> >> >> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,
>>> >> >> > 0.2%si,
>>> >> >> > 0.0%st
>>> >> >> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k
>>> >> >> > buffers
>>> >> >> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k
>>> >> >> > cached
>>> >> >> >
>>> >> >> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
>>> >> >> > COMMAND
>>> >> >> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
>>> >> >> > freeswitch
>>> >> >> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
>>> >> >> > freeswitch
>>> >> >> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
>>> >> >> > freeswitch
>>> >> >> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
>>> >> >> > freeswitch
>>> >> >> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
>>> >> >> > freeswitch
>>> >> >> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
>>> >> >> > freeswitch
>>> >> >> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
>>> >> >> > freeswitch
>>> >> >> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
>>> >> >> > freeswitch
>>> >> >> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
>>> >> >> > freeswitch
>>> >> >> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50
>>> >> >> > freeswitch
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > cat /proc/cpuinfo
>>> >> >> > processor       : 0
>>> >> >> > vendor_id       : GenuineIntel
>>> >> >> > cpu family      : 6
>>> >> >> > model           : 15
>>> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
>>> >> >> > stepping        : 6
>>> >> >> > cpu MHz         : 2333.560
>>> >> >> > cache size      : 4096 KB
>>> >> >> > physical id     : 0
>>> >> >> > siblings        : 2
>>> >> >> > core id         : 0
>>> >> >> > cpu cores       : 2
>>> >> >> > apicid          : 0
>>> >> >> > initial apicid  : 0
>>> >> >> > fdiv_bug        : no
>>> >> >> > hlt_bug         : no
>>> >> >> > f00f_bug        : no
>>> >> >> > coma_bug        : no
>>> >> >> > fpu             : yes
>>> >> >> > fpu_exception   : yes
>>> >> >> > cpuid level     : 10
>>> >> >> > wp              : yes
>>> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep
>>> mtrr
>>> >> >> > pge
>>> >> >> > mca
>>> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>>> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
>>> >> >> > ssse3
>>> >> >> > cx16
>>> >> >> > xtpr dca lahf_lm
>>> >> >> > bogomips        : 4670.78
>>> >> >> > clflush size    : 64
>>> >> >> > power management:
>>> >> >> >
>>> >> >> > processor       : 1
>>> >> >> > vendor_id       : GenuineIntel
>>> >> >> > cpu family      : 6
>>> >> >> > model           : 15
>>> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @ 2.33GHz
>>> >> >> > stepping        : 6
>>> >> >> > cpu MHz         : 2333.560
>>> >> >> > cache size      : 4096 KB
>>> >> >> > physical id     : 0
>>> >> >> > siblings        : 2
>>> >> >> > core id         : 1
>>> >> >> > cpu cores       : 2
>>> >> >> > apicid          : 1
>>> >> >> > initial apicid  : 1
>>> >> >> > fdiv_bug        : no
>>> >> >> > hlt_bug         : no
>>> >> >> > f00f_bug        : no
>>> >> >> > coma_bug        : no
>>> >> >> > fpu             : yes
>>> >> >> > fpu_exception   : yes
>>> >> >> > cpuid level     : 10
>>> >> >> > wp              : yes
>>> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep
>>> mtrr
>>> >> >> > pge
>>> >> >> > mca
>>> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe lm
>>> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est tm2
>>> >> >> > ssse3
>>> >> >> > cx16
>>> >> >> > xtpr dca lahf_lm
>>> >> >> > bogomips        : 4666.82
>>> >> >> > clflush size    : 64
>>> >> >> > power management:
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > uname -a
>>> >> >> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC
>>> 2009
>>> >> >> > i686
>>> >> >> > GNU/Linux
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > Of course, i've tuned the machine up
>>> >> >> >
>>> >> >> > ulimit -c unlimited
>>> >> >> > ulimit -d unlimited
>>> >> >> > ulimit -f unlimited
>>> >> >> > ulimit -i unlimited
>>> >> >> > ulimit -n 999999
>>> >> >> > ulimit -q unlimited
>>> >> >> > ulimit -u unlimited
>>> >> >> > ulimit -v unlimited
>>> >> >> > ulimit -x unlimited
>>> >> >> > ulimit -s 240
>>> >> >> > ulimit -l unlimited
>>> >> >> > ulimit -a
>>> >> >> >
>>> >> >> >
>>> >> >> > Started FS with minimum modules but still 40 CPS seems to be the
>>> >> >> > limit.
>>> >> >> >
>>> >> >> >
>>> >> >> > So, is there any way to improve performance?
>>> >> >> >
>>> >> >> >
>>> >> >> > Tihomir.
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> >
>>> >> >> > _______________________________________________
>>> >> >> > FreeSWITCH-users mailing list
>>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >> >
>>> >> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> > http://www.freeswitch.org
>>> >> >> >
>>> >> >> >
>>> >> >>
>>> >> >> _______________________________________________
>>> >> >> FreeSWITCH-users mailing list
>>> >> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> >>
>>> >> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> >> http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >> > _______________________________________________
>>> >> > FreeSWITCH-users mailing list
>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> > http://www.freeswitch.org
>>> >> >
>>> >> >
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >> UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> > UNSUBSCRIBE:
>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From csa at nowthor.com  Wed Aug 26 10:48:11 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Wed, 26 Aug 2009 13:48:11 -0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <191c3a030908260825n73d6ffc2kbc87296197e72c90@mail.gmail.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>	<4A93EFE1.6010007@nowthor.com>	<191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>	<4A94A2E0.6040306@nowthor.com>
	<191c3a030908260825n73d6ffc2kbc87296197e72c90@mail.gmail.com>
Message-ID: <4A95755B.9020201@nowthor.com>

Anthony Minessale wrote:
> Unfortunately it's still a problem because the core bridge/originate 
> mechanism does not parse XML it's only parses the dial string markup. 

Wouldn't it be possible to "translate" the markup to a dial string prior 
to hitting the core mechanism?

> In the end you must end up with one string of application and one 
> string of data.
>

How about something like

<action application="bridge"><data> ... </data></action>

being somewhat* equivalent to

<action application="bridge" data=" ... ">?

(* somewhat to be properly define later)

> I am very strict about maintaining scope boundaries between abstract 
> concepts.  I think it's the number 1 breakdown that led to asterisk 
> getting so messed up over the years and I want to avoid that same plague.

I totally agree with you on this.

> I hope you understand and if you want to continue to ponder some more 
> functionality I welcome it.

Don't worry, I rarely stop having ideas. Feel free to shoot them down at 
will! :)

Carlos




From msc at freeswitch.org  Wed Aug 26 10:50:31 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Aug 2009 10:50:31 -0700
Subject: [Freeswitch-users] HOW-TO: being on "trunk" of FreePBX,
	starting 	from the ISO
In-Reply-To: <7b197bef0908260608l2c3f454cn8c2b8a12bc3d30ce@mail.gmail.com>
References: <7b197bef0908260608l2c3f454cn8c2b8a12bc3d30ce@mail.gmail.com>
Message-ID: <87f2f3b90908261050i25a6fbeeke30d21145d679616@mail.gmail.com>

On Wed, Aug 26, 2009 at 6:08 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>wrote:

> Instructions for being on "trunk" of FreePBX, starting from the ISO
> (cut and paste to the ssh console after ISO install):
>
> /etc/init.d/httpd stop
> cd /var/www/html
> mv freepbx freepbx-originale
> svn co http://www.freepbx.org/v3/svn/trunk/ freepbx
> chown -R apache.apache freepbx
> ln -s freepbx  freepbx-v3
> cd freepbx-v3/
> ln -s freepbx  freepbx-v3
> /etc/init.d/httpd start
>
> then browse to:
>
> http://192.168.0.12/freepbx-v3/index.php/installer
>
> it will work! :-)


Giovanni, you're such a swell guy! :P
-MC
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From aep.lists at it46.se  Wed Aug 26 11:25:22 2009
From: aep.lists at it46.se (Alberto Escudero-Pascual (lists))
Date: Wed, 26 Aug 2009 20:25:22 +0200
Subject: [Freeswitch-users] Internal Profile sends RTP to external IP
 when using two network interfaces
In-Reply-To: <56E6F9F8-82FB-4A4D-9B7A-44D9BE2BDE43@freeswitch.org>
References: <46b96649f81b0bfcb111928a55450211.squirrel@correo.nodo50.org>
	<3694FC90-3FD1-4F36-8CA9-A3C18EBDEF34@freeswitch.org>
	<7682ba3b2cbb1cf578dcc2226e7d4787.squirrel@correo.nodo50.org>
	<56E6F9F8-82FB-4A4D-9B7A-44D9BE2BDE43@freeswitch.org>
Message-ID: <131017dbd9f5c898b164231856682f14.squirrel@correo.nodo50.org>


Yes, I did fix the internal IP in the internal profile overwritting the
local_ipv4 var in sip-ip and rtp-ip. When the internal clients

>sofia status profile internet


User:       	1000 at 216.82.231.69
Contact:    	"user"
<sip:1000 at 10.0.46.51;user=phone;fs_nat=yes;fs_path=sip%3A1000%4010.0.46.51%3A5060%3Buser%3Dphone>
Agent:      	Grandstream BT110 1.0.8.12
Status:     	Registered(UDP-NAT)(unknown) EXP(2009-08-26 22:22:32)
Host:       	open46
IP:         	10.0.46.51
Instead of

User: 1000 at 10.0.46.51


I can see SIP/SDP messages announcing the external IP and port 5060 in the
INVITE.


It is a very old phone... will try with different gadgets tomorrow.

/aep
-- 
Stopping junk mailers is good for the environment

> Well actually you do have issues you're going to have to specify the
> rtp-ip and sip-ip on BOTH profiles yourself because you're config is
> currently putting your external IP into the internal's settings.
> Please correct that and I'm sure it'll work then.
>
> /b
>
> On Aug 26, 2009, at 9:18 AM, Alberto Escudero-Pascual (lists) wrote:
>
>>
>>
>> ASCII art follows
>>
>> voip-phones..... 10.0.46.1 [ FS ] 216.82.231.69 ----> Internet
>>
>>
>> I have two profiles binded to two different IPs
>>
>> internal is binded to 10.0.46.1 port 5060 /24 network
>> external is binded to 216.82.231.69 port 5080
>>
>> My IP phones are in the 10.0.46.X range and they register @
>> 10.0.46.1 IP
>> address.
>>
>> When I place a call between the two phones in the 10.0.46.0/24 network
>> the phones register but audio RTP is sent to 216.82.231.69 instead of
>> 10.0.46.1 (FS)
>>
>> In this scenario i have FS without any NAT configuration.
>>
>> --
>> The way that i solved was to run both profiles (internal and
>> external) in
>> the public IP (local_ipv4) and force the phones to register with
>> 216.82.231.69 instead of 10.0.46.1
>>
>> Thanks looong time!
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>





From anthony.minessale at gmail.com  Wed Aug 26 11:26:40 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 13:26:40 -0500
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <4A95755B.9020201@nowthor.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
	<4A93EFE1.6010007@nowthor.com>
	<191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>
	<4A94A2E0.6040306@nowthor.com>
	<191c3a030908260825n73d6ffc2kbc87296197e72c90@mail.gmail.com>
	<4A95755B.9020201@nowthor.com>
Message-ID: <191c3a030908261126o1d0e4255k6f31a8de84e8fc44@mail.gmail.com>

did you see my other email on this thread with the xml macro idea?
When it comes to displacing the complexity there are many choices.

if you want to make some api call that can look in xml and generate a big |
sep dial string that's entirely possible.
you could do something similar with lua or some other embedded language as
well.


On Wed, Aug 26, 2009 at 12:48 PM, Carlos S. Antunes <csa at nowthor.com> wrote:

> Anthony Minessale wrote:
> > Unfortunately it's still a problem because the core bridge/originate
> > mechanism does not parse XML it's only parses the dial string markup.
>
> Wouldn't it be possible to "translate" the markup to a dial string prior
> to hitting the core mechanism?
>
> > In the end you must end up with one string of application and one
> > string of data.
> >
>
> How about something like
>
> <action application="bridge"><data> ... </data></action>
>
> being somewhat* equivalent to
>
> <action application="bridge" data=" ... ">?
>
> (* somewhat to be properly define later)
>
> > I am very strict about maintaining scope boundaries between abstract
> > concepts.  I think it's the number 1 breakdown that led to asterisk
> > getting so messed up over the years and I want to avoid that same plague.
>
> I totally agree with you on this.
>
> > I hope you understand and if you want to continue to ponder some more
> > functionality I welcome it.
>
> Don't worry, I rarely stop having ideas. Feel free to shoot them down at
> will! :)
>
> Carlos
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anatoliy at kounitskiy.com  Wed Aug 26 11:27:52 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Wed, 26 Aug 2009 21:27:52 +0300
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
Message-ID: <1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>

After several hours of testing I was able to answer myself the
previous mentioned questions.

It appears that # and the 0 option work _only_ if user C has answered
the call OR voicemail system answers it.

user A ---call---> user B----attended xfer---> user C

At this point I have new question. In example user C does not have a
voicemail and the call timeout is not an option to wait for. How can
user B go back to the user A, who is listening to MOH?
Could someone help me with an advice/tip?

At the moment I have just one idea for accomplishing it:
1) try to use bind_meta_app in the extension with the att_xfer (not
sure if it can be done). To have a key feature that takes the user A
call leg id and bridging it with user B

Thank you in advnace,
Anatoliy Kounitskiy


On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy
Kounitskiy<anatoliy at kounitskiy.com> wrote:
> Hello everybody!
> I have few questions about the att_xfer application. First, what i want
> to accomplish is: user A calls user B, after that user B makes attended
> transfer to user C.
> In the dialplan i have:
>
> <context name="vpbx">
> ?<extension name="local_number">
> ...
> ? ? ?<action application="bind_meta_app" data="1 b s
> execute_extension::dx XML features"/>
> ? ? ?<action application="bind_meta_app" data="2 b s
> record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> ? ? ?<action application="bind_meta_app" data="3 b s
> execute_extension::cf XML features"/>
> ? ? ?<action application="bind_meta_app" data="4 b s
> execute_extension::attented_xfer XML features"/>
> ?....
> ? ?</condition>
> ?</extension>
>
> So when user B answers the call, he sends *4 and the extensions for the
> attended transfer is started - the usual - plays message and read the
> input dtmf:
>
> features.xml
> ...
> ? ?<extension name="attented_xfer">
> ? ? ?<condition field="${toll_allow}" expression="local"/>
> ? ? ?<condition field="destination_number" expression="^attented_xfer$">
> ? ? ? ?<action application="info"/>
> ? ? ? ?<action application="read" data="3 4 ivr/ivr-enter_ext.wav
> attxfer_callthis 30000 #"/>
> ? ? ? ?<action application="set" data="call_timeout=15"/>
> ? ? ? ?<action application="att_xfer"
> data="user/${attxfer_callthis}@${domain_name}"/>
> ? ? ?</condition>
> ? ?</extension>
> ...
>
> To this problems everything is perfect. But here comes the questions, so
> if you can give some tips would be great.
>
> 1) when user B enters the extension number of C - the C's phone starts
> ringing in the tcpdump i can see that the phone is sending 180 ringing,
> BUT user B does not hear the ringing.
> 2) as mentioned in the
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
> quote: "If the other leg is a voicemail or doesn't answered you can
> hangup that leg by pressing dtmf # (fixed in r14438) "
> It doesn't work. The option 0 is working even before C answering the
> phone - after he answers it's a threeway conference :) - i like this
> feature.
>
> I'm using FreeSWITCH Version 1.0.trunk (14633M)
>
> Also I tried to set call timeout to see if I can go back the user A, who
> is listening to MOH - no luck here.
>
> Probably I'm missing something. Tried to look in the source of att_xfer
> to understand why the feature i want is not working - but it seems my
> C/C++ skills are not so good, as i want :( .
>
> Thank you in advance,
> Anatoliy Kounitskiy
>



-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540



From anthony.minessale at gmail.com  Wed Aug 26 11:39:46 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 13:39:46 -0500
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
Message-ID: <191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>

maybe we can make an origination_cancel_key=# you could set on the dial
string to be able to cancel that originate with dtmf


On Wed, Aug 26, 2009 at 1:27 PM, Anatoliy Kounitskiy <
anatoliy at kounitskiy.com> wrote:

> After several hours of testing I was able to answer myself the
> previous mentioned questions.
>
> It appears that # and the 0 option work _only_ if user C has answered
> the call OR voicemail system answers it.
>
> user A ---call---> user B----attended xfer---> user C
>
> At this point I have new question. In example user C does not have a
> voicemail and the call timeout is not an option to wait for. How can
> user B go back to the user A, who is listening to MOH?
> Could someone help me with an advice/tip?
>
> At the moment I have just one idea for accomplishing it:
> 1) try to use bind_meta_app in the extension with the att_xfer (not
> sure if it can be done). To have a key feature that takes the user A
> call leg id and bridging it with user B
>
> Thank you in advnace,
> Anatoliy Kounitskiy
>
>
> On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy
> Kounitskiy<anatoliy at kounitskiy.com> wrote:
> > Hello everybody!
> > I have few questions about the att_xfer application. First, what i want
> > to accomplish is: user A calls user B, after that user B makes attended
> > transfer to user C.
> > In the dialplan i have:
> >
> > <context name="vpbx">
> >  <extension name="local_number">
> > ...
> >      <action application="bind_meta_app" data="1 b s
> > execute_extension::dx XML features"/>
> >      <action application="bind_meta_app" data="2 b s
> >
> record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
> >      <action application="bind_meta_app" data="3 b s
> > execute_extension::cf XML features"/>
> >      <action application="bind_meta_app" data="4 b s
> > execute_extension::attented_xfer XML features"/>
> >  ....
> >    </condition>
> >  </extension>
> >
> > So when user B answers the call, he sends *4 and the extensions for the
> > attended transfer is started - the usual - plays message and read the
> > input dtmf:
> >
> > features.xml
> > ...
> >    <extension name="attented_xfer">
> >      <condition field="${toll_allow}" expression="local"/>
> >      <condition field="destination_number" expression="^attented_xfer$">
> >        <action application="info"/>
> >        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
> > attxfer_callthis 30000 #"/>
> >        <action application="set" data="call_timeout=15"/>
> >        <action application="att_xfer"
> > data="user/${attxfer_callthis}@${domain_name}"/>
> >      </condition>
> >    </extension>
> > ...
> >
> > To this problems everything is perfect. But here comes the questions, so
> > if you can give some tips would be great.
> >
> > 1) when user B enters the extension number of C - the C's phone starts
> > ringing in the tcpdump i can see that the phone is sending 180 ringing,
> > BUT user B does not hear the ringing.
> > 2) as mentioned in the
> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
> > quote: "If the other leg is a voicemail or doesn't answered you can
> > hangup that leg by pressing dtmf # (fixed in r14438) "
> > It doesn't work. The option 0 is working even before C answering the
> > phone - after he answers it's a threeway conference :) - i like this
> > feature.
> >
> > I'm using FreeSWITCH Version 1.0.trunk (14633M)
> >
> > Also I tried to set call timeout to see if I can go back the user A, who
> > is listening to MOH - no luck here.
> >
> > Probably I'm missing something. Tried to look in the source of att_xfer
> > to understand why the feature i want is not working - but it seems my
> > C/C++ skills are not so good, as i want :( .
> >
> > Thank you in advance,
> > Anatoliy Kounitskiy
> >
>
>
>
> --
> Anatoliy Kounitskiy
> -------------------------
> E-mail: anatoliy at kounitskiy.com
> Mobile: +359898913540
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From csa at nowthor.com  Wed Aug 26 11:54:52 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Wed, 26 Aug 2009 14:54:52 -0400
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <191c3a030908261126o1d0e4255k6f31a8de84e8fc44@mail.gmail.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>	<4A93EFE1.6010007@nowthor.com>	<191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>	<4A94A2E0.6040306@nowthor.com>	<191c3a030908260825n73d6ffc2kbc87296197e72c90@mail.gmail.com>	<4A95755B.9020201@nowthor.com>
	<191c3a030908261126o1d0e4255k6f31a8de84e8fc44@mail.gmail.com>
Message-ID: <4A9584FC.9030305@nowthor.com>

Anthony Minessale wrote:
> did you see my other email on this thread with the xml macro idea?
> When it comes to displacing the complexity there are many choices.

Yes, I did see that macro idea. It looks indeed very good in the case of 
a failover scenario. But what about something like I proposed initially, 
that is, a mix of simultaneous and sequential dialing in arbitrary 
order? I guess this is also what Phillip Jones is after with his feature 
request (http://jira.freeswitch.org/browse/FSCORE-422).

In any case, I am going to adopt your solution for the failover scenario 
as the most elegant so far! :)

> if you want to make some api call that can look in xml and generate a 
> big | sep dial string that's entirely possible. you could do something 
> similar with lua or some other embedded language as well.

Yes, that is indeed another possibility. However, I am not sure it would 
allow that mix of simultaneous and sequential dialing, would it?

Carlos



From anthony.minessale at gmail.com  Wed Aug 26 11:56:35 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 13:56:35 -0500
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
Message-ID: <191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>

which revision are you on?
The defaults on the latest code and examples should be configured to
minimize delay.
Some of the older revisions built up some delay issues from udp buffering
when timers were not synced.


On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net> wrote:

>  When running conferences with users dialed in from a PSTN gateway (SIP)
> and directly from remote SIP endpoints there is an ever longer buildup in
> delay, reaching up to multiple seconds. Is there any way to limit the delay
> ?
>
>
>
> I am not 100% sure whether the delays is caused by the SIP jitter buffer of
> freeswitch or directly by the conference module.
>
>
>
> Any advice?
>
>
>
> Thanks
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
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From foxb at abv.bg  Wed Aug 26 12:02:13 2009
From: foxb at abv.bg (Hristo Benev)
Date: Wed, 26 Aug 2009 22:02:13 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <925579051.122056.1251313341898.JavaMail.apache@mail22.abv.bg>

 I think that the problem is here:
-------------------------
2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
--------------------------

Actually Regex FAIL

I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
Here is my  default.xml:
----------------
<?xml version="1.0" encoding="utf-8"?>
<!-- http://wiki.freeswitch.org/wiki/Dialplan_XML -->
<include>
  <context name="default">

     <extension name="LOOKUP_ROUTE">
      <condition field="destination_number" expression="(\d+)$">
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" data="continue_on_fail=true"/>
         <action application="export" data="sip_h_X-ROUTE=LOOKUP"/>
         <action application="bridge" data="sofia/internal/${destination_number}@127.0.0.1:5062"/>
         <action application="set" data="ROUTE_GW=${sip_redirect_contact_user_0}"/>
         <action application="set" data="AREA=${sip_redirect_contact_user_0}"/>
         <action application="transfer" data="${destination_number} XML ROUTING"/>
      </condition>
    </extension>

   </context>

  <context name="ROUTING">

    <extension name="PEER_01">
      <condition field="${sip_h_X-ROUTE}" expression="PEER_01">
        <action application="set" data="hangup_after_bridge=true"/>
        <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,TIMEOUT,NO_ROUTE_DESTINATION"/>
        <action application="set" data="PEER=1.1.1.1"/>
        <action application="bridge" data="sofia/external/${destination_number}@1.1.1.1"/>
        <action application="set" data="PEER=2.2.2.2"/>
        <action application="bridge" data="sofia/external/${destination_number}@2.2.2.2"/>
        <action application="set" data="PEER=3.3.3.3"/>
        <action application="bridge" data="sofia/external/${destination_number}@3.3.3.3"/>
      </condition>
    </extension>

  </context>

</include>

--------------------------



 >-------- ?????????? ????? --------
 >??:  Brian West 
 >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >??: freeswitch-users at lists.freeswitch.org
 >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST

 >We do not blindly follow 302's as that is a dangerous thing to do. 
You have to process all 302's in the dialplan. 
Set this on your sofia profile
You can set these variables sip_redirect_profile,
sip_redirect_context,
sip_redirect_dialplan,
When a redirect happens you get these variables - sip_redirect_contact_%d,
sip_redirected_to,
sip_redirect_contact_user_%d,
sip_redirect_contact_host_%d,
sip_redirect_contact_params_%d,
sip_redirect_dialstring_%d,
sip_redirect_dialstring,
sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
 >



From anthony.minessale at gmail.com  Wed Aug 26 12:11:22 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 14:11:22 -0500
Subject: [Freeswitch-users] how to avoid many "|" in bridge application?
In-Reply-To: <4A9584FC.9030305@nowthor.com>
References: <fbfa7b0c0908250157t35177a51qea5d0870000131ea@mail.gmail.com>
	<4A93EFE1.6010007@nowthor.com>
	<191c3a030908250745x61b5bc8eh43982d6bf4167ad7@mail.gmail.com>
	<4A94A2E0.6040306@nowthor.com>
	<191c3a030908260825n73d6ffc2kbc87296197e72c90@mail.gmail.com>
	<4A95755B.9020201@nowthor.com>
	<191c3a030908261126o1d0e4255k6f31a8de84e8fc44@mail.gmail.com>
	<4A9584FC.9030305@nowthor.com>
Message-ID: <191c3a030908261211r40698642r37d338c08679b03d@mail.gmail.com>

that simo-sequential dialing thing is highly complicated because like
highlander there can be only one in the end.
The way we do it now is all controlled by the same monitor thread and it's a
ton more complexity added to a system that many
now depend on.  The things originate can do easily outdo the leading brand
and i would hate to break anything.

You already hate the syntax now and the only real solution would be to add
even more delimiters to the originate syntax
to denote sequential within simo as sequential currently has the highest
priority.

most chars are used up already maybe ^ is still free (it has to be valid in
an xml attr too, we got rid of + for simo dial because of that)
sofia/foo1^sofia/foo2,sofia/bar1^sofia/bar2

on top of all that it's a lot of work and we are not exactly swimming with
coders blessed to hack the core volunteering their time =D





On Wed, Aug 26, 2009 at 1:54 PM, Carlos S. Antunes <csa at nowthor.com> wrote:

> Anthony Minessale wrote:
> > did you see my other email on this thread with the xml macro idea?
> > When it comes to displacing the complexity there are many choices.
>
> Yes, I did see that macro idea. It looks indeed very good in the case of
> a failover scenario. But what about something like I proposed initially,
> that is, a mix of simultaneous and sequential dialing in arbitrary
> order? I guess this is also what Phillip Jones is after with his feature
> request (http://jira.freeswitch.org/browse/FSCORE-422).
>
> In any case, I am going to adopt your solution for the failover scenario
> as the most elegant so far! :)
>
> > if you want to make some api call that can look in xml and generate a
> > big | sep dial string that's entirely possible. you could do something
> > similar with lua or some other embedded language as well.
>
> Yes, that is indeed another possibility. However, I am not sure it would
> allow that mix of simultaneous and sequential dialing, would it?
>
> Carlos
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From gshfreesw at gmail.com  Wed Aug 26 12:55:38 2009
From: gshfreesw at gmail.com (Shameem Shiek)
Date: Wed, 26 Aug 2009 15:55:38 -0400
Subject: [Freeswitch-users] caller id on origination does not work
In-Reply-To: <7e2ac3270908252007g5dfb9390w806267550726b99a@mail.gmail.com>
References: <5070fcbd0908251951j5a7163b1se346abcdf0664c94@mail.gmail.com>
	<7e2ac3270908252007g5dfb9390w806267550726b99a@mail.gmail.com>
Message-ID: <5070fcbd0908261255o677d1abbseba42569ba9ebaf0@mail.gmail.com>

This is one did the trick.

Thanks.

On Tue, Aug 25, 2009 at 11:07 PM, SP <sprice at gmail.com> wrote:

> You can try playing with this in your gateway profile
>   <param name="caller-id-in-from" value="true"/>
>
> Not sure what it'll do to your registration, give it a try
>
> On Tue, Aug 25, 2009 at 21:51, Shameem Shiek <gshfreesw at gmail.com> wrote:
>
>> I am setting the caller id like this in my ESL script:
>>
>> @con.sendRecv("api originate
>> {origination_caller_id_number=15103245678}sofia/gateway/junctionnetworks/1#{@number}
>> &park()")
>>
>> And the caller id comes out as all zeroes.  The sip trace shows the "from"
>> as shown in the sofia status command.  This is output of sofia gateway
>> status and my gateway is under
>> /usr/local/freeswitch/conf/sip_profiles/external/junctionnetworks.xml .
>> Also, adding the gateway under /conf/directory/default did not work as the
>> wiki suugests. What I need to do for the caller_id to come through?
>>
>>
>> sofia status gateway junctionnetworks
>>
>> =================================================================================================
>> Name        junctionnetworks
>> Scheme      Digest
>> Realm       jnctn.net
>> Username    username
>> Password    yes
>> From        <sip:username at jnctn.net <sip%3Ausername at jnctn.net>
>> ;transport=udp>
>> Contact     <sip:gw+junctionnetworks at 24.87.80.12:5080;transport=udp>
>> Exten       899xxxxxxx
>> To          sip:username at sip.jnctn.net <sip%3Ausername at sip.jnctn.net>
>> Proxy       sip:sip.jnctn.net
>> Context     public
>> Expires     3600
>> Freq        3600
>> Ping        0
>> PingFreq    0
>> State       REGED
>> Status      UP
>> CallsIN     0
>> CallsOUT    1
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Shannon
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Wed Aug 26 12:58:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Aug 2009 12:58:10 -0700
Subject: [Freeswitch-users] Newbie startup help. Tutorial? Learning path?
In-Reply-To: <4A9537DF.8070609@yahoo.com>
References: <4A9537DF.8070609@yahoo.com>
Message-ID: <87f2f3b90908261258r77869c3ara40dbd5a7a4259d6@mail.gmail.com>

On Wed, Aug 26, 2009 at 6:25 AM, Merle J. Ebbert <se02005-mec at yahoo.com>wrote:

>
> Hi,
>
> I'm trying to avoid taking up a lot of peoples valuable time.
>
> SIP & FS have  brought some ideas for some commercial products but I
> need to know where to start.
>
> Having once written a proprietary DOS & helped with writing a RTOS, I
> consider myself capable
> of learning.  I (we) just need to know where to start to come up to
> speed rapidly.
>
> Is there a FreeSWITCH tutorial available?
> Should someone new start with Asterisk and then possibly move to FS?
>
> Thanks,
> Merle
>

BTW, if you're in Europe then you can get a copy of the September issue of
Linux-Pro magazine. (I thinks it's just called Linux Magazine in Europe.) It
has a very gentle introduction to FS. The mag should be available any time
in US and end of August in Australia/NZ.
-MC
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From anthony.minessale at gmail.com  Wed Aug 26 12:59:35 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 14:59:35 -0500
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
Message-ID: <191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>

i added a patch to attempt to do this so try adding

{origination_cancel_key=#} before the dial string
or
<action application="set" data="origination_cancel_key=#"/>
before you bridge.

On Wed, Aug 26, 2009 at 1:39 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> maybe we can make an origination_cancel_key=# you could set on the dial
> string to be able to cancel that originate with dtmf
>
>
>
> On Wed, Aug 26, 2009 at 1:27 PM, Anatoliy Kounitskiy <
> anatoliy at kounitskiy.com> wrote:
>
>> After several hours of testing I was able to answer myself the
>> previous mentioned questions.
>>
>> It appears that # and the 0 option work _only_ if user C has answered
>> the call OR voicemail system answers it.
>>
>> user A ---call---> user B----attended xfer---> user C
>>
>> At this point I have new question. In example user C does not have a
>> voicemail and the call timeout is not an option to wait for. How can
>> user B go back to the user A, who is listening to MOH?
>> Could someone help me with an advice/tip?
>>
>> At the moment I have just one idea for accomplishing it:
>> 1) try to use bind_meta_app in the extension with the att_xfer (not
>> sure if it can be done). To have a key feature that takes the user A
>> call leg id and bridging it with user B
>>
>> Thank you in advnace,
>> Anatoliy Kounitskiy
>>
>>
>> On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy
>> Kounitskiy<anatoliy at kounitskiy.com> wrote:
>> > Hello everybody!
>> > I have few questions about the att_xfer application. First, what i want
>> > to accomplish is: user A calls user B, after that user B makes attended
>> > transfer to user C.
>> > In the dialplan i have:
>> >
>> > <context name="vpbx">
>> >  <extension name="local_number">
>> > ...
>> >      <action application="bind_meta_app" data="1 b s
>> > execute_extension::dx XML features"/>
>> >      <action application="bind_meta_app" data="2 b s
>> >
>> record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>> >      <action application="bind_meta_app" data="3 b s
>> > execute_extension::cf XML features"/>
>> >      <action application="bind_meta_app" data="4 b s
>> > execute_extension::attented_xfer XML features"/>
>> >  ....
>> >    </condition>
>> >  </extension>
>> >
>> > So when user B answers the call, he sends *4 and the extensions for the
>> > attended transfer is started - the usual - plays message and read the
>> > input dtmf:
>> >
>> > features.xml
>> > ...
>> >    <extension name="attented_xfer">
>> >      <condition field="${toll_allow}" expression="local"/>
>> >      <condition field="destination_number" expression="^attented_xfer$">
>> >        <action application="info"/>
>> >        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
>> > attxfer_callthis 30000 #"/>
>> >        <action application="set" data="call_timeout=15"/>
>> >        <action application="att_xfer"
>> > data="user/${attxfer_callthis}@${domain_name}"/>
>> >      </condition>
>> >    </extension>
>> > ...
>> >
>> > To this problems everything is perfect. But here comes the questions, so
>> > if you can give some tips would be great.
>> >
>> > 1) when user B enters the extension number of C - the C's phone starts
>> > ringing in the tcpdump i can see that the phone is sending 180 ringing,
>> > BUT user B does not hear the ringing.
>> > 2) as mentioned in the
>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
>> > quote: "If the other leg is a voicemail or doesn't answered you can
>> > hangup that leg by pressing dtmf # (fixed in r14438) "
>> > It doesn't work. The option 0 is working even before C answering the
>> > phone - after he answers it's a threeway conference :) - i like this
>> > feature.
>> >
>> > I'm using FreeSWITCH Version 1.0.trunk (14633M)
>> >
>> > Also I tried to set call timeout to see if I can go back the user A, who
>> > is listening to MOH - no luck here.
>> >
>> > Probably I'm missing something. Tried to look in the source of att_xfer
>> > to understand why the feature i want is not working - but it seems my
>> > C/C++ skills are not so good, as i want :( .
>> >
>> > Thank you in advance,
>> > Anatoliy Kounitskiy
>> >
>>
>>
>>
>> --
>> Anatoliy Kounitskiy
>> -------------------------
>> E-mail: anatoliy at kounitskiy.com
>> Mobile: +359898913540
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anatoliy at kounitskiy.com  Wed Aug 26 13:10:59 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Wed, 26 Aug 2009 23:10:59 +0300
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
Message-ID: <1cd828b60908261310m3cca0e56u98c8ccc29a54b21c@mail.gmail.com>

Thank you for the fast patch :)

I'm going to test it right away and report you if there are problems or not.


On Wed, Aug 26, 2009 at 10:59 PM, Anthony
Minessale<anthony.minessale at gmail.com> wrote:
> i added a patch to attempt to do this so try adding
>
> {origination_cancel_key=#} before the dial string
> or
> <action application="set" data="origination_cancel_key=#"/>
> before you bridge.
>
> On Wed, Aug 26, 2009 at 1:39 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> maybe we can make an origination_cancel_key=# you could set on the dial
>> string to be able to cancel that originate with dtmf
>>
>>
>> On Wed, Aug 26, 2009 at 1:27 PM, Anatoliy Kounitskiy
>> <anatoliy at kounitskiy.com> wrote:
>>>
>>> After several hours of testing I was able to answer myself the
>>> previous mentioned questions.
>>>
>>> It appears that # and the 0 option work _only_ if user C has answered
>>> the call OR voicemail system answers it.
>>>
>>> user A ---call---> user B----attended xfer---> user C
>>>
>>> At this point I have new question. In example user C does not have a
>>> voicemail and the call timeout is not an option to wait for. How can
>>> user B go back to the user A, who is listening to MOH?
>>> Could someone help me with an advice/tip?
>>>
>>> At the moment I have just one idea for accomplishing it:
>>> 1) try to use bind_meta_app in the extension with the att_xfer (not
>>> sure if it can be done). To have a key feature that takes the user A
>>> call leg id and bridging it with user B
>>>
>>> Thank you in advnace,
>>> Anatoliy Kounitskiy
>>>
>>>
>>> On Wed, Aug 26, 2009 at 5:51 PM, Anatoliy
>>> Kounitskiy<anatoliy at kounitskiy.com> wrote:
>>> > Hello everybody!
>>> > I have few questions about the att_xfer application. First, what i want
>>> > to accomplish is: user A calls user B, after that user B makes attended
>>> > transfer to user C.
>>> > In the dialplan i have:
>>> >
>>> > <context name="vpbx">
>>> > ?<extension name="local_number">
>>> > ...
>>> > ? ? ?<action application="bind_meta_app" data="1 b s
>>> > execute_extension::dx XML features"/>
>>> > ? ? ?<action application="bind_meta_app" data="2 b s
>>> >
>>> > record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
>>> > ? ? ?<action application="bind_meta_app" data="3 b s
>>> > execute_extension::cf XML features"/>
>>> > ? ? ?<action application="bind_meta_app" data="4 b s
>>> > execute_extension::attented_xfer XML features"/>
>>> > ?....
>>> > ? ?</condition>
>>> > ?</extension>
>>> >
>>> > So when user B answers the call, he sends *4 and the extensions for the
>>> > attended transfer is started - the usual - plays message and read the
>>> > input dtmf:
>>> >
>>> > features.xml
>>> > ...
>>> > ? ?<extension name="attented_xfer">
>>> > ? ? ?<condition field="${toll_allow}" expression="local"/>
>>> > ? ? ?<condition field="destination_number"
>>> > expression="^attented_xfer$">
>>> > ? ? ? ?<action application="info"/>
>>> > ? ? ? ?<action application="read" data="3 4 ivr/ivr-enter_ext.wav
>>> > attxfer_callthis 30000 #"/>
>>> > ? ? ? ?<action application="set" data="call_timeout=15"/>
>>> > ? ? ? ?<action application="att_xfer"
>>> > data="user/${attxfer_callthis}@${domain_name}"/>
>>> > ? ? ?</condition>
>>> > ? ?</extension>
>>> > ...
>>> >
>>> > To this problems everything is perfect. But here comes the questions,
>>> > so
>>> > if you can give some tips would be great.
>>> >
>>> > 1) when user B enters the extension number of C - the C's phone starts
>>> > ringing in the tcpdump i can see that the phone is sending 180 ringing,
>>> > BUT user B does not hear the ringing.
>>> > 2) as mentioned in the
>>> > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer
>>> > quote: "If the other leg is a voicemail or doesn't answered you can
>>> > hangup that leg by pressing dtmf # (fixed in r14438) "
>>> > It doesn't work. The option 0 is working even before C answering the
>>> > phone - after he answers it's a threeway conference :) - i like this
>>> > feature.
>>> >
>>> > I'm using FreeSWITCH Version 1.0.trunk (14633M)
>>> >
>>> > Also I tried to set call timeout to see if I can go back the user A,
>>> > who
>>> > is listening to MOH - no luck here.
>>> >
>>> > Probably I'm missing something. Tried to look in the source of att_xfer
>>> > to understand why the feature i want is not working - but it seems my
>>> > C/C++ skills are not so good, as i want :( .
>>> >
>>> > Thank you in advance,
>>> > Anatoliy Kounitskiy
>>> >
>>>
>>>
>>>
>>> --
>>> Anatoliy Kounitskiy
>>> -------------------------
>>> E-mail: anatoliy at kounitskiy.com
>>> Mobile: +359898913540
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org
>> pstn:213-799-1400
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540



From msc at freeswitch.org  Wed Aug 26 13:11:14 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Aug 2009 13:11:14 -0700
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
Message-ID: <87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>

On Wed, Aug 26, 2009 at 12:59 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> i added a patch to attempt to do this so try adding
>

Nice work!

>
> {origination_cancel_key=#} before the dial string
> or
> <action application="set" data="origination_cancel_key=#"/>
> before you bridge.
>

Anatoliy,

Please try this and let us know if it works for you as expected. If so,
please write up the exact procedure that you used and include dialplan
examples. I'd like to get this on the wiki while it's fresh in my mind. If
you feel comfortable editing the wiki on your own then just let me know so
that we can coordinate efforts.

Thanks,
MC
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From msc at freeswitch.org  Wed Aug 26 13:28:17 2009
From: msc at freeswitch.org (Michael Collins)
Date: Wed, 26 Aug 2009 13:28:17 -0700
Subject: [Freeswitch-users] Timers/DTMFs During a Call
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>
Message-ID: <87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com>

Do you actually need Python for the IVR, or is it that you're comfortable
using a scripting lang for an IVR? I like using XML for IVRs, but using
scripting langs does give you a bit more power & flexibility at the cost of
some resources.

For the record, you can do this in the dialplan using XML and sched_api
without touching a scripting language. Checkout the sched_api channel
variable on the wiki - it may give you the functionality you need.

-MC

On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM <delianspam at gmail.com> wrote:

>  Hello Everybody!
>
>
>
> 1.       Scenario.
>
> I am writing an IVR in Python that gets a destination from the calling
> party (party A) and then connects to the destination (party B).
>
> When the call is CONNECTED, I want to:
>
> -          Receive DTMFs
>
> -          Have a timer that can call a certain function in my script. The
> script will have to play a message to party A.
>
> -          Have a timer that can call a certain function in my script. The
> script will have to drop the call.
>
> Please notice that I want to do the things after the two parties are
> connected, and not after I send the Invite to party B.
>
>
>
> 2.       Problem.
>
> I will be happy to receive help on:
>
> -          Which methods should I look for to implement this.
>
>
>
> 3.       Details
>
> Here is how I connect the call currently:
>
> session.execute("bridge",?sofia/internal/" + destination_number + "@
> domain.com?)
>
>
>
> I have tried to create a timer callback function ?my_method()? using:
>
> ivr_timer =threading.Timer(30,my_method)
>
> This never called the function ?my_method()?.
>
>
>
> Maybe I am wrong in using threading.Timer and the ?bridge? application?
> Maybe I need to create a new thread and a new timer using the API of
> freeswitch, plus to use the ?session.setInputCallback?, plus use a
> conference rather than a bridge? Can you please provide any suggestions or
> examples?
>
>
>
> Thank you!
>
> Best Regards, Delian Tashev
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From pbd at suspiria.net  Wed Aug 26 13:30:07 2009
From: pbd at suspiria.net (Public Dump)
Date: Wed, 26 Aug 2009 22:30:07 +0200
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
Message-ID: <13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>

Hi,



I am on a quite recent version (i assume):



FreeSWITCH Version 1.0.trunk (14461)



Should the bug be fixed in this revision ? What config settings would a have to check to limit delay (even at the cost of reduction in quality).



Thanks



Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 20:57
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] delay buildup in conference



which revision are you on?
The defaults on the latest code and examples should be configured to minimize delay.
Some of the older revisions built up some delay issues from udp buffering when timers were not synced.



On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net<mailto:pbd at suspiria.net>> wrote:

When running conferences with users dialed in from a PSTN gateway (SIP) and directly from remote SIP endpoints there is an ever longer buildup in delay, reaching up to multiple seconds. Is there any way to limit the delay ?



I am not 100% sure whether the delays is caused by the SIP jitter buffer of freeswitch or directly by the conference module.



Any advice?



Thanks


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400

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From anthony.minessale at gmail.com  Wed Aug 26 13:45:29 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 15:45:29 -0500
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
Message-ID: <191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>

There is no bug, it's all dependent on your network conditions.
I spend literally 8-12 hours a day on a conference and there is no delay.

The important param is
<param name="rtp-autoflush-during-bridge" value="true"/>

in the sofia profile in question.

if you have delay with that in place, then it's probably not FS


On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd at suspiria.net> wrote:

>  Hi,
>
>
>
> I am on a quite recent version (i assume):
>
>
>
> FreeSWITCH Version 1.0.trunk (14461)
>
>
>
> Should the bug be fixed in this revision ? What config settings would a
> have to check to limit delay (even at the cost of reduction in quality).
>
>
>
> Thanks
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Anthony
> Minessale
> *Gesendet:* Mittwoch, 26. August 2009 20:57
> *An:* freeswitch-users at lists.freeswitch.org
> *Betreff:* Re: [Freeswitch-users] delay buildup in conference
>
>
>
> which revision are you on?
> The defaults on the latest code and examples should be configured to
> minimize delay.
> Some of the older revisions built up some delay issues from udp buffering
> when timers were not synced.
>
>  On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net> wrote:
>
> When running conferences with users dialed in from a PSTN gateway (SIP) and
> directly from remote SIP endpoints there is an ever longer buildup in delay,
> reaching up to multiple seconds. Is there any way to limit the delay ?
>
>
>
> I am not 100% sure whether the delays is caused by the SIP jitter buffer of
> freeswitch or directly by the conference module.
>
>
>
> Any advice?
>
>
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From harry at vangberg.name  Wed Aug 26 13:59:02 2009
From: harry at vangberg.name (Harry Vangberg)
Date: Wed, 26 Aug 2009 22:59:02 +0200
Subject: [Freeswitch-users] bind_meta_app: "already
	broadcasting..broadcast aborted" if bound via event socket
In-Reply-To: <74d41a3d0908260654x52bf9d94g124ceaa27f6df6bb@mail.gmail.com>
References: <74d41a3d0908260654x52bf9d94g124ceaa27f6df6bb@mail.gmail.com>
Message-ID: <74d41a3d0908261359sa78607bx20852471dfc3aa49@mail.gmail.com>

For your information, revision 14644 committed by anthm fixes this.
Again, thanks!

2009/8/26 Harry Vangberg <harry at vangberg.name>:
> Hello.
>
> After pestering the IRC channel for a few days with useless
> information, I think I've finally made a breakthrough, but I have no
> idea why it works this way.
>
> Basically, what I want to achieve is: A calls in and is bridged to B
> (1234). If B presses *1, he should be taken out, and A be bridged to C
> (8888) instead. Pretty simple.
>
> With the following dialplan it works very nicely:
>
> ? ?<extension name="ivr">
> ? ? ?<condition field="destination_number" expression="^(\d+)$">
> ? ? ? ?<action application="set" data="bypass_media=false"/>
> ? ? ? ?<action application="answer"/>
> ? ? ? ?<action application="bind_meta_app" data="1 b a
> bridge::sofia/gateway/secretgw/8888"/>
> ? ? ? ?<action application="bridge" data="sofia/gateway/secretgw/1234" />
> ? ? ?</condition>
> ? ?</extension>
>
> On the other hand, if I try to replicate the above via an outbound
> event socket it fails with a: "already broadcasting...broadcast
> aborted". In this case my dialplan XML looks like this:
>
> ? ?<extension name="ff-ivr">
> ? ? ?<condition field="destination_number" expression="^(.*)$">
> ? ? ? ?<action application="set" data="bypass_media=false"/>
> ? ? ? ?<action application="socket" data="127.0.0.1:8084 full"/>
> ? ? ?</condition>
> ? ?</extension>
>
> And this is the event socket communication (in reality I use a Ruby
> framework, do a bunch of database calls etc., but for the sake of
> clarity I'm mimicking it with a netcat session):
>
> --------------------------------
> $ nc -v -l 127.0.0.1 8084
> connect
>
> Event-Name: CHANNEL_DATA
> Core-UUID: ba78636a-9241-11de-ac94-b736da546252
> FreeSWITCH-Hostname: ip-10-226-231-225
> FreeSWITCH-IPv4: 10.226.231.225
> FreeSWITCH-IPv6: %3A%3A1
> Event-Date-Local: 2009-08-26%2013%3A43%3A20
> Event-Date-GMT: Wed,%2026%20Aug%202009%2013%3A43%3A20%20GMT
> Event-Date-Timestamp: 1251294200244673
> Event-Calling-File: mod_event_socket.c
> Event-Calling-Function: parse_command
> Event-Calling-Line-Number: 1482
> Channel-Username: hemmeligt
> Channel-Dialplan: XML
> Channel-Caller-ID-Name: hemmeligt
> Channel-Caller-ID-Number: hemmeligt
> Channel-Network-Addr: 129.142.224.250
> Channel-Destination-Number: 77344541
> Channel-Unique-ID: 681102bc-9246-11de-ac94-b736da546252
> Channel-Source: mod_sofia
> Channel-Context: public
> Channel-Channel-Name: sofia/external/hemmeligt%40129.142.224.250
> Channel-Profile-Index: 1
> Channel-Profile-Created-Time: 1251294195778388
> Channel-Channel-Created-Time: 1251294195778388
> Channel-Channel-Answered-Time: 0
> Channel-Channel-Progress-Time: 0
> Channel-Channel-Progress-Media-Time: 0
> Channel-Channel-Hangup-Time: 0
> Channel-Channel-Transfer-Time: 0
> Channel-Screen-Bit: false
> Channel-Privacy-Hide-Name: true
> Channel-Privacy-Hide-Number: true
> Channel-State: CS_EXECUTE
> Channel-State-Number: 4
> Channel-Name: sofia/external/hemmeligt%40129.142.224.250
> Unique-ID: 681102bc-9246-11de-ac94-b736da546252
> Call-Direction: inbound
> Presence-Call-Direction: inbound
> Answer-State: ringing
> Channel-Read-Codec-Name: PCMA
> Channel-Read-Codec-Rate: 8000
> Channel-Write-Codec-Name: PCMA
> Channel-Write-Codec-Rate: 8000
> Caller-Username: hemmeligt
> Caller-Dialplan: XML
> Caller-Caller-ID-Name: hemmeligt
> Caller-Caller-ID-Number: hemmeligt
> Caller-Network-Addr: 129.142.224.250
> Caller-Destination-Number: 77344541
> Caller-Unique-ID: 681102bc-9246-11de-ac94-b736da546252
> Caller-Source: mod_sofia
> Caller-Context: public
> Caller-Channel-Name: sofia/external/hemmeligt%40129.142.224.250
> Caller-Profile-Index: 1
> Caller-Profile-Created-Time: 1251294195778388
> Caller-Channel-Created-Time: 1251294195778388
> Caller-Channel-Answered-Time: 0
> Caller-Channel-Progress-Time: 0
> Caller-Channel-Progress-Media-Time: 0
> Caller-Channel-Hangup-Time: 0
> Caller-Channel-Transfer-Time: 0
> Caller-Screen-Bit: false
> Caller-Privacy-Hide-Name: true
> Caller-Privacy-Hide-Number: true
> variable_sip_received_ip: 129.142.224.250
> variable_sip_received_port: 5060
> variable_sip_via_protocol: udp
> variable_sip_from_user: hemmeligt
> variable_sip_from_uri: hemmeligt%40129.142.224.250
> variable_sip_from_host: 129.142.224.250
> variable_sip_from_user_stripped: hemmeligt
> variable_sip_from_tag: as19641342
> variable_sofia_profile_name: external
> variable_sip_Remote-Party-ID:
> %22hemmeligt%22%20%3Csip%3Ahemmeligt%40129.142.224.250%3E%3Bprivacy%3Dfull%3Bscreen%3Dno
> variable_sip_cid_type: rpid
> variable_sip_req_user: 77344541
> variable_sip_req_port: 5080
> variable_sip_req_uri: 77344541%4079.125.42.248%3A5080
> variable_sip_req_host: 79.125.42.248
> variable_sip_to_user: 77344541
> variable_sip_to_port: 5080
> variable_sip_to_uri: 77344541%4079.125.42.248%3A5080
> variable_sip_to_host: 79.125.42.248
> variable_sip_contact_user: hemmeligt
> variable_sip_contact_uri: hemmeligt%40129.142.224.250
> variable_sip_contact_host: 129.142.224.250
> variable_channel_name: sofia/external/hemmeligt%40129.142.224.250
> variable_sip_call_id: 1aa24dab25cf03f07933f7136822ceff%40129.142.224.250
> variable_sip_user_agent: Condor_gw1
> variable_sip_via_host: 129.142.224.250
> variable_sip_via_port: 5060
> variable_sip_via_rport: 5060
> variable_max_forwards: 70
> variable_switch_r_sdp:
> v%3D0%0D%0Ao%3Droot%2018982%2018982%20IN%20IP4%20129.142.224.250%0D%0As%3Dsession%0D%0Ac%3DIN%20IP4%20129.142.224.250%0D%0At%3D0%200%0D%0Am%3Daudio%2015430%20RTP/AVP%208%200%203%2097%20101%0D%0Aa%3Drtpmap%3A8%20PCMA/8000%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A3%20GSM/8000%0D%0Aa%3Drtpmap%3A97%20iLBC/8000%0D%0Aa%3Dfmtp%3A97%20mode%3D30%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3DsilenceSupp%3Aoff%20-%20-%20-%20-%0D%0Aa%3Dptime%3A20%0D%0A
> variable_remote_media_ip: 129.142.224.250
> variable_remote_media_port: 15430
> variable_read_codec: PCMA
> variable_read_rate: 8000
> variable_write_codec: PCMA
> variable_write_rate: 8000
> variable_endpoint_disposition: RECEIVED
> variable_bypass_media: false
> variable_current_application_data: 127.0.0.1%3A8084%20full
> variable_current_application: socket
> variable_socket_host: 127.0.0.1
> Content-Type: command/reply
> Reply-Text: %2BOK%0A
> Socket-Mode: static
> Control: full
>
> sendmsg
> call-command: execute
> execute-app-name: answer
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> sendmsg
> call-command: execute
> execute-app-name: bind_meta_app
> execute-app-arg: 1 b a bridge::sofia/gateway/secretgw/8888
>
> Content-Type: command/reply
> Reply-Text: +OK
>
> sendmsg
> call-command: execute
> execute-app-name: bridge
> execute-app-arg: sofia/gateway/secretgw/1234
>
> Content-Type: command/reply
> Reply-Text: +OK
> --------------------------------
>
> And on the FreeSWITCH side:
>
> 2009-08-26 13:43:15.778388 [NOTICE] switch_channel.c:602 New Channel
> sofia/external/hemmeligt at 129.142.224.250
> [681102bc-9246-11de-ac94-b736da546252]
> 2009-08-26 13:43:15.778388 [DEBUG] sofia.c:3302 Channel
> sofia/external/hemmeligt at 129.142.224.250 entering state
> [received][100]
> 2009-08-26 13:43:15.778388 [DEBUG] sofia.c:3309 Remote SDP:
> v=0
> o=root 18982 18982 IN IP4 129.142.224.250
> s=session
> c=IN IP4 129.142.224.250
> t=0 0
> m=audio 15430 RTP/AVP 8 0 3 97 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=30
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2009-08-26 13:43:15.778388 [DEBUG] sofia_glue.c:3132 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
> 2009-08-26 13:43:15.778388 [DEBUG] sofia_glue.c:2090 Set Codec
> sofia/external/hemmeligt at 129.142.224.250 PCMA/8000 20 ms 160 samples
> 2009-08-26 13:43:15.778388 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
> payload to 101
> 2009-08-26 13:43:15.778388 [DEBUG] sofia.c:3468
> (sofia/external/hemmeligt at 129.142.224.250) State Change CS_NEW ->
> CS_INIT
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/hemmeligt at 129.142.224.250) Running State Change
> CS_INIT
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:481
> (sofia/external/hemmeligt at 129.142.224.250) State INIT
> 2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:83
> sofia/external/hemmeligt at 129.142.224.250 SOFIA INIT
> 2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:111
> (sofia/external/hemmeligt at 129.142.224.250) State Change CS_INIT ->
> CS_ROUTING
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:481
> (sofia/external/hemmeligt at 129.142.224.250) State INIT going to sleep
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/hemmeligt at 129.142.224.250) Running State Change
> CS_ROUTING
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:484
> (sofia/external/hemmeligt at 129.142.224.250) State ROUTING
> 2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:130
> sofia/external/hemmeligt at 129.142.224.250 SOFIA ROUTING
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:78
> sofia/external/hemmeligt at 129.142.224.250 Standard ROUTING
> 2009-08-26 13:43:15.778388 [INFO] mod_dialplan_xml.c:315 Processing
> hemmeligt->77344541 in context public
> Dialplan: sofia/external/hemmeligt at 129.142.224.250 parsing
> [public->ff-ivr] continue=false
> Dialplan: sofia/external/hemmeligt at 129.142.224.250 Regex (PASS)
> [ff-ivr] destination_number(77344541) =~ /^(.*)$/ break=on-false
> Dialplan: sofia/external/hemmeligt at 129.142.224.250 Action
> set(bypass_media=false)
> Dialplan: sofia/external/hemmeligt at 129.142.224.250 Action
> socket(127.0.0.1:8084 full)
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:114
> (sofia/external/hemmeligt at 129.142.224.250) State Change CS_ROUTING ->
> CS_EXECUTE
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:484
> (sofia/external/hemmeligt at 129.142.224.250) State ROUTING going to
> sleep
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/hemmeligt at 129.142.224.250) Running State Change
> CS_EXECUTE
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:491
> (sofia/external/hemmeligt at 129.142.224.250) State EXECUTE
> 2009-08-26 13:43:15.778388 [DEBUG] mod_sofia.c:173
> sofia/external/hemmeligt at 129.142.224.250 SOFIA EXECUTE
> 2009-08-26 13:43:15.778388 [DEBUG] switch_core_state_machine.c:151
> sofia/external/hemmeligt at 129.142.224.250 Standard EXECUTE
> EXECUTE sofia/external/hemmeligt at 129.142.224.250 set(bypass_media=false)
> 2009-08-26 13:43:15.778388 [DEBUG] mod_dptools.c:748
> sofia/external/hemmeligt at 129.142.224.250 SET [bypass_media]=[false]
> EXECUTE sofia/external/hemmeligt at 129.142.224.250 socket(127.0.0.1:8084 full)
> 2009-08-26 13:43:21.701447 [DEBUG] switch_ivr.c:540
> sofia/external/hemmeligt at 129.142.224.250 Command Execute answer()
> EXECUTE sofia/external/hemmeligt at 129.142.224.250 answer()
> 2009-08-26 13:43:21.701447 [DEBUG] mod_dptools.c:649
> sofia/external/hemmeligt at 129.142.224.250 receive message [ANSWER]
> 2009-08-26 13:43:21.701447 [DEBUG] sofia_glue.c:2324 AUDIO RTP
> [sofia/external/hemmeligt at 129.142.224.250] 10.226.231.225 port 19016
> -> 129.142.224.250 port 15430 codec: 8 ms: 20
> 2009-08-26 13:43:21.701447 [DEBUG] switch_rtp.c:1138 Starting timer
> [soft] 160 bytes per 20ms
> 2009-08-26 13:43:21.702447 [DEBUG] mod_sofia.c:536 Local SDP
> sofia/external/hemmeligt at 129.142.224.250:
> v=0
> o=FreeSWITCH 1251275185 1251275186 IN IP4 79.125.42.248
> s=FreeSWITCH
> c=IN IP4 79.125.42.248
> t=0 0
> m=audio 19016 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> 2009-08-26 13:43:21.702447 [DEBUG] switch_core_session.c:630 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:21.702447 [NOTICE] mod_dptools.c:649 Channel
> [sofia/external/hemmeligt at 129.142.224.250] has been answered
> 2009-08-26 13:43:21.702447 [DEBUG] switch_channel.c:182
> sofia/external/hemmeligt at 129.142.224.250 receive message [AUDIO_SYNC]
> 2009-08-26 13:43:21.702447 [DEBUG] sofia.c:3302 Channel
> sofia/external/hemmeligt at 129.142.224.250 entering state
> [completed][200]
> 2009-08-26 13:43:21.782616 [DEBUG] sofia.c:3302 Channel
> sofia/external/hemmeligt at 129.142.224.250 entering state [ready][200]
> 2009-08-26 13:43:26.388737 [DEBUG] switch_ivr.c:540
> sofia/external/hemmeligt at 129.142.224.250 Command Execute
> bind_meta_app(1 b a bridge::sofia/gateway/secretgw/8888)
> EXECUTE sofia/external/hemmeligt at 129.142.224.250 bind_meta_app(1 b a
> bridge::sofia/gateway/secretgw/8888)
> 2009-08-26 13:43:26.388737 [INFO] switch_ivr_async.c:1795 Bound B-Leg:
> 1 bridge::sofia/gateway/secretgw/8888
> 2009-08-26 13:43:29.684236 [DEBUG] switch_ivr.c:540
> sofia/external/hemmeligt at 129.142.224.250 Command Execute
> bridge(sofia/gateway/secretgw/1234)
> EXECUTE sofia/external/hemmeligt at 129.142.224.250
> bridge(sofia/gateway/secretgw/1234)
> 2009-08-26 13:43:29.684236 [NOTICE] switch_channel.c:602 New Channel
> sofia/external/1234 [705ad5d8-9246-11de-ac94-b736da546252]
> 2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:2809
> (sofia/external/1234) State Change CS_NEW -> CS_INIT
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/1234) Running State Change CS_INIT
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:481
> (sofia/external/1234) State INIT
> 2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:83 sofia/external/1234 SOFIA INIT
> 2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:111
> (sofia/external/1234) State Change CS_INIT -> CS_ROUTING
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:29.684236 [DEBUG] sofia.c:3302 Channel
> sofia/external/1234 entering state [calling][0]
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:481
> (sofia/external/1234) State INIT going to sleep
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/1234) Running State Change CS_ROUTING
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:484
> (sofia/external/1234) State ROUTING
> 2009-08-26 13:43:29.684236 [DEBUG] mod_sofia.c:130 sofia/external/1234
> SOFIA ROUTING
> 2009-08-26 13:43:29.684236 [DEBUG] switch_ivr_originate.c:63
> (sofia/external/1234) State Change CS_ROUTING -> CS_CONSUME_MEDIA
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:484
> (sofia/external/1234) State ROUTING going to sleep
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/1234) Running State Change CS_CONSUME_MEDIA
> 2009-08-26 13:43:29.684236 [DEBUG] switch_core_state_machine.c:503
> (sofia/external/1234) State CONSUME_MEDIA
> 2009-08-26 13:43:29.731221 [DEBUG] sofia.c:3302 Channel
> sofia/external/1234 entering state [calling][0]
> 2009-08-26 13:43:30.660221 [DEBUG] sofia.c:3302 Channel
> sofia/external/1234 entering state [proceeding][183]
> 2009-08-26 13:43:30.660221 [DEBUG] sofia.c:3309 Remote SDP:
> v=0
> o=root 18982 18982 IN IP4 129.142.224.250
> s=session
> c=IN IP4 129.142.224.250
> t=0 0
> m=audio 12480 RTP/AVP 8 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
>
> 2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:3132 Audio Codec
> Compare [PCMA:8:8000:20]/[PCMA:8:8000:20]
> 2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:2090 Set Codec
> sofia/external/1234 PCMA/8000 20 ms 160 samples
> 2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:3092 Set 2833 dtmf
> payload to 101
> 2009-08-26 13:43:30.660221 [DEBUG] sofia_glue.c:2324 AUDIO RTP
> [sofia/external/1234] 10.226.231.225 port 25540 -> 129.142.224.250
> port 12480 codec: 8 ms: 20
> 2009-08-26 13:43:30.660221 [DEBUG] switch_rtp.c:1138 Starting timer
> [soft] 160 bytes per 20ms
> 2009-08-26 13:43:30.661084 [NOTICE] sofia_glue.c:2759 Pre-Answer
> sofia/external/1234!
> 2009-08-26 13:43:30.661084 [DEBUG] switch_channel.c:1778 Send signal
> sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_originate.c:2061
> Originate Resulted in Success: [sofia/external/1234]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_channel.c:182
> sofia/external/1234 receive message [AUDIO_SYNC]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_channel.c:182
> sofia/external/hemmeligt at 129.142.224.250 receive message [AUDIO_SYNC]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_bridge.c:889
> sofia/external/1234 receive message [BRIDGE]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_core_session.c:630 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_bridge.c:896
> sofia/external/hemmeligt at 129.142.224.250 receive message [BRIDGE]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_core_session.c:630 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:30.665170 [DEBUG] switch_ivr_bridge.c:940
> (sofia/external/1234) State Change CS_CONSUME_MEDIA ->
> CS_EXCHANGE_MEDIA
> 2009-08-26 13:43:30.665170 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:30.705074 [DEBUG] switch_core_state_machine.c:503
> (sofia/external/1234) State CONSUME_MEDIA going to sleep
> 2009-08-26 13:43:30.705074 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/1234) Running State Change CS_EXCHANGE_MEDIA
> 2009-08-26 13:43:30.705074 [DEBUG] switch_core_state_machine.c:494
> (sofia/external/1234) State EXCHANGE_MEDIA
> 2009-08-26 13:43:30.705074 [DEBUG] mod_sofia.c:430 SOFIA LOOPBACK
> 2009-08-26 13:43:36.642173 [DEBUG] sofia.c:3302 Channel
> sofia/external/1234 entering state [ready][200]
> 2009-08-26 13:43:36.642173 [DEBUG] switch_channel.c:1891 Send signal
> sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:36.642173 [NOTICE] sofia.c:3752 Channel
> [sofia/external/1234] has been answered
> 2009-08-26 13:43:36.642173 [DEBUG] switch_channel.c:182
> sofia/external/1234 receive message [AUDIO_SYNC]
> 2009-08-26 13:43:40.309616 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF *:2000
> 2009-08-26 13:43:40.970513 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF 1:2000
> 2009-08-26 13:43:40.970513 [DEBUG] switch_ivr_async.c:1711
> sofia/external/hemmeligt at 129.142.224.250 Processing meta digit '1'
> [bridge::sofia/gateway/secretgw/8888]
> 2009-08-26 13:43:40.970513 [WARNING] switch_ivr_async.c:2310 Channel
> [sofia/external/hemmeligt at 129.142.224.250][bridge::sofia/gateway/secretgw/8888]
> already broadcasting...broadcast aborted
> 2009-08-26 13:43:46.272706 [NOTICE] sofia.c:327 Hangup
> sofia/external/1234 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING]
> 2009-08-26 13:43:46.272706 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/1234 [KILL]
> 2009-08-26 13:43:46.272706 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:371
> sofia/external/1234 ending bridge by request from write function
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:426
> sofia/external/hemmeligt at 129.142.224.250 receive message [UNBRIDGE]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_session.c:630 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:452 BRIDGE
> THREAD DONE [sofia/external/hemmeligt at 129.142.224.250]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:454 Send signal
> sofia/external/1234 [BREAK]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:377
> sofia/external/1234 ending bridge by request from read function
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:452 BRIDGE
> THREAD DONE [sofia/external/1234]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_ivr_bridge.c:454 Send signal
> sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:494
> (sofia/external/1234) State EXCHANGE_MEDIA going to sleep
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/1234) Running State Change CS_HANGUP
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:434
> (sofia/external/1234) State HANGUP
> 2009-08-26 13:43:46.284704 [DEBUG] mod_sofia.c:338 Channel
> sofia/external/1234 hanging up, cause: NORMAL_CLEARING
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:46
> sofia/external/1234 Standard HANGUP, cause: NORMAL_CLEARING
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:434
> (sofia/external/1234) State HANGUP going to sleep
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:476
> (sofia/external/1234) State Change CS_HANGUP -> CS_REPORTING
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/1234 [BREAK]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/1234) Running State Change CS_REPORTING
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:612
> (sofia/external/1234) State REPORTING
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:53
> sofia/external/1234 Standard REPORTING, cause: NORMAL_CLEARING
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:612
> (sofia/external/1234) State REPORTING going to sleep
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:411
> (sofia/external/1234) State Change CS_REPORTING -> CS_DESTROY
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_session.c:1068 Session
> 5 (sofia/external/1234) Locked, Waiting on external entities
> 2009-08-26 13:43:46.284704 [NOTICE] switch_core_session.c:1086 Session
> 5 (sofia/external/1234) Ended
> 2009-08-26 13:43:46.284704 [NOTICE] switch_core_session.c:1088 Close
> Channel sofia/external/1234 [CS_DESTROY]
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:564
> (sofia/external/1234) State DESTROY
> 2009-08-26 13:43:46.284704 [DEBUG] mod_sofia.c:255 sofia/external/1234
> SOFIA DESTROY
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:60
> sofia/external/1234 Standard DESTROY
> 2009-08-26 13:43:46.284704 [DEBUG] switch_core_state_machine.c:564
> (sofia/external/1234) State DESTROY going to sleep
> 2009-08-26 13:44:46.746524 [NOTICE] sofia.c:327 Hangup
> sofia/external/hemmeligt at 129.142.224.250 [CS_EXECUTE]
> [NORMAL_CLEARING]
> 2009-08-26 13:44:46.746524 [DEBUG] switch_channel.c:1683 Send signal
> sofia/external/hemmeligt at 129.142.224.250 [KILL]
> 2009-08-26 13:44:46.746524 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:491
> (sofia/external/hemmeligt at 129.142.224.250) State EXECUTE going to
> sleep
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/hemmeligt at 129.142.224.250) Running State Change
> CS_HANGUP
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:434
> (sofia/external/hemmeligt at 129.142.224.250) State HANGUP
> 2009-08-26 13:44:46.747796 [DEBUG] mod_sofia.c:338 Channel
> sofia/external/hemmeligt at 129.142.224.250 hanging up, cause:
> NORMAL_CLEARING
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:46
> sofia/external/hemmeligt at 129.142.224.250 Standard HANGUP, cause:
> NORMAL_CLEARING
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:434
> (sofia/external/hemmeligt at 129.142.224.250) State HANGUP going to sleep
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:476
> (sofia/external/hemmeligt at 129.142.224.250) State Change CS_HANGUP ->
> CS_REPORTING
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_session.c:932 Send
> signal sofia/external/hemmeligt at 129.142.224.250 [BREAK]
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:398
> (sofia/external/hemmeligt at 129.142.224.250) Running State Change
> CS_REPORTING
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:612
> (sofia/external/hemmeligt at 129.142.224.250) State REPORTING
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:53
> sofia/external/hemmeligt at 129.142.224.250 Standard REPORTING, cause:
> NORMAL_CLEARING
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:612
> (sofia/external/hemmeligt at 129.142.224.250) State REPORTING going to
> sleep
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:411
> (sofia/external/hemmeligt at 129.142.224.250) State Change CS_REPORTING
> -> CS_DESTROY
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_session.c:1068 Session
> 4 (sofia/external/hemmeligt at 129.142.224.250) Locked, Waiting on
> external entities
> 2009-08-26 13:44:46.747796 [NOTICE] switch_core_session.c:1086 Session
> 4 (sofia/external/hemmeligt at 129.142.224.250) Ended
> 2009-08-26 13:44:46.747796 [NOTICE] switch_core_session.c:1088 Close
> Channel sofia/external/hemmeligt at 129.142.224.250 [CS_DESTROY]
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:564
> (sofia/external/hemmeligt at 129.142.224.250) State DESTROY
> 2009-08-26 13:44:46.747796 [DEBUG] mod_sofia.c:255
> sofia/external/hemmeligt at 129.142.224.250 SOFIA DESTROY
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:60
> sofia/external/hemmeligt at 129.142.224.250 Standard DESTROY
> 2009-08-26 13:44:46.747796 [DEBUG] switch_core_state_machine.c:564
> (sofia/external/hemmeligt at 129.142.224.250) State DESTROY going to
> sleep
>
> Of which I mainly find these interesting:
>
> EXECUTE sofia/external/hemmeligt at 129.142.224.250 bind_meta_app(1 b a
> bridge::sofia/gateway/secretgw/8888)
> 2009-08-26 13:43:26.388737 [INFO] switch_ivr_async.c:1795 Bound B-Leg:
> 1 bridge::sofia/gateway/secretgw/8888
> [...]
> 2009-08-26 13:43:40.309616 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF *:2000
> 2009-08-26 13:43:40.970513 [DEBUG] switch_rtp.c:2222 RTP RECV DTMF 1:2000
> 2009-08-26 13:43:40.970513 [DEBUG] switch_ivr_async.c:1711
> sofia/external/hemmeligt at 129.142.224.250 Processing meta digit '1'
> [bridge::sofia/gateway/secretgw/8888]
> 2009-08-26 13:43:40.970513 [WARNING] switch_ivr_async.c:2310 Channel
> [sofia/external/hemmeligt at 129.142.224.250][bridge::sofia/gateway/secretgw/8888]
> already broadcasting...broadcast aborted
>
> Can anybody explain what is going on here? It seems like the
> bind_meta_app is bound perfectly via event socket, but when it comes
> down, it doesn't work.
>
> Best regards,
> Harry
>



From foxb at abv.bg  Wed Aug 26 14:03:07 2009
From: foxb at abv.bg (Hristo Benev)
Date: Thu, 27 Aug 2009 00:03:07 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <784150205.123595.1251320595248.JavaMail.apache@mail22.abv.bg>

 
It seems that the problem is on kamailio configuration.
Will ask on their list.

But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.

I have it setup as friend in asterisk, but still ???

Any ideas?

Thanks,


 >-------- ?????????? ????? --------
 >??:  Hristo Benev 
 >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >??: freeswitch-users at lists.freeswitch.org
 >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST

 > I think that the problem is here:
 >-------------------------
 >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
 >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
 >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
 >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
 >--------------------------
 >
 >Actually Regex FAIL
 >
 >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
 >Here is my  default.xml:
 >----------------
 >
 >
 >
 >  
 >
 >     
 >      
 >         
 >         
 >         
 >         
 >         
 >         
 >         
 >      
 >    
 >
 >   
 >
 >  
 >
 >    
 >      
 >        
 >        
 >        
 >        
 >        
 >        
 >        
 >        
 >      
 >    
 >
 >  
 >
 >
 >
 >--------------------------
 >
 >
 >
 > >-------- ?????????? ????? --------
 > >??:  Brian West 
 > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 > >??: freeswitch-users at lists.freeswitch.org
 > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
 >
 > >We do not blindly follow 302's as that is a dangerous thing to do. 
 >You have to process all 302's in the dialplan. 
 >Set this on your sofia profile
 >You can set these variables sip_redirect_profile,
 >sip_redirect_context,
 >sip_redirect_dialplan,
 >When a redirect happens you get these variables - sip_redirect_contact_%d,
 >sip_redirected_to,
 >sip_redirect_contact_user_%d,
 >sip_redirect_contact_host_%d,
 >sip_redirect_contact_params_%d,
 >sip_redirect_dialstring_%d,
 >sip_redirect_dialstring,
 >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
 >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
 > >
 >
 >_______________________________________________
 >FreeSWITCH-users mailing list
 >FreeSWITCH-users at lists.freeswitch.org
 >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >http://www.freeswitch.org
 >



From pbd at suspiria.net  Wed Aug 26 14:36:41 2009
From: pbd at suspiria.net (Public Dump)
Date: Wed, 26 Aug 2009 23:36:41 +0200
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
	<191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>
Message-ID: <13C421883438EB42B9E2C30069FD4AB77004E8C0A9@crushinator.central.local>

I am quite sure that it's not a problem with the network connection, since the delay builds up gradually over time, while the host is reachable without any problems.



I have added the suggested config lines and will check the results.



By the way, can I remap the hash key, so that is does not end the conference ? I've tried remapping it, but freeswitch seems to ignore the setting (the console shows: hangup mapped to '*' ... but # still ends the call).

It is irritating for users, since most users try to enter the conference pin the #-key at the end.



Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 22:45
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] delay buildup in conference



There is no bug, it's all dependent on your network conditions.
I spend literally 8-12 hours a day on a conference and there is no delay.

The important param is
<param name="rtp-autoflush-during-bridge" value="true"/>

in the sofia profile in question.

if you have delay with that in place, then it's probably not FS



On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd at suspiria.net<mailto:pbd at suspiria.net>> wrote:

Hi,



I am on a quite recent version (i assume):



FreeSWITCH Version 1.0.trunk (14461)



Should the bug be fixed in this revision ? What config settings would a have to check to limit delay (even at the cost of reduction in quality).



Thanks



Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 20:57
An: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] delay buildup in conference



which revision are you on?
The defaults on the latest code and examples should be configured to minimize delay.
Some of the older revisions built up some delay issues from udp buffering when timers were not synced.

On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net<mailto:pbd at suspiria.net>> wrote:

When running conferences with users dialed in from a PSTN gateway (SIP) and directly from remote SIP endpoints there is an ever longer buildup in delay, reaching up to multiple seconds. Is there any way to limit the delay ?



I am not 100% sure whether the delays is caused by the SIP jitter buffer of freeswitch or directly by the conference module.



Any advice?



Thanks


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org




--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400

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From anthony.minessale at gmail.com  Wed Aug 26 15:00:32 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Wed, 26 Aug 2009 17:00:32 -0500
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <13C421883438EB42B9E2C30069FD4AB77004E8C0A9@crushinator.central.local>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
	<191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A9@crushinator.central.local>
Message-ID: <191c3a030908261500q78cc50c6gaf4d469722e1edf4@mail.gmail.com>

being unreachable does not cause delay, sending the audio too fast does.
the timer in the rtp stack will slow down the audio stream to match the
clocked rate and if there is a difference
it can translate to delay over time.

This is been especially prevalent in cisco phones but other things have
caused a similar problem.

you can also try disabling rtp timer by setting rtp_timer_name=none before
you answer the call.

the keys are controlled by the mappings in the conference profile


it's a little unnerving talking to someone named public dump, do you have an
identity?



On Wed, Aug 26, 2009 at 4:36 PM, Public Dump <pbd at suspiria.net> wrote:

>  I am quite sure that it?s not a problem with the network connection,
> since the delay builds up gradually over time, while the host is reachable
> without any problems.
>
>
>
> I have added the suggested config lines and will check the results.
>
>
>
> By the way, can I remap the hash key, so that is does not end the
> conference ? I?ve tried remapping it, but freeswitch seems to ignore the
> setting (the console shows: hangup mapped to ?*? ? but # still ends the
> call).
>
> It is irritating for users, since most users try to enter the conference
> pin the #-key at the end.
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Anthony
> Minessale
> *Gesendet:* Mittwoch, 26. August 2009 22:45
>
> *An:* freeswitch-users at lists.freeswitch.org
> *Betreff:* Re: [Freeswitch-users] delay buildup in conference
>
>
>
> There is no bug, it's all dependent on your network conditions.
> I spend literally 8-12 hours a day on a conference and there is no delay.
>
> The important param is
> <param name="rtp-autoflush-during-bridge" value="true"/>
>
> in the sofia profile in question.
>
> if you have delay with that in place, then it's probably not FS
>
>  On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd at suspiria.net> wrote:
>
> Hi,
>
>
>
> I am on a quite recent version (i assume):
>
>
>
> FreeSWITCH Version 1.0.trunk (14461)
>
>
>
> Should the bug be fixed in this revision ? What config settings would a
> have to check to limit delay (even at the cost of reduction in quality).
>
>
>
> Thanks
>
>
>
> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Anthony
> Minessale
> *Gesendet:* Mittwoch, 26. August 2009 20:57
> *An:* freeswitch-users at lists.freeswitch.org
> *Betreff:* Re: [Freeswitch-users] delay buildup in conference
>
>
>
> which revision are you on?
> The defaults on the latest code and examples should be configured to
> minimize delay.
> Some of the older revisions built up some delay issues from udp buffering
> when timers were not synced.
>
> On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net> wrote:
>
> When running conferences with users dialed in from a PSTN gateway (SIP) and
> directly from remote SIP endpoints there is an ever longer buildup in delay,
> reaching up to multiple seconds. Is there any way to limit the delay ?
>
>
>
> I am not 100% sure whether the delays is caused by the SIP jitter buffer of
> freeswitch or directly by the conference module.
>
>
>
> Any advice?
>
>
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jaybinks at gmail.com  Wed Aug 26 15:45:21 2009
From: jaybinks at gmail.com (jay binks)
Date: Thu, 27 Aug 2009 08:45:21 +1000
Subject: [Freeswitch-users] Freeswitch performance as a redirecting
	server
In-Reply-To: <65d96fc80908261042l42437ed9xc4fab87b757adc80@mail.gmail.com>
References: <65d96fc80908250154w1c582478t85b21b155d6b5f67@mail.gmail.com>
	<7b197bef0908250200r2c76250ejdc77748f24ec2e1f@mail.gmail.com>
	<65d96fc80908250619t718f8dedl6e6d0dc8202dfbfd@mail.gmail.com>
	<7b197bef0908250641na1d8c8xa83aebd6417b38fd@mail.gmail.com>
	<65d96fc80908260108x452b6785tefa5155b2699dce6@mail.gmail.com>
	<7b197bef0908260125m56c3d681l26cb18a5e34aaa33@mail.gmail.com>
	<65d96fc80908260129xebbf460raffaa38c72c3281a@mail.gmail.com>
	<681a20520908260453q57877b1eg8f0949061cd129ee@mail.gmail.com>
	<65d96fc80908261042l42437ed9xc4fab87b757adc80@mail.gmail.com>
Message-ID: <ec2ce17e0908261545i2c2a8537pf36984945fe8b823@mail.gmail.com>

Just for fun...
Im personally interested how this holds up on Debian 5 - 64 bit..

Im more of a debian person... not a huge centos fan and it seems you had a
pref for debian at first also
so if you have the time to play, let us know how that goes :)
surly its just the change to a 64 bit OS....

Jay



On Thu, Aug 27, 2009 at 3:42 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> Guys you made a monster!!
>
>
> so, i moved the machine to 64bit CentOS 5.3... recompiled the latest trunk
> and did my tests again.
>
> The old tests on 32bit debian 5 on the same hardware shown a CPS rate of
> 120 as 75 - 80% CPU.... and after some time on that 120 CPS rate the CPU
> goes to 100% without any chance FS recovers at all.
> New tests on 64bit CentOS shown a monster.... 400 CPS rate at 75% CPU....
> during the tests FS was really stable and responsive... i placed few calls
> that went through as a charm :).
>
>
> Also, i didn't optimize the machine at all ... as it is after CentOS
> install!.... not even db files are on ramdisk.
>
> What did it really happen? .. did you guys change something in the trunk
> overnight or it is just moving to CentOS 64bit that boosted drastically?
>
>
>
>
> Here are some details:
>
> ?nmon?12a??????[H for help]???Hostname=l01sipindir2?Refresh= 2secs
> ???19:17.48??????????????????????????????????????????????????????
> ? CPU
> +-------------------------------------------------------------------------+
> ?
> ?100%-|
> |
> ?
> ? 95%-|
> |
> ?
> ? 90%-|
> |
> ?
> ? 85%-|
> |
> ?
> ? 80%-|  | w              s  w      www  w        s   sw    s
> s           s                                                     ?
> ? 75%-|
> |ssssssssssssssssssssswssssssssssssssssssssssssssssssssssssssssssssss
> ?
> ? 70%-|
> +sssssssssssssssssssssssssssssssUssssssssUsssUssssssssssUssssssssssss
> ?
> ? 65%-|
> |UUUUUsUsUUUUUUUUUUUUUsUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUsUUU
> ?
> ? 60%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 55%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 50%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 45%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 40%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUw
> ?
> ? 35%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUs
> ?
> ? 30%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 25%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 20%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 15%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ? 10%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ?  5%-|
> |UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUU
> ?
> ?
> +-------------------------------------------------------------------------+
> ?
> ? CPU Utilisation
> ??????????????????????????????????????????????????????????????????????????????????????????????????????????????????
> ?
> +-------------------------------------------------+
> ?
> ?CPU  User%  Sys% Wait% Idle|0          |25         |50          |75
> 100|                                                    ?
> ? 1   1.0   0.5   0.0   98.5|
> >          |                                                    ?
> ? 2   1.5   1.0   0.0   97.5|
> >  |                                                    ?
> ?
> +-------------------------------------------------+
> ?
> ?Avg  1.2   0.5   0.0   98.3|
> >         |                                                    ?
> ?
> +-------------------------------------------------+
> ?
> ? Disk I/O ?????(/proc/diskstats)????????all data is Kbytes per
> second??????????????????????????????????????????????????????????????
> ?DiskName Busy  Read WriteKB|0          |25         |50          |75
> 100|                                                    ?
> ?iss/c0d0   0%    0.0    0.0|
> >   |                                                    ?
> ?s/c0d0p1   0%    0.0
> 0.0|>
> |                                                    ?
> ?s/c0d0p2   0%    0.0    0.0|
> >  |                                                    ?
> ?dm-0       0%    0.0    0.0|
> >  |                                                    ?
> ?dm-1       0%    0.0
> 0.0|>
> |                                                    ?
>
> ????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????????
>
>
>
>
>
>
>
> ------------------------------ Scenario Screen -------- [1-9]: Change
> Screen --
>   Call-rate(length)   Port   Total-time  Total-calls  Remote-host
>   40.0(0 ms)/0.100s   5060     488.20 s       193555  10.4.4.252:5060(UDP)
>
>   402 new calls during 1.002 s period    0 ms scheduler resolution
>   3 calls (limit 4000)                   Peak was 53 calls, after 351 s
>   0 Running, 13216 Paused, 670 Woken up
>   0 dead call msg (discarded)            0 out-of-call msg
> (discarded)
>   3 open sockets
>
>                                  Messages  Retrans   Timeout
> Unexpected-Msg
>       INVITE ---------->  B-RTD1 193553    0         0
>          100 <----------  E-RTD1 193553    0         0         0
>          302 <----------  E-RTD2 193552    0         0         0
>          ACK ---------->         193552    0
> ------ [+|-|*|/]: Adjust rate ---- [q]: Soft exit ---- [p]: Pause traffic
> -----
>
> ------------------------------ Scenario Screen -------- [1-9]: Change
> Screen --
>   Call-rate(length)   Port   Total-time  Total-calls  Remote-host
>   40.0(0 ms)/0.100s   5060     489.10 s       193917  10.4.4.252:5060(UDP)
>
>   362 new calls during 0.906 s period    1 ms scheduler resolution
>   2 calls (limit 4000)                   Peak was 53 calls, after 351 s
>   0 Running, 13215 Paused, 623 Woken up
>   0 dead call msg (discarded)            0 out-of-call msg
> (discarded)
>   3 open sockets
>
>                                  Messages  Retrans   Timeout
> Unexpected-Msg
>       INVITE ---------->  B-RTD1 193917    0         0
>          100 <----------  E-RTD1 193917    0         0         0
>          302 <----------  E-RTD2 193915    0         0         0
>          ACK ---------->         193915    0
> ------------------------------ Test Terminated
> --------------------------------
>
>
> ----------------------------- Statistics Screen ------- [1-9]: Change
> Screen --
>   Start Time             | 2009-08-26   19:09:34:575
> 1251306574.575684
>   Last Reset Time        | 2009-08-26   19:17:42:779
> 1251307062.779468
>   Current Time           | 2009-08-26   19:17:43:685
> 1251307063.685281
>
> -------------------------+---------------------------+--------------------------
>   Counter Name           | Periodic value            | Cumulative value
>
> -------------------------+---------------------------+--------------------------
>   Elapsed Time           | 00:00:00:905              |
> 00:08:09:109
>   Call Rate              |  400.000 cps              |  396.470
> cps
>
> -------------------------+---------------------------+--------------------------
>   Incoming call created  |        0                  |
> 0
>   OutGoing call created  |      362                  |
> 193917
>   Total Call created     |                           |
> 193917
>   Current Call           |        2
> |
>
> -------------------------+---------------------------+--------------------------
>   Successful call        |      363                  |
> 193915
>   Failed call            |        0                  |
> 0
>
> -------------------------+---------------------------+--------------------------
>   Response Time 1        | 00:00:00:001              |
> 00:00:00:000
>   Response Time 2        | 00:00:00:010              |
> 00:00:00:008
>   Call Length            | 00:00:00:010              |
> 00:00:00:008
> ------------------------------ Test Terminated
> --------------------------------
>
>
>
> ...i didn't beleive to SIPp and i went to FS console issuing status command
> to conferm the results.
>
>
> freeswitch at l01sipindir2.ot.hr> status
> API CALL [status()] output:
> UP 0 years, 0 days, 0 hours, 8 minutes, 13 seconds, 703 milliseconds, 971
> microseconds
> 183382 session(s) since startup
> 1 session(s) 410/800
> 8000 session(s) max
>
> freeswitch at l01sipindir2.ot.hr> status
> API CALL [status()] output:
> UP 0 years, 0 days, 0 hours, 8 minutes, 15 seconds, 109 milliseconds, 891
> microseconds
> 183944 session(s) since startup
> 1 session(s) 401/800
> 8000 session(s) max
>
> freeswitch at l01sipindir2.ot.hr> status
> API CALL [status()] output:
> UP 0 years, 0 days, 0 hours, 8 minutes, 16 seconds, 139 milliseconds, 412
> microseconds
> 184356 session(s) since startup
> 2 session(s) 389/800
> 8000 session(s) max
>
> freeswitch at l01sipindir2.ot.hr> status
> API CALL [status()] output:
> UP 0 years, 0 days, 0 hours, 8 minutes, 17 seconds, 62 milliseconds, 16
> microseconds
> 184717 session(s) since startup
> 6 session(s) 410/800
> 8000 session(s) max
>
> freeswitch at l01sipindir2.ot.hr> status
> API CALL [status()] output:
> UP 0 years, 0 days, 0 hours, 8 minutes, 35 seconds, 150 milliseconds, 253
> microseconds
> 191959 session(s) since startup
> 1 session(s) 400/800
> 8000 session(s) max
>
> freeswitch at l01sipindir2.ot.hr> status
> API CALL [status()] output:
> UP 0 years, 0 days, 0 hours, 8 minutes, 36 seconds, 892 milliseconds, 672
> microseconds
> 192657 session(s) since startup
> 1 session(s) 393/800
> 8000 session(s) max
>
>
>
>
> On Wed, Aug 26, 2009 at 1:53 PM, Dmitry Kadantsev <kadantsev.d at gmail.com>wrote:
>
>> Hi all,
>>
>> is there same situation with FS for Windows? I mean 64bit is more
>> preferable than 32bit, isn't it?
>>
>> Any performance test on Win 32/64 were done?
>>
>> --
>> Best regards,
>> Dmitry Kadantsev
>>
>>
>>
>> On Wed, Aug 26, 2009 at 10:29 AM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> intanto e il centos che si sta installando :)
>>>
>>> grazie.
>>>
>>> T.
>>>
>>>
>>> On Wed, Aug 26, 2009 at 10:25 AM, Giovanni Maruzzelli <
>>> gmaruzz at celliax.org> wrote:
>>>
>>>> netbook remix
>>>>
>>>>
>>>> joking! Server 64bit :-)
>>>>
>>>> -gm
>>>>
>>>>
>>>>
>>>> On Wed, Aug 26, 2009 at 10:08 AM, Tihomir Culjaga<tculjaga at gmail.com>
>>>> wrote:
>>>> > Hi Giovanny,
>>>> >
>>>> > regarding ubuntu, did you mean 8.04 server or desktop ?
>>>> >
>>>> >
>>>> > On Tue, Aug 25, 2009 at 3:41 PM, Giovanni Maruzzelli <
>>>> gmaruzz at celliax.org>
>>>> > wrote:
>>>> >>
>>>> >> Definitely go for 64 bit OS.
>>>> >>
>>>> >> If you want to be safe and sure,  go for CentOS 5.2 64bit. Is the one
>>>> >> used both for development and for heavy duty production.
>>>> >>
>>>> >> Also Ubuntu 8.04 is good.
>>>> >>
>>>> >> Other versions/distros are less used by the community.
>>>> >>
>>>> >> Adding RAM and CPUs helps to scale up.
>>>> >>
>>>> >> -gm
>>>> >>
>>>> >>
>>>> >>
>>>> >> Sincerely,
>>>> >>
>>>> >> Giovanni Maruzzelli
>>>> >> Cell : +39-347-2665618
>>>> >>
>>>> >>
>>>> >>
>>>> >>
>>>> >> On Tue, Aug 25, 2009 at 3:19 PM, Tihomir Culjaga<tculjaga at gmail.com>
>>>> >> wrote:
>>>> >> > Hey Giovanni,
>>>> >> >
>>>> >> > thanks for the tip... indeed the db files were heavily used
>>>> regardless
>>>> >> > if i
>>>> >> > started freeswitch with nosql option (freeswitch -nosql)... FS was
>>>> not
>>>> >> > writing anything into that files ... instead it was just accessing
>>>> >> > it....
>>>> >> > This behaviour leads to a waste of 40% CPU time... waiting for
>>>> other
>>>> >> > processes (mainly disk access) to finish!!!
>>>> >> >
>>>> >> > I moved freeswitch/db/ to a ramdisk and the performance got a boost
>>>> to
>>>> >> > 140
>>>> >> > CPS with a CPU load of 80%. I was keeping the machine for a while
>>>> (20 -
>>>> >> > 30
>>>> >> > minutes) on that rate when i sow CPU suddenly went to 100% and FS
>>>> >> > becoming
>>>> >> > irresponsive :).
>>>> >> >
>>>> >> >
>>>> >> > What can be wrong?
>>>> >> > What are the limits in CPU usage (50%, 60%, 70%, 80%...) we should
>>>> not
>>>> >> > cross?
>>>> >> > What fine tuning do we need in order to asure a long high load run?
>>>> >> >
>>>> >> >
>>>> >> >
>>>> >> > Also, I'm running 32-bit OS (debian 5) on a 64 bit CPU... does it
>>>> have
>>>> >> > sense
>>>> >> > to move my OS to 64 bit? ... will FS gain more preformance ?... I
>>>> mean
>>>> >> > will
>>>> >> > FS perofomr drastically better 20%+ ?
>>>> >> >
>>>> >> >
>>>> >> > Tihomir.
>>>> >> >
>>>> >> >
>>>> >> > On Tue, Aug 25, 2009 at 11:00 AM, Giovanni Maruzzelli
>>>> >> > <gmaruzz at celliax.org>
>>>> >> > wrote:
>>>> >> >>
>>>> >> >> Maybe your load comes from disk access?
>>>> >> >>
>>>> >> >> Try putting the sql and log directories on a ramdisk.
>>>> >> >>
>>>> >> >> OTH,
>>>> >> >>
>>>> >> >> -giovanni
>>>> >> >>
>>>> >> >> On Tue, Aug 25, 2009 at 10:54 AM, Tihomir Culjaga<
>>>> tculjaga at gmail.com>
>>>> >> >> wrote:
>>>> >> >> > Hello,
>>>> >> >> >
>>>> >> >> > i'm trying to use freeswitch as a redirecting server meaning FS
>>>> has
>>>> >> >> > to
>>>> >> >> > receive an INVITE and according to some rules it will redirect
>>>> calls
>>>> >> >> > to
>>>> >> >> > other destinations.
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > CALLING_USER                FREESWITCH
>>>> >> >> > SOMEWHERE
>>>> >> >> >
>>>> >> >> > INVITE ------------------------------->
>>>> >> >> >            <------------------------------ 100 Trying
>>>> >> >> >            <------------------------------ 302 Moved Temporary
>>>> >> >> > ACK    ------------------------------->
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> INVITE--------------------------------------------------------------------------------->
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > Well, wverything works well except i have perfromance issues
>>>> .... on
>>>> >> >> > my
>>>> >> >> > HW
>>>> >> >> > FS cannot do more than 40 CPS (INVITE answered by 302 Moved
>>>> >> >> > Temporary).
>>>> >> >> > When
>>>> >> >> > i increase the rate, FS starts delaying 302 response. Right at
>>>> 50 CPS
>>>> >> >> > i
>>>> >> >> > see
>>>> >> >> > "calls" being build up in FS and the delay begining to grow.
>>>> >> >> >
>>>> >> >> > When i observe the machine, load average is almost nothing (load
>>>> >> >> > average:
>>>> >> >> > 1.41, 0.61, 0.60) CPU never goes to 100%, and i see only one
>>>> thread
>>>> >> >> > taking
>>>> >> >> > most load... all others are just sitting there with 1-5 % CPU
>>>> time.
>>>> >> >> > This looks to me as FS handles 302 messages in a single
>>>> thread?!?!
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > tculjaga at FS:/usr/local/freeswitch/conf/dialplan$ top -H
>>>> >> >> >
>>>> >> >> > top - 10:41:37 up 167 days, 20:42,  3 users,  load average:
>>>> 1.41,
>>>> >> >> > 0.61,
>>>> >> >> > 0.60
>>>> >> >> > Tasks:  83 total,   2 running,  81 sleeping,   0 stopped,   0
>>>> zombie
>>>> >> >> > Cpu(s): 25.3%us,  1.5%sy,  0.0%ni, 30.3%id, 42.7%wa,  0.0%hi,
>>>> >> >> > 0.2%si,
>>>> >> >> > 0.0%st
>>>> >> >> > Mem:   2074520k total,   571244k used,  1503276k free,   259604k
>>>> >> >> > buffers
>>>> >> >> > Swap:  2650684k total,     3020k used,  2647664k free,   153868k
>>>> >> >> > cached
>>>> >> >> >
>>>> >> >> >   PID USER      PR  NI  VIRT  RES  SHR S %CPU %MEM    TIME+
>>>> >> >> > COMMAND
>>>> >> >> >  4814 root      20   0 34188  20m 3780 S   38  1.0   3:10.29
>>>> >> >> > freeswitch
>>>> >> >> >  4800 root      20   0 34188  20m 3780 S    6  1.0   0:08.26
>>>> >> >> > freeswitch
>>>> >> >> >  4798 root      20   0 34188  20m 3780 R    5  1.0   0:24.46
>>>> >> >> > freeswitch
>>>> >> >> >  4787 root      20   0 34188  20m 3780 S    2  1.0   0:11.24
>>>> >> >> > freeswitch
>>>> >> >> >  4794 root      20   0 34188  20m 3780 S    1  1.0   0:11.42
>>>> >> >> > freeswitch
>>>> >> >> >  4803 root      20   0 34188  20m 3780 S    1  1.0   0:11.74
>>>> >> >> > freeswitch
>>>> >> >> >  4788 root      20   0 34188  20m 3780 S    1  1.0   0:02.96
>>>> >> >> > freeswitch
>>>> >> >> >  4804 root      20   0 34188  20m 3780 S    1  1.0   0:01.64
>>>> >> >> > freeswitch
>>>> >> >> >  4807 root      20   0 34188  20m 3780 S    1  1.0   0:01.68
>>>> >> >> > freeswitch
>>>> >> >> >  4811 root      20   0 34188  20m 3780 S    1  1.0   0:02.50
>>>> >> >> > freeswitch
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > cat /proc/cpuinfo
>>>> >> >> > processor       : 0
>>>> >> >> > vendor_id       : GenuineIntel
>>>> >> >> > cpu family      : 6
>>>> >> >> > model           : 15
>>>> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @
>>>> 2.33GHz
>>>> >> >> > stepping        : 6
>>>> >> >> > cpu MHz         : 2333.560
>>>> >> >> > cache size      : 4096 KB
>>>> >> >> > physical id     : 0
>>>> >> >> > siblings        : 2
>>>> >> >> > core id         : 0
>>>> >> >> > cpu cores       : 2
>>>> >> >> > apicid          : 0
>>>> >> >> > initial apicid  : 0
>>>> >> >> > fdiv_bug        : no
>>>> >> >> > hlt_bug         : no
>>>> >> >> > f00f_bug        : no
>>>> >> >> > coma_bug        : no
>>>> >> >> > fpu             : yes
>>>> >> >> > fpu_exception   : yes
>>>> >> >> > cpuid level     : 10
>>>> >> >> > wp              : yes
>>>> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep
>>>> mtrr
>>>> >> >> > pge
>>>> >> >> > mca
>>>> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe
>>>> lm
>>>> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est
>>>> tm2
>>>> >> >> > ssse3
>>>> >> >> > cx16
>>>> >> >> > xtpr dca lahf_lm
>>>> >> >> > bogomips        : 4670.78
>>>> >> >> > clflush size    : 64
>>>> >> >> > power management:
>>>> >> >> >
>>>> >> >> > processor       : 1
>>>> >> >> > vendor_id       : GenuineIntel
>>>> >> >> > cpu family      : 6
>>>> >> >> > model           : 15
>>>> >> >> > model name      : Intel(R) Xeon(R) CPU            5140  @
>>>> 2.33GHz
>>>> >> >> > stepping        : 6
>>>> >> >> > cpu MHz         : 2333.560
>>>> >> >> > cache size      : 4096 KB
>>>> >> >> > physical id     : 0
>>>> >> >> > siblings        : 2
>>>> >> >> > core id         : 1
>>>> >> >> > cpu cores       : 2
>>>> >> >> > apicid          : 1
>>>> >> >> > initial apicid  : 1
>>>> >> >> > fdiv_bug        : no
>>>> >> >> > hlt_bug         : no
>>>> >> >> > f00f_bug        : no
>>>> >> >> > coma_bug        : no
>>>> >> >> > fpu             : yes
>>>> >> >> > fpu_exception   : yes
>>>> >> >> > cpuid level     : 10
>>>> >> >> > wp              : yes
>>>> >> >> > flags           : fpu vme de pse tsc msr pae mce cx8 apic sep
>>>> mtrr
>>>> >> >> > pge
>>>> >> >> > mca
>>>> >> >> > cmov pat pse36 clflush dts acpi mmx fxsr sse sse2 ss ht tm pbe
>>>> lm
>>>> >> >> > constant_tsc arch_perfmon pebs bts pni monitor ds_cpl vmx est
>>>> tm2
>>>> >> >> > ssse3
>>>> >> >> > cx16
>>>> >> >> > xtpr dca lahf_lm
>>>> >> >> > bogomips        : 4666.82
>>>> >> >> > clflush size    : 64
>>>> >> >> > power management:
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > uname -a
>>>> >> >> > Linux l01sipindir1 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:31 UTC
>>>> 2009
>>>> >> >> > i686
>>>> >> >> > GNU/Linux
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > Of course, i've tuned the machine up
>>>> >> >> >
>>>> >> >> > ulimit -c unlimited
>>>> >> >> > ulimit -d unlimited
>>>> >> >> > ulimit -f unlimited
>>>> >> >> > ulimit -i unlimited
>>>> >> >> > ulimit -n 999999
>>>> >> >> > ulimit -q unlimited
>>>> >> >> > ulimit -u unlimited
>>>> >> >> > ulimit -v unlimited
>>>> >> >> > ulimit -x unlimited
>>>> >> >> > ulimit -s 240
>>>> >> >> > ulimit -l unlimited
>>>> >> >> > ulimit -a
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > Started FS with minimum modules but still 40 CPS seems to be the
>>>> >> >> > limit.
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > So, is there any way to improve performance?
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > Tihomir.
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> >
>>>> >> >> > _______________________________________________
>>>> >> >> > FreeSWITCH-users mailing list
>>>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> >> >
>>>> >> >> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> >> > http://www.freeswitch.org
>>>> >> >> >
>>>> >> >> >
>>>> >> >>
>>>> >> >> _______________________________________________
>>>> >> >> FreeSWITCH-users mailing list
>>>> >> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> >>
>>>> >> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> >> http://www.freeswitch.org
>>>> >> >
>>>> >> >
>>>> >> > _______________________________________________
>>>> >> > FreeSWITCH-users mailing list
>>>> >> > FreeSWITCH-users at lists.freeswitch.org
>>>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> > http://www.freeswitch.org
>>>> >> >
>>>> >> >
>>>> >>
>>>> >> _______________________________________________
>>>> >> FreeSWITCH-users mailing list
>>>> >> FreeSWITCH-users at lists.freeswitch.org
>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> >> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> >> http://www.freeswitch.org
>>>> >
>>>> >
>>>> > _______________________________________________
>>>> > FreeSWITCH-users mailing list
>>>> > FreeSWITCH-users at lists.freeswitch.org
>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> > UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> > http://www.freeswitch.org
>>>> >
>>>> >
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:
>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Sincerely

Jay
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From larclap at yahoo.com  Wed Aug 26 18:00:00 2009
From: larclap at yahoo.com (Lars Zeb)
Date: Wed, 26 Aug 2009 18:00:00 -0700
Subject: [Freeswitch-users] Dial phone number + extension?
Message-ID: <001c01ca26b1$b45bc360$1d134a20$@com>

Is there a way to dial an external 10-digit phone number, wait a second or
two after connecting, and then dial a 4-digit extension?

 

Thanks, Lars

 

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From Prometheus001 at gmx.net  Wed Aug 26 18:05:49 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 27 Aug 2009 03:05:49 +0200
Subject: [Freeswitch-users] sofia profile external register gwname via
	XML-Curl?
Message-ID: <4A95DBED.2060505@gmx.net>

Hello,

I am using XML-Curl to handle the configuration of freeswitch
When I try to register a gateway via event-socket with
    sofia profile external register <gw-name>
I receive back "invalid gateway".

After reload mod_sofia the gateway is there. Question: Does this command
work with xml-curl or only with local files??
At least I see no xml-curl request when grepping network traffic.

Best regards
Peter



From msc at freeswitch.org  Wed Aug 26 19:15:45 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Wed, 26 Aug 2009 19:15:45 -0700
Subject: [Freeswitch-users] Dial phone number + extension?
Message-ID: <A057C640-E152-4218-A4FB-CFF82FC48FE1@freeswitch.org>



Sent from my iPhone

On Aug 26, 2009, at 6:00 PM, "Lars Zeb" <larclap at yahoo.com> wrote:

> Is there a way to dial an external 10-digit phone number, wait a  
> second or two after connecting, and then dial a 4-digit extension?
>
>
 From the CLI:
originate sofia/gateway/mygw/1234567890 1234

As for the short pause you could put that in the Dialplan entry for  
ext 1234 or you could even do an inline Dialplan instead of just 1234  
as the second arg to originate.

originate sofia/foo/1234567890 sleep:2000,transfer:1234 inline

Search the wiki for "inline dialplan" to learn more about this handy  
feature
-MC
>
>
> Thanks, Lars
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From ahmedmunir007 at gmail.com  Wed Aug 26 20:57:51 2009
From: ahmedmunir007 at gmail.com (Ahmed Munir)
Date: Thu, 27 Aug 2009 09:57:51 +0600
Subject: [Freeswitch-users] Define Condition in FreeSwitch
Message-ID: <a60fed5e0908262057k5c49cb91t68466a69d6a7df56@mail.gmail.com>

Hi,

   I'm newbie. How can we translate the asterisk's condition in freeswitch
as listed below;

   1. NoOp ("Remote Conference Call")
   2. GotoIf ($[${LEN(${DIALSTR})}=0]?3:4)
   3. Hangup()
   4. NoOp(Finish if-CONFERENCE-430)

   Kindly reply soon.

-- 
Regards,

Ahmed Munir
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From lakindia89 at gmail.com  Wed Aug 26 21:38:43 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Wed, 26 Aug 2009 21:38:43 -0700 (PDT)
Subject: [Freeswitch-users]  How to make a call back
Message-ID: <25166083.post@talk.nabble.com>


When I give the following from the command line it calls to 1010 extension
and once answered, it calls to 1000 and bridge the connection.
            originate user/1010 &bridge(user/1000)
But I want to do this in perl. So I have given as follows
            $session->originate($session,"user/1010 &bridge user/1000");
But it is not working. It says "user/1010 &bridge user/1000 is invalid
user". 
How to do this in perl. pls help.

-- 
View this message in context: http://www.nabble.com/How-to-make-a-call-back-tp25166083p25166083.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From delianspam at gmail.com  Wed Aug 26 23:35:02 2009
From: delianspam at gmail.com (delianSPAM)
Date: Thu, 27 Aug 2009 09:35:02 +0300
Subject: [Freeswitch-users] Timers/DTMFs During a Call
In-Reply-To: <87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>
	<87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAPzw61gTNaNJsjaEGYPk5BEBAAAAAA==@gmail.com>

Hello Michael!

 

Thank you for your reply! I know about sched_api and I use Python as a
scripting language. Also "sched_api" can be used inside python, using
session.execute. However the problem is that the "sched_api" timer starts
right after you initiate the SETUP of the second call leg. What I need is to
call something, after a call CONNECT instead. One workaround would be if I
can check what time it took to connect the call, but I do not know/see how
to do this. I do not see a CONNECT callback function either.

 

Best Regards, Delian Tashev

P.S. Dear enlightened people, thank you for providing help to the community
by replying to the list e-mails.

 

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Wednesday, August 26, 2009 11:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call

 

Do you actually need Python for the IVR, or is it that you're comfortable
using a scripting lang for an IVR? I like using XML for IVRs, but using
scripting langs does give you a bit more power & flexibility at the cost of
some resources.

For the record, you can do this in the dialplan using XML and sched_api
without touching a scripting language. Checkout the sched_api channel
variable on the wiki - it may give you the functionality you need.

-MC

On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM <delianspam at gmail.com> wrote:

Hello Everybody!

 

1.       Scenario.

I am writing an IVR in Python that gets a destination from the calling party
(party A) and then connects to the destination (party B).

When the call is CONNECTED, I want to:

-          Receive DTMFs

-          Have a timer that can call a certain function in my script. The
script will have to play a message to party A.

-          Have a timer that can call a certain function in my script. The
script will have to drop the call.

Please notice that I want to do the things after the two parties are
connected, and not after I send the Invite to party B.

 

2.       Problem.

I will be happy to receive help on:

-          Which methods should I look for to implement this.

 

3.       Details

Here is how I connect the call currently:

session.execute("bridge","sofia/internal/" + destination_number +
"@domain.com")

 

I have tried to create a timer callback function "my_method()" using:

ivr_timer =threading.Timer(30,my_method)

This never called the function "my_method()".

 

Maybe I am wrong in using threading.Timer and the "bridge" application?
Maybe I need to create a new thread and a new timer using the API of
freeswitch, plus to use the "session.setInputCallback", plus use a
conference rather than a bridge? Can you please provide any suggestions or
examples? 

 

Thank you!

Best Regards, Delian Tashev


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From kawarod at laposte.net  Wed Aug 26 23:41:05 2009
From: kawarod at laposte.net (rod)
Date: Thu, 27 Aug 2009 10:41:05 +0400
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
In-Reply-To: <784150205.123595.1251320595248.JavaMail.apache@mail22.abv.bg>
References: <784150205.123595.1251320595248.JavaMail.apache@mail22.abv.bg>
Message-ID: <4A962A81.3010709@laposte.net>

Hi Hristo,

I'm the author of this setup and wiki page. I did a lot of modifications 
on this setup (alternative routing if failure essentially) but don't 
have too much time to update the wiki.

May you please send me an ngrep trace when you call 1000:

ngrep -d any -nn -i '1000' port 5060 -W byline 

I will check what's happening.
Do you have an entry for 1000 in your mysql database ?

regards,
rod

Hristo Benev a ?crit :
>  
> It seems that the problem is on kamailio configuration.
> Will ask on their list.
>
> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
>
> I have it setup as friend in asterisk, but still ???
>
> Any ideas?
>
> Thanks,
>
>
>  >-------- ?????????? ????? --------
>  >??:  Hristo Benev 
>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >??: freeswitch-users at lists.freeswitch.org
>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
>
>  > I think that the problem is here:
>  >-------------------------
>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
>  >--------------------------
>  >
>  >Actually Regex FAIL
>  >
>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
>  >Here is my  default.xml:
>  >----------------
>  >
>  >
>  >
>  >  
>  >
>  >     
>  >      
>  >         
>  >         
>  >         
>  >         
>  >         
>  >         
>  >         
>  >      
>  >    
>  >
>  >   
>  >
>  >  
>  >
>  >    
>  >      
>  >        
>  >        
>  >        
>  >        
>  >        
>  >        
>  >        
>  >        
>  >      
>  >    
>  >
>  >  
>  >
>  >
>  >
>  >--------------------------
>  >
>  >
>  >
>  > >-------- ?????????? ????? --------
>  > >??:  Brian West 
>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  > >??: freeswitch-users at lists.freeswitch.org
>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
>  >
>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
>  >You have to process all 302's in the dialplan. 
>  >Set this on your sofia profile
>  >You can set these variables sip_redirect_profile,
>  >sip_redirect_context,
>  >sip_redirect_dialplan,
>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
>  >sip_redirected_to,
>  >sip_redirect_contact_user_%d,
>  >sip_redirect_contact_host_%d,
>  >sip_redirect_contact_params_%d,
>  >sip_redirect_dialstring_%d,
>  >sip_redirect_dialstring,
>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
>  > >
>  >
>  >_______________________________________________
>  >FreeSWITCH-users mailing list
>  >FreeSWITCH-users at lists.freeswitch.org
>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >http://www.freeswitch.org
>  >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From delianspam at gmail.com  Wed Aug 26 23:52:35 2009
From: delianspam at gmail.com (delianSPAM)
Date: Thu, 27 Aug 2009 09:52:35 +0300
Subject: [Freeswitch-users] Timers/DTMFs During a Call
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>
	<87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com> 
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAANZUwRmfIlpOtIAK9HYapUoBAAAAAA==@gmail.com>

Looks like I was wrong about using the native Python timers. Here is how
they can be used in your script:

 

# Imports - add these new imports

import time

import threading

 

# class definitions - add this new class

class Timer(threading.Thread):

                def __init__(self, seconds):

                               self.runTime = seconds

                               threading.Thread.__init__(self)

                def run(self):

                               time.sleep(self.runTime)

                               console_log("debug", "TIMER
********************")

 

# entry point - add two rows in the entry point function that is called from
freeswitch

def handler(session, args):

.            

 t = Timer(10)

 t.start()

.

 

So what is now remaining is to get when the call CONNECTS and how to get
DTMFs during the call. 

 

From: delianSPAM [mailto:delianspam at gmail.com] 
Sent: Thursday, August 27, 2009 9:35 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call

 

Hello Michael!

 

Thank you for your reply! I know about sched_api and I use Python as a
scripting language. Also "sched_api" can be used inside python, using
session.execute. However the problem is that the "sched_api" timer starts
right after you initiate the SETUP of the second call leg. What I need is to
call something, after a call CONNECT instead. One workaround would be if I
can check what time it took to connect the call, but I do not know/see how
to do this. I do not see a CONNECT callback function either.

 

Best Regards, Delian Tashev

P.S. Dear enlightened people, thank you for providing help to the community
by replying to the list e-mails.

 

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Wednesday, August 26, 2009 11:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call

 

Do you actually need Python for the IVR, or is it that you're comfortable
using a scripting lang for an IVR? I like using XML for IVRs, but using
scripting langs does give you a bit more power & flexibility at the cost of
some resources.

For the record, you can do this in the dialplan using XML and sched_api
without touching a scripting language. Checkout the sched_api channel
variable on the wiki - it may give you the functionality you need.

-MC

On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM <delianspam at gmail.com> wrote:

Hello Everybody!

 

1.       Scenario.

I am writing an IVR in Python that gets a destination from the calling party
(party A) and then connects to the destination (party B).

When the call is CONNECTED, I want to:

-          Receive DTMFs

-          Have a timer that can call a certain function in my script. The
script will have to play a message to party A.

-          Have a timer that can call a certain function in my script. The
script will have to drop the call.

Please notice that I want to do the things after the two parties are
connected, and not after I send the Invite to party B.

 

2.       Problem.

I will be happy to receive help on:

-          Which methods should I look for to implement this.

 

3.       Details

Here is how I connect the call currently:

session.execute("bridge","sofia/internal/" + destination_number +
"@domain.com")

 

I have tried to create a timer callback function "my_method()" using:

ivr_timer =threading.Timer(30,my_method)

This never called the function "my_method()".

 

Maybe I am wrong in using threading.Timer and the "bridge" application?
Maybe I need to create a new thread and a new timer using the API of
freeswitch, plus to use the "session.setInputCallback", plus use a
conference rather than a bridge? Can you please provide any suggestions or
examples? 

 

Thank you!

Best Regards, Delian Tashev


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From delianspam at gmail.com  Thu Aug 27 00:01:09 2009
From: delianspam at gmail.com (delianSPAM)
Date: Thu, 27 Aug 2009 10:01:09 +0300
Subject: [Freeswitch-users] Timers/DTMFs During a Call
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>
	<87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com> 
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAPlzV3BY/cZHrlu/Yf4kc9QBAAAAAA==@gmail.com>

I will try to execute and parse from python:

freeswitch at internal> show channels

uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,des
t,application,application_data,dialplan,context,read_codec,read_rate,write_c
odec,write_rate

53a62ebd-156c-4684-b616-740d7a5b609b,inbound,2009-04-23
11:14:09,1240510449,sofia/internal/1000 at ...,CS_EXECUTE,Mikey,1000,10.15.0.21
3,9999,playback,local_stream://moh,XML,default,PCMU,8000,PCMU,8000

 

Hoping that this will get the state of the call. If I call this check
frequently I will catch the call connect I trust.

 

From: delianSPAM [mailto:delianspam at gmail.com] 
Sent: Thursday, August 27, 2009 9:53 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call

 

Looks like I was wrong about using the native Python timers. Here is how
they can be used in your script:

 

# Imports - add these new imports

import time

import threading

 

# class definitions - add this new class

class Timer(threading.Thread):

                def __init__(self, seconds):

                               self.runTime = seconds

                               threading.Thread.__init__(self)

                def run(self):

                               time.sleep(self.runTime)

                               console_log("debug", "TIMER
********************")

 

# entry point - add two rows in the entry point function that is called from
freeswitch

def handler(session, args):

.            

 t = Timer(10)

 t.start()

.

 

So what is now remaining is to get when the call CONNECTS and how to get
DTMFs during the call. 

 

From: delianSPAM [mailto:delianspam at gmail.com] 
Sent: Thursday, August 27, 2009 9:35 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call

 

Hello Michael!

 

Thank you for your reply! I know about sched_api and I use Python as a
scripting language. Also "sched_api" can be used inside python, using
session.execute. However the problem is that the "sched_api" timer starts
right after you initiate the SETUP of the second call leg. What I need is to
call something, after a call CONNECT instead. One workaround would be if I
can check what time it took to connect the call, but I do not know/see how
to do this. I do not see a CONNECT callback function either.

 

Best Regards, Delian Tashev

P.S. Dear enlightened people, thank you for providing help to the community
by replying to the list e-mails.

 

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Wednesday, August 26, 2009 11:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call

 

Do you actually need Python for the IVR, or is it that you're comfortable
using a scripting lang for an IVR? I like using XML for IVRs, but using
scripting langs does give you a bit more power & flexibility at the cost of
some resources.

For the record, you can do this in the dialplan using XML and sched_api
without touching a scripting language. Checkout the sched_api channel
variable on the wiki - it may give you the functionality you need.

-MC

On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM <delianspam at gmail.com> wrote:

Hello Everybody!

 

1.       Scenario.

I am writing an IVR in Python that gets a destination from the calling party
(party A) and then connects to the destination (party B).

When the call is CONNECTED, I want to:

-          Receive DTMFs

-          Have a timer that can call a certain function in my script. The
script will have to play a message to party A.

-          Have a timer that can call a certain function in my script. The
script will have to drop the call.

Please notice that I want to do the things after the two parties are
connected, and not after I send the Invite to party B.

 

2.       Problem.

I will be happy to receive help on:

-          Which methods should I look for to implement this.

 

3.       Details

Here is how I connect the call currently:

session.execute("bridge","sofia/internal/" + destination_number +
"@domain.com")

 

I have tried to create a timer callback function "my_method()" using:

ivr_timer =threading.Timer(30,my_method)

This never called the function "my_method()".

 

Maybe I am wrong in using threading.Timer and the "bridge" application?
Maybe I need to create a new thread and a new timer using the API of
freeswitch, plus to use the "session.setInputCallback", plus use a
conference rather than a bridge? Can you please provide any suggestions or
examples? 

 

Thank you!

Best Regards, Delian Tashev


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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From msc at freeswitch.org  Thu Aug 27 00:01:41 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 00:01:41 -0700
Subject: [Freeswitch-users] Define Condition in FreeSwitch
In-Reply-To: <a60fed5e0908262057k5c49cb91t68466a69d6a7df56@mail.gmail.com>
References: <a60fed5e0908262057k5c49cb91t68466a69d6a7df56@mail.gmail.com>
Message-ID: <87f2f3b90908270001o1e0611fsa7a2f61893571333@mail.gmail.com>

On Wed, Aug 26, 2009 at 8:57 PM, Ahmed Munir <ahmedmunir007 at gmail.com>wrote:

> Hi,
>
>    I'm newbie. How can we translate the asterisk's condition in freeswitch
> as listed below;
>
>    1. NoOp ("Remote Conference Call")
>    2. GotoIf ($[${LEN(${DIALSTR})}=0]?3:4)
>    3. Hangup()
>    4. NoOp(Finish if-CONFERENCE-430)
>
>    Kindly reply soon.


Before I answer this question I just want you to know that there's probably
a more elegant way of doing whatever it is you're trying to do. This
dialplan snippet is pretty short. My first question would be: how does a
call get to this point? Also, what is the big picture, that is, what's the
application you're creating? Remember the golden rule: Anything that you do
in Asterisk is easier to do in FreeSWITCH, but you need to learn the ropes a
bit.

The answer to your question is, of course, "It depends." :P Give us the
background on what you're doing so that we can give you an educated answer.
You could use the <condition> tags with actions and anti-actions. You could
also call a Lua/JavaScript/Perl/Python/etc. script to handle the logic but
that's probably overkill.

Tell us more and we'll tell you more. ;)

Thanks!
-Michael Collins
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From msc at freeswitch.org  Thu Aug 27 00:04:50 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 00:04:50 -0700
Subject: [Freeswitch-users] Timers/DTMFs During a Call
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAPzw61gTNaNJsjaEGYPk5BEBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>
	<87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com>
	<!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAPzw61gTNaNJsjaEGYPk5BEBAAAAAA==@gmail.com>
Message-ID: <87f2f3b90908270004u2845df79t4bf908c7556a4b6b@mail.gmail.com>

On Wed, Aug 26, 2009 at 11:35 PM, delianSPAM <delianspam at gmail.com> wrote:

>  Hello Michael!
>
>
>
> Thank you for your reply! I know about sched_api and I use Python as a
> scripting language. Also ?sched_api? can be used inside python, using
> session.execute. However the problem is that the ?sched_api? timer starts
> right after you initiate the SETUP of the second call leg. What I need is to
> call something, after a call CONNECT instead. One workaround would be if I
> can check what time it took to connect the call, but I do not know/see how
> to do this. I do not see a CONNECT callback function either.
>
>
>
Perhaps you could set this channel variable to the sched_api call?
http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer
-MC
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From delianspam at gmail.com  Thu Aug 27 00:35:03 2009
From: delianspam at gmail.com (delianSPAM)
Date: Thu, 27 Aug 2009 10:35:03 +0300
Subject: [Freeswitch-users] Timers/DTMFs During a Call
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAPlzV3BY/cZHrlu/Yf4kc9QBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAABsO49ptDaFNno6JSTkx9tIBAAAAAA==@gmail.com>	<87f2f3b90908261328l1b1c72dgcd6f85d57dc500e2@mail.gmail.com>
	<!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAPlzV3BY/cZHrlu/Yf4kc9QBAAAAAA==@gmail.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAACPZt1oMIAxCnItIAzX5gwYBAAAAAA==@gmail.com>

To get the state use: 

session.getVariable("state")

 

I hope that I will solve the Timers puzzle soon. I am still looking on
getting the DTMFs during a bridged call.

 

Best Regards, Delian Tashev

 

From: delianSPAM [mailto:delianspam at gmail.com] 
Sent: Thursday, August 27, 2009 10:01 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call

 

I will try to execute and parse from python:

freeswitch at internal> show channels

uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,des
t,application,application_data,dialplan,context,read_codec,read_rate,write_c
odec,write_rate

53a62ebd-156c-4684-b616-740d7a5b609b,inbound,2009-04-23
11:14:09,1240510449,sofia/internal/1000 at ...,CS_EXECUTE,Mikey,1000,10.15.0.21
3,9999,playback,local_stream://moh,XML,default,PCMU,8000,PCMU,8000

 

Hoping that this will get the state of the call. If I call this check
frequently I will catch the call connect I trust.

 

From: delianSPAM [mailto:delianspam at gmail.com] 
Sent: Thursday, August 27, 2009 9:53 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call

 

Looks like I was wrong about using the native Python timers. Here is how
they can be used in your script:

 

# Imports - add these new imports

import time

import threading

 

# class definitions - add this new class

class Timer(threading.Thread):

                def __init__(self, seconds):

                               self.runTime = seconds

                               threading.Thread.__init__(self)

                def run(self):

                               time.sleep(self.runTime)

                               console_log("debug", "TIMER
********************")

 

# entry point - add two rows in the entry point function that is called from
freeswitch

def handler(session, args):

.            

 t = Timer(10)

 t.start()

.

 

So what is now remaining is to get when the call CONNECTS and how to get
DTMFs during the call. 

 

From: delianSPAM [mailto:delianspam at gmail.com] 
Sent: Thursday, August 27, 2009 9:35 AM
To: 'freeswitch-users at lists.freeswitch.org'
Subject: RE: [Freeswitch-users] Timers/DTMFs During a Call

 

Hello Michael!

 

Thank you for your reply! I know about sched_api and I use Python as a
scripting language. Also "sched_api" can be used inside python, using
session.execute. However the problem is that the "sched_api" timer starts
right after you initiate the SETUP of the second call leg. What I need is to
call something, after a call CONNECT instead. One workaround would be if I
can check what time it took to connect the call, but I do not know/see how
to do this. I do not see a CONNECT callback function either.

 

Best Regards, Delian Tashev

P.S. Dear enlightened people, thank you for providing help to the community
by replying to the list e-mails.

 

From: Michael Collins [mailto:msc at freeswitch.org] 
Sent: Wednesday, August 26, 2009 11:28 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Timers/DTMFs During a Call

 

Do you actually need Python for the IVR, or is it that you're comfortable
using a scripting lang for an IVR? I like using XML for IVRs, but using
scripting langs does give you a bit more power & flexibility at the cost of
some resources.

For the record, you can do this in the dialplan using XML and sched_api
without touching a scripting language. Checkout the sched_api channel
variable on the wiki - it may give you the functionality you need.

-MC

On Wed, Aug 26, 2009 at 7:59 AM, delianSPAM <delianspam at gmail.com> wrote:

Hello Everybody!

 

1.       Scenario.

I am writing an IVR in Python that gets a destination from the calling party
(party A) and then connects to the destination (party B).

When the call is CONNECTED, I want to:

-          Receive DTMFs

-          Have a timer that can call a certain function in my script. The
script will have to play a message to party A.

-          Have a timer that can call a certain function in my script. The
script will have to drop the call.

Please notice that I want to do the things after the two parties are
connected, and not after I send the Invite to party B.

 

2.       Problem.

I will be happy to receive help on:

-          Which methods should I look for to implement this.

 

3.       Details

Here is how I connect the call currently:

session.execute("bridge","sofia/internal/" + destination_number +
"@domain.com")

 

I have tried to create a timer callback function "my_method()" using:

ivr_timer =threading.Timer(30,my_method)

This never called the function "my_method()".

 

Maybe I am wrong in using threading.Timer and the "bridge" application?
Maybe I need to create a new thread and a new timer using the API of
freeswitch, plus to use the "session.setInputCallback", plus use a
conference rather than a bridge? Can you please provide any suggestions or
examples? 

 

Thank you!

Best Regards, Delian Tashev


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From juanbackson at gmail.com  Thu Aug 27 01:56:09 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 27 Aug 2009 16:56:09 +0800
Subject: [Freeswitch-users] need help with mod_xml_odbc
Message-ID: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>

Hi,

I tried to make install mod_xml_odbc and load it in freeswitch, but I am
getting:

2009-08-28 00:46:55.848087 [CRIT] switch_loadable_module.c:871 Error Loading
module /usr/local/freeswitch/mod/mod_xml_odbc.so
**/usr/local/freeswitch/mod/mod_xml_odbc.so: invalid ELF header**

I had to manually copy the .so file from the mod_xml_odbc dir to
/usr/local/freeswitch/mod

Does anyone know what is wrong?

Thanks,
JB
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From sylver_b at yahoo.com  Wed Aug 26 17:14:55 2009
From: sylver_b at yahoo.com (sylver_b)
Date: Wed, 26 Aug 2009 17:14:55 -0700 (PDT)
Subject: [Freeswitch-users] upstream Registrar / Mirror proxy
Message-ID: <252645.39333.qm@web39706.mail.mud.yahoo.com>

Hello,

We would like to use FreeSwitch to test interoperability with our sip registrar platform - as our registrar doesnt handle NAT Traversal we would like to use FS as a mirror proxy . 

If a call goes through, FS should transparently handle nat traversal functionalities while forwarding the REGISTER/INVITE requests to our registrar server.

Below some detailed requirements:
Upper Registration is the capability of a SBC to proxy Registrations towards an upstream Registrar. While it is necessary that all SIP requests traverse the SBC for NAT continuity, Registrations should be allowed to be relayed towards our upstream Registrar while the SBC retains a copy of the AOR and masquerade on behalf of the UA that send the registration request.

Please let us know the best way to configure FS to achieve this type of configuration.

Thank you


      
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From krice at freeswitch.org  Thu Aug 27 02:02:14 2009
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 27 Aug 2009 04:02:14 -0500
Subject: [Freeswitch-users] upstream Registrar / Mirror proxy
In-Reply-To: <252645.39333.qm@web39706.mail.mud.yahoo.com>
Message-ID: <C6BBB5C6.5DDCE%krice@freeswitch.org>

FreeSWITCH is a B2BUA and NOT a proxy and will not proxy any requests
(REGISTER or INVITE)

If you want a Proxy you should look toward OpenSIPS



From: sylver_b <sylver_b at yahoo.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Wed, 26 Aug 2009 17:14:55 -0700 (PDT)
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] upstream Registrar / Mirror proxy

Hello,

We would like to use FreeSwitch to test interoperability with our sip
registrar platform - as our registrar doesnt handle NAT Traversal we would
like to use FS as a mirror proxy .

If a call goes through, FS should transparently handle nat traversal
functionalities while forwarding the REGISTER/INVITE requests to our
registrar server.

Below some detailed requirements:
Upper Registration is the capability of a SBC to proxy Registrations towards
an upstream Registrar. While it is necessary that all SIP requests traverse
the SBC for NAT continuity, Registrations should be allowed to be relayed
towards our upstream Registrar while the SBC retains a copy of the AOR and
masquerade on behalf of the UA that send the registration request.

Please let us know the best way to configure FS to achieve this type of
configuration.

Thank you

 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From krice at freeswitch.org  Thu Aug 27 02:03:50 2009
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 27 Aug 2009 04:03:50 -0500
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
Message-ID: <C6BBB626.5DDCF%krice@freeswitch.org>

You can not just copy the .so file from the directory it is just a pointer
file to the real .so ... You should always use the make install target or
make {module_name}-install target to get it properly installed



From: Juan Backson <juanbackson at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Thu, 27 Aug 2009 16:56:09 +0800
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] need help with mod_xml_odbc

Hi,

I tried to make install mod_xml_odbc and load it in freeswitch, but I am
getting:

2009-08-28 00:46:55.848087 [CRIT] switch_loadable_module.c:871 Error Loading
module /usr/local/freeswitch/mod/mod_xml_odbc.so
**/usr/local/freeswitch/mod/mod_xml_odbc.so: invalid ELF header**

I had to manually copy the .so file from the mod_xml_odbc dir to
/usr/local/freeswitch/mod

Does anyone know what is wrong?

Thanks,
JB


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From steve.kurzeja at gmail.com  Thu Aug 27 02:21:47 2009
From: steve.kurzeja at gmail.com (Steve Kurzeja)
Date: Thu, 27 Aug 2009 21:21:47 +1200
Subject: [Freeswitch-users] upstream Registrar / Mirror proxy
In-Reply-To: <C6BBB5C6.5DDCE%krice@freeswitch.org>
References: <252645.39333.qm@web39706.mail.mud.yahoo.com>
	<C6BBB5C6.5DDCE%krice@freeswitch.org>
Message-ID: <5f7152000908270221g5b37ffbdi851288874e28f95d@mail.gmail.com>

A proxy won't do what the original poster is asking for. Upper registration
is a special type of function performed by SBCs and not defined by any RFC
yet but there are drafts out there. This question comes up quite often in
various mailing lists and has been asked this list before. The answer is no
freeswitch can't be configured to do this as it stands today.

The only opensource SBC I know of that attempts to do upper registration at
the moment is OpenSBC.  Otherwise there's vendor SBCs like ACME packet,
Nextone (now Genband -  they call it Mirror Proxy mode) etc which do it.

Regards,
Steve

On Thu, Aug 27, 2009 at 9:02 PM, Ken Rice <krice at freeswitch.org> wrote:

>  FreeSWITCH is a B2BUA and NOT a proxy and will not proxy any requests
> (REGISTER or INVITE)
>
> If you want a Proxy you should look toward OpenSIPS
>
>
> ------------------------------
> *From: *sylver_b <sylver_b at yahoo.com>
> *Reply-To: *<freeswitch-users at lists.freeswitch.org>
> *Date: *Wed, 26 Aug 2009 17:14:55 -0700 (PDT)
> *To: *<freeswitch-users at lists.freeswitch.org>
> *Subject: *[Freeswitch-users] upstream Registrar / Mirror proxy
>
> Hello,
>
> We would like to use FreeSwitch to test interoperability with our sip
> registrar platform - as our registrar doesnt handle NAT Traversal we would
> like to use FS as a mirror proxy .
>
> If a call goes through, FS should transparently handle nat traversal
> functionalities while forwarding the REGISTER/INVITE requests to our
> registrar server.
>
> Below some detailed requirements:
> Upper Registration is the capability of a SBC to proxy Registrations
> towards an upstream Registrar. While it is necessary that all SIP requests
> traverse the SBC for NAT continuity, Registrations should be allowed to be
> relayed towards our upstream Registrar while the SBC retains a copy of the
> AOR and masquerade on behalf of the UA that send the registration request.
>
> Please let us know the best way to configure FS to achieve this type of
> configuration.
>
> Thank you
>
>
> ------------------------------
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From se02005-mec at yahoo.com  Thu Aug 27 02:46:47 2009
From: se02005-mec at yahoo.com (Merle J. Ebbert)
Date: Thu, 27 Aug 2009 02:46:47 -0700
Subject: [Freeswitch-users] Can an Instant Message (IM) be sent to a cell
	phone with FS?
Message-ID: <4A965607.7000003@yahoo.com>

Can an Instant Message (IM) be sent to a cell phone with FS?
Any of these:
    AOL Instant Messenger (AIM)
    Yahoo (Y!)
    Blackberry Messenger
    Google Talk
    Windows Live Messenger

Thanks,
Merle




From se02005-mec at yahoo.com  Thu Aug 27 02:50:54 2009
From: se02005-mec at yahoo.com (Merle J. Ebbert)
Date: Thu, 27 Aug 2009 02:50:54 -0700
Subject: [Freeswitch-users] Can a chat message be sent to a cell phone with
	FS?
Message-ID: <4A9656FE.9030504@yahoo.com>

Can a chat message be sent to a cell phone with FS?

Thanks,
Merle



From delianspam at gmail.com  Thu Aug 27 03:23:49 2009
From: delianspam at gmail.com (delianSPAM)
Date: Thu, 27 Aug 2009 13:23:49 +0300
Subject: [Freeswitch-users] Get the Session State
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAI0pCHFyxT5Mn9pdicKxxSMBAAAAAA==@gmail.com>

Hello Everybody!

 

How do you get the call state in Python?

 

I have tried:

.

session.answer()

state_result=str(session.getVariable("state"))

console_log("debug",state_result)

.

 

But it returns: "None"

 

Thank you!

 

Best Regards, Delian Tashev

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From juanbackson at gmail.com  Thu Aug 27 03:37:41 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 27 Aug 2009 18:37:41 +0800
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <C6BBB626.5DDCF%krice@freeswitch.org>
References: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
	<C6BBB626.5DDCF%krice@freeswitch.org>
Message-ID: <27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>

Hi,

ok, the module can be loaded, but it now complains about odbc.  I can't find
anything missing in my odbc.ini.  Could someone please point me to the right
direction?

2009-08-28 02:22:35.670284 [ERR] switch_odbc.c:188 STATE: IM002 CODE 201
ERROR: [unixODBC]Missing server name, port, or database name in call to
CC_connect.

2009-08-28 02:22:35.670312 [CRIT] mod_xml_odbc.c:577 Cannot Open ODBC
Database!
2009-08-28 02:22:35.670319 [ERR] mod_xml_odbc.c:612 Unable to load xml_odbc
config file
2009-08-28 02:22:35.670326 [CRIT] switch_loadable_module.c:871 Error Loading
module /usr/local/freeswitch/mod/mod_xml_odbc.so
**Module load routine returned an error**


I already have xml_odbc.conf
 cat /etc/odbc.ini

; begin odbc.ini
[ODBC Data Sources]
test = PostgreSQL ODBC Driver

[test]
Driver = /usr/local/lib/libodbcpsql.so
Description = PostgreSQL Data Source
DSN = test
Servername = 192.168.1.133
Server = 192.168.1.133
Port = 5432
;Socket = 4096
Protocol = 6.4 # 7.2 or other values
UserName = root
Password = JdqB-S
Database = freeswitch
ReadOnly = no
ServerType = Postgres
FetchBufferSize = 99
ServerOptions =
ConnectOptions =
;Options = 3
Trace = 0
TraceFile = /var/log/PostgreSQL_test_trace.log
Debug = 0
DebugFile = /var/log/PostgreSQL_test_debug.log
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From odermann at googlemail.com  Thu Aug 27 04:06:08 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 13:06:08 +0200
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <5e414ed0908260910h55122d77o5013ef1795d0c75@mail.gmail.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
	<5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
	<274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>
	<5e414ed0908260910h55122d77o5013ef1795d0c75@mail.gmail.com>
Message-ID: <5e414ed0908270406m628342d5ya0e9f25514ed09d2@mail.gmail.com>

sorry, but i do not know i which category i have to set this problem.
could you help me with that?



From juanbackson at gmail.com  Thu Aug 27 04:40:37 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 27 Aug 2009 19:40:37 +0800
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>
References: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
	<C6BBB626.5DDCF%krice@freeswitch.org>
	<27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>
Message-ID: <27c25bc40908270440o5f1b7a6csf675716677baef48@mail.gmail.com>

Hi,

Finally, I got xml_odbc running, but it does not really work well for me.  I
am getting:
2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:325 Stopped rendering
template, called xml_odbc_render_template more than [32] times, probably
looping.
2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:408 Something went horribly
wrong while generating an XML template!

My config is:
<configuration name="xml_odbc.conf" description="XML ODBC Configuration">

  <settings>
        <param name="binding" value="directory"/>
    <param name="odbc-dsn" value="class5_odbc:root:JdqB-S"/>
    <param name="debug" value="true"/>
    <param name="keep-files-around" value="true"/>
  </settings>

  <templates>

    <template name="default">
      <document type="freeswitch/xml">
        <section name="directory">
     <xml-odbc-do name="query" on-empty-result-break-to="not-found"
value="SELECT enabled, sip_password FROM agent WHERE sip_user =  ${user}"/>
        <domain name="${domain}">
          <user id="${user}">
            <params>
              <xml-odbc-do name="query" value="SELECT sip_password FROM
agent WHERE sip_user = ${user}">
                <param name="password", value="${sip_password}"/>
              </xml-odbc-do>
            </params>
              </user>
        </domain>


        </section>
      </document>
    </template>

  </templates>

</configuration>


Since these two queries get data from the same table, I tried to merge them,
but could not get it to work.

Anyone has any idea?

Thanks,
JB
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From foxb at abv.bg  Thu Aug 27 04:58:53 2009
From: foxb at abv.bg (Hristo Benev)
Date: Thu, 27 Aug 2009 14:58:53 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <877409356.2721.1251374341513.JavaMail.apache@mail23.abv.bg>

 
Bojnour,

I'll send a trace ASAP.

What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)

I'm really interested to see the updates of the project.

Thank you for the good tutorial.

Hristo


 >-------- ?????????? ????? --------
 >??:  rod 
 >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >??: freeswitch-users at lists.freeswitch.org
 >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST

 >Hi Hristo,
 >
 >I'm the author of this setup and wiki page. I did a lot of modifications 
 >on this setup (alternative routing if failure essentially) but don't 
 >have too much time to update the wiki.
 >
 >May you please send me an ngrep trace when you call 1000:
 >
 >ngrep -d any -nn -i '1000' port 5060 -W byline 
 >
 >I will check what's happening.
 >Do you have an entry for 1000 in your mysql database ?
 >
 >regards,
 >rod
 >
 >Hristo Benev a ?crit :
 >>  
 >> It seems that the problem is on kamailio configuration.
 >> Will ask on their list.
 >>
 >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
 >>
 >> I have it setup as friend in asterisk, but still ???
 >>
 >> Any ideas?
 >>
 >> Thanks,
 >>
 >>
 >>  >-------- ?????????? ????? --------
 >>  >??:  Hristo Benev 
 >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >??: freeswitch-users at lists.freeswitch.org
 >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
 >>
 >>  > I think that the problem is here:
 >>  >-------------------------
 >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
 >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
 >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
 >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
 >>  >--------------------------
 >>  >
 >>  >Actually Regex FAIL
 >>  >
 >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
 >>  >Here is my  default.xml:
 >>  >----------------
 >>  >
 >>  >
 >>  >
 >>  >  
 >>  >
 >>  >     
 >>  >      
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >      
 >>  >    
 >>  >
 >>  >   
 >>  >
 >>  >  
 >>  >
 >>  >    
 >>  >      
 >>  >        
 >>  >        
 >>  >        
 >>  >        
 >>  >        
 >>  >        
 >>  >        
 >>  >        
 >>  >      
 >>  >    
 >>  >
 >>  >  
 >>  >
 >>  >
 >>  >
 >>  >--------------------------
 >>  >
 >>  >
 >>  >
 >>  > >-------- ?????????? ????? --------
 >>  > >??:  Brian West 
 >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  > >??: freeswitch-users at lists.freeswitch.org
 >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
 >>  >
 >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
 >>  >You have to process all 302's in the dialplan. 
 >>  >Set this on your sofia profile
 >>  >You can set these variables sip_redirect_profile,
 >>  >sip_redirect_context,
 >>  >sip_redirect_dialplan,
 >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
 >>  >sip_redirected_to,
 >>  >sip_redirect_contact_user_%d,
 >>  >sip_redirect_contact_host_%d,
 >>  >sip_redirect_contact_params_%d,
 >>  >sip_redirect_dialstring_%d,
 >>  >sip_redirect_dialstring,
 >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
 >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
 >>  > >
 >>  >
 >>  >_______________________________________________
 >>  >FreeSWITCH-users mailing list
 >>  >FreeSWITCH-users at lists.freeswitch.org
 >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >http://www.freeswitch.org
 >>  >
 >>
 >> _______________________________________________
 >> FreeSWITCH-users mailing list
 >> FreeSWITCH-users at lists.freeswitch.org
 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >> http://www.freeswitch.org
 >>   
 >
 >_______________________________________________
 >FreeSWITCH-users mailing list
 >FreeSWITCH-users at lists.freeswitch.org
 >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >http://www.freeswitch.org
 >



From rs at runsolutions.com  Thu Aug 27 05:10:36 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Thu, 27 Aug 2009 14:10:36 +0200
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
	<191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>
Message-ID: <225AA30C-8455-4D4A-A888-5AC7D8C64E7B@runsolutions.com>

very offtopic, but, if this is not to personally, but how do you get  
to spend literally 8-12 hours a day on a conference?

:-)

best

-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-
Email: rs at runsolutions.com
tel: 625 40 32 08

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 26, 2009, at 10:45 PM, Anthony Minessale wrote:

> There is no bug, it's all dependent on your network conditions.
> I spend literally 8-12 hours a day on a conference and there is no  
> delay.
>
> The important param is
> <param name="rtp-autoflush-during-bridge" value="true"/>
>
> in the sofia profile in question.
>
> if you have delay with that in place, then it's probably not FS
>
>
> On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd at suspiria.net> wrote:
> Hi,
>
>
> I am on a quite recent version (i assume):
>
>
> FreeSWITCH Version 1.0.trunk (14461)
>
>
> Should the bug be fixed in this revision ? What config settings  
> would a have to check to limit delay (even at the cost of reduction  
> in quality).
>
>
> Thanks
>
>
> Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org 
> ] Im Auftrag von Anthony Minessale
> Gesendet: Mittwoch, 26. August 2009 20:57
> An: freeswitch-users at lists.freeswitch.org
> Betreff: Re: [Freeswitch-users] delay buildup in conference
>
>
> which revision are you on?
> The defaults on the latest code and examples should be configured to  
> minimize delay.
> Some of the older revisions built up some delay issues from udp  
> buffering when timers were not synced.
>
> On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net>  
> wrote:
>
> When running conferences with users dialed in from a PSTN gateway  
> (SIP) and directly from remote SIP endpoints there is an ever longer  
> buildup in delay, reaching up to multiple seconds. Is there any way  
> to limit the delay ?
>
>
> I am not 100% sure whether the delays is caused by the SIP jitter  
> buffer of freeswitch or directly by the conference module.
>
>
> Any advice?
>
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
>
> -- 
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From leon at scarlet-internet.nl  Thu Aug 27 05:57:20 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 27 Aug 2009 14:57:20 +0200
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <27c25bc40908270440o5f1b7a6csf675716677baef48@mail.gmail.com>
References: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
	<C6BBB626.5DDCF%krice@freeswitch.org>
	<27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>
	<27c25bc40908270440o5f1b7a6csf675716677baef48@mail.gmail.com>
Message-ID: <352125CC-DE10-4665-AD62-ABB3B4B2A1DE@scarlet-internet.nl>

Hi Juan,

Perhaps it loops because you didn't include the "not-found" template ?  
Actually, I see there's a bug in the example xml_odbc.conf.xml file  
where it's defined with an underscore instead of a dash, will change  
that tonight..

The not-found template needs to be specified as a template in the  
configuration. I think I'll define that template statically in the  
module itself later.
Because it's the 'fall-through' template when it can't find a  
template, you get a loop.

So, something like this should probably work for you:

<templates>

   <template name="default">
     <xml-odbc-do name="query" on-empty-result-break-to="not-found"  
value="SELECT sip_password FROM agent WHERE sip_user = '${user}';"/>
     <document type="freeswitch/xml">
       <section name="directory">
         <domain name="${domain}">
           <user id="${user}">
             <params>
               <param name="password", value="${sip_password}"/>
             </params>
           </user>
         </domain>
       </section>
     </document>
   </template>

   <template name="not-found">
     <document type="freeswitch/xml">
       <section name="result">
         <result status="not found"/>
       </section>
     </document>
   </template>

</templates>

(I didn't include the enabled field in your select statement, as you  
don't use it later, perhaps you need it in the where clause ?)

Also, note that this way the template will also be used at startup  
when FS tries to get a list of all ACL's - I believe for something  
else as well, have to check it - but those lookups probably don't give  
a ${user} so will render the "not-found" anyway..

One last thing, you didn't have ${user} enclosed in quotes in your  
query, so if no ${user} was given with the lookup to the module, then  
your query becomes invalid, which probably breaks things as well.

Let me know if it works..

regards,

Leon



On Aug 27, 2009, at 1:40 PM, Juan Backson wrote:

> Hi,
>
> Finally, I got xml_odbc running, but it does not really work well  
> for me.  I am getting:
> 2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:325 Stopped  
> rendering template, called xml_odbc_render_template more than [32]  
> times, probably looping.
> 2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:408 Something went  
> horribly wrong while generating an XML template!
>
> My config is:
> <configuration name="xml_odbc.conf" description="XML ODBC  
> Configuration">
>
>   <settings>
>         <param name="binding" value="directory"/>
>     <param name="odbc-dsn" value="class5_odbc:root:JdqB-S"/>
>     <param name="debug" value="true"/>
>     <param name="keep-files-around" value="true"/>
>   </settings>
>
>   <templates>
>
>     <template name="default">
>       <document type="freeswitch/xml">
>         <section name="directory">
>      <xml-odbc-do name="query" on-empty-result-break-to="not-found"  
> value="SELECT enabled, sip_password FROM agent WHERE sip_user =  $ 
> {user}"/>
>         <domain name="${domain}">
>           <user id="${user}">
>             <params>
>               <xml-odbc-do name="query" value="SELECT sip_password  
> FROM agent WHERE sip_user = ${user}">
>                 <param name="password", value="${sip_password}"/>
>               </xml-odbc-do>
>             </params>
>               </user>
>         </domain>
>
>
>         </section>
>       </document>
>     </template>
>
>   </templates>
>
> </configuration>
>
>
> Since these two queries get data from the same table, I tried to  
> merge them, but could not get it to work.
>
> Anyone has any idea?
>
> Thanks,
> JB
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From fraunhofer.lists.freeswitch-001 at traced.net  Thu Aug 27 06:15:21 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Thu, 27 Aug 2009 15:15:21 +0200
Subject: [Freeswitch-users] memory leak
Message-ID: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>

Hello *,

a memory leak showed up in our loadtests. It's (still) the same setup as in the
http://jira.freeswitch.org/browse/MODSOFIA-22 bugfix.

One thing I'd like to add is that "fsctl shutdown restart" was unable
to shutdown freeswitch.
The last line printed is "switch_core_memory.c:567 Stopping memory pool queue."

attached file is a the collected and graphed output of some "ps waux"
command. "sz" should be
in "physical pages" (2k?).

I rerun the test and this time it coredumped trying to malloc() space
for some playback.

Anything else you need (full backtraces?) to dig into it?

Cheers
  Beni.
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From anatoliy at kounitskiy.com  Thu Aug 27 06:19:27 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Thu, 27 Aug 2009 16:19:27 +0300
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
Message-ID: <1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>

Just for information - the idea :
A---calls---> B ---att_xfer---> C    , B hangs C up and goes back to A
( I`m sorry C is not answering :) )

dialplan:
-----------
 <extension name="local_number">
    <condition field="${toll_allow}" expression="local"/>
    <condition field="destination_number" expression="^(\d{3})$">
      <action application="set" data="dialed_extension=$1"/>
      <action application="export" data="dialed_extension=$1"/>
      <action application="bind_meta_app" data="1 b s
execute_extension::dx XML features"/>
      <action application="bind_meta_app" data="2 b s
record_session::$${base_dir}/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav"/>
      <action application="bind_meta_app" data="3 b s
execute_extension::cf XML features"/>
      <action application="bind_meta_app" data="4 b s
execute_extension::attented_xfer XML features"/>
      <action application="set" data="transfer_ringback=$${hold_music}"/>
      <action application="set" data="call_timeout=10"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="set"
data="continue_on_fail=USER_NOT_REGISTERED,SUBSCRIBER_ABSENT"/>
      <action application="info"/>
      <action application="set"
data="bringback=${user_data(${dialed_extension}@${domain_name} var
cringback )}"/>
      <action application="set" data="ringback=${${bringback}}"/>
      <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>
      <action application="info"/>
      ...
    </condition>
  </extension>

features.xml
---------------------
  <extension name="attented_xfer">
      <condition field="${toll_allow}" expression="local"/>
      <condition field="destination_number" expression="^attented_xfer$">
        <action application="set" data="continue_on_fail=true"/>
        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 30000 #"/>
        <action application="set"
data="bringback=${user_data(${attxfer_callthis}@${domain_name} var
cringback )}"/>
        <action application="set" data="ringback=${${bringback}}"/>
        <action application="info"/>
        <action application="set" data="origination_cancel_key=#"/>
        <action application="att_xfer"
data="user/${attxfer_callthis}@${domain_name}"/>
      </condition>
    </extension>

After testing the patch the results are:

1) B answers the call from A and executes the feature code *4
(attented_xfer), sends the desired number ( in this case the number of
C). A goes to MusicOnHold, B is waiting for C to pick up - B sends # -
sends SIP CANCEL to C and SIP BUY to A - so all calls are dropped.

2) B answers the call from A and executes the feature code *4
(attented_xfer), but without entering the number of C - so the read
aplication timeouts after 30 sec - after that att_xfer is executed
with empty string (nothing is entered) - B and A are bridged together.

On the console without entering the extension of C:
2009-08-27 16:00:58.929756 [NOTICE] switch_core_session.c:1576 Execute
set(origination_cancel_key=#)
EXECUTE sofia/internal/sip:102 at 10.17.4.107:5060 set(origination_cancel_key=#)
2009-08-27 16:00:58.929756 [DEBUG] mod_dptools.c:748
sofia/internal/sip:userB at 10.17.4.107:5060 SET
[origination_cancel_key]=[#]
2009-08-27 16:00:58.929756 [NOTICE] switch_core_session.c:1576 Execute
att_xfer(user/${attxfer_callthis}@${domain_name})
EXECUTE sofia/internal/sip:102 at 10.17.4.107:5060 att_xfer(user/@1.1.1.1)
2009-08-27 16:00:58.929756 [WARNING] mod_dptools.c:2373 Can't find
user [@1.1.1.1]
2009-08-27 16:00:58.929756 [ERR] switch_ivr_originate.c:1527 Cannot
create outgoing channel of type [user] cause: [SUBSCRIBER_ABSENT]
2009-08-27 16:00:58.929756 [DEBUG] switch_ivr_originate.c:2167
Originate Resulted in Error Cause: 20 [SUBSCRIBER_ABSENT]
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_play_say.c:1402 done playing file
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:231
sofia/internal/userA at 1.1.1.1 receive message [BRIDGE]
2009-08-27 16:00:58.949842 [DEBUG] switch_core_session.c:630 Send
signal sofia/internal/userA at 1.1.1.1 [BREAK]
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:233 Send signal
sofia/internal/sip:userB at 10.17.4.107:5060 [BREAK]
2009-08-27 16:00:58.949842 [DEBUG] switch_ivr_bridge.c:231
sofia/internal/sip:userB at 10.17.4.107:5060 receive message [BRIDGE]
2009-08-27 16:00:58.949842 [DEBUG] switch_core_session.c:630 Send
signal sofia/internal/sip:userB at 10.17.4.107:5060 [BREAK]
2009-08-27 16:00:58.954052 [DEBUG] switch_ivr_bridge.c:233 Send signal
sofia/internal/userA at 1.1.1.1 [BREAK]
2009-08-27 16:01:22.881501 [DEBUG] sofia.c:3302 Channel
sofia/internal/userA at 1.1.1.1 entering state [calling][0]
2009-08-27 16:01:22.886122 [DEBUG] sofia.c:3302 Channel
sofia/internal/userA at 1.1.1.1 entering state [ready][200]

I'll try to summarize my idea - could it be possible when the # (in
the att_xfer) is executed to behave as if you're trying to make
attended transfer to non existing subscriber( SUBSCRIBER_ABSENT ), so
B can be bridged back with A.

Thank you in advance,
Anatoliy Kounitskiy

On Wed, Aug 26, 2009 at 11:11 PM, Michael Collins<msc at freeswitch.org> wrote:
>
>
> On Wed, Aug 26, 2009 at 12:59 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> i added a patch to attempt to do this so try adding
>
> Nice work!
>>
>> {origination_cancel_key=#} before the dial string
>> or
>> <action application="set" data="origination_cancel_key=#"/>
>> before you bridge.
>
> Anatoliy,
>
> Please try this and let us know if it works for you as expected. If so,
> please write up the exact procedure that you used and include dialplan
> examples. I'd like to get this on the wiki while it's fresh in my mind. If
> you feel comfortable editing the wiki on your own then just let me know so
> that we can coordinate efforts.
>
> Thanks,
> MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From leon at scarlet-internet.nl  Thu Aug 27 06:31:21 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 27 Aug 2009 15:31:21 +0200
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <352125CC-DE10-4665-AD62-ABB3B4B2A1DE@scarlet-internet.nl>
References: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
	<C6BBB626.5DDCF%krice@freeswitch.org>
	<27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>
	<27c25bc40908270440o5f1b7a6csf675716677baef48@mail.gmail.com>
	<352125CC-DE10-4665-AD62-ABB3B4B2A1DE@scarlet-internet.nl>
Message-ID: <F062600C-BDA4-4B70-B684-5837EF5E4066@scarlet-internet.nl>

Made a typo in the <param>, you have to leave out the comma..

regards,

Leon


On Aug 27, 2009, at 2:57 PM, Leon de Rooij wrote:

> Hi Juan,
>
> Perhaps it loops because you didn't include the "not-found" template ?
> Actually, I see there's a bug in the example xml_odbc.conf.xml file
> where it's defined with an underscore instead of a dash, will change
> that tonight..
>
> The not-found template needs to be specified as a template in the
> configuration. I think I'll define that template statically in the
> module itself later.
> Because it's the 'fall-through' template when it can't find a
> template, you get a loop.
>
> So, something like this should probably work for you:
>
> <templates>
>
>   <template name="default">
>     <xml-odbc-do name="query" on-empty-result-break-to="not-found"
> value="SELECT sip_password FROM agent WHERE sip_user = '${user}';"/>
>     <document type="freeswitch/xml">
>       <section name="directory">
>         <domain name="${domain}">
>           <user id="${user}">
>             <params>
>               <param name="password", value="${sip_password}"/>
>             </params>
>           </user>
>         </domain>
>       </section>
>     </document>
>   </template>
>
>   <template name="not-found">
>     <document type="freeswitch/xml">
>       <section name="result">
>         <result status="not found"/>
>       </section>
>     </document>
>   </template>
>
> </templates>
>
> (I didn't include the enabled field in your select statement, as you
> don't use it later, perhaps you need it in the where clause ?)
>
> Also, note that this way the template will also be used at startup
> when FS tries to get a list of all ACL's - I believe for something
> else as well, have to check it - but those lookups probably don't give
> a ${user} so will render the "not-found" anyway..
>
> One last thing, you didn't have ${user} enclosed in quotes in your
> query, so if no ${user} was given with the lookup to the module, then
> your query becomes invalid, which probably breaks things as well.
>
> Let me know if it works..
>
> regards,
>
> Leon
>
>
>
> On Aug 27, 2009, at 1:40 PM, Juan Backson wrote:
>
>> Hi,
>>
>> Finally, I got xml_odbc running, but it does not really work well
>> for me.  I am getting:
>> 2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:325 Stopped
>> rendering template, called xml_odbc_render_template more than [32]
>> times, probably looping.
>> 2009-08-28 03:33:47.459383 [ERR] mod_xml_odbc.c:408 Something went
>> horribly wrong while generating an XML template!
>>
>> My config is:
>> <configuration name="xml_odbc.conf" description="XML ODBC
>> Configuration">
>>
>>  <settings>
>>        <param name="binding" value="directory"/>
>>    <param name="odbc-dsn" value="class5_odbc:root:JdqB-S"/>
>>    <param name="debug" value="true"/>
>>    <param name="keep-files-around" value="true"/>
>>  </settings>
>>
>>  <templates>
>>
>>    <template name="default">
>>      <document type="freeswitch/xml">
>>        <section name="directory">
>>     <xml-odbc-do name="query" on-empty-result-break-to="not-found"
>> value="SELECT enabled, sip_password FROM agent WHERE sip_user =  $
>> {user}"/>
>>        <domain name="${domain}">
>>          <user id="${user}">
>>            <params>
>>              <xml-odbc-do name="query" value="SELECT sip_password
>> FROM agent WHERE sip_user = ${user}">
>>                <param name="password", value="${sip_password}"/>
>>              </xml-odbc-do>
>>            </params>
>>              </user>
>>        </domain>
>>
>>
>>        </section>
>>      </document>
>>    </template>
>>
>>  </templates>
>>
>> </configuration>
>>
>>
>> Since these two queries get data from the same table, I tried to
>> merge them, but could not get it to work.
>>
>> Anyone has any idea?
>>
>> Thanks,
>> JB
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From woodydickson at gmail.com  Thu Aug 27 06:49:35 2009
From: woodydickson at gmail.com (Woody Dickson)
Date: Thu, 27 Aug 2009 21:49:35 +0800
Subject: [Freeswitch-users] mod_limit and memcache
Message-ID: <a44688a0908270649y6cde3c26pb739c7d144e1b86b@mail.gmail.com>

Hello,

I read something that talks about using memcache for mod_limit before.   Is
it something that is available now?

If I have multiple instances of freeswitch that need to share the same limit
status, it there any existing solution?

If no existing solution is available, what is the best way to go about
modifying mod_limit to accomplish limiting for multiple freeswitch servers
together?

Thanks,
Woody
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From juanbackson at gmail.com  Thu Aug 27 06:53:42 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Thu, 27 Aug 2009 21:53:42 +0800
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <F062600C-BDA4-4B70-B684-5837EF5E4066@scarlet-internet.nl>
References: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
	<C6BBB626.5DDCF%krice@freeswitch.org>
	<27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>
	<27c25bc40908270440o5f1b7a6csf675716677baef48@mail.gmail.com>
	<352125CC-DE10-4665-AD62-ABB3B4B2A1DE@scarlet-internet.nl>
	<F062600C-BDA4-4B70-B684-5837EF5E4066@scarlet-internet.nl>
Message-ID: <27c25bc40908270653q47345f17s8d78e99ee5cc4b35@mail.gmail.com>

Hi Leon,

Thanks for your help.

I have changed it according to your comment but I am still getting the
looping error.

Would you please take a look see what else I did wrong?

Also, sip_user is an integer field, so I can't really use ''.  Is there
anyway to get around that?


<configuration name="xml_odbc.conf" description="XML ODBC Configuration">

  <settings>
        <param name="binding" value="directory"/>
    <param name="odbc-dsn" value="class5_odbc:root:abcd"/>
    <param name="debug" value="true"/>
    <param name="keep-files-around" value="true"/>
  </settings>

  <templates>

    <template name="default">
      <document type="freeswitch/xml">
        <section name="directory">
     <xml-odbc-do name="query" on-empty-result-break-to="not-found"
value="SELECT enabled, sip_password FROM agent WHERE sip_user =  ${user} and
enabled='t'"/>
        <domain name="${domain}">
          <user id="${user}">
            <params>
              <xml-odbc-do name="query" value="SELECT sip_password FROM
agent WHERE sip_user = ${user}">
                <param name="password" value="${sip_password}"/>
              </xml-odbc-do>
            </params>
              </user>
        </domain>


        </section>
      </document>
    </template>


    <template>

        <document type="freeswitch/xml">
            <section name="result">
                <result status="not found"/>
            </section>
        </document>
    </template>
  </templates>

</configuration>
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From fraunhofer.lists.freeswitch-001 at traced.net  Thu Aug 27 07:08:23 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Thu, 27 Aug 2009 16:08:23 +0200
Subject: [Freeswitch-users] fscore mutex locking question
Message-ID: <a37510170908270708g1e3236ao1f16a6835a87e8d7@mail.gmail.com>

Hello *,

while looking at the code i came across a region of code which is
unclear to me regarding locking issues.
One example is switch_ivr_broadcast in switch_ivr_async.c. This should
be the function called by
uuid_broadcast() and others.

in line 2341 it tries to queue an event to the bleg if it has to...
-----
.   if ((flags & SMF_ECHO_BLEG) && (other_uuid =
switch_channel_get_variable(channel, SWITCH_SIGNAL_BOND_VARIABLE))
.   .   && (other_session = switch_core_session_locate(other_uuid))) {

---
switch_core_session_locate() does a "readonly trylock" on the channel
mutex returning NULL if it's unable to
aquire the lock, which brings up the following question:
If some other thread is currently holding a writelock on the channel,
the broadcast is not queued and not retried at a later time at all?

I guess it's pretty easy to cause some unexpected behaviour using some
endless loop calling "uuid_setvar" or some other race condition where
the channel-mutex is write-locked while calling uuid_broadcast (eg.
uuid_media?).

Could this lead to a problem in "real live" scenarios, or are there
other countermeasures despite "chances are one in a million that you
hit that small time frame"?

thx in advance
  Beni.



From anthony.minessale at gmail.com  Thu Aug 27 07:32:27 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 09:32:27 -0500
Subject: [Freeswitch-users] delay buildup in conference
In-Reply-To: <225AA30C-8455-4D4A-A888-5AC7D8C64E7B@runsolutions.com>
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
	<191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com>
	<225AA30C-8455-4D4A-A888-5AC7D8C64E7B@runsolutions.com>
Message-ID: <191c3a030908270732q114fc3d4h5f2c1f407182e43e@mail.gmail.com>

Our whole development team works from 6 different states so we use a 24x7
conference as part of our virtual office.
There is a public conference for freeswitch users as well.

http://conference.freeswitch.org
sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>



On Thu, Aug 27, 2009 at 7:10 AM, Raimund Sacherer <rs at runsolutions.com>wrote:

> very offtopic, but, if this is not to personally, but how do you get to
> spend literally 8-12 hours a day on a conference?
> :-)
>
> best
>
> --
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
> Email: rs at runsolutions.com
> tel: 625 40 32 08
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
> On Aug 26, 2009, at 10:45 PM, Anthony Minessale wrote:
>
> There is no bug, it's all dependent on your network conditions.
> I spend literally 8-12 hours a day on a conference and there is no delay.
>
> The important param is
> <param name="rtp-autoflush-during-bridge" value="true"/>
>
> in the sofia profile in question.
>
> if you have delay with that in place, then it's probably not FS
>
>
> On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd at suspiria.net> wrote:
>
>>  Hi,
>>
>>
>> I am on a quite recent version (i assume):
>>
>>
>> FreeSWITCH Version 1.0.trunk (14461)
>>
>>
>> Should the bug be fixed in this revision ? What config settings would a
>> have to check to limit delay (even at the cost of reduction in quality).
>>
>>
>> Thanks
>>
>>
>> *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto:
>> freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Anthony
>> Minessale
>> *Gesendet:* Mittwoch, 26. August 2009 20:57
>> *An:* freeswitch-users at lists.freeswitch.org
>> *Betreff:* Re: [Freeswitch-users] delay buildup in conference
>>
>>
>> which revision are you on?
>> The defaults on the latest code and examples should be configured to
>> minimize delay.
>> Some of the older revisions built up some delay issues from udp buffering
>> when timers were not synced.
>>
>>  On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net> wrote:
>>
>> When running conferences with users dialed in from a PSTN gateway (SIP)
>> and directly from remote SIP endpoints there is an ever longer buildup in
>> delay, reaching up to multiple seconds. Is there any way to limit the delay
>> ?
>>
>>
>> I am not 100% sure whether the delays is caused by the SIP jitter buffer
>> of freeswitch or directly by the conference module.
>>
>>
>> Any advice?
>>
>>
>> Thanks
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Aug 27 07:38:58 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 09:38:58 -0500
Subject: [Freeswitch-users] memory leak
In-Reply-To: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
Message-ID: <191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>

600k is not a leak? FS can use as much as a gig of ram or more depending on
what you are doing.
you may want to install a fresh copy of FS, removing all your old files etc
and make sure they build clean.
We also have not had much luck running on ubuntu which is more of a desktop
centric OS.
I recommend you try your application on 64 bit CentOS which is the platform
all of our paid customers use.





On Thu, Aug 27, 2009 at 8:15 AM, Benedikt Fraunhofer <
fraunhofer.lists.freeswitch-001 at traced.net> wrote:

> Hello *,
>
> a memory leak showed up in our loadtests. It's (still) the same setup as in
> the
> http://jira.freeswitch.org/browse/MODSOFIA-22 bugfix.
>
> One thing I'd like to add is that "fsctl shutdown restart" was unable
> to shutdown freeswitch.
> The last line printed is "switch_core_memory.c:567 Stopping memory pool
> queue."
>
> attached file is a the collected and graphed output of some "ps waux"
> command. "sz" should be
> in "physical pages" (2k?).
>
> I rerun the test and this time it coredumped trying to malloc() space
> for some playback.
>
> Anything else you need (full backtraces?) to dig into it?
>
> Cheers
>  Beni.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From rupa at rupa.com  Thu Aug 27 07:42:38 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Thu, 27 Aug 2009 09:42:38 -0500
Subject: [Freeswitch-users] mod_limit and memcache
In-Reply-To: <a44688a0908270649y6cde3c26pb739c7d144e1b86b@mail.gmail.com>
References: <a44688a0908270649y6cde3c26pb739c7d144e1b86b@mail.gmail.com>
Message-ID: <b39273530908270742i1e018ed2x953203e4eee7bb2f@mail.gmail.com>

limit and memcache haven't been introduced to each other yet -- it is
on my (semi-long) list of things to do.

If you want it you can:

1) do it yourself and submit the patches
2) open a jira and hope someone does it
3) open a jira + bounty and someone will probably do it

It will get done eventually, just hasn't been a itch for ME to scratch yet.


To do it:

1) I need to make it possible to call inc/dec methods of mod_memcache
to support an expiration time.

2) mod_limit.c - use the hash limit as a guide

Initial pitfalls:

hash limits concurrent access/modification of the hash and by
implication limit_hash_item_t (hash data) by using a mutex.  We can't
mutex across FS instances.

So perhaps split up limit_has_item_t and spread it across multiple
keys.  So instead of one key marked as realm_id, we could have
realm_id_total_usage realm_id_rate_usage and realm_id_last_check.

This does mean that rate_usage and total_usage can inc/dec independent
of each other, but I think the logic will still be "ok" *IF* we
remember to decrement earlier incremented items in the event a later
item is failed.   (so, say we increment rate but fail on total we need
to remember to decrement rate so that we have no net effect on the
counters)

Alternatively, we could use CAS support and pull the limit_hash_item_t
item from memcache, twiddle it and then try to put it back only if the
check info is the same (no one else has changed the entry).  If the
entry has changed, pull the new version, do the limit logic, and try
again.  Loop that a few times until you succeed or give up.  Problem
is that CAS needs to be explicitly turned on in memcache (some distros
compile with it off), is relatively new in memcache (hint: may have
issues) and has some performance/memory downsides though by how much
I'm not sure.

Thoughts?

On Thu, Aug 27, 2009 at 8:49 AM, Woody Dickson<woodydickson at gmail.com> wrote:
> Hello,
>
> I read something that talks about using memcache for mod_limit before.?? Is
> it something that is available now?
>
> If I have multiple instances of freeswitch that need to share the same limit
> status, it there any existing solution?
>
> If no existing solution is available, what is the best way to go about
> modifying mod_limit to accomplish limiting for multiple freeswitch servers
> together?
>
> Thanks,
> Woody
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
-Rupa



From sylver_b at yahoo.com  Thu Aug 27 07:56:23 2009
From: sylver_b at yahoo.com (sylver_b)
Date: Thu, 27 Aug 2009 07:56:23 -0700 (PDT)
Subject: [Freeswitch-users] Re :  upstream Registrar / Mirror proxy
In-Reply-To: <5f7152000908270221g5b37ffbdi851288874e28f95d@mail.gmail.com>
References: <252645.39333.qm@web39706.mail.mud.yahoo.com>
	<C6BBB5C6.5DDCE%krice@freeswitch.org>
	<5f7152000908270221g5b37ffbdi851288874e28f95d@mail.gmail.com>
Message-ID: <608767.97175.qm@web39705.mail.mud.yahoo.com>

Thanks for the feedback

Steve: we are using Genband/Nextone MSX , but we're looking for a cheaper alternative to avoid using our vports for this purpose.

We'll check OpenSBC .. thanks



________________________________
De : Steve Kurzeja <steve.kurzeja at gmail.com>
? : freeswitch-users at lists.freeswitch.org
Envoy? le : Jeudi, 27 Ao?t 2009, 10h21mn 47s
Objet : Re: [Freeswitch-users] upstream Registrar / Mirror proxy

A proxy won't do what the original poster is asking for. Upper registration is a special type of function performed by SBCs and not defined by any RFC yet but there are drafts out there. This question comes up quite often in various mailing lists and has been asked this list before. The answer is no freeswitch can't be configured to do this as it stands today.

The only opensource SBC I know of that attempts to do upper registration at the moment is OpenSBC.  Otherwise there's vendor SBCs like ACME packet, Nextone (now Genband -  they call it Mirror Proxy mode) etc which do it.

Regards,
Steve


On Thu, Aug 27, 2009 at 9:02 PM, Ken Rice <krice at freeswitch.org> wrote:

FreeSWITCH is a B2BUA and NOT a proxy and will not proxy any requests (REGISTER or INVITE) 
>
>>If you want a Proxy you should look toward OpenSIPS 
>
>
>________________________________
From: sylver_b <sylver_b at yahoo.com>
>Reply-To: <freeswitch-users at lists.freeswitch.org>
>Date: Wed, 26 Aug 2009 17:14:55 -0700 (PDT)
>To: <freeswitch-users at lists.freeswitch.org>
>Subject: [Freeswitch-users] upstream Registrar / Mirror proxy
>
>
>Hello,
>
>>We would like to use FreeSwitch to test interoperability with our sip registrar platform - as our registrar doesnt handle NAT Traversal we would like to use FS as a mirror proxy . 
>
>>If a call goes through, FS should transparently handle nat traversal functionalities while forwarding the REGISTER/INVITE requests to our registrar server.
>
>>Below some detailed requirements:
>>Upper Registration is the capability of a SBC to proxy Registrations towards an upstream Registrar. While it is necessary that all SIP requests traverse the SBC for NAT continuity, Registrations should be allowed to be relayed towards our upstream Registrar while the SBC retains a copy of the AOR and masquerade on behalf of the UA that send the registration request.
>
>>Please let us know the best way to configure FS to achieve this type of configuration.
>
>>Thank you
>
>> 
>________________________________
_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
> 
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>



      
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From leon at scarlet-internet.nl  Thu Aug 27 07:57:08 2009
From: leon at scarlet-internet.nl (Leon de Rooij)
Date: Thu, 27 Aug 2009 16:57:08 +0200
Subject: [Freeswitch-users] need help with mod_xml_odbc
In-Reply-To: <27c25bc40908270653q47345f17s8d78e99ee5cc4b35@mail.gmail.com>
References: <27c25bc40908270156x192163aw85c4740e55ad427f@mail.gmail.com>
	<C6BBB626.5DDCF%krice@freeswitch.org>
	<27c25bc40908270337k78f9c6e5gc7a4860d1f074a8f@mail.gmail.com>
	<27c25bc40908270440o5f1b7a6csf675716677baef48@mail.gmail.com>
	<352125CC-DE10-4665-AD62-ABB3B4B2A1DE@scarlet-internet.nl>
	<F062600C-BDA4-4B70-B684-5837EF5E4066@scarlet-internet.nl>
	<27c25bc40908270653q47345f17s8d78e99ee5cc4b35@mail.gmail.com>
Message-ID: <E2C7D974-6301-4AC5-BC57-E854C35400F1@scarlet-internet.nl>

Hi Juan,

With debug=true you should be able to see what template it's trying to  
render in a loop, can you tell which one that is ? (I'm guessing it  
says 32 times it wants to render "not-found")

In the xml you pasted in your mail, you didn't specify the name of the  
"not-found" template, just <template>, that could be the problem..

To get around the problem of generating a wrong query when the ${user}  
is not given, you could simply encapsulate everything with a check- 
event-header, like this:

<template name="default">
   <xml-odbc-do name="check-event-header" if-name="user">
     <xml-odbc-do name="query" on-empty-result-break-to="not-found"  
value="...etc">
       ..rest of your template goes here..
     </xml-odbc-do>
   </xml-odbc-do>
</template>

But that would mean that an "empty" xml is returned whenever no $ 
{user} is provided. I didn't test how fs will react to that when it's  
looking up all the users for generating the acl list..

A way around that would be to only render your template when ${user}  
is set and otherwise render the not-found template, like this:

<template name="default">
   <xml-odbc-do name="check-event-header" if-name="user">
     <xml-odbc-do name="break-to" value="directory-user"/>
   </xml-odbc-do>
   <xml-odbc-do name="break-to" value="not-found"/>
</template>

<template name="directory-user">
   <xml-odbc-do name="query" on-empty-result-break-to="not-found"  
value="...etc">
     ..rest of your template goes here..
   </xml-odbc-do>
</template>

<template name="not-found">
etc...
</template>

---

Which comes close to how it's defined in the xml_odbc.conf.xml  
example, which solves it the other way around:

...
     <template name="default">
       <xml-odbc-do name="break-to" value="${section}"/>
     </template>

     <template name="directory"> -->
       <xml-odbc-do name="check-event-header" if-name="purpose">
         <!-- catches purpose gateways and network-list (any more?) -->
         <xml-odbc-do name="break-to" value="directory-${purpose}"/>
       </xml-odbc-do>
       <xml-odbc-do name="break-to" value="directory-user"/>
     </template>
...

- First it checks what section is requested, which is nice but not  
really necessary in your case, since you only bind to directory.
- Then it checks whether a purpose header is set (which was only the  
case for looking up acl's and gateways and NOT for regular user  
lookups last time I checked out the svn)
- If a purpose header IS set, then it tries to render directory-$ 
{purpose}, which isn't specified, so it falls back to "not-found"  
which is what you want in this case
- And as a last resort - this is where regular user lookups end up, it  
renders "directory-user" where you can put the template as specified  
above..

Hope that helps ?

regards,

Leon


On Aug 27, 2009, at 3:53 PM, Juan Backson wrote:

> Hi Leon,
>
> Thanks for your help.
>
> I have changed it according to your comment but I am still getting  
> the looping error.
>
> Would you please take a look see what else I did wrong?
>
> Also, sip_user is an integer field, so I can't really use ''.  Is  
> there anyway to get around that?
>
>
> <configuration name="xml_odbc.conf" description="XML ODBC  
> Configuration">
>
>   <settings>
>         <param name="binding" value="directory"/>
>     <param name="odbc-dsn" value="class5_odbc:root:abcd"/>
>     <param name="debug" value="true"/>
>     <param name="keep-files-around" value="true"/>
>   </settings>
>
>   <templates>
>
>     <template name="default">
>       <document type="freeswitch/xml">
>         <section name="directory">
>      <xml-odbc-do name="query" on-empty-result-break-to="not-found"  
> value="SELECT enabled, sip_password FROM agent WHERE sip_user =  $ 
> {user} and enabled='t'"/>
>         <domain name="${domain}">
>           <user id="${user}">
>             <params>
>               <xml-odbc-do name="query" value="SELECT sip_password  
> FROM agent WHERE sip_user = ${user}">
>                 <param name="password" value="${sip_password}"/>
>               </xml-odbc-do>
>             </params>
>               </user>
>         </domain>
>
>
>         </section>
>       </document>
>     </template>
>
>
>     <template>
>
>         <document type="freeswitch/xml">
>             <section name="result">
>                 <result status="not found"/>
>             </section>
>         </document>
>     </template>
>   </templates>
>
> </configuration>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From gregt at cgicommunications.com  Thu Aug 27 08:06:26 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Thu, 27 Aug 2009 11:06:26 -0400
Subject: [Freeswitch-users] Odd sonus warning
Message-ID: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>

Hi, I have some DIDs from Bandwidth.com and when they call in I see  
this in the console:

2009-08-27 11:01:47 [WARNING] sofia_glue.c:2701  
sofia_glue_negotiate_sdp() Hello,
I see you have a Sonus!
FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
Sadly, my creator had to spend several hours figuring this out so I  
thought you'd like to know that!
Don't worry, DTMF will work but you may want to ask them to fix it......

Anything I should worry about?
--
Greg


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From damin at nacs.net  Thu Aug 27 08:14:55 2009
From: damin at nacs.net (Gregory Boehnlein)
Date: Thu, 27 Aug 2009 11:14:55 -0400
Subject: [Freeswitch-users] Odd sonus warning
In-Reply-To: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>
References: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>
Message-ID: <00e801ca2729$254f0b00$6fed2100$@net>

> Anything I should worry about?
 
Nothing to worry about. Freeswitch will deal with the issue, but there are a
few caveats.

I believe this is the most up-to date resource:

http://wiki.freeswitch.org/wiki/RTP_Issues#Sonus




From brian at freeswitch.org  Thu Aug 27 08:15:36 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 27 Aug 2009 10:15:36 -0500
Subject: [Freeswitch-users] Odd sonus warning
In-Reply-To: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>
References: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>
Message-ID: <03399852-40FF-45E8-A69A-FE9DBEA963EB@freeswitch.org>

Does your DTMF work?

/b

On Aug 27, 2009, at 10:06 AM, Greg Thoen wrote:

> Anything I should worry about?




From foxb at abv.bg  Thu Aug 27 08:23:00 2009
From: foxb at abv.bg (Hristo Benev)
Date: Thu, 27 Aug 2009 18:23:00 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <999373713.8871.1251386588649.JavaMail.apache@mail22.abv.bg>

 I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).

If you need additional info I'll provide it.

Here is trace:

ngrep -d any -nn -i '1000' port 5062 -W byline
interface: any
filter: (ip or ip6) and ( port 5062 )
match: 1000
#
U 10.10.10.10:5090 -> 127.0.0.1:5062
INVITE sip:1000 at 127.0.0.1:5062 SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
Max-Forwards: 69.
From: "Extension 1001" <sip:1001 at 10.10.10.10>;tag=978g69jZaFpBD.
To: <sip:1000 at 127.0.0.1:5062>.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 INVITE.
Contact: <sip:mod_sofia at 10.10.10.10:5090>.
User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
Supported: timer, precondition, path, replaces.
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
Content-Type: application/sdp.
Content-Disposition: session.
Content-Length: 429.
X-ROUTE: LOOKUP.
Remote-Party-ID: "Extension 1001" <sip:1001 at 10.10.10.10>;party=calling;screen=yes;privacy=off.
.
v=0.
o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
s=FreeSWITCH.
c=IN IP4 10.10.10.10.
t=0 0.
m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
a=rtpmap:0 PCMU/8000.
a=rtpmap:115 G7221/32000.
a=fmtp:115 bitrate=48000.
a=rtpmap:107 G7221/16000.
a=fmtp:107 bitrate=32000.
a=rtpmap:9 G722/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-16.
a=rtpmap:13 CN/8000.
a=ptime:20.

#
U 127.0.0.1:5062 -> 10.10.10.10:5090
SIP/2.0 302 PEER_01.
Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
From: "Extension 1001" <sip:1001 at 10.10.10.10>;tag=978g69jZaFpBD.
To: <sip:1000 at 127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 INVITE.
Contact: sip:France at PEER_01.
Server: Kamailio (1.5.2-notls (i386/linux)).
Content-Length: 0.
.

#
U 10.10.10.10:5090 -> 127.0.0.1:5062
ACK sip:1000 at 127.0.0.1:5062 SIP/2.0.
Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
Max-Forwards: 69.
From: "Extension 1001" <sip:1001 at 10.10.10.10>;tag=978g69jZaFpBD.
To: <sip:1000 at 127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8.
Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
CSeq: 119574771 ACK.
Content-Length: 0.
.





 >-------- ?????????? ????? --------
 >??:  Hristo Benev 
 >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >??: freeswitch-users at lists.freeswitch.org
 >????????? ??: ?????????, 2009, ?????? 27 14:58:53 EEST

 > 
 >Bojnour,
 >
 >I'll send a trace ASAP.
 >
 >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
 >
 >I'm really interested to see the updates of the project.
 >
 >Thank you for the good tutorial.
 >
 >Hristo
 >
 >
 > >-------- ?????????? ????? --------
 > >??:  rod 
 > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 > >??: freeswitch-users at lists.freeswitch.org
 > >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST
 >
 > >Hi Hristo,
 > >
 > >I'm the author of this setup and wiki page. I did a lot of modifications 
 > >on this setup (alternative routing if failure essentially) but don't 
 > >have too much time to update the wiki.
 > >
 > >May you please send me an ngrep trace when you call 1000:
 > >
 > >ngrep -d any -nn -i '1000' port 5060 -W byline 
 > >
 > >I will check what's happening.
 > >Do you have an entry for 1000 in your mysql database ?
 > >
 > >regards,
 > >rod
 > >
 > >Hristo Benev a ?crit :
 > >>  
 > >> It seems that the problem is on kamailio configuration.
 > >> Will ask on their list.
 > >>
 > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
 > >>
 > >> I have it setup as friend in asterisk, but still ???
 > >>
 > >> Any ideas?
 > >>
 > >> Thanks,
 > >>
 > >>
 > >>  >-------- ?????????? ????? --------
 > >>  >??:  Hristo Benev 
 > >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 > >>  >??: freeswitch-users at lists.freeswitch.org
 > >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
 > >>
 > >>  > I think that the problem is here:
 > >>  >-------------------------
 > >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
 > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
 > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
 > >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
 > >>  >--------------------------
 > >>  >
 > >>  >Actually Regex FAIL
 > >>  >
 > >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
 > >>  >Here is my  default.xml:
 > >>  >----------------
 > >>  >
 > >>  >
 > >>  >
 > >>  >  
 > >>  >
 > >>  >     
 > >>  >      
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >         
 > >>  >      
 > >>  >    
 > >>  >
 > >>  >   
 > >>  >
 > >>  >  
 > >>  >
 > >>  >    
 > >>  >      
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >        
 > >>  >      
 > >>  >    
 > >>  >
 > >>  >  
 > >>  >
 > >>  >
 > >>  >
 > >>  >--------------------------
 > >>  >
 > >>  >
 > >>  >
 > >>  > >-------- ?????????? ????? --------
 > >>  > >??:  Brian West 
 > >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 > >>  > >??: freeswitch-users at lists.freeswitch.org
 > >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
 > >>  >
 > >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
 > >>  >You have to process all 302's in the dialplan. 
 > >>  >Set this on your sofia profile
 > >>  >You can set these variables sip_redirect_profile,
 > >>  >sip_redirect_context,
 > >>  >sip_redirect_dialplan,
 > >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
 > >>  >sip_redirected_to,
 > >>  >sip_redirect_contact_user_%d,
 > >>  >sip_redirect_contact_host_%d,
 > >>  >sip_redirect_contact_params_%d,
 > >>  >sip_redirect_dialstring_%d,
 > >>  >sip_redirect_dialstring,
 > >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
 > >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
 > >>  > >
 > >>  >
 > >>  >_______________________________________________
 > >>  >FreeSWITCH-users mailing list
 > >>  >FreeSWITCH-users at lists.freeswitch.org
 > >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > >>  >http://www.freeswitch.org
 > >>  >
 > >>
 > >> _______________________________________________
 > >> FreeSWITCH-users mailing list
 > >> FreeSWITCH-users at lists.freeswitch.org
 > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > >> http://www.freeswitch.org
 > >>   
 > >
 > >_______________________________________________
 > >FreeSWITCH-users mailing list
 > >FreeSWITCH-users at lists.freeswitch.org
 > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 > >http://www.freeswitch.org
 > >
 >
 >_______________________________________________
 >FreeSWITCH-users mailing list
 >FreeSWITCH-users at lists.freeswitch.org
 >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >http://www.freeswitch.org
 >



From dave at 3c.co.uk  Thu Aug 27 08:38:13 2009
From: dave at 3c.co.uk (David Knell)
Date: Thu, 27 Aug 2009 18:38:13 +0300
Subject: [Freeswitch-users] Odd sonus warning
In-Reply-To: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>
References: <93551649-2C9F-4718-9835-781108F7FE71@cgicommunications.com>
Message-ID: <1251387493.27240.105.camel@dk-d820>

On Thu, 2009-08-27 at 11:06 -0400, Greg Thoen wrote:
> Hi, I have some DIDs from Bandwidth.com and when they call in I see
> this in the console:
> 
> 
> 2009-08-27 11:01:47 [WARNING] sofia_glue.c:2701
> sofia_glue_negotiate_sdp() Hello,
> I see you have a Sonus!
> FYI, Sonus cannot follow the RFC on the proper way to send DTMF.
> Sadly, my creator had to spend several hours figuring this out so I
> thought you'd like to know that!
> Don't worry, DTMF will work but you may want to ask them to fix
> it......
> 
> 
> Anything I should worry about?

Not unless you're a Sonus shareholder - see
http://messages.finance.yahoo.com/Stocks_(A_to_Z)/Stocks_S/threadview?m=te&bn=16942&tid=827769&mid=-1&tof=15&rt=1&frt=2
for an amusing read ;-)

--Dave


-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk




From odermann at googlemail.com  Thu Aug 27 08:39:16 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 17:39:16 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
Message-ID: <5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>

> Is the snom firmware up to the latest?
yes, the firmware is the latest.

> I believe session timers should work properly with snom?
i don't know, but i think so. we tested with session timers "off" and
with setting the session timer to 0. both does not change anything.


we played with using a stun-server and had a small success. if we are
connected with a stund-server, the hangup will not come after 120
seconds. we can also call the 5900 and listen to the moh. this works
fine.
here we have a totally different problem. if another voip phone is
connected to the server using stun (for example the 1000), both sides
can not talk to each other. one won't hear, if the other side is
talking.

very strange (for us), but we have no idea, what we can do about that.



From brian at freeswitch.org  Thu Aug 27 08:42:28 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 27 Aug 2009 10:42:28 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
Message-ID: <32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>

Are the phones behind the same nat as FS?

/b

On Aug 27, 2009, at 10:39 AM, Dennis wrote:

> we played with using a stun-server and had a small success. if we are
> connected with a stund-server, the hangup will not come after 120
> seconds. we can also call the 5900 and listen to the moh. this works
> fine.
> here we have a totally different problem. if another voip phone is
> connected to the server using stun (for example the 1000), both sides
> can not talk to each other. one won't hear, if the other side is
> talking.




From odermann at googlemail.com  Thu Aug 27 09:01:06 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 18:01:06 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
Message-ID: <5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>

> Are the phones behind the same nat as FS?

no, both phones are behind the same nat, fs is behind another nat (and
the internet is inbetween).

both phones are in our office and we are sitting behind a nat. we
connect through our nat over the internet into the other nat, where fs
is behind.



From csa at nowthor.com  Thu Aug 27 09:04:56 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 27 Aug 2009 12:04:56 -0400
Subject: [Freeswitch-users] Who is "callie"?
Message-ID: <4A96AEA8.3030308@nowthor.com>

Hi!

I searched the wiki but couldn't find the answer. Is "callie" a real 
woman one may ask to record additional sounds?

Thanks!

Carlos



From msc at freeswitch.org  Thu Aug 27 09:16:30 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 09:16:30 -0700
Subject: [Freeswitch-users] How to make a call back
In-Reply-To: <25166083.post@talk.nabble.com>
References: <25166083.post@talk.nabble.com>
Message-ID: <87f2f3b90908270916l7cc7f3cap36bcf8725fa5fa5f@mail.gmail.com>

On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com> wrote:

>
> When I give the following from the command line it calls to 1010 extension
> and once answered, it calls to 1000 and bridge the connection.
>            originate user/1010 &bridge(user/1000)
> But I want to do this in perl. So I have given as follows
>            $session->originate($session,"user/1010 &bridge user/1000");
> But it is not working. It says "user/1010 &bridge user/1000 is invalid
> user".
> How to do this in perl. pls help.


Are you calling this perl script from the CLI? If so you won't have the
$session object because a channel does not exist for a simple API call.
-MC
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From msc at freeswitch.org  Thu Aug 27 09:18:10 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 09:18:10 -0700
Subject: [Freeswitch-users] Who is "callie"?
In-Reply-To: <4A96AEA8.3030308@nowthor.com>
References: <4A96AEA8.3030308@nowthor.com>
Message-ID: <87f2f3b90908270918u66439bd2wd046a4dc485caeb8@mail.gmail.com>

On Thu, Aug 27, 2009 at 9:04 AM, Carlos S. Antunes <csa at nowthor.com> wrote:

> Hi!
>
> I searched the wiki but couldn't find the answer. Is "callie" a real
> woman one may ask to record additional sounds?
>

"Callie" is one of the voices from GM Voices. She is definitely available
for custom work. Visit www.gmvoices.com for more info. Tell them that the
FreeSWITCH project sent you. :)
-MC


>
> Thanks!
>
> Carlos
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From fraunhofer.lists.freeswitch-001 at traced.net  Thu Aug 27 09:22:15 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Thu, 27 Aug 2009 18:22:15 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
Message-ID: <a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>

Hello Anthony,

2009/8/27 Anthony Minessale <anthony.minessale at gmail.com>:

> 600k is not a leak? FS can use as much as a gig of ram or more depending on
> what you are doing.

it's 600*1024 * 4* 1024 /1024/1024 => ~ 2.4 Gig
that's what i tried to express with the "physical pages" in my first post.

> you may want to install a fresh copy of FS, removing all your old files etc
> and make sure they build clean.

i "make current" to get the SOFIA-22 bugfix.

> We also have not had much luck running on ubuntu which is more of a desktop
> centric OS.

really? even the "server" edition? (not the netbook remix :)
Is 64bit so grave different if i don't need to address more than 4 Gig
of ram per process?

> I recommend you try your application on 64 bit CentOS which is the platform
> all of our paid customers use.

hmm. our operations-people wont be happy with that.

But as i carefully followed the other mailing-list-threads it came to
my mind that i could be the culprit: somehow the original example.xml
came back into the directory tree and included a lot of things, most
notably the hash-inserts for last-dailed-numbers and stuff.
I stripped it down to what it was before and gave it another try.

thx again and (hopefully) sorry

  Beni.



From msc at freeswitch.org  Thu Aug 27 09:25:46 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 09:25:46 -0700
Subject: [Freeswitch-users] Get the Session State
In-Reply-To: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAI0pCHFyxT5Mn9pdicKxxSMBAAAAAA==@gmail.com>
References: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAAI0pCHFyxT5Mn9pdicKxxSMBAAAAAA==@gmail.com>
Message-ID: <87f2f3b90908270925i8815f33qdbe60758f4645631@mail.gmail.com>

When you say call state, do you mean something like "ringing" or "answered"?
-MC

On Thu, Aug 27, 2009 at 3:23 AM, delianSPAM <delianspam at gmail.com> wrote:

>  Hello Everybody!
>
>
>
> How do you get the call state in Python?
>
>
>
> I have tried:
>
> ?
>
> session.answer()
>
> state_result=str(session.getVariable("state"))
>
> console_log("debug",state_result)
>
> ?
>
>
>
> But it returns: ?None?
>
>
>
> Thank you!
>
>
>
> Best Regards, Delian Tashev
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jmesquita at freeswitch.org  Thu Aug 27 09:32:23 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Thu, 27 Aug 2009 13:32:23 -0300
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <5e414ed0908270406m628342d5ya0e9f25514ed09d2@mail.gmail.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
	<5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
	<274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>
	<5e414ed0908260910h55122d77o5013ef1795d0c75@mail.gmail.com>
	<5e414ed0908270406m628342d5ya0e9f25514ed09d2@mail.gmail.com>
Message-ID: <ebbc2d550908270932q391deaffwd03fc7a6aa42dde4@mail.gmail.com>

Open on endpoint modules. We will relocate if needed.

jmesquita

On Thu, Aug 27, 2009 at 8:06 AM, Dennis <odermann at googlemail.com> wrote:

> sorry, but i do not know i which category i have to set this problem.
> could you help me with that?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mrene_lists at avgs.ca  Thu Aug 27 09:41:25 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Thu, 27 Aug 2009 12:41:25 -0400
Subject: [Freeswitch-users] fscore mutex locking question
In-Reply-To: <a37510170908270708g1e3236ao1f16a6835a87e8d7@mail.gmail.com>
References: <a37510170908270708g1e3236ao1f16a6835a87e8d7@mail.gmail.com>
Message-ID: <83AEC845-DFE5-47EF-8C67-80A9CD73F1B1@avgs.ca>

The only time the session is  write locked is when its about to be  
free'd, to make sure nothing holds a valid pointer.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 27-Aug-09, at 10:08 AM, Benedikt Fraunhofer wrote:

> Hello *,
>
> while looking at the code i came across a region of code which is
> unclear to me regarding locking issues.
> One example is switch_ivr_broadcast in switch_ivr_async.c. This should
> be the function called by
> uuid_broadcast() and others.
>
> in line 2341 it tries to queue an event to the bleg if it has to...
> -----
> .   if ((flags & SMF_ECHO_BLEG) && (other_uuid =
> switch_channel_get_variable(channel, SWITCH_SIGNAL_BOND_VARIABLE))
> .   .   && (other_session = switch_core_session_locate(other_uuid))) {
>
> ---
> switch_core_session_locate() does a "readonly trylock" on the channel
> mutex returning NULL if it's unable to
> aquire the lock, which brings up the following question:
> If some other thread is currently holding a writelock on the channel,
> the broadcast is not queued and not retried at a later time at all?
>
> I guess it's pretty easy to cause some unexpected behaviour using some
> endless loop calling "uuid_setvar" or some other race condition where
> the channel-mutex is write-locked while calling uuid_broadcast (eg.
> uuid_media?).
>
> Could this lead to a problem in "real live" scenarios, or are there
> other countermeasures despite "chances are one in a million that you
> hit that small time frame"?
>
> thx in advance
>  Beni.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From anthony.minessale at gmail.com  Thu Aug 27 09:48:37 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 11:48:37 -0500
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
	<1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
Message-ID: <191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>

please retest with latest SVN trunk

On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy <
anatoliy at kounitskiy.com> wrote:

> Just for information - the idea :
> A---calls---> B ---att_xfer---> C    , B hangs C up and goes back to A
> ( I`m sorry C is not answering :) )
>
>
-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Aug 27 09:51:33 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 11:51:33 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<E0F6F39C90934BB9AE3204B43A55D9C6@voztovoice>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
Message-ID: <191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>

Can you paste in the line that actually hangs up your call
it's BLUE and says something like

2009-08-27 11:39:04.903202 [NOTICE] sofia.c:327 Hangup sofia/internal/
1004 at dev.bkw.org [CS_EXECUTE] [NORMAL_CLEARING]

find the line like this for each leg of you call and put them in your reply


On Thu, Aug 27, 2009 at 11:01 AM, Dennis <odermann at googlemail.com> wrote:

> > Are the phones behind the same nat as FS?
>
> no, both phones are behind the same nat, fs is behind another nat (and
> the internet is inbetween).
>
> both phones are in our office and we are sitting behind a nat. we
> connect through our nat over the internet into the other nat, where fs
> is behind.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From anthony.minessale at gmail.com  Thu Aug 27 09:54:22 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 11:54:22 -0500
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <ebbc2d550908270932q391deaffwd03fc7a6aa42dde4@mail.gmail.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
	<5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
	<274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>
	<5e414ed0908260910h55122d77o5013ef1795d0c75@mail.gmail.com>
	<5e414ed0908270406m628342d5ya0e9f25514ed09d2@mail.gmail.com>
	<ebbc2d550908270932q391deaffwd03fc7a6aa42dde4@mail.gmail.com>
Message-ID: <191c3a030908270954n4b24fea4tf6a7b8815943193c@mail.gmail.com>

are you setting presence_id=user at domain variable on the outbound leg?
This is done for you in the DP via the user/ channel in the defaults but if
you are not using this
you have to set it manually.


2009/8/27 Jo?o Mesquita <jmesquita at freeswitch.org>

> Open on endpoint modules. We will relocate if needed.
>
> jmesquita
>
>
> On Thu, Aug 27, 2009 at 8:06 AM, Dennis <odermann at googlemail.com> wrote:
>
>> sorry, but i do not know i which category i have to set this problem.
>> could you help me with that?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
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From odermann at googlemail.com  Thu Aug 27 10:02:52 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 19:02:52 +0200
Subject: [Freeswitch-users] Question about presence
In-Reply-To: <191c3a030908270954n4b24fea4tf6a7b8815943193c@mail.gmail.com>
References: <5e414ed0908260618u4176fdf8tc73e97695974ef05@mail.gmail.com>
	<0AC13636-8F96-4536-B3C3-1F0C8FBE63D4@jerris.com>
	<5e414ed0908260834x4cbffb4dve8182c8647bb8226@mail.gmail.com>
	<274739BE-9B3F-4B3B-AEFA-891ADD89AB83@jerris.com>
	<5e414ed0908260910h55122d77o5013ef1795d0c75@mail.gmail.com>
	<5e414ed0908270406m628342d5ya0e9f25514ed09d2@mail.gmail.com>
	<ebbc2d550908270932q391deaffwd03fc7a6aa42dde4@mail.gmail.com>
	<191c3a030908270954n4b24fea4tf6a7b8815943193c@mail.gmail.com>
Message-ID: <5e414ed0908271002y6f30ab24w2ed5b5532df758a8@mail.gmail.com>

> are you setting presence_id=user at domain variable on the outbound leg?
> This is done for you in the DP via the user/ channel in the defaults but if
> you are not using this
> you have to set it manually.

in directory default we have the following:

<params>
     <param name="dial-string"
value="{presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(${dialed_user}@${dialed_domain})}"/>
   </params>

everthing works fine with the led lights - the only problem is the
described above.



From msc at freeswitch.org  Thu Aug 27 10:21:09 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 10:21:09 -0700
Subject: [Freeswitch-users] fifo question
In-Reply-To: <27c25bc40908240725k5e0c92a2u2def5bd679b7b666@mail.gmail.com>
References: <27c25bc40908240725k5e0c92a2u2def5bd679b7b666@mail.gmail.com>
Message-ID: <87f2f3b90908271021i2ca380euf41e233c6a8ea232@mail.gmail.com>

On Mon, Aug 24, 2009 at 7:25 AM, Juan Backson <juanbackson at gmail.com> wrote:

> Hi,
>
> Does anyone know the purpose of fifo_orbit_announce?
>
> When does fifo_orbit_announce get played?
>

I believe this is what gets played to a caller when he/she has been in queue
too long and it times out. I'm assuming this gets played and then the caller
is transferred to the fifo_orbit_extension. (See also fifo_orbit_context and
fifo_orbit_dialplan.)
-MC
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From msc at freeswitch.org  Thu Aug 27 10:23:21 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 10:23:21 -0700
Subject: [Freeswitch-users] FXO and analogue phones
In-Reply-To: <C281504B-3D79-465E-ADA9-25C140FCF50F@mac.com>
References: <C281504B-3D79-465E-ADA9-25C140FCF50F@mac.com>
Message-ID: <87f2f3b90908271023u10abd4c3n6012bc7817a73782@mail.gmail.com>

Merul,

My apologies, I had meant to follow up much earlier. Would you mind going
ahead and capturing full debug log output of a call in as well as a call
out? Please use pastebin.freeswitch.org to post the logs and then reply here
with the link to the pb post.

Thanks,
MC

On Sun, Aug 23, 2009 at 10:27 AM, Merul Patel <merul at mac.com> wrote:

> I have a Freeswitch setup working on an Alix embedded platform in
> conjunction with a USB FXO device from Sangoma. My goal is to be able to
> either answer incoming calls on a softphone or on a POTS handset elsewhere
> in the building, and to also be able to make outgoing calls from either. For
> clarity, the analogue line has two physical extensions, one connected to the
> POTS and the other to the FXO.
>
> I can make and receive calls fine, but have problems when the call is
> answered on the POTS handset.
>
> Here is the dialplan I initially used  in
> /opt/freeswitch/conf/dialplans/public/01_incoming.xml:
>
> <include>
>  <extension name="public_did">
>    <condition field="${strftime(%w)}" expression="^(\d)$">
>      <!-- There seems to be a delay of 7 seconds from when FS starts
> dealing with the call and from when it starts ringing -->
>      <action application="sleep" data="23000"/>
>      <action application="set" data="domain_name=$${domain}"/>
>      <action application="transfer" data="1001 XML default"/>
>    </condition>
>  </extension>
> </include>
>
> It's pretty basic, and if the softphone is not registered or does not
> answer then the call goes to voicemail. However the call will always go to
> voicemail, and the voicemail application will begin to execute after the
> call has been answered on the POTS handset.
>
> I've been trying to make the dialplan more useful, by having it ring the
> softphone immediately, and only transfer the call to the voicemail
> application if the line is still ringing. I'm in the UK, hence my choice of
> frequencies in the tone_detect application:
>
> <include>
>  <extension name="public_did">
>    <condition field="${strftime(%w)}" expression="^(\d)$">
>      <!-- There seems to be a delay of 7 seconds from when FS starts
> dealing with the call and from when it starts ringing -->
>      <action application="set" data="call_timeout=23"/>
>      <action application="set" data="continue_on_fail=true"/>
>      <action application="set" data="hangup_after_bridge=true"/>
>      <action application="bridge" data="sofia/internal/1001%$${domain}"/>
>      <action application="sleep" data="23000"/>
>      <action application="tone_detect" data="ring 400,450 r +5000 set
> RING=true"/>
>      <action application="transfer" data="public_answer_and_email"/>
>    </condition>
>  </extension>
>
> <extension name="public_answer_and_email">
>    <condition field="RING" expression="true">
>       <action application="answer"/>
>       <action application="voicemail" data="default $${domain} 1001"/>
>    </condition>
>  </extension>
> </include>
>
> Unfortunately, this is not working, and the logs are not yielding anything
> the is helpful to me. Is my use of the tone_detect application and the basic
> dialplan correct?
>
> Merul
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From odermann at googlemail.com  Thu Aug 27 10:25:56 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 19:25:56 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<ddad80610908232119g670f72efw76a5c9e46aaa0df2@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
Message-ID: <5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>

this is the line (without stun - so we only have one leg) and we
called the 5900 to moh:

2009-08-27 19:18:02.348232 [NOTICE] sofia.c:3863 Hangup
sofia/internal/1004 at 212.18.215.102 [CS_EXECUTE]
[RECOVERY_ON_TIMER_EXPIRE]

we called the 5900 and waited 2 minutes...

or did you mean something different?



From Nick.Lemberger at lkfd.net  Thu Aug 27 10:41:39 2009
From: Nick.Lemberger at lkfd.net (Nick Lemberger)
Date: Thu, 27 Aug 2009 12:41:39 -0500
Subject: [Freeswitch-users] mod_voicemail email template variables
Message-ID: <4A967F21.2C9A.00FE.0@lkfd.net>

Is there a way to use dialplan variables in the email that gets sent with the voicemail attachement.  I tried using some but nothing seems to show up, I'm guessing it's a different channel or something...

Any ideas?

Thanks,
-Nick




From anthony.minessale at gmail.com  Thu Aug 27 10:41:18 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 12:41:18 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<5e414ed0908260559o6435f712oc5a0587cb90c3c5b@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
Message-ID: <191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>

do the following:

1) restart FS
2) type sofia profile internal siptrace on
3) type console loglevel debug
4) reproduce the issue
5) upload the resulting output to http://pastebin.freeswitch.org (*hint* the
pass is in the dialog box)
6) let me know the pastebin number


On Thu, Aug 27, 2009 at 12:25 PM, Dennis <odermann at googlemail.com> wrote:

> this is the line (without stun - so we only have one leg) and we
> called the 5900 to moh:
>
> 2009-08-27 19:18:02.348232 [NOTICE] sofia.c:3863 Hangup
> sofia/internal/1004 at 212.18.215.102 [CS_EXECUTE]
> [RECOVERY_ON_TIMER_EXPIRE]
>
> we called the 5900 and waited 2 minutes...
>
> or did you mean something different?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From anthony.minessale at gmail.com  Thu Aug 27 10:44:22 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 12:44:22 -0500
Subject: [Freeswitch-users] mod_voicemail email template variables
In-Reply-To: <4A967F21.2C9A.00FE.0@lkfd.net>
References: <4A967F21.2C9A.00FE.0@lkfd.net>
Message-ID: <191c3a030908271044k63973088xeec12c578d02ef14@mail.gmail.com>

all variables referenced in the template should expand when sending the
email.


On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger <Nick.Lemberger at lkfd.net>wrote:

> Is there a way to use dialplan variables in the email that gets sent with
> the voicemail attachement.  I tried using some but nothing seems to show up,
> I'm guessing it's a different channel or something...
>
> Any ideas?
>
> Thanks,
> -Nick
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From Prometheus001 at gmx.net  Thu Aug 27 10:53:42 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 27 Aug 2009 19:53:42 +0200
Subject: [Freeswitch-users] sofia profile external register gwname via
 XML-Curl?
In-Reply-To: <4A95DBED.2060505@gmx.net>
References: <4A95DBED.2060505@gmx.net>
Message-ID: <4A96C826.3020204@gmx.net>

I got it,

gateways have to be preloaded (rescanned) before they can be registered.

Best regards
peter

Peter P GMX schrieb:
> Hello,
>
> I am using XML-Curl to handle the configuration of freeswitch
> When I try to register a gateway via event-socket with
>     sofia profile external register <gw-name>
> I receive back "invalid gateway".
>
> After reload mod_sofia the gateway is there. Question: Does this command
> work with xml-curl or only with local files??
> At least I see no xml-curl request when grepping network traffic.
>
> Best regards
> Peter
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   



From Prometheus001 at gmx.net  Thu Aug 27 11:02:31 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 27 Aug 2009 20:02:31 +0200
Subject: [Freeswitch-users] Calls from registered gateway try to lookup
	Directory
Message-ID: <4A96CA37.8060303@gmx.net>

I have found a strange thing in my FS installation,

FS is registered via a Gateway to an external provider (QSC) in the
external context.
But when a call is coming in, FS does not seem to go to any context, but
tries to lookup the user, as I receive the following message
    2009-08-27 18:44:23.287782 [WARNING] sofia_reg.c:1773 Can't find
user [026xxxxxxxxx at my.domain@my.domain]
    You must define a domain called 'my.domain' in your directory and
add a user with the id="026xxxxxxxxx at my.domain" attribute
    and you must configure your device to use the proper domain in it's
authentication credentials.

I learnt that a call from an external gateway should go to the public
context. But (in CLI debug mode) there are no other messages, except the
3 lines above.

What am I doing wrong?
Best regards
Peter

Here the invite message.

INVITE sip:gw+gw_xxxxxxxxxx at xx.xxx.xx.xxx:5080;transport=udp SIP/2.0.
Via:SIP/2.0/UDP
62.206.3.xxx;branch=z9hG4bK-BroadWorks.as1-xx.xxx.xx.xxxV5080-0-778271239-1616003581-1251392025611-.
From:"0xxxxXXXXXX"<sip:0xxxxXXXXXX at 62.206.3.xxx;user=phone>;tag=1616003581-1251392025611-.
To:"Me"<sip:026xxxxxxxxx at my.domain>.
Call-ID:BW185345611270809356816303 at 62.206.3.xxx.
CSeq:778271239 INVITE.
Contact:<sip:62.206.3.xxx:5060>.
Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE.
Accept:multipart/mixed,application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp.
Supported:.
Max-Forwards:20.
Proxy-Authorization:DIGEST
cnonce="fyvqi2pf",qop=auth,uri="sip:gw+gw_026xxxxxxxxx at xx.xxx.xx.xxx:5080;transport=udp",realm="my.domain",username="026xxxxxxxxx at my.domain",nonce="21bbe70c-932a-11de-b94d-bbade892ded3",algorithm=MD5,response="6d39a2546a4aa9a1fc39e2dc07c1e934",nc=00000001.
Content-Type:application/sdp.
Content-Length:344.
.
v=0.
o=BroadWorks 1271473 1 IN IP4 87.234.9.178.
s=-.
c=IN IP4 87.234.9.178.
t=0 0.
m=audio 18534 RTP/AVP 8 0 2 99 18 110.
a=rtpmap:99 G726-24/8000.
a=rtpmap:110 X-NSE/8000.
a=fmtp:110 192-194,200-202.
a=X-sqn:0.
a=X-cap: 1 audio RTP/AVP 110.
a=X-cpar: a=rtpmap:110 X-NSE/8000.
a=X-cpar: a=fmtp:110 192-194,200-202.
a=X-cap: 2 image udptl t38.



From Nick.Lemberger at lkfd.net  Thu Aug 27 11:06:01 2009
From: Nick.Lemberger at lkfd.net (Nick Lemberger)
Date: Thu, 27 Aug 2009 13:06:01 -0500
Subject: [Freeswitch-users] mod_voicemail email template variables
In-Reply-To: <191c3a030908271044k63973088xeec12c578d02ef14@mail.gmail.com>
References: <4A967F21.2C9A.00FE.0@lkfd.net>
	<191c3a030908271044k63973088xeec12c578d02ef14@mail.gmail.com>
Message-ID: <4A9684D7.2C9A.00FE.0@lkfd.net>

Thanks for the fast reply!

I just tried 10 random variables from http://wiki.freeswitch.org/wiki/Channel_Variables  and I only see the whitespace where the variable should be.  I've only been able to get the ones that are set in mod_voicemail.c circa line 1600 to work.

-Nick

>>> On 8/27/2009 at 12:44 PM, in message
<191c3a030908271044k63973088xeec12c578d02ef14 at mail.gmail.com>, Anthony
Minessale <anthony.minessale at gmail.com> wrote:
> all variables referenced in the template should expand when sending the
> email.
> 
> 
> On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger 
> <Nick.Lemberger at lkfd.net>wrote:
> 
>> Is there a way to use dialplan variables in the email that gets sent with
>> the voicemail attachement.  I tried using some but nothing seems to show up,
>> I'm guessing it's a different channel or something...
>>
>> Any ideas?
>>
>> Thanks,
>> -Nick
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users 
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org 
>>
> 
> 




From odermann at googlemail.com  Thu Aug 27 11:11:36 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 20:11:36 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
Message-ID: <5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>

the pastebin number is 10129



From odermann at googlemail.com  Thu Aug 27 11:11:36 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 20:11:36 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<191c3a030908260844w1419941an1b680cc990c889b@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
Message-ID: <5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>

the pastebin number is 10129



From asterisk at dotr.com  Thu Aug 27 11:16:33 2009
From: asterisk at dotr.com (Julian Lyndon-Smith)
Date: Thu, 27 Aug 2009 19:16:33 +0100
Subject: [Freeswitch-users] newbie alert. Help with dialplan
Message-ID: <f9c510070908271116n33c3f35di44eea1790f62bfc8@mail.gmail.com>

I am a long-time asterisk user (2005), so I come from an unfortunate
position of having to unlearn an awful lot of stuff in order to make
freeswitch do the things I want ;(

I *think* i've got my head around using mod_curl_xml (?) to read all
the config stuff from my webserver. I *think* I've got my head around
setting up the sip clients etc etc ...

However, where I am really struggling is the dialplan. For the life of
me I simply cannot seem to grasp the fs way - that's no disrespect to
the fs way, but perhaps the failure of this old brain to change !

 have pastebinned an example of show 1234546 at inboundq at
http://www.pastebin.ca/1544890. If possible, would someone be able to
show me how to convert this dialplan to the fs way ? If I could be
given a little foot-up I'm sure that I would be able to convert the
rest of the dialplan !

Thanks in advance, and please go easy !

Julian



From anatoliy at kounitskiy.com  Thu Aug 27 11:18:53 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Thu, 27 Aug 2009 21:18:53 +0300
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
	<1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
	<191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>
Message-ID: <1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>

Thank you all for the great job :)
Now it works as I wanted!!

Tomorrow I'll  will try to update the wiki :)

But with few words ( I tried to strip the extensions to minimum for
more easy read for users).
In the dialplan I have created the extension showed below:

 <extension name="local_number">
    <condition field="destination_number" expression="^(\d{3})$">
      <action application="set" data="dialed_extension=$1"/>
      <action application="export" data="dialed_extension=$1"/>
      <action application="bind_meta_app" data="1 b s
execute_extension::attented_xfer XML features"/>
      <action application="set" data="transfer_ringback=$${hold_music}"/>
      <action application="set" data="call_timeout=10"/>
      <action application="set" data="hangup_after_bridge=true"/>
      <action application="bridge"
data="user/${dialed_extension}@${domain_name}"/>
    </condition>
  </extension>

In the features.xml file I have extension attented_xfer, that is using
the att_xfer application :

 <extension name="attented_xfer">
      <condition field="destination_number" expression="^attented_xfer$">
        <action application="set" data="continue_on_fail=true"/>
        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
attxfer_callthis 30000 #"/>
        <action application="set" data="origination_cancel_key=#"/>
        <action application="att_xfer"
data="user/${attxfer_callthis}@${domain_name}"/>
      </condition>
    </extension>

Also I have set the the origination cancel key will be # - so if B
wants to cancel the call to C and to go back to A - just uses the #
key :)

Again - thank you very much :) :)

Anatoliy Kounitskiy

On Thu, Aug 27, 2009 at 7:48 PM, Anthony
Minessale<anthony.minessale at gmail.com> wrote:
> please retest with latest SVN trunk
>
> On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy
> <anatoliy at kounitskiy.com> wrote:
>>
>> Just for information - the idea :
>> A---calls---> B ---att_xfer---> C ? ?, B hangs C up and goes back to A
>> ( I`m sorry C is not answering :) )
>>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540



From Prometheus001 at gmx.net  Thu Aug 27 11:23:12 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Thu, 27 Aug 2009 20:23:12 +0200
Subject: [Freeswitch-users] Calls from registered gateway try to lookup
	Directory
In-Reply-To: <4A96CA37.8060303@gmx.net>
References: <4A96CA37.8060303@gmx.net>
Message-ID: <4A96CF10.8090009@gmx.net>

And yes,

external profile is on Port 5080 and all request go to 5080.

Best regards
Peter

Peter P GMX schrieb:
> I have found a strange thing in my FS installation,
>
> FS is registered via a Gateway to an external provider (QSC) in the
> external context.
> But when a call is coming in, FS does not seem to go to any context, but
> tries to lookup the user, as I receive the following message
>     2009-08-27 18:44:23.287782 [WARNING] sofia_reg.c:1773 Can't find
> user [026xxxxxxxxx at my.domain@my.domain]
>     You must define a domain called 'my.domain' in your directory and
> add a user with the id="026xxxxxxxxx at my.domain" attribute
>     and you must configure your device to use the proper domain in it's
> authentication credentials.
>
> I learnt that a call from an external gateway should go to the public
> context. But (in CLI debug mode) there are no other messages, except the
> 3 lines above.
>
> What am I doing wrong?
> Best regards
> Peter
>
> Here the invite message.
>
> INVITE sip:gw+gw_xxxxxxxxxx at xx.xxx.xx.xxx:5080;transport=udp SIP/2.0.
> Via:SIP/2.0/UDP
> 62.206.3.xxx;branch=z9hG4bK-BroadWorks.as1-xx.xxx.xx.xxxV5080-0-778271239-1616003581-1251392025611-.
> From:"0xxxxXXXXXX"<sip:0xxxxXXXXXX at 62.206.3.xxx;user=phone>;tag=1616003581-1251392025611-.
> To:"Me"<sip:026xxxxxxxxx at my.domain>.
> Call-ID:BW185345611270809356816303 at 62.206.3.xxx.
> CSeq:778271239 INVITE.
> Contact:<sip:62.206.3.xxx:5060>.
> Allow:ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE.
> Accept:multipart/mixed,application/dtmf,application/dtmf-relay,application/media_control+xml,application/sdp.
> Supported:.
> Max-Forwards:20.
> Proxy-Authorization:DIGEST
> cnonce="fyvqi2pf",qop=auth,uri="sip:gw+gw_026xxxxxxxxx at xx.xxx.xx.xxx:5080;transport=udp",realm="my.domain",username="026xxxxxxxxx at my.domain",nonce="21bbe70c-932a-11de-b94d-bbade892ded3",algorithm=MD5,response="6d39a2546a4aa9a1fc39e2dc07c1e934",nc=00000001.
> Content-Type:application/sdp.
> Content-Length:344.
> .
> v=0.
> o=BroadWorks 1271473 1 IN IP4 87.234.9.178.
> s=-.
> c=IN IP4 87.234.9.178.
> t=0 0.
> m=audio 18534 RTP/AVP 8 0 2 99 18 110.
> a=rtpmap:99 G726-24/8000.
> a=rtpmap:110 X-NSE/8000.
> a=fmtp:110 192-194,200-202.
> a=X-sqn:0.
> a=X-cap: 1 audio RTP/AVP 110.
> a=X-cpar: a=rtpmap:110 X-NSE/8000.
> a=X-cpar: a=fmtp:110 192-194,200-202.
> a=X-cap: 2 image udptl t38.
>
>   



From brian at freeswitch.org  Thu Aug 27 11:23:39 2009
From: brian at freeswitch.org (Brian West)
Date: Thu, 27 Aug 2009 13:23:39 -0500
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
	<1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
	<191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>
	<1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>
Message-ID: <417A7E0B-E053-4E1C-9848-93E101656E49@freeswitch.org>

/me points to the first question on the FAQ  ;)

/b

On Aug 27, 2009, at 1:18 PM, Anatoliy Kounitskiy wrote:

> Thank you all for the great job :)
> Now it works as I wanted!!
>
> Tomorrow I'll  will try to update the wiki :)




From msc at freeswitch.org  Thu Aug 27 11:30:11 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 11:30:11 -0700
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
	<1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
	<191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>
	<1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>
Message-ID: <87f2f3b90908271130v4bb7deads45d5f1c190ea0661@mail.gmail.com>

Thanks! I'll be in touch to help you if you need any wiki editing
assistance.
-MC

On Thu, Aug 27, 2009 at 11:18 AM, Anatoliy Kounitskiy <
anatoliy at kounitskiy.com> wrote:

> Thank you all for the great job :)
> Now it works as I wanted!!
>
> Tomorrow I'll  will try to update the wiki :)
>
> But with few words ( I tried to strip the extensions to minimum for
> more easy read for users).
> In the dialplan I have created the extension showed below:
>
>  <extension name="local_number">
>     <condition field="destination_number" expression="^(\d{3})$">
>      <action application="set" data="dialed_extension=$1"/>
>      <action application="export" data="dialed_extension=$1"/>
>      <action application="bind_meta_app" data="1 b s
> execute_extension::attented_xfer XML features"/>
>      <action application="set" data="transfer_ringback=$${hold_music}"/>
>      <action application="set" data="call_timeout=10"/>
>      <action application="set" data="hangup_after_bridge=true"/>
>       <action application="bridge"
> data="user/${dialed_extension}@${domain_name}"/>
>     </condition>
>  </extension>
>
> In the features.xml file I have extension attented_xfer, that is using
> the att_xfer application :
>
>  <extension name="attented_xfer">
>       <condition field="destination_number" expression="^attented_xfer$">
>        <action application="set" data="continue_on_fail=true"/>
>        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
> attxfer_callthis 30000 #"/>
>         <action application="set" data="origination_cancel_key=#"/>
>        <action application="att_xfer"
> data="user/${attxfer_callthis}@${domain_name}"/>
>      </condition>
>    </extension>
>
> Also I have set the the origination cancel key will be # - so if B
> wants to cancel the call to C and to go back to A - just uses the #
> key :)
>
> Again - thank you very much :) :)
>
> Anatoliy Kounitskiy
>
> On Thu, Aug 27, 2009 at 7:48 PM, Anthony
> Minessale<anthony.minessale at gmail.com> wrote:
> > please retest with latest SVN trunk
> >
> > On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy
> > <anatoliy at kounitskiy.com> wrote:
> >>
> >> Just for information - the idea :
> >> A---calls---> B ---att_xfer---> C    , B hangs C up and goes back to A
> >> ( I`m sorry C is not answering :) )
> >>
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anatoliy Kounitskiy
> -------------------------
> E-mail: anatoliy at kounitskiy.com
> Mobile: +359898913540
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From mike at jerris.com  Thu Aug 27 11:44:30 2009
From: mike at jerris.com (Michael Jerris)
Date: Thu, 27 Aug 2009 14:44:30 -0400
Subject: [Freeswitch-users] Who is "callie"?
In-Reply-To: <87f2f3b90908270918u66439bd2wd046a4dc485caeb8@mail.gmail.com>
References: <4A96AEA8.3030308@nowthor.com>
	<87f2f3b90908270918u66439bd2wd046a4dc485caeb8@mail.gmail.com>
Message-ID: <D206B5F9-2D91-4652-A944-09CAD425ABBD@jerris.com>

Also, note, we have a few sound files we would like to add to our  
default set.  If you are doing a session anyways and are willing  
please let me know if we can throw a few other prompts in your list to  
be recorded.

Mike

On Aug 27, 2009, at 12:18 PM, Michael Collins wrote:

>
>
> On Thu, Aug 27, 2009 at 9:04 AM, Carlos S. Antunes <csa at nowthor.com>  
> wrote:
> Hi!
>
> I searched the wiki but couldn't find the answer. Is "callie" a real
> woman one may ask to record additional sounds?
>
> "Callie" is one of the voices from GM Voices. She is definitely  
> available for custom work. Visit www.gmvoices.com for more info.  
> Tell them that the FreeSWITCH project sent you. :)
> -MC
>
>
> Thanks!
>
> Carlos
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From anthony.minessale at gmail.com  Thu Aug 27 11:45:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 13:45:03 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<5e414ed0908260923o262aee87ufa4ef3ef997639aa@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
	<5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>
Message-ID: <191c3a030908271145g1995e642s92f6979c207987ef@mail.gmail.com>

I think the problem is the session-timeout is too long and your nat mapping
is being deleted.

try setting it to a smaller value like 20 or 40 sec
or try setting those phones to register more frequently (every 10 or 20 sec)


On Thu, Aug 27, 2009 at 1:11 PM, Dennis <odermann at googlemail.com> wrote:

> the pastebin number is 10129
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From anthony.minessale at gmail.com  Thu Aug 27 11:54:09 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 13:54:09 -0500
Subject: [Freeswitch-users] mod_voicemail email template variables
In-Reply-To: <4A9684D7.2C9A.00FE.0@lkfd.net>
References: <4A967F21.2C9A.00FE.0@lkfd.net>
	<191c3a030908271044k63973088xeec12c578d02ef14@mail.gmail.com>
	<4A9684D7.2C9A.00FE.0@lkfd.net>
Message-ID: <191c3a030908271154t155e844dy459acb700316182c@mail.gmail.com>

you should be able to for instance put

<action application="set" data="test_var=this is a test"/>

right before the voicemail app is called

then put

${test_var} in your template.

making sure to issue reloadxml or restart FS


On Thu, Aug 27, 2009 at 1:06 PM, Nick Lemberger <Nick.Lemberger at lkfd.net>wrote:

> Thanks for the fast reply!
>
> I just tried 10 random variables from
> http://wiki.freeswitch.org/wiki/Channel_Variables  and I only see the
> whitespace where the variable should be.  I've only been able to get the
> ones that are set in mod_voicemail.c circa line 1600 to work.
>
> -Nick
>
> >>> On 8/27/2009 at 12:44 PM, in message
> <191c3a030908271044k63973088xeec12c578d02ef14 at mail.gmail.com>, Anthony
> Minessale <anthony.minessale at gmail.com> wrote:
> > all variables referenced in the template should expand when sending the
> > email.
> >
> >
> > On Thu, Aug 27, 2009 at 12:41 PM, Nick Lemberger
> > <Nick.Lemberger at lkfd.net>wrote:
> >
> >> Is there a way to use dialplan variables in the email that gets sent
> with
> >> the voicemail attachement.  I tried using some but nothing seems to show
> up,
> >> I'm guessing it's a different channel or something...
> >>
> >> Any ideas?
> >>
> >> Thanks,
> >> -Nick
> >>
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >>
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From odermann at googlemail.com  Thu Aug 27 12:09:12 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 21:09:12 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908271145g1995e642s92f6979c207987ef@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<191c3a030908260953h2e0e95es4d3a97758354e681@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
	<5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>
	<191c3a030908271145g1995e642s92f6979c207987ef@mail.gmail.com>
Message-ID: <5e414ed0908271209j15cb8cc7v2a21d02b6831eb1f@mail.gmail.com>

> I think the problem is the session-timeout is too long and your nat mapping
> is being deleted.

we changed the param session-timeout (in internal) to 20 and 40,
without success.

we changed the minimum-session-expires to 20, although we knew, the
allowed minimum is 90 and 90 was shown in the console - no success.


> try setting it to a smaller value like 20 or 40 sec
> or try setting those phones to register more frequently (every 10 or 20 sec)

we changed all possible options to 10 and 20, again without success :(



From gmaruzz at celliax.org  Thu Aug 27 12:20:36 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Thu, 27 Aug 2009 21:20:36 +0200
Subject: [Freeswitch-users] Can a chat message be sent to a cell phone
	with FS?
In-Reply-To: <4A9656FE.9030504@yahoo.com>
References: <4A9656FE.9030504@yahoo.com>
Message-ID: <7b197bef0908271220w2dca29fase62f50fc5419d073@mail.gmail.com>

In a short while (for any value of "short") will be available for
testing mod-celliax, an interface to the cellular phones networks for
voice calls and SMSs.

-giovanni


On 8/27/09, Merle J. Ebbert <se02005-mec at yahoo.com> wrote:
> Can a chat message be sent to a cell phone with FS?
>
> Thanks,
> Merle
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>

-- 
Sent from my mobile device

Sincerely,

Giovanni Maruzzelli
Cell : +39-347-2665618



From anthony.minessale at gmail.com  Thu Aug 27 12:21:59 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 14:21:59 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908271209j15cb8cc7v2a21d02b6831eb1f@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<5e414ed0908270839y73184603re5c60378ea4f7aba@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
	<5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>
	<191c3a030908271145g1995e642s92f6979c207987ef@mail.gmail.com>
	<5e414ed0908271209j15cb8cc7v2a21d02b6831eb1f@mail.gmail.com>
Message-ID: <191c3a030908271221tdf76f28g657cb8fd2b39406@mail.gmail.com>

If you look at your trace the call is sending a re-invite over and over and
over again with no reply

you need to examine your network topology and find out why the packets FS is
sending to your phone
never make it.

also try disabling session-timers on the snom


On Thu, Aug 27, 2009 at 2:09 PM, Dennis <odermann at googlemail.com> wrote:

> > I think the problem is the session-timeout is too long and your nat
> mapping
> > is being deleted.
>
> we changed the param session-timeout (in internal) to 20 and 40,
> without success.
>
> we changed the minimum-session-expires to 20, although we knew, the
> allowed minimum is 90 and 90 was shown in the console - no success.
>
>
> > try setting it to a smaller value like 20 or 40 sec
> > or try setting those phones to register more frequently (every 10 or 20
> sec)
>
> we changed all possible options to 10 and 20, again without success :(
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From odermann at googlemail.com  Thu Aug 27 12:42:34 2009
From: odermann at googlemail.com (Dennis)
Date: Thu, 27 Aug 2009 21:42:34 +0200
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <191c3a030908271221tdf76f28g657cb8fd2b39406@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<32B0CA69-48AB-44C0-B964-5F55ED423BC6@freeswitch.org>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
	<5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>
	<191c3a030908271145g1995e642s92f6979c207987ef@mail.gmail.com>
	<5e414ed0908271209j15cb8cc7v2a21d02b6831eb1f@mail.gmail.com>
	<191c3a030908271221tdf76f28g657cb8fd2b39406@mail.gmail.com>
Message-ID: <5e414ed0908271242v214eeb4eofa47f796290b41df@mail.gmail.com>

> If you look at your trace the call is sending a re-invite over and over and
> over again with no reply
>
> you need to examine your network topology and find out why the packets FS is
> sending to your phone
> never make it.
>
> also try disabling session-timers on the snom

are you talking about our problem, that we get a hangup after 120
seconds? i just ask to avoid, that we are talking about different
things.

if we connect without stun and call the 5900, we can hear the music,
so i assume, that we are receiving packets from fs!? but after about
120 seconds, we receive the hangup. before everything was fine and as
expected.

we also tried disabling session-timers in the snom - same problem.

thant's the problem, we tried soooooo many things, but simply nothing
changes the problem or makes something different (like sending the
hangup later or earlier).



From anthony.minessale at gmail.com  Thu Aug 27 13:06:24 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 15:06:24 -0500
Subject: [Freeswitch-users] Call exits after 120 seconds with hangup
	cause
In-Reply-To: <5e414ed0908271242v214eeb4eofa47f796290b41df@mail.gmail.com>
References: <5e414ed0908210546j66c4ddfchacde3f59dde9674e@mail.gmail.com>
	<5e414ed0908270901m74cfb2c4g55018238e557f689@mail.gmail.com>
	<191c3a030908270951t391e0ef0ocab9541692a258a9@mail.gmail.com>
	<5e414ed0908271025i6e6d8531w27921c15ba60ee91@mail.gmail.com>
	<191c3a030908271041i1ec2010anb0f5b3c93fdcf83e@mail.gmail.com>
	<5e414ed0908271111i45d76215q32e7308e5c02e412@mail.gmail.com>
	<191c3a030908271145g1995e642s92f6979c207987ef@mail.gmail.com>
	<5e414ed0908271209j15cb8cc7v2a21d02b6831eb1f@mail.gmail.com>
	<191c3a030908271221tdf76f28g657cb8fd2b39406@mail.gmail.com>
	<5e414ed0908271242v214eeb4eofa47f796290b41df@mail.gmail.com>
Message-ID: <191c3a030908271306s6e656caeqea86343184e4bc33@mail.gmail.com>

yes i am talking about that same problem.
I am not talking about audio I am talking about sip messages.

look at the trace you made from my earlier request
You can see the same invite being sent 10 times and never got any answer.
so you need to investigate it.


On Thu, Aug 27, 2009 at 2:42 PM, Dennis <odermann at googlemail.com> wrote:

> > If you look at your trace the call is sending a re-invite over and over
> and
> > over again with no reply
> >
> > you need to examine your network topology and find out why the packets FS
> is
> > sending to your phone
> > never make it.
> >
> > also try disabling session-timers on the snom
>
> are you talking about our problem, that we get a hangup after 120
> seconds? i just ask to avoid, that we are talking about different
> things.
>
> if we connect without stun and call the 5900, we can hear the music,
> so i assume, that we are receiving packets from fs!? but after about
> 120 seconds, we receive the hangup. before everything was fine and as
> expected.
>
> we also tried disabling session-timers in the snom - same problem.
>
> thant's the problem, we tried soooooo many things, but simply nothing
> changes the problem or makes something different (like sending the
> hangup later or earlier).
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
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From pjintheusa at gmail.com  Thu Aug 27 13:09:10 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Thu, 27 Aug 2009 16:09:10 -0400
Subject: [Freeswitch-users] Can I stream a file to a parked call.
Message-ID: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>

Hi there,

I know there are other ways of doing this. I am just trying get to
know have fun with the FreeSWITCH API.

I am using originate and park a call:

fsApi.Execute("originate",
string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
&park", blegSession.Uuid, OutgoingCallerID, NumberToDial));

That works great, the phone rings. I want to play something to this
called party when they pick up.

 while (!blegSession.answered())
 {
        Log.WriteLine(LogLevel.Alert,
"Inside::CallReturns:!Session.answered::Loop");
        blegSession.sleep(500, 1);
 }

string promptFile = "prompts/whisper.wav";
blegSession.StreamFile(promptFile, 0);


This works - but the audio is choppy and slow.

Is there something I need to do to that parked call before streaming that file?


Thanks for any input.

Phil



From csa at nowthor.com  Thu Aug 27 13:13:37 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Thu, 27 Aug 2009 16:13:37 -0400
Subject: [Freeswitch-users] Who is "callie"?
In-Reply-To: <D206B5F9-2D91-4652-A944-09CAD425ABBD@jerris.com>
References: <4A96AEA8.3030308@nowthor.com>	<87f2f3b90908270918u66439bd2wd046a4dc485caeb8@mail.gmail.com>
	<D206B5F9-2D91-4652-A944-09CAD425ABBD@jerris.com>
Message-ID: <4A96E8F1.8010106@nowthor.com>

Mike,

Sure! I am planning on doing a session "soon", maybe in a couple of 
weeks or so. My only question is whether GM Voices license will allow 
this kind of thing. Do you or anyone else know?

Carlos

Michael Jerris wrote:
> Also, note, we have a few sound files we would like to add to our 
> default set.  If you are doing a session anyways and are willing 
> please let me know if we can throw a few other prompts in your list to 
> be recorded.
>
> Mike
>
> On Aug 27, 2009, at 12:18 PM, Michael Collins wrote:
>
>>
>>
>> On Thu, Aug 27, 2009 at 9:04 AM, Carlos S. Antunes <csa at nowthor.com 
>> <mailto:csa at nowthor.com>> wrote:
>>
>>     Hi!
>>
>>     I searched the wiki but couldn't find the answer. Is "callie" a real
>>     woman one may ask to record additional sounds?
>>
>>
>> "Callie" is one of the voices from GM Voices. She is definitely 
>> available for custom work. Visit www.gmvoices.com 
>> <http://www.gmvoices.com/> for more info. Tell them that the 
>> FreeSWITCH project sent you. :)
>> -MC
>>  
>>
>>
>>     Thanks!
>>
>>     Carlos
>>
>>     _______________________________________________
>>     FreeSWITCH-users mailing list
>>     FreeSWITCH-users at lists.freeswitch.org
>>     <mailto:FreeSWITCH-users at lists.freeswitch.org>
>>     http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>     UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>     http://www.freeswitch.org <http://www.freeswitch.org/>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org 
>> <mailto:FreeSWITCH-users at lists.freeswitch.org>
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   
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From anthony.minessale at gmail.com  Thu Aug 27 13:30:03 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 15:30:03 -0500
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
Message-ID: <191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>

try wedging {ignore_early_media=true} before the first [ in your dial string
and eliminate the code waiting for answer.


On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> Hi there,
>
> I know there are other ways of doing this. I am just trying get to
> know have fun with the FreeSWITCH API.
>
> I am using originate and park a call:
>
> fsApi.Execute("originate",
>
> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
>
> That works great, the phone rings. I want to play something to this
> called party when they pick up.
>
>  while (!blegSession.answered())
>  {
>        Log.WriteLine(LogLevel.Alert,
> "Inside::CallReturns:!Session.answered::Loop");
>        blegSession.sleep(500, 1);
>  }
>
> string promptFile = "prompts/whisper.wav";
> blegSession.StreamFile(promptFile, 0);
>
>
> This works - but the audio is choppy and slow.
>
> Is there something I need to do to that parked call before streaming that
> file?
>
>
> Thanks for any input.
>
> Phil
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Thu Aug 27 13:40:04 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 13:40:04 -0700
Subject: [Freeswitch-users] Can a chat message be sent to a cell phone
	with FS?
In-Reply-To: <7b197bef0908271220w2dca29fase62f50fc5419d073@mail.gmail.com>
References: <4A9656FE.9030504@yahoo.com>
	<7b197bef0908271220w2dca29fase62f50fc5419d073@mail.gmail.com>
Message-ID: <87f2f3b90908271340l1a794686x55100f4da32c73f8@mail.gmail.com>

On Thu, Aug 27, 2009 at 12:20 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> In a short while (for any value of "short") will be available for
> testing mod-celliax, an interface to the cellular phones networks for
> voice calls and SMSs.
>
> -giovanni


We eagerly anticipate it's arrival! :)
-MC
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From gregt at cgicommunications.com  Thu Aug 27 14:13:42 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Thu, 27 Aug 2009 17:13:42 -0400
Subject: [Freeswitch-users] Mixing mod_curl_xml dynamic dialplans and static
	ones
Message-ID: <C62AA2C3-DBE4-47D1-95FA-A4DF92C2CE9E@cgicommunications.com>

I have mod_curl_xml working, but I also have several static dialplans  
in dialplan/public/ and I can't seem to get it to search those if my  
php generated xml page does not return a result.

I know in the static public.xml there is this
     <extension name="unloop">
       <condition field="${unroll_loops}" expression="^true$"/>
       <condition field="${sip_looped_call}" expression="^true$">
         <action application="deflect" data="${destination_number}"/>
       </condition>
     </extension>

Do I need my php page to generate something like that so it will  
continue looking in the static pages if it does not find a match in  
the xml_curl dynamic dialplan?

I don't know if I have given enough data for someone to point me in  
the right direction...
This is in the console when I have mod_xml_curl enabled:

2009-08-27 16:39:52 [INFO] mod_dialplan_xml.c:252 dialplan_hunt()  
Processing  ->15854199896 in context public
Dialplan: sofia/internal/585xxxxxxx at 208.34.86.39 parsing [public- 
 >curl_test] continue=false
Dialplan: sofia/internal/585xxxxxxx at 208.34.86.39 Regex (FAIL)  
[curl_test] destination_number(15854199896) =~ /^(18775844111)$/  
break=on-false
2009-08-27 16:39:52 [INFO] switch_core_state_machine.c:136  
switch_core_standard_on_routing() No Route, Aborting

--
Greg



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From krice at freeswitch.org  Thu Aug 27 14:32:28 2009
From: krice at freeswitch.org (Ken Rice)
Date: Thu, 27 Aug 2009 16:32:28 -0500
Subject: [Freeswitch-users] Mixing mod_curl_xml dynamic dialplans and
 static ones
In-Reply-To: <C62AA2C3-DBE4-47D1-95FA-A4DF92C2CE9E@cgicommunications.com>
Message-ID: <C6BC659C.5DE68%krice@freeswitch.org>

Your php should return a not found, also you can stack static and non-static
XML dialplans on your technology profile settings

For example, on your internal.xml (in the default sip profiles) the dialplan
setting can be set to ?XML,XML:/path/to/some/static.xml? . Once it processes
thru the xml_curl responses it will hit the static.xml



From: Greg Thoen <gregt at cgicommunications.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Thu, 27 Aug 2009 17:13:42 -0400
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Mixing mod_curl_xml dynamic dialplans and static
ones

I have mod_curl_xml working, but I also have several static dialplans in
dialplan/public/ and I can't seem to get it to search those if my php
generated xml page does not return a result.

I know in the static public.xml there is this
    <extension name="unloop">
      <condition field="${unroll_loops}" expression="^true$"/>
      <condition field="${sip_looped_call}" expression="^true$">
        <action application="deflect" data="${destination_number}"/>
      </condition>
    </extension>

Do I need my php page to generate something like that so it will continue
looking in the static pages if it does not find a match in the xml_curl
dynamic dialplan?

I don't know if I have given enough data for someone to point me in the
right direction...
This is in the console when I have mod_xml_curl enabled:

2009-08-27 16:39:52 [INFO] mod_dialplan_xml.c:252 dialplan_hunt() Processing
->15854199896 in context public
Dialplan: sofia/internal/585xxxxxxx at 208.34.86.39 parsing [public->curl_test]
continue=false
Dialplan: sofia/internal/585xxxxxxx at 208.34.86.39 Regex (FAIL) [curl_test]
destination_number(15854199896) =~ /^(18775844111)$/ break=on-false
2009-08-27 16:39:52 [INFO] switch_core_state_machine.c:136
switch_core_standard_on_routing() No Route, Aborting

 
--

Greg 



 



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From msc at freeswitch.org  Thu Aug 27 14:39:44 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 14:39:44 -0700
Subject: [Freeswitch-users] Mixing mod_curl_xml dynamic dialplans and
	static ones
In-Reply-To: <C6BC659C.5DE68%krice@freeswitch.org>
References: <C62AA2C3-DBE4-47D1-95FA-A4DF92C2CE9E@cgicommunications.com>
	<C6BC659C.5DE68%krice@freeswitch.org>
Message-ID: <87f2f3b90908271439u466a2634xd5b1ff62dd41ebfe@mail.gmail.com>

On Thu, Aug 27, 2009 at 2:32 PM, Ken Rice <krice at freeswitch.org> wrote:

>  Your php should return a not found, also you can stack static and
> non-static XML dialplans on your technology profile settings
>
> For example, on your internal.xml (in the default sip profiles) the
> dialplan setting can be set to ?XML,XML:/path/to/some/static.xml? . Once it
> processes thru the xml_curl responses it will hit the static.xml
>

Ken,

That's a very cool feature. I wasn't aware of it. I will see about adding
this knowledge to the wiki if it isn't there already.
-MC
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From tina at a2unlimited.com  Thu Aug 27 14:47:35 2009
From: tina at a2unlimited.com (Tina Martinez)
Date: Thu, 27 Aug 2009 17:47:35 -0400
Subject: [Freeswitch-users] Using a "Virtual" Extension
Message-ID: <47111.1251409655@a2unlimited.com>

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From jaybinks at gmail.com  Thu Aug 27 14:55:27 2009
From: jaybinks at gmail.com (Jay Binks)
Date: Fri, 28 Aug 2009 07:55:27 +1000
Subject: [Freeswitch-users] memory leak
In-Reply-To: <a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
Message-ID: <3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>

Anthony can you ( or anyone else alao ). Please elaborate on what  
makes centos 5.3 o much better for Freeswitch.

Is there some specific library vesiion on centos that makes a massive  
difference ?

Reason I ask ...   I personally only have a preference for debian,   
but others may have policy mandated Os's
For their companies, and it would be great to have some info about this.

I'd imagine this actually boils down to a requirement for libfoo  
version X ...  And if we ran those library versions on other OS's we'd  
be fine ?

Jay



>
>> I recommend you try your application on 64 bit CentOS which is the  
>> platform
>> all of our paid customers use.
>
> hmm. our operations-people wont be happy with that.



From msc at freeswitch.org  Thu Aug 27 14:59:56 2009
From: msc at freeswitch.org (Michael Collins)
Date: Thu, 27 Aug 2009 14:59:56 -0700
Subject: [Freeswitch-users] Using a "Virtual" Extension
In-Reply-To: <47111.1251409655@a2unlimited.com>
References: <47111.1251409655@a2unlimited.com>
Message-ID: <87f2f3b90908271459s57e571d3ldb716d1913549edc@mail.gmail.com>

> So far this works great!  However, my question is whether or not using the
> same extension is a bad idea?  does it matter how many times I setup the
> link between the servers using the same extension for every conference?
>
> - T
>

Tina,

Glad to hear your multi-machine conference thing is working. Most likely I'd
say that you are okay to use the same extension number. It's really just an
administrative thing. As long as people know what number to dial it should
be just fine.

-MC
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From anthony.minessale at gmail.com  Thu Aug 27 15:43:39 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Thu, 27 Aug 2009 17:43:39 -0500
Subject: [Freeswitch-users] memory leak
In-Reply-To: <3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
	<3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
Message-ID: <191c3a030908271543r70c0f03cw88e4fba21df2b61b@mail.gmail.com>

Mostly based on that is what we developed it with and continue to use and
have had the best luck and least problems with.

I am sure you are right about it being certain version of the core toolchain
I will guess it's the combo of libc and kernel and the fact that CentOS has
a purist attitude towards patching such code.

Some people use stable debian with no issues, other have reported problems
resolved by switching to CentOS

so presumably you could roll your own debian that was similar but we don't
have the resources to try to find out what makes what happen etc.

If you can configure debian to work, it's fine, if we get some inexplicable
problem on debian we will start to wonder though.
it's happened more than once with people doing nothing other than changing
OS =D



On Thu, Aug 27, 2009 at 4:55 PM, Jay Binks <jaybinks at gmail.com> wrote:

> Anthony can you ( or anyone else alao ). Please elaborate on what
> makes centos 5.3 o much better for Freeswitch.
>
> Is there some specific library vesiion on centos that makes a massive
> difference ?
>
> Reason I ask ...   I personally only have a preference for debian,
> but others may have policy mandated Os's
> For their companies, and it would be great to have some info about this.
>
> I'd imagine this actually boils down to a requirement for libfoo
> version X ...  And if we ran those library versions on other OS's we'd
> be fine ?
>
> Jay
>
>
>
> >
> >> I recommend you try your application on 64 bit CentOS which is the
> >> platform
> >> all of our paid customers use.
> >
> > hmm. our operations-people wont be happy with that.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From jason at jasonjgw.net  Thu Aug 27 17:10:50 2009
From: jason at jasonjgw.net (Jason White)
Date: Fri, 28 Aug 2009 10:10:50 +1000
Subject: [Freeswitch-users] memory leak
In-Reply-To: <3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
	<3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
Message-ID: <20090828001050.GA8108@jdc.jasonjgw.net>

Jay Binks <jaybinks at gmail.com> wrote:
 
> Reason I ask ...   I personally only have a preference for debian,   
> but others may have policy mandated Os's
> For their companies, and it would be great to have some info about this.

The only problem I've had with FreeSWITCH under Debian Squeeze and Sid
involves TLS-related segmentation faults that appear to be related to
something in the version of libssl supplied with Debian. The same problem
can't be reproduced on Fedora, for example, but it does occur under Debian
Lenny as well as Debian Squeeze and Sid (i.e., "testing" and "unstable",
respectively).

besides this, all of the issues that I have encountered turned out to be
(usually short-lived) bugs in FreeSWITCH or one of the libraries included in
the source tree - they're mostly FreeSWITCH issues.

I should point out that the FreeSWITCH developers are very good at avoiding
the introduction of bugs into their code and that known bugs get fixed. It
appears to be an unchanging fact about programming that with a large and
complex project, even given highly knowledgeable, experienced, committed and
talented developers (as we have with FreeSWITCH), sometimes, bugs do slip
through.




From lakindia89 at gmail.com  Thu Aug 27 22:01:44 2009
From: lakindia89 at gmail.com (lakshmanan ganapathy)
Date: Fri, 28 Aug 2009 10:31:44 +0530
Subject: [Freeswitch-users] How to make a call back
In-Reply-To: <87f2f3b90908270916l7cc7f3cap36bcf8725fa5fa5f@mail.gmail.com>
References: <25166083.post@talk.nabble.com>
	<87f2f3b90908270916l7cc7f3cap36bcf8725fa5fa5f@mail.gmail.com>
Message-ID: <7d79b3930908272201i3267aa54nafcc0171975cb023@mail.gmail.com>

No. In the dial plan I said, application="perl" data="The perl script".
I also checked $session->execute("bridge","user/1010"). This is working
fine.
But originate is not working as I expected.

On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com> wrote:
>
>>
>> When I give the following from the command line it calls to 1010 extension
>> and once answered, it calls to 1000 and bridge the connection.
>>            originate user/1010 &bridge(user/1000)
>> But I want to do this in perl. So I have given as follows
>>            $session->originate($session,"user/1010 &bridge user/1000");
>> But it is not working. It says "user/1010 &bridge user/1000 is invalid
>> user".
>> How to do this in perl. pls help.
>
>
> Are you calling this perl script from the CLI? If so you won't have the
> $session object because a channel does not exist for a simple API call.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From csa at nowthor.com  Thu Aug 27 22:13:38 2009
From: csa at nowthor.com (Carlos S. Antunes)
Date: Fri, 28 Aug 2009 01:13:38 -0400
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <DEBF02AF-E36C-47FA-88C7-363BCC91282F@freeswitch.org>
References: <4A8D87D8.8080001@nowthor.com>	<700500D9-4508-4848-A198-76692574AF15@jerris.com>
	<DEBF02AF-E36C-47FA-88C7-363BCC91282F@freeswitch.org>
Message-ID: <4A976782.5000101@nowthor.com>

Brian,

You've been vindicated. Callcentric is now advertising zero weighted SRV 
records! :)

I've re-enabled SRV lookups for the Callcentric profile and will monitor 
to see if I get any errors.

Carlos

Brian West wrote:
> Or as I have argued today they should fix their SRV records to be zero  
> weighted.
>
> /b
>
> On Aug 20, 2009, at 1:36 PM, Michael Jerris wrote:
>
>   
>> You can bypass the srv records if you like by passing a :port with the
>> hostname where you use it in freeswitch.
>>     
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>   
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From msc at freeswitch.org  Thu Aug 27 22:57:59 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Thu, 27 Aug 2009 22:57:59 -0700
Subject: [Freeswitch-users] How to make a call back
Message-ID: <5F197C4F-2AF5-4DFA-AED5-045B8B08E342@freeswitch.org>



Sent from my iPhone

On Aug 27, 2009, at 10:01 PM, lakshmanan ganapathy  
<lakindia89 at gmail.com> wrote:

> No. In the dial plan I said, application="perl" data="The perl  
> script".
> I also checked $session->execute("bridge","user/1010"). This is  
> working fine.
> But originate is not working as I expected.
>
I think you might be confusing Dialplan apps with API commands. The  
$session object represents an existing channel and therefore it uses  
Dialplan apps. Originate is an API, that is, it is a command that you  
type at the CLI.

You need an API object to use originate from a script:
my $api = new $freeswitch::API();
my $res = $api->executeString("originate user/1010 &bridge(user/1000);

What kind of application are you developing? I'm curious why you need  
an originate to create a whole new call.

-MC

> On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins  
> <msc at freeswitch.org> wrote:
>
>
> On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com>  
> wrote:
>
> When I give the following from the command line it calls to 1010  
> extension
> and once answered, it calls to 1000 and bridge the connection.
>            originate user/1010 &bridge(user/1000)
> But I want to do this in perl. So I have given as follows
>            $session->originate($session,"user/1010 &bridge user/ 
> 1000");
> But it is not working. It says "user/1010 &bridge user/1000 is invalid
> user".
> How to do this in perl. pls help.
>
> Are you calling this perl script from the CLI? If so you won't have  
> the $session object because a channel does not exist for a simple  
> API call.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
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From jmesquita at freeswitch.org  Thu Aug 27 23:09:26 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Fri, 28 Aug 2009 03:09:26 -0300
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A976782.5000101@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>
	<700500D9-4508-4848-A198-76692574AF15@jerris.com>
	<DEBF02AF-E36C-47FA-88C7-363BCC91282F@freeswitch.org>
	<4A976782.5000101@nowthor.com>
Message-ID: <ebbc2d550908272309i49620e97qdbf01b0efb2c8061@mail.gmail.com>

Bkw, I would recommend charging a fee from callcentric for the
consultancy. This consultant thing can get you going someday! LOL

Jmesquita funny joke

On 8/28/09, Carlos S. Antunes <csa at nowthor.com> wrote:
> Brian,
>
> You've been vindicated. Callcentric is now advertising zero weighted SRV
> records! :)
>
> I've re-enabled SRV lookups for the Callcentric profile and will monitor
> to see if I get any errors.
>
> Carlos
>
> Brian West wrote:
>> Or as I have argued today they should fix their SRV records to be zero
>> weighted.
>>
>> /b
>>
>> On Aug 20, 2009, at 1:36 PM, Michael Jerris wrote:
>>
>>
>>> You can bypass the srv records if you like by passing a :port with the
>>> hostname where you use it in freeswitch.
>>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>

-- 
Sent from my mobile device



From lakindia89 at gmail.com  Thu Aug 27 23:43:08 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Thu, 27 Aug 2009 23:43:08 -0700 (PDT)
Subject: [Freeswitch-users] How to make a call back
In-Reply-To: <5F197C4F-2AF5-4DFA-AED5-045B8B08E342@freeswitch.org>
References: <25166083.post@talk.nabble.com>
	<5F197C4F-2AF5-4DFA-AED5-045B8B08E342@freeswitch.org>
Message-ID: <25184832.post@talk.nabble.com>


The following is the requirement.
Let say I have 3 extensions in my freeswitch. (777,1000,1010).
When 1000 made a call to 777, I'll execute a perl application.
In that, I'll get the DTMF (1010) from the caller. Then I'll make a call to
the entered digit and I've to bridge 1000 and 1010.

This is what my requirement.
I also tried the following.
#!/usr/bin/perl
use strict;
use freeswitch;
our $session;
my $sess=new freeswitch::Session("user/1000");
$sess->answer();
if($sess->ready())
{
        freeswitch::consoleLog("INFO","Session is answered\n");
       
$sess->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav");
        my $dtmf = $sess->getDigits(4,"", 5000);
        freeswitch::consoleLog("INFO","I received $dtmf\n");
        my $new_sess=new freeswitch::Session("user/$dtmf");
        $new_sess->answer();
        if($new_sess->ready())
        {
                $session->bridge($sess,$new_sess);
        }
}

In the above program I got usage error on $session->bridge.



mercutioviz wrote:
> 
> 
> 
> Sent from my iPhone
> 
> On Aug 27, 2009, at 10:01 PM, lakshmanan ganapathy  
> <lakindia89 at gmail.com> wrote:
> 
>> No. In the dial plan I said, application="perl" data="The perl  
>> script".
>> I also checked $session->execute("bridge","user/1010"). This is  
>> working fine.
>> But originate is not working as I expected.
>>
> I think you might be confusing Dialplan apps with API commands. The  
> $session object represents an existing channel and therefore it uses  
> Dialplan apps. Originate is an API, that is, it is a command that you  
> type at the CLI.
> 
> You need an API object to use originate from a script:
> my $api = new $freeswitch::API();
> my $res = $api->executeString("originate user/1010 &bridge(user/1000);
> 
> What kind of application are you developing? I'm curious why you need  
> an originate to create a whole new call.
> 
> -MC
> 
>> On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins  
>> <msc at freeswitch.org> wrote:
>>
>>
>> On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com>  
>> wrote:
>>
>> When I give the following from the command line it calls to 1010  
>> extension
>> and once answered, it calls to 1000 and bridge the connection.
>>            originate user/1010 &bridge(user/1000)
>> But I want to do this in perl. So I have given as follows
>>            $session->originate($session,"user/1010 &bridge user/ 
>> 1000");
>> But it is not working. It says "user/1010 &bridge user/1000 is invalid
>> user".
>> How to do this in perl. pls help.
>>
>> Are you calling this perl script from the CLI? If so you won't have  
>> the $session object because a channel does not exist for a simple  
>> API call.
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/How-to-make-a-call-back-tp25166083p25184832.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From delianspam at gmail.com  Thu Aug 27 23:43:48 2009
From: delianspam at gmail.com (delianSPAM)
Date: Fri, 28 Aug 2009 09:43:48 +0300
Subject: [Freeswitch-users] How to make a call back
In-Reply-To: <7d79b3930908272201i3267aa54nafcc0171975cb023@mail.gmail.com>
References: <25166083.post@talk.nabble.com>	<87f2f3b90908270916l7cc7f3cap36bcf8725fa5fa5f@mail.gmail.com>
	<7d79b3930908272201i3267aa54nafcc0171975cb023@mail.gmail.com>
Message-ID: <!&!AAAAAAAAAAAYAAAAAAAAANMBDFPaQalKlwrBu4ftg1vCgAAAEAAAALucXIH9fG1HosZ+if323WMBAAAAAA==@gmail.com>

Hello Michael!

 

#Does this Python code helps:

new_api_obj = API()

result = new_api_obj.executeString("show channels"))             # Replace
"show channels" with a command of your choice

 

# By the way, I use the command below to connect an existing and answered
leg to a second one:

destination=1000

session.execute("bridge", "sofia/internal/" + destination +
"@yourpbx.domain.com")

 

Best Regards, 

Delian Tashev

 

 

From: lakshmanan ganapathy [mailto:lakindia89 at gmail.com] 
Sent: Friday, August 28, 2009 8:02 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] How to make a call back

 

No. In the dial plan I said, application="perl" data="The perl script".
I also checked $session->execute("bridge","user/1010"). This is working
fine.
But originate is not working as I expected.

On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins <msc at freeswitch.org> wrote:

 

On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com> wrote:


When I give the following from the command line it calls to 1010 extension
and once answered, it calls to 1000 and bridge the connection.
           originate user/1010 &bridge(user/1000)
But I want to do this in perl. So I have given as follows
           $session->originate($session,"user/1010 &bridge user/1000");
But it is not working. It says "user/1010 &bridge user/1000 is invalid
user".
How to do this in perl. pls help.


Are you calling this perl script from the CLI? If so you won't have the
$session object because a channel does not exist for a simple API call. 
-MC


_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

 

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From juanbackson at gmail.com  Thu Aug 27 23:51:03 2009
From: juanbackson at gmail.com (Juan Backson)
Date: Fri, 28 Aug 2009 14:51:03 +0800
Subject: [Freeswitch-users] freeswitch odbc problem
Message-ID: <27c25bc40908272351g345ad1e2w31dc945e8f9a0331@mail.gmail.com>

Hi,
In a new environment, I am getting the following error when building the
latest freeswitch from svn.  Does anyone know how to resolve it?

gcc -I/usr/src/freeswitch-snapshot/src/include
-I/usr/src/freeswitch-snapshot/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-g -O2 -Wall -std=c99 -pedantic -o .libs/freeswitch freeswitch-switch.o  -lm
./.libs/libfreeswitch.so
-L/usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib
/usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/usr/src/freeswitch-snapshot/libs/apr/.libs/libapr-1.a
-L/usr/src/freeswitch-snapshot/libs/srtp -L/usr/kerberos/lib64
libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt -lpthread
libs/libedit/src/.libs/libedit.a -lssl -lcrypto -ldl -lz -lncurses
-Wl,--rpath -Wl,/usr/local/freeswitch/lib
./.libs/libfreeswitch.so: undefined reference to `SQLGetData'
./.libs/libfreeswitch.so: undefined reference to `SQLDisconnect'
./.libs/libfreeswitch.so: undefined reference to `SQLSetConnectAttr'
./.libs/libfreeswitch.so: undefined reference to `SQLGetDiagRec'
./.libs/libfreeswitch.so: undefined reference to `SQLConnect'
./.libs/libfreeswitch.so: undefined reference to `SQLExecute'
./.libs/libfreeswitch.so: undefined reference to `SQLNumResultCols'
./.libs/libfreeswitch.so: undefined reference to `SQLFreeHandle'
./.libs/libfreeswitch.so: undefined reference to `SQLGetInfo'
./.libs/libfreeswitch.so: undefined reference to `SQLAllocHandle'
./.libs/libfreeswitch.so: undefined reference to `SQLDriverConnect'
./.libs/libfreeswitch.so: undefined reference to `SQLPrepare'
./.libs/libfreeswitch.so: undefined reference to `SQLError'
./.libs/libfreeswitch.so: undefined reference to `SQLRowCount'
./.libs/libfreeswitch.so: undefined reference to `SQLDescribeCol'
./.libs/libfreeswitch.so: undefined reference to `SQLFetch'
./.libs/libfreeswitch.so: undefined reference to `SQLSetEnvAttr'
collect2: ld returned 1 exit status
make[2]: *** [freeswitch] Error 1
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From kawarod at laposte.net  Thu Aug 27 23:54:00 2009
From: kawarod at laposte.net (rod)
Date: Fri, 28 Aug 2009 10:54:00 +0400
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
In-Reply-To: <999373713.8871.1251386588649.JavaMail.apache@mail22.abv.bg>
References: <999373713.8871.1251386588649.JavaMail.apache@mail22.abv.bg>
Message-ID: <4A977F08.9040005@laposte.net>

Hello,

the trace seems good.
If you check the answer from Kamailio, you'll see that Kamailio answers 
with "302 PEER_01".

As Michael Collins stated before, you can get the variable containing 
"PEER_01", then this variable is stored in a custom variable.

In your dialplan, may you please add:

<action application="info"/>, just before the transfer line, eg:

     <condition field="destination_number" expression="(\d+)$">
         <action application="set" data="hangup_after_bridge=true"/>
         <action application="set" data="continue_on_fail=true"/>
         <action application="export" data="sip_h_X-ROUTE=LOOKUP"/>
         <action application="bridge" data="sofia/internal/${destination_number}@127.0.0.1:5062"/>
         <action application="set" data="ROUTE_GW=${sip_redirect_contact_user_0}"/>
         <action application="set" data="AREA=${sip_redirect_contact_user_0}"/>
	 <action application="info"/>
         <action application="transfer" data="${destination_number} XML ROUTING"/>
      </condition>


Using application Info, you'll see on the console (or CDR) the list of 
variables used for this call.  You should see the content of 
"${sip_redirect_contact_user_0}" that should contain  the value 
"PEER_01". Please check this and let me know.

For the new configuration file, no problem for sharing. But as I wrote 
on the wiki page, I worked on this setup cause LCR module was not 
available when I start working on FS, nor I'm a good programmer to write 
a server side HTTP script (used by xml_curl) that could scale to my needs.

I enhanced a bit this configuration with support for fallback routing, 
almost realtime graph (every minutes using www.cacti.net and some 
functions in FS like limit_hash) of number of concurrent calls per AREA, 
PEER...

Thanks for "the good tutorial", but don't forget the dev team who did 
this great product ;-)

rod.


Hristo Benev a ?crit :
>  I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).
>
> If you need additional info I'll provide it.
>
> Here is trace:
>
> ngrep -d any -nn -i '1000' port 5062 -W byline
> interface: any
> filter: (ip or ip6) and ( port 5062 )
> match: 1000
> #
> U 10.10.10.10:5090 -> 127.0.0.1:5062
> INVITE sip:1000 at 127.0.0.1:5062 SIP/2.0.
> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
> Max-Forwards: 69.
> From: "Extension 1001" <sip:1001 at 10.10.10.10>;tag=978g69jZaFpBD.
> To: <sip:1000 at 127.0.0.1:5062>.
> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
> CSeq: 119574771 INVITE.
> Contact: <sip:mod_sofia at 10.10.10.10:5090>.
> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
> Supported: timer, precondition, path, replaces.
> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
> Content-Type: application/sdp.
> Content-Disposition: session.
> Content-Length: 429.
> X-ROUTE: LOOKUP.
> Remote-Party-ID: "Extension 1001" <sip:1001 at 10.10.10.10>;party=calling;screen=yes;privacy=off.
> .
> v=0.
> o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
> s=FreeSWITCH.
> c=IN IP4 10.10.10.10.
> t=0 0.
> m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:115 G7221/32000.
> a=fmtp:115 bitrate=48000.
> a=rtpmap:107 G7221/16000.
> a=fmtp:107 bitrate=32000.
> a=rtpmap:9 G722/8000.
> a=rtpmap:8 PCMA/8000.
> a=rtpmap:3 GSM/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=rtpmap:13 CN/8000.
> a=ptime:20.
>
> #
> U 127.0.0.1:5062 -> 10.10.10.10:5090
> SIP/2.0 302 PEER_01.
> Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
> From: "Extension 1001" <sip:1001 at 10.10.10.10>;tag=978g69jZaFpBD.
> To: <sip:1000 at 127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8.
> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
> CSeq: 119574771 INVITE.
> Contact: sip:France at PEER_01.
> Server: Kamailio (1.5.2-notls (i386/linux)).
> Content-Length: 0.
> .
>
> #
> U 10.10.10.10:5090 -> 127.0.0.1:5062
> ACK sip:1000 at 127.0.0.1:5062 SIP/2.0.
> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
> Max-Forwards: 69.
> From: "Extension 1001" <sip:1001 at 10.10.10.10>;tag=978g69jZaFpBD.
> To: <sip:1000 at 127.0.0.1:5062>;tag=458fb4012080e656b6742c09466dabcd.31c8.
> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
> CSeq: 119574771 ACK.
> Content-Length: 0.
> .
>
>
>
>
>
>  >-------- ?????????? ????? --------
>  >??:  Hristo Benev 
>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >??: freeswitch-users at lists.freeswitch.org
>  >????????? ??: ?????????, 2009, ?????? 27 14:58:53 EEST
>
>  > 
>  >Bojnour,
>  >
>  >I'll send a trace ASAP.
>  >
>  >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
>  >
>  >I'm really interested to see the updates of the project.
>  >
>  >Thank you for the good tutorial.
>  >
>  >Hristo
>  >
>  >
>  > >-------- ?????????? ????? --------
>  > >??:  rod 
>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  > >??: freeswitch-users at lists.freeswitch.org
>  > >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST
>  >
>  > >Hi Hristo,
>  > >
>  > >I'm the author of this setup and wiki page. I did a lot of modifications 
>  > >on this setup (alternative routing if failure essentially) but don't 
>  > >have too much time to update the wiki.
>  > >
>  > >May you please send me an ngrep trace when you call 1000:
>  > >
>  > >ngrep -d any -nn -i '1000' port 5060 -W byline 
>  > >
>  > >I will check what's happening.
>  > >Do you have an entry for 1000 in your mysql database ?
>  > >
>  > >regards,
>  > >rod
>  > >
>  > >Hristo Benev a ?crit :
>  > >>  
>  > >> It seems that the problem is on kamailio configuration.
>  > >> Will ask on their list.
>  > >>
>  > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
>  > >>
>  > >> I have it setup as friend in asterisk, but still ???
>  > >>
>  > >> Any ideas?
>  > >>
>  > >> Thanks,
>  > >>
>  > >>
>  > >>  >-------- ?????????? ????? --------
>  > >>  >??:  Hristo Benev 
>  > >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  > >>  >??: freeswitch-users at lists.freeswitch.org
>  > >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
>  > >>
>  > >>  > I think that the problem is here:
>  > >>  >-------------------------
>  > >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
>  > >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
>  > >>  >--------------------------
>  > >>  >
>  > >>  >Actually Regex FAIL
>  > >>  >
>  > >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
>  > >>  >Here is my  default.xml:
>  > >>  >----------------
>  > >>  >
>  > >>  >
>  > >>  >
>  > >>  >  
>  > >>  >
>  > >>  >     
>  > >>  >      
>  > >>  >         
>  > >>  >         
>  > >>  >         
>  > >>  >         
>  > >>  >         
>  > >>  >         
>  > >>  >         
>  > >>  >      
>  > >>  >    
>  > >>  >
>  > >>  >   
>  > >>  >
>  > >>  >  
>  > >>  >
>  > >>  >    
>  > >>  >      
>  > >>  >        
>  > >>  >        
>  > >>  >        
>  > >>  >        
>  > >>  >        
>  > >>  >        
>  > >>  >        
>  > >>  >        
>  > >>  >      
>  > >>  >    
>  > >>  >
>  > >>  >  
>  > >>  >
>  > >>  >
>  > >>  >
>  > >>  >--------------------------
>  > >>  >
>  > >>  >
>  > >>  >
>  > >>  > >-------- ?????????? ????? --------
>  > >>  > >??:  Brian West 
>  > >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  > >>  > >??: freeswitch-users at lists.freeswitch.org
>  > >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
>  > >>  >
>  > >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
>  > >>  >You have to process all 302's in the dialplan. 
>  > >>  >Set this on your sofia profile
>  > >>  >You can set these variables sip_redirect_profile,
>  > >>  >sip_redirect_context,
>  > >>  >sip_redirect_dialplan,
>  > >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
>  > >>  >sip_redirected_to,
>  > >>  >sip_redirect_contact_user_%d,
>  > >>  >sip_redirect_contact_host_%d,
>  > >>  >sip_redirect_contact_params_%d,
>  > >>  >sip_redirect_dialstring_%d,
>  > >>  >sip_redirect_dialstring,
>  > >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
>  > >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
>  > >>  > >
>  > >>  >
>  > >>  >_______________________________________________
>  > >>  >FreeSWITCH-users mailing list
>  > >>  >FreeSWITCH-users at lists.freeswitch.org
>  > >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  > >>  >http://www.freeswitch.org
>  > >>  >
>  > >>
>  > >> _______________________________________________
>  > >> FreeSWITCH-users mailing list
>  > >> FreeSWITCH-users at lists.freeswitch.org
>  > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  > >> http://www.freeswitch.org
>  > >>   
>  > >
>  > >_______________________________________________
>  > >FreeSWITCH-users mailing list
>  > >FreeSWITCH-users at lists.freeswitch.org
>  > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  > >http://www.freeswitch.org
>  > >
>  >
>  >_______________________________________________
>  >FreeSWITCH-users mailing list
>  >FreeSWITCH-users at lists.freeswitch.org
>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >http://www.freeswitch.org
>  >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From lakindia89 at gmail.com  Fri Aug 28 00:06:40 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Fri, 28 Aug 2009 00:06:40 -0700 (PDT)
Subject: [Freeswitch-users] How to make a call back
In-Reply-To: <25184832.post@talk.nabble.com>
References: <25166083.post@talk.nabble.com>
	<5F197C4F-2AF5-4DFA-AED5-045B8B08E342@freeswitch.org>
	<25184832.post@talk.nabble.com>
Message-ID: <25185052.post@talk.nabble.com>


Hey I'm sorry. I've solved this as follows
my $sess=new freeswitch::Session("user/1000");
$sess->answer();
#$sess->waitForAnswer($session);
if($sess->ready())
{
    freeswitch::consoleLog("INFO","Session is answered\n");
   
$sess->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav");
    my $dtmf = $sess->getDigits(4,"", 5000);
    freeswitch::consoleLog("INFO","I received $dtmf\n");
    $sess->execute("bridge","user/$dtmf");
}


lakshmanan wrote:
> 
> I also tried the following.
> #!/usr/bin/perl
> use strict;
> use freeswitch;
> our $session;
> my $sess=new freeswitch::Session("user/1000");
> $sess->answer();
> if($sess->ready())
> {
>         freeswitch::consoleLog("INFO","Session is answered\n");
>        
> $sess->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav");
>         my $dtmf = $sess->getDigits(4,"", 5000);
>         freeswitch::consoleLog("INFO","I received $dtmf\n");
>         my $new_sess=new freeswitch::Session("user/$dtmf");
>         $new_sess->answer();
>         if($new_sess->ready())
>         {
>                 $session->bridge($sess,$new_sess);
>         }
> }
> 
> In the above program I got usage error on $session->bridge.
> 
> 
> 
> mercutioviz wrote:
>> 
>> 
>> 
>> Sent from my iPhone
>> 
>> On Aug 27, 2009, at 10:01 PM, lakshmanan ganapathy  
>> <lakindia89 at gmail.com> wrote:
>> 
>>> No. In the dial plan I said, application="perl" data="The perl  
>>> script".
>>> I also checked $session->execute("bridge","user/1010"). This is  
>>> working fine.
>>> But originate is not working as I expected.
>>>
>> I think you might be confusing Dialplan apps with API commands. The  
>> $session object represents an existing channel and therefore it uses  
>> Dialplan apps. Originate is an API, that is, it is a command that you  
>> type at the CLI.
>> 
>> You need an API object to use originate from a script:
>> my $api = new $freeswitch::API();
>> my $res = $api->executeString("originate user/1010 &bridge(user/1000);
>> 
>> What kind of application are you developing? I'm curious why you need  
>> an originate to create a whole new call.
>> 
>> -MC
>> 
>>> On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins  
>>> <msc at freeswitch.org> wrote:
>>>
>>>
>>> On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com>  
>>> wrote:
>>>
>>> When I give the following from the command line it calls to 1010  
>>> extension
>>> and once answered, it calls to 1000 and bridge the connection.
>>>            originate user/1010 &bridge(user/1000)
>>> But I want to do this in perl. So I have given as follows
>>>            $session->originate($session,"user/1010 &bridge user/ 
>>> 1000");
>>> But it is not working. It says "user/1010 &bridge user/1000 is invalid
>>> user".
>>> How to do this in perl. pls help.
>>>
>>> Are you calling this perl script from the CLI? If so you won't have  
>>> the $session object because a channel does not exist for a simple  
>>> API call.
>>> -MC
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 

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From lakindia89 at gmail.com  Fri Aug 28 00:08:50 2009
From: lakindia89 at gmail.com (lakshmanan)
Date: Fri, 28 Aug 2009 00:08:50 -0700 (PDT)
Subject: [Freeswitch-users] How to make a call back
Message-ID: <25185052.post@talk.nabble.com>


Hey I'm sorry. I've solved this as follows
my $sess=new freeswitch::Session("user/1000");
$sess->answer();
#$sess->waitForAnswer($session);
if($sess->ready())
{
    freeswitch::consoleLog("INFO","Session is answered\n");
   
$sess->execute("playback","/usr/local/freeswitch/sounds/en/us/callie/time/8000/day-1.wav");
    my $dtmf = $sess->getDigits(4,"", 5000);
    freeswitch::consoleLog("INFO","I received $dtmf\n");
    $sess->execute("bridge","user/$dtmf");
}




mercutioviz wrote:
> 
> 
> 
> Sent from my iPhone
> 
> On Aug 27, 2009, at 10:01 PM, lakshmanan ganapathy  
> <lakindia89 at gmail.com> wrote:
> 
>> No. In the dial plan I said, application="perl" data="The perl  
>> script".
>> I also checked $session->execute("bridge","user/1010"). This is  
>> working fine.
>> But originate is not working as I expected.
>>
> I think you might be confusing Dialplan apps with API commands. The  
> $session object represents an existing channel and therefore it uses  
> Dialplan apps. Originate is an API, that is, it is a command that you  
> type at the CLI.
> 
> You need an API object to use originate from a script:
> my $api = new $freeswitch::API();
> my $res = $api->executeString("originate user/1010 &bridge(user/1000);
> 
> What kind of application are you developing? I'm curious why you need  
> an originate to create a whole new call.
> 
> -MC
> 
>> On Thu, Aug 27, 2009 at 9:46 PM, Michael Collins  
>> <msc at freeswitch.org> wrote:
>>
>>
>> On Wed, Aug 26, 2009 at 9:38 PM, lakshmanan <lakindia89 at gmail.com>  
>> wrote:
>>
>> When I give the following from the command line it calls to 1010  
>> extension
>> and once answered, it calls to 1000 and bridge the connection.
>>            originate user/1010 &bridge(user/1000)
>> But I want to do this in perl. So I have given as follows
>>            $session->originate($session,"user/1010 &bridge user/ 
>> 1000");
>> But it is not working. It says "user/1010 &bridge user/1000 is invalid
>> user".
>> How to do this in perl. pls help.
>>
>> Are you calling this perl script from the CLI? If so you won't have  
>> the $session object because a channel does not exist for a simple  
>> API call.
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 



-- 
View this message in context: http://www.nabble.com/How-to-make-a-call-back-tp25166083p25185052.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From cstomi.levlist at gmail.com  Fri Aug 28 03:59:57 2009
From: cstomi.levlist at gmail.com (Tamas Cseke)
Date: Fri, 28 Aug 2009 12:59:57 +0200
Subject: [Freeswitch-users] timeout while waiting for CF_BRIDGED flag
Message-ID: <4A97B8AD.3030009@gmail.com>

Hello,

we have a strange problem, sometimes calls are hung up with 
DESTINATION_OUT_OF_ORDER cause.
I added some extra log lines, and it turned out that

switch_ivr_bridge.c:140 a timeout occurs while waiting for CF_BRIDGED flag
switch_channel_wait_for_flag(chan_b, CF_BRIDGED, SWITCH_TRUE, 10000, 
chan_a); returns SWITCH_STATUS_TIMEOUT

As far as I understand the flag on B channel is set in another thread, 
so maybe it happens slow sometimes.
We are using r14144. I know it is quiet old, but maybe this code hasn't 
changed since them

Do you have any idea, what could cause this?
Do you think it is a bug?

Thanks in advance,
Tamas




From diego.viola at gmail.com  Fri Aug 28 03:59:50 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 28 Aug 2009 10:59:50 +0000
Subject: [Freeswitch-users] memory leak
In-Reply-To: <20090828001050.GA8108@jdc.jasonjgw.net>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
	<3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
	<20090828001050.GA8108@jdc.jasonjgw.net>
Message-ID: <86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com>

Cool, feel free to report these bugs in Jira. If they are indeed FreeSWITCH
bugs.

On Fri, Aug 28, 2009 at 12:10 AM, Jason White <jason at jasonjgw.net> wrote:

> Jay Binks <jaybinks at gmail.com> wrote:
>
> > Reason I ask ...   I personally only have a preference for debian,
> > but others may have policy mandated Os's
> > For their companies, and it would be great to have some info about this.
>
> The only problem I've had with FreeSWITCH under Debian Squeeze and Sid
> involves TLS-related segmentation faults that appear to be related to
> something in the version of libssl supplied with Debian. The same problem
> can't be reproduced on Fedora, for example, but it does occur under Debian
> Lenny as well as Debian Squeeze and Sid (i.e., "testing" and "unstable",
> respectively).
>
> besides this, all of the issues that I have encountered turned out to be
> (usually short-lived) bugs in FreeSWITCH or one of the libraries included
> in
> the source tree - they're mostly FreeSWITCH issues.
>
> I should point out that the FreeSWITCH developers are very good at avoiding
> the introduction of bugs into their code and that known bugs get fixed. It
> appears to be an unchanging fact about programming that with a large and
> complex project, even given highly knowledgeable, experienced, committed
> and
> talented developers (as we have with FreeSWITCH), sometimes, bugs do slip
> through.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From anatoliy at kounitskiy.com  Fri Aug 28 04:13:14 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Fri, 28 Aug 2009 14:13:14 +0300
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <87f2f3b90908271130v4bb7deads45d5f1c190ea0661@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
	<1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
	<191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>
	<1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>
	<87f2f3b90908271130v4bb7deads45d5f1c190ea0661@mail.gmail.com>
Message-ID: <1cd828b60908280413v6e5726cck72a5dcaa94ce3705@mail.gmail.com>

The wiki page is updated (
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer ). Hope
it is understandable :)

Anatoliy Kounitskiy

On Thu, Aug 27, 2009 at 9:30 PM, Michael Collins<msc at freeswitch.org> wrote:
> Thanks! I'll be in touch to help you if you need any wiki editing
> assistance.
> -MC
>
> On Thu, Aug 27, 2009 at 11:18 AM, Anatoliy Kounitskiy
> <anatoliy at kounitskiy.com> wrote:
>>
>> Thank you all for the great job :)
>> Now it works as I wanted!!
>>
>> Tomorrow I'll ?will try to update the wiki :)
>>
>> But with few words ( I tried to strip the extensions to minimum for
>> more easy read for users).
>> In the dialplan I have created the extension showed below:
>>
>> ?<extension name="local_number">
>> ? ?<condition field="destination_number" expression="^(\d{3})$">
>> ? ? ?<action application="set" data="dialed_extension=$1"/>
>> ? ? ?<action application="export" data="dialed_extension=$1"/>
>> ? ? ?<action application="bind_meta_app" data="1 b s
>> execute_extension::attented_xfer XML features"/>
>> ? ? ?<action application="set" data="transfer_ringback=$${hold_music}"/>
>> ? ? ?<action application="set" data="call_timeout=10"/>
>> ? ? ?<action application="set" data="hangup_after_bridge=true"/>
>> ? ? ?<action application="bridge"
>> data="user/${dialed_extension}@${domain_name}"/>
>> ? ?</condition>
>> ?</extension>
>>
>> In the features.xml file I have extension attented_xfer, that is using
>> the att_xfer application :
>>
>> ?<extension name="attented_xfer">
>> ? ? ?<condition field="destination_number" expression="^attented_xfer$">
>> ? ? ? ?<action application="set" data="continue_on_fail=true"/>
>> ? ? ? ?<action application="read" data="3 4 ivr/ivr-enter_ext.wav
>> attxfer_callthis 30000 #"/>
>> ? ? ? ?<action application="set" data="origination_cancel_key=#"/>
>> ? ? ? ?<action application="att_xfer"
>> data="user/${attxfer_callthis}@${domain_name}"/>
>> ? ? ?</condition>
>> ? ?</extension>
>>
>> Also I have set the the origination cancel key will be # - so if B
>> wants to cancel the call to C and to go back to A - just uses the #
>> key :)
>>
>> Again - thank you very much :) :)
>>
>> Anatoliy Kounitskiy
>>
>> On Thu, Aug 27, 2009 at 7:48 PM, Anthony
>> Minessale<anthony.minessale at gmail.com> wrote:
>> > please retest with latest SVN trunk
>> >
>> > On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy
>> > <anatoliy at kounitskiy.com> wrote:
>> >>
>> >> Just for information - the idea :
>> >> A---calls---> B ---att_xfer---> C ? ?, B hangs C up and goes back to A
>> >> ( I`m sorry C is not answering :) )
>> >>
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>>
>> --
>> Anatoliy Kounitskiy
>> -------------------------
>> E-mail: anatoliy at kounitskiy.com
>> Mobile: +359898913540
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540



From chad at apartmentlines.com  Fri Aug 28 04:58:59 2009
From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines)
Date: Fri, 28 Aug 2009 07:58:59 -0400
Subject: [Freeswitch-users] Who is "callie"?
In-Reply-To: <4A96E8F1.8010106@nowthor.com>
References: <4A96AEA8.3030308@nowthor.com>	<87f2f3b90908270918u66439bd2wd046a4dc485caeb8@mail.gmail.com>
	<D206B5F9-2D91-4652-A944-09CAD425ABBD@jerris.com>
	<4A96E8F1.8010106@nowthor.com>
Message-ID: <BAD6BA9A-2192-4676-A03E-4B5B02AA2063@apartmentlines.com>

i've also got a list of voicemail-related prompts that i'm going to  
use callie for.  i'm willing to throw my stuff into the pot, too, if  
we can have this turned around in two weeks.  otherwise i'll probably  
have to press on, as i have a deadline to make.  lemme know...


On Aug 27, 2009, at 4:13 PM, Carlos S. Antunes wrote:

> Mike,
>
> Sure! I am planning on doing a session "soon", maybe in a couple of  
> weeks or so. My only question is whether GM Voices license will  
> allow this kind of thing. Do you or anyone else know?
>
> Carlos
>
> Michael Jerris wrote:
>>
>> Also, note, we have a few sound files we would like to add to our  
>> default set.  If you are doing a session anyways and are willing  
>> please let me know if we can throw a few other prompts in your list  
>> to be recorded.
>>
>> Mike
>>
>> On Aug 27, 2009, at 12:18 PM, Michael Collins wrote:
>>
>>>
>>>
>>> On Thu, Aug 27, 2009 at 9:04 AM, Carlos S. Antunes  
>>> <csa at nowthor.com> wrote:
>>> Hi!
>>>
>>> I searched the wiki but couldn't find the answer. Is "callie" a real
>>> woman one may ask to record additional sounds?
>>>
>>> "Callie" is one of the voices from GM Voices. She is definitely  
>>> available for custom work. Visit www.gmvoices.com for more info.  
>>> Tell them that the FreeSWITCH project sent you. :)
>>> -MC
>>>
>>>
>>> Thanks!
>>>
>>> Carlos
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

-------------- next part --------------
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From brian at freeswitch.org  Fri Aug 28 06:10:59 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 28 Aug 2009 08:10:59 -0500
Subject: [Freeswitch-users] Authorizations when using DNS SRV bug?
In-Reply-To: <4A976782.5000101@nowthor.com>
References: <4A8D87D8.8080001@nowthor.com>	<700500D9-4508-4848-A198-76692574AF15@jerris.com>
	<DEBF02AF-E36C-47FA-88C7-363BCC91282F@freeswitch.org>
	<4A976782.5000101@nowthor.com>
Message-ID: <C8F7DF8A-C219-4745-893D-48D39ACA040D@freeswitch.org>

Thats still debatable.... The RFC's on these things read like chinese  
stereo instructions.

/b


On Aug 28, 2009, at 12:13 AM, Carlos S. Antunes wrote:

> Brian,
>
> You've been vindicated. Callcentric is now advertising zero weighted  
> SRV records! :)
>
> I've re-enabled SRV lookups for the Callcentric profile and will  
> monitor to see if I get any errors.
>
> Carlos




From nagalenoj at gmail.com  Thu Aug 27 23:44:31 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Thu, 27 Aug 2009 23:44:31 -0700 (PDT)
Subject: [Freeswitch-users]  How can it be done with freeswitch?
Message-ID: <25184845.post@talk.nabble.com>


Dear friends,
    I am using freeswitch for the last one month and trying to write perl
scripts to do some tasks. 
I am trying a new thing, which is something that I don't how to do it and I
don't whether it can be done with freeswitch.   my requirement is as
follows,
             A calls B[This extension has to execute a script].
             The script has to do the following,
              * Cut the call and callback to the number.
              * Get a phone number[DTMF] from A.
              * Dial the number(C) and bridge A and C.   
              * When C cuts the call, the control has to comeback to script.
              * Ask another number to A to dial out, get another phone
number[DTMF] from A.
              * Dial the number(D) and bridge A and D.

Please, help me to do this. Is there any other documentation to read about
the perl freeswitch functions.
-- 
View this message in context: http://www.nabble.com/How-can-it-be-done-with-freeswitch--tp25184845p25184845.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From math.parent at gmail.com  Fri Aug 28 02:15:04 2009
From: math.parent at gmail.com (Mathieu Parent)
Date: Fri, 28 Aug 2009 11:15:04 +0200
Subject: [Freeswitch-users] mod_fax not working
Message-ID: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com>

Hi,

I just started testing Freeswitch, this works well. This is powerfull
and easy to administrate.

I just have one problem and one request. The request is BOUNTY-5: IMAP
integration of voicemail (I will try to propose a patch if this is
within my competencies)

The problem is with mod_fax not receiving any fax.

Relevant extension is:
    <extension name="fax2mail">
        <condition field="destination_number" expression="^(9998)$">
            <action application="set" data="fax_verbose=true"/>
            <action application="set" data="fax_disable_v17=false"/>
            <action application="set" data="fax_use_ecm=true"/>
            <action application="answer" />
            <action application="playback" data="silence_stream://2000"/>
            <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/>
            <action application="hangup"/>
        </condition>
    </extension>

Test case is a 1-page fax coming from a Minolta printer connected to a
Cisco VG248 (via RJ11). The VG248 port is registred on a Cisco Call
Manager communicating with FS via a SIP trunk.

Log is attached:
- first try from 11:04:05 to 11:04:31
- 2nd try from 11:04:57 to 11:05:22
- 3rd try from 11:05:48 to 11:06:08

Do you need more information?

Regards

Mathieu Parent
-------------- next part --------------
2009-08-28 11:04:05.852705 [NOTICE] switch_channel.c:602 New Channel sofia/external/7629 at 85.0.30.241 [bd163f2c-93b1-11de-ae8e-27b1291d9722]
2009-08-28 11:04:05.852705 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_NEW
2009-08-28 11:04:05.852705 [DEBUG] switch_core_state_machine.c:404 (sofia/external/7629 at 85.0.30.241) State NEW
2009-08-28 11:04:05.852705 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [received][100]
2009-08-28 11:04:05.852705 [DEBUG] sofia.c:3296 Remote SDP:
v=0
o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 85.0.30.241
s=SIP Call
c=IN IP4 85.0.30.240
t=0 0
m=audio 28086 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2009-08-28 11:04:05.852705 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2009-08-28 11:04:05.852705 [DEBUG] sofia_glue.c:2029 Set Codec sofia/external/7629 at 85.0.30.241 PCMU/8000 20 ms 160 samples
2009-08-28 11:04:05.852705 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 101
2009-08-28 11:04:05.852705 [DEBUG] sofia.c:3455 (sofia/external/7629 at 85.0.30.241) State Change CS_NEW -> CS_INIT
2009-08-28 11:04:05.852705 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_INIT
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:481 (sofia/external/7629 at 85.0.30.241) State INIT
2009-08-28 11:04:05.856715 [DEBUG] mod_sofia.c:83 sofia/external/7629 at 85.0.30.241 SOFIA INIT
2009-08-28 11:04:05.856715 [DEBUG] mod_sofia.c:111 (sofia/external/7629 at 85.0.30.241) State Change CS_INIT -> CS_ROUTING
2009-08-28 11:04:05.856715 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:481 (sofia/external/7629 at 85.0.30.241) State INIT going to sleep
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_ROUTING
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:484 (sofia/external/7629 at 85.0.30.241) State ROUTING
2009-08-28 11:04:05.856715 [DEBUG] mod_sofia.c:130 sofia/external/7629 at 85.0.30.241 SOFIA ROUTING
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:78 sofia/external/7629 at 85.0.30.241 Standard ROUTING
2009-08-28 11:04:05.856715 [INFO] mod_dialplan_xml.c:315 Processing FAX D.I.C. 7629->9998 in context public
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->unloop] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->outside_call] continue=true
Dialplan: sofia/external/7629 at 85.0.30.241 Absolute Condition [outside_call]
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(outside_call=true)
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->call_debug] continue=true
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->public_extensions] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [public_extensions] destination_number(9998) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->vmain] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [vmain] destination_number(9998) =~ /^(9999)$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->fax2mail] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (PASS) [fax2mail] destination_number(9998) =~ /^(9998)$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_verbose=true)
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_disable_v17=false)
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_use_ecm=true)
Dialplan: sofia/external/7629 at 85.0.30.241 Action answer()
Dialplan: sofia/external/7629 at 85.0.30.241 Action playback(silence_stream://2000)
Dialplan: sofia/external/7629 at 85.0.30.241 Action rxfax(/tmp/${uuid}.rxfax.tiff)
Dialplan: sofia/external/7629 at 85.0.30.241 Action hangup()
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:114 (sofia/external/7629 at 85.0.30.241) State Change CS_ROUTING -> CS_EXECUTE
2009-08-28 11:04:05.856715 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:484 (sofia/external/7629 at 85.0.30.241) State ROUTING going to sleep
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_EXECUTE
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:491 (sofia/external/7629 at 85.0.30.241) State EXECUTE
2009-08-28 11:04:05.856715 [DEBUG] mod_sofia.c:173 sofia/external/7629 at 85.0.30.241 SOFIA EXECUTE
2009-08-28 11:04:05.856715 [DEBUG] switch_core_state_machine.c:151 sofia/external/7629 at 85.0.30.241 Standard EXECUTE
EXECUTE sofia/external/7629 at 85.0.30.241 set(outside_call=true)
2009-08-28 11:04:05.856715 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [outside_call]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_verbose=true)
2009-08-28 11:04:05.856715 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_verbose]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_disable_v17=false)
2009-08-28 11:04:05.856715 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_disable_v17]=[false]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_use_ecm=true)
2009-08-28 11:04:05.856715 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_use_ecm]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 answer()
2009-08-28 11:04:05.856715 [DEBUG] mod_dptools.c:649 sofia/external/7629 at 85.0.30.241 receive message [ANSWER]
2009-08-28 11:04:05.856715 [DEBUG] sofia_glue.c:2263 AUDIO RTP [sofia/external/7629 at 85.0.30.241] 10.30.15.137 port 31604 -> 85.0.30.240 port 28086 codec: 0 ms: 20
2009-08-28 11:04:05.856715 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms
2009-08-28 11:04:05.860708 [DEBUG] mod_sofia.c:552 Local SDP sofia/external/7629 at 85.0.30.241:
v=0
o=FreeSWITCH 1251418641 1251418642 IN IP4 10.30.15.137
s=FreeSWITCH
c=IN IP4 10.30.15.137
t=0 0
m=audio 31604 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-08-28 11:04:05.860708 [DEBUG] switch_core_session.c:630 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:05.860708 [NOTICE] mod_dptools.c:649 Channel [sofia/external/7629 at 85.0.30.241] has been answered
2009-08-28 11:04:05.860708 [DEBUG] switch_channel.c:182 sofia/external/7629 at 85.0.30.241 receive message [AUDIO_SYNC]
EXECUTE sofia/external/7629 at 85.0.30.241 playback(silence_stream://2000)
2009-08-28 11:04:05.860708 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 8000hz 1 channels 20ms
2009-08-28 11:04:05.860708 [DEBUG] switch_core_io.c:649 sofia/external/7629 at 85.0.30.241 receive message [TRANSCODING_NECESSARY]
2009-08-28 11:04:05.860708 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [completed][200]
2009-08-28 11:04:05.872721 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [ready][200]
2009-08-28 11:04:07.868730 [DEBUG] switch_ivr_play_say.c:1391 done playing file
EXECUTE sofia/external/7629 at 85.0.30.241 rxfax(/tmp/bd163f2c-93b1-11de-ae8e-27b1291d9722.rxfax.tiff)
2009-08-28 11:04:07.868730 [DEBUG] mod_fax.c:591 Raw read codec activation Success L16 20000
2009-08-28 11:04:07.868730 [DEBUG] mod_fax.c:607 Raw write codec activation Success L16
2009-08-28 11:04:07.868730 [DEBUG] switch_channel.c:182 sofia/external/7629 at 85.0.30.241 receive message [AUDIO_SYNC]
2009-08-28 11:04:07.908721 [DEBUG] switch_core_io.c:232 sofia/external/7629 at 85.0.30.241 receive message [TRANSCODING_NECESSARY]
2009-08-28 11:04:08.468727 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2009-08-28 11:04:08.488736 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1
2009-08-28 11:04:10.68454 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2009-08-28 11:04:10.568485 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_A_CED, state 1
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Starting answer mode
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_A_CED to T30_PHASE_B_TX
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 1 to 17
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  CSI without final frame tag
2009-08-28 11:04:10.948466 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30 DIS:
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Can receive fax: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= 2-D coding: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Polled sub-address: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= T.43 coding: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Plane interleave: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Selective polling: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Sub-addressing: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Password: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Character mode: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Digital network capability: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Duplex capability: Half only
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= JPEG coding: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Full colour mode: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Custom illuminant: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Custom gamut range: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DIS with final frame tag
2009-08-28 11:04:12.528716 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2009-08-28 11:04:12.988723 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:04:13.68696 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:04:13.68696 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2009-08-28 11:04:13.68696 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 4
2009-08-28 11:04:13.68696 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:04:13.68696 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4
2009-08-28 11:04:13.88698 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:13.108696 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:13.128697 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:13.148696 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:13.148696 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:13.168483 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:15.228722 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:15.728724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Retry number 1
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  CSI without final frame tag
2009-08-28 11:04:16.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30 DIS:
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Can receive fax: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= 2-D coding: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Polled sub-address: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= T.43 coding: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Plane interleave: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Selective polling: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Sub-addressing: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Password: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Character mode: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Digital network capability: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Duplex capability: Half only
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= JPEG coding: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Full colour mode: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Custom illuminant: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Custom gamut range: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DIS with final frame tag
2009-08-28 11:04:18.108708 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2009-08-28 11:04:18.588474 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:04:18.668474 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:04:18.668474 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2009-08-28 11:04:18.668474 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 4
2009-08-28 11:04:18.668474 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:04:18.668474 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4
2009-08-28 11:04:18.688478 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:18.708475 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:18.728726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:18.748726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:18.968729 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:19.188723 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:04:19.208726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:04:19.508723 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 17
2009-08-28 11:04:19.508723 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4A
2009-08-28 11:04:20.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:04:20.528725 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 7
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T4A (13920 remaining)
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  DCS with final frame tag
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 In state 17
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 DCS:
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Receive fax: Set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 00..= Selected data signalling rate: V.17 14400bps
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= 2-D coding: Not set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..00= Recording width: 215mm +- 1%
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Minimum scan line time: 0ms
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Frame size: 256 octets
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Selected compression 3
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Get document at 14400bps, modem 7
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 17 to 7
2009-08-28 11:04:20.788727 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-08-28 11:04:20.848726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:04:20.848726 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_C_NON_ECM_RX
2009-08-28 11:04:20.848726 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:04:20.848726 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 7
2009-08-28 11:04:20.848726 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:04:20.928727 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:04:20.928727 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:04:21.68725 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:04:21.128725 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2009-08-28 11:04:21.128725 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:04:21.128725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:04:21.128725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:04:21.248725 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training in progress (-3) in state 7
2009-08-28 11:04:21.748711 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:04:22.168711 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:04:22.508710 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:04:22.808710 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:04:22.868712 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:04:23.648711 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:04:24.48712 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2009-08-28 11:04:24.68714 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:04:24.68714 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:04:27.108725 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:04:27.108725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:04:27.268725 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:04:27.468727 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 7
2009-08-28 11:04:27.468727 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2A
2009-08-28 11:04:28.648725 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:04:28.648725 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 3
2009-08-28 11:04:28.648725 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T2A (14720 remaining)
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Stop none (0 remaining)
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  DCS with final frame tag
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 In state 7
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected DCS received in state 7
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 7 to 3
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DCN with final frame tag
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 fa
2009-08-28 11:04:28.928723 [DEBUG] mod_fax.c:137 FLOW FAX Switching from V.17 + V.21 to V.21 (-14.65dBm0)
2009-08-28 11:04:28.988725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 3
2009-08-28 11:04:28.988725 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_C_NON_ECM_RX to T30_PHASE_D_TX
2009-08-28 11:04:28.988725 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:04:28.988725 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:04:30.48729 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2009-08-28 11:04:30.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2009-08-28 11:04:30.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Disconnecting
2009-08-28 11:04:30.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E
2009-08-28 11:04:30.128724 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:04:30.128724 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 1
2009-08-28 11:04:30.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 3 to 2
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_E, state 2
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:167 ==============================================================================
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:180 Fax processing not successful - result (13) Unexpected message received.
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:185 Remote station id: 
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:186 Local station id:  SpanDSP Fax Ident
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:187 Pages transferred: 0
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:189 Total fax pages:   0
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:190 Image resolution:  8031x7700
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:191 Transfer Rate:     14400
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:193 ECM status         on
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:194 remote country:   
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:195 remote vendor:    
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:196 remote model:     
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:198 ==============================================================================
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 2 to 32
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 8
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 8
2009-08-28 11:04:31.128724 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete
2009-08-28 11:04:31.148726 [DEBUG] switch_core_codec.c:122 Restore original codec.
EXECUTE sofia/external/7629 at 85.0.30.241 hangup()
2009-08-28 11:04:31.148726 [NOTICE] mod_dptools.c:633 Hangup sofia/external/7629 at 85.0.30.241 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-28 11:04:31.148726 [DEBUG] switch_channel.c:1683 Send signal sofia/external/7629 at 85.0.30.241 [KILL]
2009-08-28 11:04:31.148726 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:491 (sofia/external/7629 at 85.0.30.241) State EXECUTE going to sleep
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_HANGUP
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:434 (sofia/external/7629 at 85.0.30.241) State HANGUP
2009-08-28 11:04:31.148726 [DEBUG] mod_sofia.c:338 Channel sofia/external/7629 at 85.0.30.241 hanging up, cause: NORMAL_CLEARING
2009-08-28 11:04:31.148726 [DEBUG] mod_sofia.c:396 Sending BYE to sofia/external/7629 at 85.0.30.241
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:46 sofia/external/7629 at 85.0.30.241 Standard HANGUP, cause: NORMAL_CLEARING
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:434 (sofia/external/7629 at 85.0.30.241) State HANGUP going to sleep
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:476 (sofia/external/7629 at 85.0.30.241) State Change CS_HANGUP -> CS_REPORTING
2009-08-28 11:04:31.148726 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_REPORTING
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:612 (sofia/external/7629 at 85.0.30.241) State REPORTING
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:53 sofia/external/7629 at 85.0.30.241 Standard REPORTING, cause: NORMAL_CLEARING
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:612 (sofia/external/7629 at 85.0.30.241) State REPORTING going to sleep
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:411 (sofia/external/7629 at 85.0.30.241) State Change CS_REPORTING -> CS_DESTROY
2009-08-28 11:04:31.148726 [DEBUG] switch_core_session.c:1068 Session 24 (sofia/external/7629 at 85.0.30.241) Locked, Waiting on external entities
2009-08-28 11:04:31.148726 [NOTICE] switch_core_session.c:1086 Session 24 (sofia/external/7629 at 85.0.30.241) Ended
2009-08-28 11:04:31.148726 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/7629 at 85.0.30.241 [CS_DESTROY]
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:564 (sofia/external/7629 at 85.0.30.241) State DESTROY
2009-08-28 11:04:31.148726 [DEBUG] mod_sofia.c:255 sofia/external/7629 at 85.0.30.241 SOFIA DESTROY
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:60 sofia/external/7629 at 85.0.30.241 Standard DESTROY
2009-08-28 11:04:31.148726 [DEBUG] switch_core_state_machine.c:564 (sofia/external/7629 at 85.0.30.241) State DESTROY going to sleep
2009-08-28 11:04:57.372710 [NOTICE] switch_channel.c:602 New Channel sofia/external/7629 at 85.0.30.241 [dbcb8bfc-93b1-11de-ae8e-27b1291d9722]
2009-08-28 11:04:57.372710 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_NEW
2009-08-28 11:04:57.372710 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [received][100]
2009-08-28 11:04:57.372710 [DEBUG] sofia.c:3296 Remote SDP:
v=0
o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 85.0.30.241
s=SIP Call
c=IN IP4 85.0.30.240
t=0 0
m=audio 28090 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2009-08-28 11:04:57.372710 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2009-08-28 11:04:57.372710 [DEBUG] sofia_glue.c:2029 Set Codec sofia/external/7629 at 85.0.30.241 PCMU/8000 20 ms 160 samples
2009-08-28 11:04:57.372710 [DEBUG] switch_core_state_machine.c:404 (sofia/external/7629 at 85.0.30.241) State NEW
2009-08-28 11:04:57.372710 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 101
2009-08-28 11:04:57.372710 [DEBUG] sofia.c:3455 (sofia/external/7629 at 85.0.30.241) State Change CS_NEW -> CS_INIT
2009-08-28 11:04:57.372710 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_INIT
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:481 (sofia/external/7629 at 85.0.30.241) State INIT
2009-08-28 11:04:57.376739 [DEBUG] mod_sofia.c:83 sofia/external/7629 at 85.0.30.241 SOFIA INIT
2009-08-28 11:04:57.376739 [DEBUG] mod_sofia.c:111 (sofia/external/7629 at 85.0.30.241) State Change CS_INIT -> CS_ROUTING
2009-08-28 11:04:57.376739 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:481 (sofia/external/7629 at 85.0.30.241) State INIT going to sleep
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_ROUTING
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:484 (sofia/external/7629 at 85.0.30.241) State ROUTING
2009-08-28 11:04:57.376739 [DEBUG] mod_sofia.c:130 sofia/external/7629 at 85.0.30.241 SOFIA ROUTING
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:78 sofia/external/7629 at 85.0.30.241 Standard ROUTING
2009-08-28 11:04:57.376739 [INFO] mod_dialplan_xml.c:315 Processing FAX D.I.C. 7629->9998 in context public
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->unloop] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->outside_call] continue=true
Dialplan: sofia/external/7629 at 85.0.30.241 Absolute Condition [outside_call]
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(outside_call=true)
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->call_debug] continue=true
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->public_extensions] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [public_extensions] destination_number(9998) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->vmain] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [vmain] destination_number(9998) =~ /^(9999)$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->fax2mail] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (PASS) [fax2mail] destination_number(9998) =~ /^(9998)$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_verbose=true)
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_disable_v17=false)
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_use_ecm=true)
Dialplan: sofia/external/7629 at 85.0.30.241 Action answer()
Dialplan: sofia/external/7629 at 85.0.30.241 Action playback(silence_stream://2000)
Dialplan: sofia/external/7629 at 85.0.30.241 Action rxfax(/tmp/${uuid}.rxfax.tiff)
Dialplan: sofia/external/7629 at 85.0.30.241 Action hangup()
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:114 (sofia/external/7629 at 85.0.30.241) State Change CS_ROUTING -> CS_EXECUTE
2009-08-28 11:04:57.376739 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:484 (sofia/external/7629 at 85.0.30.241) State ROUTING going to sleep
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_EXECUTE
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:491 (sofia/external/7629 at 85.0.30.241) State EXECUTE
2009-08-28 11:04:57.376739 [DEBUG] mod_sofia.c:173 sofia/external/7629 at 85.0.30.241 SOFIA EXECUTE
2009-08-28 11:04:57.376739 [DEBUG] switch_core_state_machine.c:151 sofia/external/7629 at 85.0.30.241 Standard EXECUTE
EXECUTE sofia/external/7629 at 85.0.30.241 set(outside_call=true)
2009-08-28 11:04:57.376739 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [outside_call]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_verbose=true)
2009-08-28 11:04:57.376739 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_verbose]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_disable_v17=false)
2009-08-28 11:04:57.376739 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_disable_v17]=[false]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_use_ecm=true)
2009-08-28 11:04:57.376739 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_use_ecm]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 answer()
2009-08-28 11:04:57.376739 [DEBUG] mod_dptools.c:649 sofia/external/7629 at 85.0.30.241 receive message [ANSWER]
2009-08-28 11:04:57.376739 [DEBUG] sofia_glue.c:2263 AUDIO RTP [sofia/external/7629 at 85.0.30.241] 10.30.15.137 port 32446 -> 85.0.30.240 port 28090 codec: 0 ms: 20
2009-08-28 11:04:57.376739 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms
2009-08-28 11:04:57.380744 [DEBUG] mod_sofia.c:552 Local SDP sofia/external/7629 at 85.0.30.241:
v=0
o=FreeSWITCH 1251417851 1251417852 IN IP4 10.30.15.137
s=FreeSWITCH
c=IN IP4 10.30.15.137
t=0 0
m=audio 32446 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-08-28 11:04:57.380744 [DEBUG] switch_core_session.c:630 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:04:57.380744 [NOTICE] mod_dptools.c:649 Channel [sofia/external/7629 at 85.0.30.241] has been answered
2009-08-28 11:04:57.380744 [DEBUG] switch_channel.c:182 sofia/external/7629 at 85.0.30.241 receive message [AUDIO_SYNC]
2009-08-28 11:04:57.380744 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [completed][200]
EXECUTE sofia/external/7629 at 85.0.30.241 playback(silence_stream://2000)
2009-08-28 11:04:57.380744 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 8000hz 1 channels 20ms
2009-08-28 11:04:57.380744 [DEBUG] switch_core_io.c:649 sofia/external/7629 at 85.0.30.241 receive message [TRANSCODING_NECESSARY]
2009-08-28 11:04:57.388729 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [ready][200]
2009-08-28 11:04:59.388741 [DEBUG] switch_ivr_play_say.c:1391 done playing file
EXECUTE sofia/external/7629 at 85.0.30.241 rxfax(/tmp/dbcb8bfc-93b1-11de-ae8e-27b1291d9722.rxfax.tiff)
2009-08-28 11:04:59.388741 [DEBUG] mod_fax.c:591 Raw read codec activation Success L16 20000
2009-08-28 11:04:59.388741 [DEBUG] mod_fax.c:607 Raw write codec activation Success L16
2009-08-28 11:04:59.388741 [DEBUG] switch_channel.c:182 sofia/external/7629 at 85.0.30.241 receive message [AUDIO_SYNC]
2009-08-28 11:04:59.428463 [DEBUG] switch_core_io.c:232 sofia/external/7629 at 85.0.30.241 receive message [TRANSCODING_NECESSARY]
2009-08-28 11:04:59.988724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2009-08-28 11:05:00.8724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1
2009-08-28 11:05:01.568724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2009-08-28 11:05:02.68697 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_A_CED, state 1
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Starting answer mode
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_A_CED to T30_PHASE_B_TX
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 1 to 17
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  CSI without final frame tag
2009-08-28 11:05:02.468726 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30 DIS:
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Can receive fax: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= 2-D coding: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Polled sub-address: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= T.43 coding: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Plane interleave: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Selective polling: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Sub-addressing: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Password: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Character mode: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Digital network capability: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Duplex capability: Half only
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= JPEG coding: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Full colour mode: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Custom illuminant: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Custom gamut range: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DIS with final frame tag
2009-08-28 11:05:04.48696 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2009-08-28 11:05:04.508695 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:04.588710 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:04.588710 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2009-08-28 11:05:04.588710 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 4
2009-08-28 11:05:04.588710 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:05:04.588710 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4
2009-08-28 11:05:04.608711 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:04.628710 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:04.648711 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:04.648711 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:04.668726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:04.688724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:06.748695 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:07.248745 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW T.30 T4 expired in phase T30_PHASE_B_RX, state 17
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW T.30 Retry number 1
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_B_TX
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  CSI without final frame tag
2009-08-28 11:05:08.48695 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30 DIS:
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Can receive fax: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= 2-D coding: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Polled sub-address: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= T.43 coding: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Plane interleave: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Selective polling: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Sub-addressing: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Password: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Character mode: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Digital network capability: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Duplex capability: Half only
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= JPEG coding: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Full colour mode: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Custom illuminant: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Custom gamut range: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DIS with final frame tag
2009-08-28 11:05:09.628461 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2009-08-28 11:05:10.108472 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:10.188469 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:10.188469 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2009-08-28 11:05:10.188469 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 4
2009-08-28 11:05:10.188469 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:05:10.188469 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4
2009-08-28 11:05:10.208465 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:10.228473 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:10.228473 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:10.248465 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:10.268468 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:10.268468 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:10.288466 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:10.288466 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:10.468464 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:10.688461 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 17
2009-08-28 11:05:10.688461 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4A
2009-08-28 11:05:10.788458 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:05:10.788458 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 7
2009-08-28 11:05:10.888470 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:05:10.888470 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 7
2009-08-28 11:05:12.28461 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:05:12.28461 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 7
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T4A (11840 remaining)
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  DCS with final frame tag
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 In state 17
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 DCS:
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Receive fax: Set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 00..= Selected data signalling rate: V.17 14400bps
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= 2-D coding: Not set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..00= Recording width: 215mm +- 1%
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Minimum scan line time: 0ms
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Frame size: 256 octets
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 Selected compression 3
2009-08-28 11:05:12.308695 [DEBUG] mod_fax.c:137 FLOW T.30 Get document at 14400bps, modem 7
2009-08-28 11:05:12.320684 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 17 to 7
2009-08-28 11:05:12.320684 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-08-28 11:05:12.368695 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:05:12.368695 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_C_NON_ECM_RX
2009-08-28 11:05:12.368695 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:12.368695 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 7
2009-08-28 11:05:12.368695 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:05:12.428696 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:05:12.428696 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:05:12.588696 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:05:12.628696 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2009-08-28 11:05:12.628696 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:05:12.628696 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:05:12.628696 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:05:12.748696 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training in progress (-3) in state 7
2009-08-28 11:05:13.268481 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:05:14.8720 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:05:14.328710 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:05:14.648695 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:05:15.568732 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2009-08-28 11:05:15.568732 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:05:18.608475 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:05:18.608475 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:05:18.748726 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:05:19.8726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 7
2009-08-28 11:05:19.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2A
2009-08-28 11:05:20.168726 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:05:20.168726 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 3
2009-08-28 11:05:20.168726 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T2A (14880 remaining)
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Stop none (0 remaining)
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  DCS with final frame tag
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 In state 7
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected DCS received in state 7
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 7 to 3
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DCN with final frame tag
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 fa
2009-08-28 11:05:20.448723 [DEBUG] mod_fax.c:137 FLOW FAX Switching from V.17 + V.21 to V.21 (-14.62dBm0)
2009-08-28 11:05:20.488724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 3
2009-08-28 11:05:20.488724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_C_NON_ECM_RX to T30_PHASE_D_TX
2009-08-28 11:05:20.488724 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:20.488724 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:05:21.548725 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2009-08-28 11:05:21.628731 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2009-08-28 11:05:21.628731 [DEBUG] mod_fax.c:137 FLOW T.30 Disconnecting
2009-08-28 11:05:21.628731 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E
2009-08-28 11:05:21.628731 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:21.628731 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 1
2009-08-28 11:05:21.628731 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 3 to 2
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_E, state 2
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:167 ==============================================================================
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:180 Fax processing not successful - result (13) Unexpected message received.
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:185 Remote station id: 
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:186 Local station id:  SpanDSP Fax Ident
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:187 Pages transferred: 0
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:189 Total fax pages:   0
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:190 Image resolution:  8031x7700
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:191 Transfer Rate:     14400
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:193 ECM status         on
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:194 remote country:   
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:195 remote vendor:    
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:196 remote model:     
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:198 ==============================================================================
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 2 to 32
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 8
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 8
2009-08-28 11:05:22.628725 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete
2009-08-28 11:05:22.648728 [DEBUG] switch_core_codec.c:122 Restore original codec.
EXECUTE sofia/external/7629 at 85.0.30.241 hangup()
2009-08-28 11:05:22.648728 [NOTICE] mod_dptools.c:633 Hangup sofia/external/7629 at 85.0.30.241 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-28 11:05:22.648728 [DEBUG] switch_channel.c:1683 Send signal sofia/external/7629 at 85.0.30.241 [KILL]
2009-08-28 11:05:22.648728 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:491 (sofia/external/7629 at 85.0.30.241) State EXECUTE going to sleep
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_HANGUP
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:434 (sofia/external/7629 at 85.0.30.241) State HANGUP
2009-08-28 11:05:22.648728 [DEBUG] mod_sofia.c:338 Channel sofia/external/7629 at 85.0.30.241 hanging up, cause: NORMAL_CLEARING
2009-08-28 11:05:22.648728 [DEBUG] mod_sofia.c:396 Sending BYE to sofia/external/7629 at 85.0.30.241
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:46 sofia/external/7629 at 85.0.30.241 Standard HANGUP, cause: NORMAL_CLEARING
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:434 (sofia/external/7629 at 85.0.30.241) State HANGUP going to sleep
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:476 (sofia/external/7629 at 85.0.30.241) State Change CS_HANGUP -> CS_REPORTING
2009-08-28 11:05:22.648728 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_REPORTING
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:612 (sofia/external/7629 at 85.0.30.241) State REPORTING
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:53 sofia/external/7629 at 85.0.30.241 Standard REPORTING, cause: NORMAL_CLEARING
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:612 (sofia/external/7629 at 85.0.30.241) State REPORTING going to sleep
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:411 (sofia/external/7629 at 85.0.30.241) State Change CS_REPORTING -> CS_DESTROY
2009-08-28 11:05:22.648728 [DEBUG] switch_core_session.c:1068 Session 25 (sofia/external/7629 at 85.0.30.241) Locked, Waiting on external entities
2009-08-28 11:05:22.648728 [NOTICE] switch_core_session.c:1086 Session 25 (sofia/external/7629 at 85.0.30.241) Ended
2009-08-28 11:05:22.648728 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/7629 at 85.0.30.241 [CS_DESTROY]
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:564 (sofia/external/7629 at 85.0.30.241) State DESTROY
2009-08-28 11:05:22.648728 [DEBUG] mod_sofia.c:255 sofia/external/7629 at 85.0.30.241 SOFIA DESTROY
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:60 sofia/external/7629 at 85.0.30.241 Standard DESTROY
2009-08-28 11:05:22.648728 [DEBUG] switch_core_state_machine.c:564 (sofia/external/7629 at 85.0.30.241) State DESTROY going to sleep
2009-08-28 11:05:48.912723 [NOTICE] switch_channel.c:602 New Channel sofia/external/7629 at 85.0.30.241 [fa83c460-93b1-11de-ae8e-27b1291d9722]
2009-08-28 11:05:48.912723 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_NEW
2009-08-28 11:05:48.912723 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [received][100]
2009-08-28 11:05:48.912723 [DEBUG] sofia.c:3296 Remote SDP:
v=0
o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 85.0.30.241
s=SIP Call
c=IN IP4 85.0.30.240
t=0 0
m=audio 28094 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

2009-08-28 11:05:48.912723 [DEBUG] sofia_glue.c:3071 Audio Codec Compare [PCMU:0:8000:20]/[PCMU:0:8000:20]
2009-08-28 11:05:48.912723 [DEBUG] sofia_glue.c:2029 Set Codec sofia/external/7629 at 85.0.30.241 PCMU/8000 20 ms 160 samples
2009-08-28 11:05:48.912723 [DEBUG] switch_core_state_machine.c:404 (sofia/external/7629 at 85.0.30.241) State NEW
2009-08-28 11:05:48.912723 [DEBUG] sofia_glue.c:3031 Set 2833 dtmf payload to 101
2009-08-28 11:05:48.912723 [DEBUG] sofia.c:3455 (sofia/external/7629 at 85.0.30.241) State Change CS_NEW -> CS_INIT
2009-08-28 11:05:48.912723 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_INIT
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:481 (sofia/external/7629 at 85.0.30.241) State INIT
2009-08-28 11:05:48.916860 [DEBUG] mod_sofia.c:83 sofia/external/7629 at 85.0.30.241 SOFIA INIT
2009-08-28 11:05:48.916860 [DEBUG] mod_sofia.c:111 (sofia/external/7629 at 85.0.30.241) State Change CS_INIT -> CS_ROUTING
2009-08-28 11:05:48.916860 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:481 (sofia/external/7629 at 85.0.30.241) State INIT going to sleep
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_ROUTING
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:484 (sofia/external/7629 at 85.0.30.241) State ROUTING
2009-08-28 11:05:48.916860 [DEBUG] mod_sofia.c:130 sofia/external/7629 at 85.0.30.241 SOFIA ROUTING
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:78 sofia/external/7629 at 85.0.30.241 Standard ROUTING
2009-08-28 11:05:48.916860 [INFO] mod_dialplan_xml.c:315 Processing FAX D.I.C. 7629->9998 in context public
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->unloop] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->outside_call] continue=true
Dialplan: sofia/external/7629 at 85.0.30.241 Absolute Condition [outside_call]
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(outside_call=true)
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->call_debug] continue=true
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->public_extensions] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [public_extensions] destination_number(9998) =~ /^(10[01][0-9])$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->vmain] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (FAIL) [vmain] destination_number(9998) =~ /^(9999)$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 parsing [public->fax2mail] continue=false
Dialplan: sofia/external/7629 at 85.0.30.241 Regex (PASS) [fax2mail] destination_number(9998) =~ /^(9998)$/ break=on-false
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_verbose=true)
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_disable_v17=false)
Dialplan: sofia/external/7629 at 85.0.30.241 Action set(fax_use_ecm=true)
Dialplan: sofia/external/7629 at 85.0.30.241 Action answer()
Dialplan: sofia/external/7629 at 85.0.30.241 Action playback(silence_stream://2000)
Dialplan: sofia/external/7629 at 85.0.30.241 Action rxfax(/tmp/${uuid}.rxfax.tiff)
Dialplan: sofia/external/7629 at 85.0.30.241 Action hangup()
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:114 (sofia/external/7629 at 85.0.30.241) State Change CS_ROUTING -> CS_EXECUTE
2009-08-28 11:05:48.916860 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:484 (sofia/external/7629 at 85.0.30.241) State ROUTING going to sleep
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_EXECUTE
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:491 (sofia/external/7629 at 85.0.30.241) State EXECUTE
2009-08-28 11:05:48.916860 [DEBUG] mod_sofia.c:173 sofia/external/7629 at 85.0.30.241 SOFIA EXECUTE
2009-08-28 11:05:48.916860 [DEBUG] switch_core_state_machine.c:151 sofia/external/7629 at 85.0.30.241 Standard EXECUTE
EXECUTE sofia/external/7629 at 85.0.30.241 set(outside_call=true)
2009-08-28 11:05:48.916860 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [outside_call]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_verbose=true)
2009-08-28 11:05:48.916860 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_verbose]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_disable_v17=false)
2009-08-28 11:05:48.916860 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_disable_v17]=[false]
EXECUTE sofia/external/7629 at 85.0.30.241 set(fax_use_ecm=true)
2009-08-28 11:05:48.916860 [DEBUG] mod_dptools.c:748 sofia/external/7629 at 85.0.30.241 SET [fax_use_ecm]=[true]
EXECUTE sofia/external/7629 at 85.0.30.241 answer()
2009-08-28 11:05:48.916860 [DEBUG] mod_dptools.c:649 sofia/external/7629 at 85.0.30.241 receive message [ANSWER]
2009-08-28 11:05:48.916860 [DEBUG] sofia_glue.c:2263 AUDIO RTP [sofia/external/7629 at 85.0.30.241] 10.30.15.137 port 24154 -> 85.0.30.240 port 28094 codec: 0 ms: 20
2009-08-28 11:05:48.916860 [DEBUG] switch_rtp.c:1138 Starting timer [soft] 160 bytes per 20ms
2009-08-28 11:05:48.920709 [DEBUG] mod_sofia.c:552 Local SDP sofia/external/7629 at 85.0.30.241:
v=0
o=FreeSWITCH 1251426194 1251426195 IN IP4 10.30.15.137
s=FreeSWITCH
c=IN IP4 10.30.15.137
t=0 0
m=audio 24154 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

2009-08-28 11:05:48.920709 [DEBUG] switch_core_session.c:630 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:05:48.920709 [NOTICE] mod_dptools.c:649 Channel [sofia/external/7629 at 85.0.30.241] has been answered
2009-08-28 11:05:48.920709 [DEBUG] switch_channel.c:182 sofia/external/7629 at 85.0.30.241 receive message [AUDIO_SYNC]
2009-08-28 11:05:48.920709 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [completed][200]
EXECUTE sofia/external/7629 at 85.0.30.241 playback(silence_stream://2000)
2009-08-28 11:05:48.920709 [DEBUG] switch_ivr_play_say.c:1097 Codec Activated L16 at 8000hz 1 channels 20ms
2009-08-28 11:05:48.920709 [DEBUG] switch_core_io.c:649 sofia/external/7629 at 85.0.30.241 receive message [TRANSCODING_NECESSARY]
2009-08-28 11:05:48.928705 [DEBUG] sofia.c:3289 Channel sofia/external/7629 at 85.0.30.241 entering state [ready][200]
2009-08-28 11:05:50.928734 [DEBUG] switch_ivr_play_say.c:1391 done playing file
EXECUTE sofia/external/7629 at 85.0.30.241 rxfax(/tmp/fa83c460-93b1-11de-ae8e-27b1291d9722.rxfax.tiff)
2009-08-28 11:05:50.928734 [DEBUG] mod_fax.c:591 Raw read codec activation Success L16 20000
2009-08-28 11:05:50.928734 [DEBUG] mod_fax.c:607 Raw write codec activation Success L16
2009-08-28 11:05:50.928734 [DEBUG] switch_channel.c:182 sofia/external/7629 at 85.0.30.241 receive message [AUDIO_SYNC]
2009-08-28 11:05:50.968739 [DEBUG] switch_core_io.c:232 sofia/external/7629 at 85.0.30.241 receive message [TRANSCODING_NECESSARY]
2009-08-28 11:05:51.548728 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2009-08-28 11:05:51.548728 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1
2009-08-28 11:05:53.128723 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 1
2009-08-28 11:05:53.628727 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 1
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_A_CED, state 1
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Starting answer mode
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_A_CED to T30_PHASE_B_TX
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 1 to 17
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Sending ident 'SpanDSP Fax Ident'
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  CSI without final frame tag
2009-08-28 11:05:54.8726 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 03 40 74 6e 65 64 49 20 78 61 46 20 50 53 44 6e 61 70 53 20 20 20
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30 DIS:
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= V.8 capabilities: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Preferred octets: 256 octets
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Ready to transmit a fax document (polling): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Can receive fax: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 11..= Supported data signalling rates: V.27 ter, V.29, and V.17
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= 2-D coding: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..10= Recording width: 215mm +- 1%, 255mm +- 1% and 303mm +- 1%
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Receiver's minimum scan line time: 0ms at 3.85 l/mm; T7.7 = T3.85
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= "Field not valid" supported: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Multiple selective polling: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Polled sub-address: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= T.43 coding: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Plane interleave: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Voice coding with 32kbit/s ADPCM (Rec. G.726): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Reserved for the use of extended voice coding set: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...1= R8x15.4lines/mm: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= 300x300pels/25.4mm: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= R16x15.4lines/mm and/or 400x400pels/25.4mm: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Inch-based resolution preferred: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= Metric-based resolution preferred: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Minimum scan line time for higher resolutions: T15.4 = T7.7
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Selective polling: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Sub-addressing: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Password: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transmit a data file (polling): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Binary file transfer (BFT): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Document transfer mode (DTM): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Electronic data interchange (EDI): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Basic transfer mode (BTM): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Ready to transfer a character or mixed mode document (polling): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Character mode: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Mixed mode (Annex E/T.4): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Processable mode 26 (Rec. T.505): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Digital network capability: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Duplex capability: Half only
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= JPEG coding: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ...0 ....= Full colour mode: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= 12bits/pel component: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= No subsampling (1:1:1): Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Custom illuminant: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Custom gamut range: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .... 1...= North American Letter (215.9mm x 279.4mm): Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ...1 ....= North American Legal (215.9mm x 355.6mm): Set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   ..0. ....= Single-progression sequential coding (Rec. T.85) basic: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   .0.. ....= Single-progression sequential coding (Rec. T.85) optional L0: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DIS with final frame tag
2009-08-28 11:05:55.588716 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 80 00 ee fa c4 80 95 80 80 80 18
2009-08-28 11:05:56.48725 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:56.128732 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_B_TX, state 17
2009-08-28 11:05:56.128732 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_TX to T30_PHASE_B_RX
2009-08-28 11:05:56.128732 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 4
2009-08-28 11:05:56.128732 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:05:56.128732 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4
2009-08-28 11:05:56.148725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:56.148725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:56.148725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:56.168727 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:56.168727 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:56.208724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:56.228725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:56.228725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:56.248726 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:56.268724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 17
2009-08-28 11:05:56.448723 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 17
2009-08-28 11:05:56.668724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 17
2009-08-28 11:05:56.668724 [DEBUG] mod_fax.c:137 FLOW T.30 Start T4A
2009-08-28 11:05:56.748724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 17
2009-08-28 11:05:56.928724 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:05:56.928724 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 7
2009-08-28 11:05:58.8714 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:05:58.8714 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 7
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T4A (11840 remaining)
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  DCS with final frame tag
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 In state 17
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 DCS:
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... ...0= Store and forward Internet fax (T.37): Not set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... .0..= Real-time Internet fax (T.38): Not set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= 3G mobile network: Not set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..1.= Receive fax: Set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   ..10 00..= Selected data signalling rate: V.17 14400bps
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= R8x7.7lines/mm and/or 200x200pels/25.4mm: Set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= 2-D coding: Not set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..00= Recording width: 215mm +- 1%
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... 10..= Recording length: Unlimited
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .111 ....= Minimum scan line time: 0ms
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   1... ....= Extension indicator: Set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... ..0.= Compressed/uncompressed mode: Compressed
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... .1..= Error correction mode (ECM): ECM
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .... 0...= Frame size: 256 octets
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   .1.. ....= T.6 coding: Set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30   0... ....= Extension indicator: Not set
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 Selected compression 3
2009-08-28 11:05:58.288715 [DEBUG] mod_fax.c:137 FLOW T.30 Get document at 14400bps, modem 7
2009-08-28 11:05:58.300704 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 17 to 7
2009-08-28 11:05:58.300704 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2
2009-08-28 11:05:58.348715 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:05:58.348715 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_B_RX to T30_PHASE_C_NON_ECM_RX
2009-08-28 11:05:58.348715 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:05:58.348715 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 7
2009-08-28 11:05:58.348715 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 0
2009-08-28 11:05:58.408715 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:05:58.408715 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:05:58.568726 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:05:58.608717 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2009-08-28 11:05:58.608717 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:05:58.608717 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:05:58.608717 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:05:58.748714 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training in progress (-3) in state 7
2009-08-28 11:05:59.248723 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:05:59.648725 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:06:00.48734 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:06:00.308724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Abort (-8) in state 7
2009-08-28 11:06:01.548728 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier down (-1) in state 7
2009-08-28 11:06:01.548728 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 7
2009-08-28 11:06:04.588723 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Carrier up (-2) in state 7
2009-08-28 11:06:04.588723 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier up (-2) in state 7
2009-08-28 11:06:04.728724 [DEBUG] mod_fax.c:137 FLOW T.30 Non-ECM signal status is Training failed (-5) in state 7
2009-08-28 11:06:04.828724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Framing OK (-6) in state 7
2009-08-28 11:06:04.828724 [DEBUG] mod_fax.c:137 FLOW T.30 Start T2A
2009-08-28 11:06:06.148723 [DEBUG] mod_fax.c:137 FLOW T.30 Bad HDLC CRC received
2009-08-28 11:06:06.148723 [DEBUG] mod_fax.c:137 FLOW T.30 Bad CRC and timer is 3
2009-08-28 11:06:06.148723 [DEBUG] mod_fax.c:137 FLOW T.30 Stop T2A (13760 remaining)
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Stop none (0 remaining)
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  DCS with final frame tag
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Rx:  ff 13 83 00 62 f8 44
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 In state 7
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Unexpected DCS received in state 7
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 7 to 3
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  DCN with final frame tag
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW T.30 Tx:  ff 13 fa
2009-08-28 11:06:06.428724 [DEBUG] mod_fax.c:137 FLOW FAX Switching from V.17 + V.21 to V.21 (-14.51dBm0)
2009-08-28 11:06:06.468724 [DEBUG] mod_fax.c:137 FLOW T.30 HDLC signal status is Carrier down (-1) in state 3
2009-08-28 11:06:06.468724 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_C_NON_ECM_RX to T30_PHASE_D_TX
2009-08-28 11:06:06.468724 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:06:06.468724 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 4
2009-08-28 11:06:07.548713 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2009-08-28 11:06:07.628704 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_D_TX, state 3
2009-08-28 11:06:07.628704 [DEBUG] mod_fax.c:137 FLOW T.30 Disconnecting
2009-08-28 11:06:07.628704 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_D_TX to T30_PHASE_E
2009-08-28 11:06:07.628704 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 0
2009-08-28 11:06:07.628704 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 1
2009-08-28 11:06:07.628704 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 3 to 2
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW T.30 Send complete in phase T30_PHASE_E, state 2
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:167 ==============================================================================
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:180 Fax processing not successful - result (13) Unexpected message received.
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:185 Remote station id: 
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:186 Local station id:  SpanDSP Fax Ident
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:187 Pages transferred: 0
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:189 Total fax pages:   0
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:190 Image resolution:  8031x7700
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:191 Transfer Rate:     14400
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:193 ECM status         on
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:194 remote country:   
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:195 remote vendor:    
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:196 remote model:     
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:198 ==============================================================================
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from state 2 to 32
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW T.30 Changing from phase T30_PHASE_E to T30_PHASE_CALL_FINISHED
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW FAX Set rx type 8
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW FAX Set tx type 8
2009-08-28 11:06:08.628536 [DEBUG] mod_fax.c:137 FLOW FAX FAX exchange complete
2009-08-28 11:06:08.648545 [DEBUG] switch_core_codec.c:122 Restore original codec.
EXECUTE sofia/external/7629 at 85.0.30.241 hangup()
2009-08-28 11:06:08.648545 [NOTICE] mod_dptools.c:633 Hangup sofia/external/7629 at 85.0.30.241 [CS_EXECUTE] [NORMAL_CLEARING]
2009-08-28 11:06:08.648545 [DEBUG] switch_channel.c:1683 Send signal sofia/external/7629 at 85.0.30.241 [KILL]
2009-08-28 11:06:08.648545 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:06:08.648545 [DEBUG] switch_core_state_machine.c:491 (sofia/external/7629 at 85.0.30.241) State EXECUTE going to sleep
2009-08-28 11:06:08.648545 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_HANGUP
2009-08-28 11:06:08.648545 [DEBUG] switch_core_state_machine.c:434 (sofia/external/7629 at 85.0.30.241) State HANGUP
2009-08-28 11:06:08.648545 [DEBUG] mod_sofia.c:338 Channel sofia/external/7629 at 85.0.30.241 hanging up, cause: NORMAL_CLEARING
2009-08-28 11:06:08.648545 [DEBUG] mod_sofia.c:396 Sending BYE to sofia/external/7629 at 85.0.30.241
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:46 sofia/external/7629 at 85.0.30.241 Standard HANGUP, cause: NORMAL_CLEARING
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:434 (sofia/external/7629 at 85.0.30.241) State HANGUP going to sleep
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:476 (sofia/external/7629 at 85.0.30.241) State Change CS_HANGUP -> CS_REPORTING
2009-08-28 11:06:08.652458 [DEBUG] switch_core_session.c:932 Send signal sofia/external/7629 at 85.0.30.241 [BREAK]
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:398 (sofia/external/7629 at 85.0.30.241) Running State Change CS_REPORTING
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:612 (sofia/external/7629 at 85.0.30.241) State REPORTING
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:53 sofia/external/7629 at 85.0.30.241 Standard REPORTING, cause: NORMAL_CLEARING
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:612 (sofia/external/7629 at 85.0.30.241) State REPORTING going to sleep
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:411 (sofia/external/7629 at 85.0.30.241) State Change CS_REPORTING -> CS_DESTROY
2009-08-28 11:06:08.652458 [DEBUG] switch_core_session.c:1068 Session 26 (sofia/external/7629 at 85.0.30.241) Locked, Waiting on external entities
2009-08-28 11:06:08.652458 [NOTICE] switch_core_session.c:1086 Session 26 (sofia/external/7629 at 85.0.30.241) Ended
2009-08-28 11:06:08.652458 [NOTICE] switch_core_session.c:1088 Close Channel sofia/external/7629 at 85.0.30.241 [CS_DESTROY]
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:564 (sofia/external/7629 at 85.0.30.241) State DESTROY
2009-08-28 11:06:08.652458 [DEBUG] mod_sofia.c:255 sofia/external/7629 at 85.0.30.241 SOFIA DESTROY
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:60 sofia/external/7629 at 85.0.30.241 Standard DESTROY
2009-08-28 11:06:08.652458 [DEBUG] switch_core_state_machine.c:564 (sofia/external/7629 at 85.0.30.241) State DESTROY going to sleep

From brian at freeswitch.org  Fri Aug 28 06:29:05 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 28 Aug 2009 08:29:05 -0500
Subject: [Freeswitch-users] Who is "callie"?
In-Reply-To: <4A96E8F1.8010106@nowthor.com>
References: <4A96AEA8.3030308@nowthor.com>	<87f2f3b90908270918u66439bd2wd046a4dc485caeb8@mail.gmail.com>
	<D206B5F9-2D91-4652-A944-09CAD425ABBD@jerris.com>
	<4A96E8F1.8010106@nowthor.com>
Message-ID: <EDADECDD-5644-4854-A8A2-E3DAF9A2DC19@freeswitch.org>

Yes you own the recordings.  Its work for hire.  Do you want to  
include the recordings in our sound repo?

/b

On Aug 27, 2009, at 3:13 PM, Carlos S. Antunes wrote:

> Mike,
>
> Sure! I am planning on doing a session "soon", maybe in a couple of  
> weeks or so. My only question is whether GM Voices license will  
> allow this kind of thing. Do you or anyone else know?
>
> Carlos




From anthony.minessale at gmail.com  Fri Aug 28 07:57:47 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 09:57:47 -0500
Subject: [Freeswitch-users] timeout while waiting for CF_BRIDGED flag
In-Reply-To: <4A97B8AD.3030009@gmail.com>
References: <4A97B8AD.3030009@gmail.com>
Message-ID: <191c3a030908280757n398077e8n9889ac58fde24b4@mail.gmail.com>

10 seconds is a pretty long time for the 2nd half of the bridge to start?
You can try increasing it but you probably have a more profound problem if
you are hitting that code.

When you start the bridge the code has 2 session pointers, it sets up state
handler on the b leg and changes state to
get it to execute the audio bridge thread then it executes the same thread
on the a leg so you end up with
2 sessions running audio_bridge_thread on each other.

Each leg thread uses that timeout you mentioned to make sure it's not
waiting forever for the bridge to start in case something
happens to the other one.

You should follow what each leg is doing and find out why the other one did
not make it into audio bridge thread.

It may be not answered or some other pre-conditon for thread. (are you
uuid_bridging 2 channels where neither one is answered yet?)


On Fri, Aug 28, 2009 at 5:59 AM, Tamas Cseke <cstomi.levlist at gmail.com>wrote:

> Hello,
>
> we have a strange problem, sometimes calls are hung up with
> DESTINATION_OUT_OF_ORDER cause.
> I added some extra log lines, and it turned out that
>
> switch_ivr_bridge.c:140 a timeout occurs while waiting for CF_BRIDGED flag
> switch_channel_wait_for_flag(chan_b, CF_BRIDGED, SWITCH_TRUE, 10000,
> chan_a); returns SWITCH_STATUS_TIMEOUT
>
> As far as I understand the flag on B channel is set in another thread,
> so maybe it happens slow sometimes.
> We are using r14144. I know it is quiet old, but maybe this code hasn't
> changed since them
>
> Do you have any idea, what could cause this?
> Do you think it is a bug?
>
> Thanks in advance,
> Tamas
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Fri Aug 28 08:49:32 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 08:49:32 -0700
Subject: [Freeswitch-users] freeswitch odbc problem
In-Reply-To: <27c25bc40908272351g345ad1e2w31dc945e8f9a0331@mail.gmail.com>
References: <27c25bc40908272351g345ad1e2w31dc945e8f9a0331@mail.gmail.com>
Message-ID: <87f2f3b90908280849w34bda66dt4d6ac3227b1fddf3@mail.gmail.com>

Juan,

Is this a new OS install? Make sure that you have all of the prerequisites
installed. Hopefully you're on CentOS and can just do this:

yum -y install unixODBC-devel gcc-c++ compat-gcc-32 compat-gcc-32-c++
autoconf automake db4-devel gdbm-devel ncurses-devel wget

Let us know how it goes.
-MC

On Thu, Aug 27, 2009 at 11:51 PM, Juan Backson <juanbackson at gmail.com>wrote:

> Hi,
> In a new environment, I am getting the following error when building the
> latest freeswitch from svn.  Does anyone know how to resolve it?
>
> gcc -I/usr/src/freeswitch-snapshot/src/include
> -I/usr/src/freeswitch-snapshot/libs/libteletone/src -fPIC -Werror
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
> -g -O2 -Wall -std=c99 -pedantic -o .libs/freeswitch freeswitch-switch.o  -lm
> ./.libs/libfreeswitch.so
> -L/usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib
> /usr/src/freeswitch-snapshot/libs/apr-util/xml/expat/lib/.libs/libexpat.a
> /usr/src/freeswitch-snapshot/libs/apr/.libs/libapr-1.a
> -L/usr/src/freeswitch-snapshot/libs/srtp -L/usr/kerberos/lib64
> libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt -lpthread
> libs/libedit/src/.libs/libedit.a -lssl -lcrypto -ldl -lz -lncurses
> -Wl,--rpath -Wl,/usr/local/freeswitch/lib
> ./.libs/libfreeswitch.so: undefined reference to `SQLGetData'
> ./.libs/libfreeswitch.so: undefined reference to `SQLDisconnect'
> ./.libs/libfreeswitch.so: undefined reference to `SQLSetConnectAttr'
> ./.libs/libfreeswitch.so: undefined reference to `SQLGetDiagRec'
> ./.libs/libfreeswitch.so: undefined reference to `SQLConnect'
> ./.libs/libfreeswitch.so: undefined reference to `SQLExecute'
> ./.libs/libfreeswitch.so: undefined reference to `SQLNumResultCols'
> ./.libs/libfreeswitch.so: undefined reference to `SQLFreeHandle'
> ./.libs/libfreeswitch.so: undefined reference to `SQLGetInfo'
> ./.libs/libfreeswitch.so: undefined reference to `SQLAllocHandle'
> ./.libs/libfreeswitch.so: undefined reference to `SQLDriverConnect'
> ./.libs/libfreeswitch.so: undefined reference to `SQLPrepare'
> ./.libs/libfreeswitch.so: undefined reference to `SQLError'
> ./.libs/libfreeswitch.so: undefined reference to `SQLRowCount'
> ./.libs/libfreeswitch.so: undefined reference to `SQLDescribeCol'
> ./.libs/libfreeswitch.so: undefined reference to `SQLFetch'
> ./.libs/libfreeswitch.so: undefined reference to `SQLSetEnvAttr'
> collect2: ld returned 1 exit status
> make[2]: *** [freeswitch] Error 1
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Fri Aug 28 09:07:06 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 09:07:06 -0700
Subject: [Freeswitch-users] How can it be done with freeswitch?
In-Reply-To: <25184845.post@talk.nabble.com>
References: <25184845.post@talk.nabble.com>
Message-ID: <87f2f3b90908280907w4812a8dbr7a97385f9afdd376@mail.gmail.com>

Nagalenoj,

This is definitely possible to do with FreeSWITCH. However, it is not
exactly simple. You have some learning to do. My recommendation is to learn
about event socket. Most likely you'll need to use outbound event socket.
(Read about event socket and ESL, the event socket library, on the wiki.)

I also recommend that you look at some of the example scripts in
src/libs/esl/perl/ which show some of the ways to handle calls with a
script. You've already done a good job of breaking down the steps, now
experiment with handling each of these steps with a script. We don't have
extensive Perl script examples which means we would love to see your code
when you have it all figured out! :)

-MC

On Thu, Aug 27, 2009 at 11:44 PM, Nagalenoj <nagalenoj at gmail.com> wrote:

>
> Dear friends,
>    I am using freeswitch for the last one month and trying to write perl
> scripts to do some tasks.
> I am trying a new thing, which is something that I don't how to do it and I
> don't whether it can be done with freeswitch.   my requirement is as
> follows,
>             A calls B[This extension has to execute a script].
>             The script has to do the following,
>              * Cut the call and callback to the number.
>              * Get a phone number[DTMF] from A.
>              * Dial the number(C) and bridge A and C.
>              * When C cuts the call, the control has to comeback to script.
>              * Ask another number to A to dial out, get another phone
> number[DTMF] from A.
>              * Dial the number(D) and bridge A and D.
>
> Please, help me to do this. Is there any other documentation to read about
> the perl freeswitch functions.
> --
> View this message in context:
> http://www.nabble.com/How-can-it-be-done-with-freeswitch--tp25184845p25184845.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Fri Aug 28 09:10:22 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 09:10:22 -0700
Subject: [Freeswitch-users] mod_fax not working
In-Reply-To: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com>
References: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com>
Message-ID: <87f2f3b90908280910i3f0566f3tcae8fd7e8cff5d9d@mail.gmail.com>

On Fri, Aug 28, 2009 at 2:15 AM, Mathieu Parent <math.parent at gmail.com>wrote:

> Hi,
>
> I just started testing Freeswitch, this works well. This is powerfull
> and easy to administrate.
>
> I just have one problem and one request. The request is BOUNTY-5: IMAP
> integration of voicemail (I will try to propose a patch if this is
> within my competencies)
>
> The problem is with mod_fax not receiving any fax.
>
> Relevant extension is:
>    <extension name="fax2mail">
>        <condition field="destination_number" expression="^(9998)$">
>            <action application="set" data="fax_verbose=true"/>
>            <action application="set" data="fax_disable_v17=false"/>
>            <action application="set" data="fax_use_ecm=true"/>
>            <action application="answer" />
>            <action application="playback" data="silence_stream://2000"/>
>            <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/>
>            <action application="hangup"/>
>        </condition>
>    </extension>
>
> Test case is a 1-page fax coming from a Minolta printer connected to a
> Cisco VG248 (via RJ11). The VG248 port is registred on a Cisco Call
> Manager communicating with FS via a SIP trunk.
>
> Log is attached:
> - first try from 11:04:05 to 11:04:31
> - 2nd try from 11:04:57 to 11:05:22
> - 3rd try from 11:05:48 to 11:06:08
>
> Do you need more information?
>

Put your FS console in debug mode (press F6 or type "console loglevel 7")
and then make another test fax call, capturing the output. Look for obvious
clues. NOTE: The debug logs are extensive so don't fret if you feel like
you're buried in information. Pastebin your log output (
pastebin.freeswitch.org) and put the link in this email thread. The experts
can then help you figure it out from there...
-MC
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From msc at freeswitch.org  Fri Aug 28 09:11:05 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 09:11:05 -0700
Subject: [Freeswitch-users] mod_fax not working
In-Reply-To: <87f2f3b90908280910i3f0566f3tcae8fd7e8cff5d9d@mail.gmail.com>
References: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com>
	<87f2f3b90908280910i3f0566f3tcae8fd7e8cff5d9d@mail.gmail.com>
Message-ID: <87f2f3b90908280911k3d7ab400l7858f69e0bd70a4a@mail.gmail.com>

Doh! I just noticed you already attached the log. My bad. Let the experts
check it out.
-MC

On Fri, Aug 28, 2009 at 9:10 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Fri, Aug 28, 2009 at 2:15 AM, Mathieu Parent <math.parent at gmail.com>wrote:
>
>> Hi,
>>
>> I just started testing Freeswitch, this works well. This is powerfull
>> and easy to administrate.
>>
>> I just have one problem and one request. The request is BOUNTY-5: IMAP
>> integration of voicemail (I will try to propose a patch if this is
>> within my competencies)
>>
>> The problem is with mod_fax not receiving any fax.
>>
>> Relevant extension is:
>>    <extension name="fax2mail">
>>        <condition field="destination_number" expression="^(9998)$">
>>            <action application="set" data="fax_verbose=true"/>
>>            <action application="set" data="fax_disable_v17=false"/>
>>            <action application="set" data="fax_use_ecm=true"/>
>>            <action application="answer" />
>>            <action application="playback" data="silence_stream://2000"/>
>>            <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/>
>>            <action application="hangup"/>
>>        </condition>
>>    </extension>
>>
>> Test case is a 1-page fax coming from a Minolta printer connected to a
>> Cisco VG248 (via RJ11). The VG248 port is registred on a Cisco Call
>> Manager communicating with FS via a SIP trunk.
>>
>> Log is attached:
>> - first try from 11:04:05 to 11:04:31
>> - 2nd try from 11:04:57 to 11:05:22
>> - 3rd try from 11:05:48 to 11:06:08
>>
>> Do you need more information?
>>
>
> Put your FS console in debug mode (press F6 or type "console loglevel 7")
> and then make another test fax call, capturing the output. Look for obvious
> clues. NOTE: The debug logs are extensive so don't fret if you feel like
> you're buried in information. Pastebin your log output (
> pastebin.freeswitch.org) and put the link in this email thread. The
> experts can then help you figure it out from there...
> -MC
>
>
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From msc at freeswitch.org  Fri Aug 28 09:18:17 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 09:18:17 -0700
Subject: [Freeswitch-users] Questions about att_xfer
In-Reply-To: <1cd828b60908280413v6e5726cck72a5dcaa94ce3705@mail.gmail.com>
References: <1cd828b60908260751g105ef051xe9441d903f04a697@mail.gmail.com>
	<1cd828b60908261127i3fb17004s304826d4744a4cd6@mail.gmail.com>
	<191c3a030908261139i2ae81777veead56e4c1c4fa62@mail.gmail.com>
	<191c3a030908261259r6f45fe1cm2a40a9618714348c@mail.gmail.com>
	<87f2f3b90908261311k7f4a7e40k454f97094c0c180f@mail.gmail.com>
	<1cd828b60908270619t3a53b38dh689772fb2efa382d@mail.gmail.com>
	<191c3a030908270948w22045e76j8d7470c9a9349795@mail.gmail.com>
	<1cd828b60908271118pd1f66bdq92aaf75dc5186564@mail.gmail.com>
	<87f2f3b90908271130v4bb7deads45d5f1c190ea0661@mail.gmail.com>
	<1cd828b60908280413v6e5726cck72a5dcaa94ce3705@mail.gmail.com>
Message-ID: <87f2f3b90908280918w77653b85jd7d4823635e1b5a4@mail.gmail.com>

Thanks! I made very minor changes. We appreciate you helping out with
documenting new features.
-MC

On Fri, Aug 28, 2009 at 4:13 AM, Anatoliy Kounitskiy <
anatoliy at kounitskiy.com> wrote:

> The wiki page is updated (
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_att_xfer ). Hope
> it is understandable :)
>
> Anatoliy Kounitskiy
>
> On Thu, Aug 27, 2009 at 9:30 PM, Michael Collins<msc at freeswitch.org>
> wrote:
> > Thanks! I'll be in touch to help you if you need any wiki editing
> > assistance.
> > -MC
> >
> > On Thu, Aug 27, 2009 at 11:18 AM, Anatoliy Kounitskiy
> > <anatoliy at kounitskiy.com> wrote:
> >>
> >> Thank you all for the great job :)
> >> Now it works as I wanted!!
> >>
> >> Tomorrow I'll  will try to update the wiki :)
> >>
> >> But with few words ( I tried to strip the extensions to minimum for
> >> more easy read for users).
> >> In the dialplan I have created the extension showed below:
> >>
> >>  <extension name="local_number">
> >>    <condition field="destination_number" expression="^(\d{3})$">
> >>      <action application="set" data="dialed_extension=$1"/>
> >>      <action application="export" data="dialed_extension=$1"/>
> >>      <action application="bind_meta_app" data="1 b s
> >> execute_extension::attented_xfer XML features"/>
> >>      <action application="set" data="transfer_ringback=$${hold_music}"/>
> >>      <action application="set" data="call_timeout=10"/>
> >>      <action application="set" data="hangup_after_bridge=true"/>
> >>      <action application="bridge"
> >> data="user/${dialed_extension}@${domain_name}"/>
> >>    </condition>
> >>  </extension>
> >>
> >> In the features.xml file I have extension attented_xfer, that is using
> >> the att_xfer application :
> >>
> >>  <extension name="attented_xfer">
> >>      <condition field="destination_number" expression="^attented_xfer$">
> >>        <action application="set" data="continue_on_fail=true"/>
> >>        <action application="read" data="3 4 ivr/ivr-enter_ext.wav
> >> attxfer_callthis 30000 #"/>
> >>        <action application="set" data="origination_cancel_key=#"/>
> >>        <action application="att_xfer"
> >> data="user/${attxfer_callthis}@${domain_name}"/>
> >>      </condition>
> >>    </extension>
> >>
> >> Also I have set the the origination cancel key will be # - so if B
> >> wants to cancel the call to C and to go back to A - just uses the #
> >> key :)
> >>
> >> Again - thank you very much :) :)
> >>
> >> Anatoliy Kounitskiy
> >>
> >> On Thu, Aug 27, 2009 at 7:48 PM, Anthony
> >> Minessale<anthony.minessale at gmail.com> wrote:
> >> > please retest with latest SVN trunk
> >> >
> >> > On Thu, Aug 27, 2009 at 8:19 AM, Anatoliy Kounitskiy
> >> > <anatoliy at kounitskiy.com> wrote:
> >> >>
> >> >> Just for information - the idea :
> >> >> A---calls---> B ---att_xfer---> C    , B hangs C up and goes back to
> A
> >> >> ( I`m sorry C is not answering :) )
> >> >>
> >> >
> >> > --
> >> > Anthony Minessale II
> >> >
> >> > FreeSWITCH http://www.freeswitch.org/
> >> > ClueCon http://www.cluecon.com/
> >> > Twitter: http://twitter.com/FreeSWITCH_wire
> >> >
> >> > AIM: anthm
> >> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> > IRC: irc.freenode.net #freeswitch
> >> >
> >> > FreeSWITCH Developer Conference
> >> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >> > iax:guest at conference.freeswitch.org/888
> >> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> > pstn:213-799-1400
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >>
> >>
> >> --
> >> Anatoliy Kounitskiy
> >> -------------------------
> >> E-mail: anatoliy at kounitskiy.com
> >> Mobile: +359898913540
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
>
> --
> Anatoliy Kounitskiy
> -------------------------
> E-mail: anatoliy at kounitskiy.com
> Mobile: +359898913540
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From foxb at abv.bg  Fri Aug 28 09:42:28 2009
From: foxb at abv.bg (Hristo Benev)
Date: Fri, 28 Aug 2009 19:42:28 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>

 Hello Rod,

I did the change.

Here is extract of console:
-----------------------------------------------------------------------------------------
variable_continue_on_fail: [true]
variable_sip_h_X-ROUTE: [LOOKUP]
variable_export_vars: [sip_h_X-ROUTE]
variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20]
variable_sip_redirect_contact_0: [sip:France at PEER_01]
variable_sip_redirected_to: [sip:France at PEER_01]
variable_sip_redirect_contact_user_0: [France] <-------------------------------------------
variable_sip_redirect_contact_host_0: [PEER_01]
variable_sip_redirect_dialstring_0: [sofia/internal/sip:France at PEER_01]
variable_sip_redirect_dialstring: [sofia/internal/sip:France at PEER_01]
variable_proto_specific_hangup_cause: [sip:503]
variable_sip_hangup_phrase: [DNS Error]
variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE]
variable_ROUTE_GW: [France]
variable_AREA: [France]
variable_current_application: [info]
------------------------------------------------------------------------------------------
I have different value it is actually the description field as shown here:

--------------------------------
 /opt/kamailio/sbin/kamctl cr show
cr carrier names
+----+---------+
| id | carrier |
+----+---------+
|  1 | default |
+----+---------+
cr domain names
+----+---------+
| id | domain  |
+----+---------+
|  1 | default |
+----+---------+
cr routes
+----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
| id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description |
+----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
|  1 |       1 |      1 | 1000        |     0 |    0 |    1 |     0 | PEER_01      |                |                | France      |
+----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
-----------------------------------------

And here is what I have in kamailio:
-------------------------------------------------------------------------------------
####### Routing Logic ########


# main request routing logic

route{

        if (!mf_process_maxfwd_header("10")) {
                sl_send_reply("483","Too Many Hops");
                exit;
        }

        t_check_trans();

        if ($rU==NULL) {
                # request with no Username in RURI
                sl_send_reply("484","Address Incomplete");
                exit;
        }

        # LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP
        if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){
                if(!cr_route("default", "default", "$rU", "$rU", "call_id","$avp(s:route_desc)")){
                     #xlog("ROUTING FAILED: no route found for $rU");
                     sl_send_reply("604", "Unable to route this call");
                     exit;
                } else {
                     xlog("LOOKUP FOUND: $rd $avp(s:route_desc)");
                     avp_pushto("$ru/username", "$avp(s:route_desc)");
                     sl_send_reply("302", "$rd");
                     exit;
                }
        }
}
----------------------------------------------------------------------------------

Another question...
In that part of FreeSwitch dialplan.xml

-----------------------------------
  <context name="ROUTING">

    <extension name="PEER_01">
      <condition field="${sip_h_X-ROUTE}" expression="PEER_01">
        <action application="set" data="hangup_after_bridge=true"/>

-----------------------------------

X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the regex is true.


As for thanks - for sure by default they are also for the developers of both apps.
I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning so it will be nice to have something working to start with.



 >-------- ?????????? ????? --------
 >??:  rod 
 >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >??: freeswitch-users at lists.freeswitch.org
 >????????? ??: ?????, 2009, ?????? 28 09:54:00 EEST

 >Hello,
 >
 >the trace seems good.
 >If you check the answer from Kamailio, you'll see that Kamailio answers 
 >with "302 PEER_01".
 >
 >As Michael Collins stated before, you can get the variable containing 
 >"PEER_01", then this variable is stored in a custom variable.
 >
 >In your dialplan, may you please add:
 >
 >, just before the transfer line, eg:
 >
 > 
 >         
 >         
 >         
 >         
 >         
 >         
 >	 
 >         
 >      
 >
 >
 >Using application Info, you'll see on the console (or CDR) the list of 
 >variables used for this call.  You should see the content of 
 >"${sip_redirect_contact_user_0}" that should contain  the value 
 >"PEER_01". Please check this and let me know.
 >
 >For the new configuration file, no problem for sharing. But as I wrote 
 >on the wiki page, I worked on this setup cause LCR module was not 
 >available when I start working on FS, nor I'm a good programmer to write 
 >a server side HTTP script (used by xml_curl) that could scale to my needs.
 >
 >I enhanced a bit this configuration with support for fallback routing, 
 >almost realtime graph (every minutes using www.cacti.net and some 
 >functions in FS like limit_hash) of number of concurrent calls per AREA, 
 >PEER...
 >
 >Thanks for "the good tutorial", but don't forget the dev team who did 
 >this great product ;-)
 >
 >rod.
 >
 >
 >Hristo Benev a ?crit :
 >>  I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).
 >>
 >> If you need additional info I'll provide it.
 >>
 >> Here is trace:
 >>
 >> ngrep -d any -nn -i '1000' port 5062 -W byline
 >> interface: any
 >> filter: (ip or ip6) and ( port 5062 )
 >> match: 1000
 >> #
 >> U 10.10.10.10:5090 -> 127.0.0.1:5062
 >> INVITE sip:1000 at 127.0.0.1:5062 SIP/2.0.
 >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
 >> Max-Forwards: 69.
 >> From: "Extension 1001" ;tag=978g69jZaFpBD.
 >> To: .
 >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
 >> CSeq: 119574771 INVITE.
 >> Contact: .
 >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
 >> Supported: timer, precondition, path, replaces.
 >> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
 >> Content-Type: application/sdp.
 >> Content-Disposition: session.
 >> Content-Length: 429.
 >> X-ROUTE: LOOKUP.
 >> Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off.
 >> .
 >> v=0.
 >> o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
 >> s=FreeSWITCH.
 >> c=IN IP4 10.10.10.10.
 >> t=0 0.
 >> m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
 >> a=rtpmap:0 PCMU/8000.
 >> a=rtpmap:115 G7221/32000.
 >> a=fmtp:115 bitrate=48000.
 >> a=rtpmap:107 G7221/16000.
 >> a=fmtp:107 bitrate=32000.
 >> a=rtpmap:9 G722/8000.
 >> a=rtpmap:8 PCMA/8000.
 >> a=rtpmap:3 GSM/8000.
 >> a=rtpmap:101 telephone-event/8000.
 >> a=fmtp:101 0-16.
 >> a=rtpmap:13 CN/8000.
 >> a=ptime:20.
 >>
 >> #
 >> U 127.0.0.1:5062 -> 10.10.10.10:5090
 >> SIP/2.0 302 PEER_01.
 >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
 >> From: "Extension 1001" ;tag=978g69jZaFpBD.
 >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
 >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
 >> CSeq: 119574771 INVITE.
 >> Contact: sip:France at PEER_01.
 >> Server: Kamailio (1.5.2-notls (i386/linux)).
 >> Content-Length: 0.
 >> .
 >>
 >> #
 >> U 10.10.10.10:5090 -> 127.0.0.1:5062
 >> ACK sip:1000 at 127.0.0.1:5062 SIP/2.0.
 >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
 >> Max-Forwards: 69.
 >> From: "Extension 1001" ;tag=978g69jZaFpBD.
 >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
 >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
 >> CSeq: 119574771 ACK.
 >> Content-Length: 0.
 >> .
 >>
 >>
 >>
 >>
 >>
 >>  >-------- ?????????? ????? --------
 >>  >??:  Hristo Benev 
 >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >??: freeswitch-users at lists.freeswitch.org
 >>  >????????? ??: ?????????, 2009, ?????? 27 14:58:53 EEST
 >>
 >>  > 
 >>  >Bojnour,
 >>  >
 >>  >I'll send a trace ASAP.
 >>  >
 >>  >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
 >>  >
 >>  >I'm really interested to see the updates of the project.
 >>  >
 >>  >Thank you for the good tutorial.
 >>  >
 >>  >Hristo
 >>  >
 >>  >
 >>  > >-------- ?????????? ????? --------
 >>  > >??:  rod 
 >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  > >??: freeswitch-users at lists.freeswitch.org
 >>  > >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST
 >>  >
 >>  > >Hi Hristo,
 >>  > >
 >>  > >I'm the author of this setup and wiki page. I did a lot of modifications 
 >>  > >on this setup (alternative routing if failure essentially) but don't 
 >>  > >have too much time to update the wiki.
 >>  > >
 >>  > >May you please send me an ngrep trace when you call 1000:
 >>  > >
 >>  > >ngrep -d any -nn -i '1000' port 5060 -W byline 
 >>  > >
 >>  > >I will check what's happening.
 >>  > >Do you have an entry for 1000 in your mysql database ?
 >>  > >
 >>  > >regards,
 >>  > >rod
 >>  > >
 >>  > >Hristo Benev a ?crit :
 >>  > >>  
 >>  > >> It seems that the problem is on kamailio configuration.
 >>  > >> Will ask on their list.
 >>  > >>
 >>  > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
 >>  > >>
 >>  > >> I have it setup as friend in asterisk, but still ???
 >>  > >>
 >>  > >> Any ideas?
 >>  > >>
 >>  > >> Thanks,
 >>  > >>
 >>  > >>
 >>  > >>  >-------- ?????????? ????? --------
 >>  > >>  >??:  Hristo Benev 
 >>  > >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  > >>  >??: freeswitch-users at lists.freeswitch.org
 >>  > >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
 >>  > >>
 >>  > >>  > I think that the problem is here:
 >>  > >>  >-------------------------
 >>  > >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
 >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
 >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
 >>  > >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
 >>  > >>  >--------------------------
 >>  > >>  >
 >>  > >>  >Actually Regex FAIL
 >>  > >>  >
 >>  > >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
 >>  > >>  >Here is my  default.xml:
 >>  > >>  >----------------
 >>  > >>  >
 >>  > >>  >
 >>  > >>  >
 >>  > >>  >  
 >>  > >>  >
 >>  > >>  >     
 >>  > >>  >      
 >>  > >>  >         
 >>  > >>  >         
 >>  > >>  >         
 >>  > >>  >         
 >>  > >>  >         
 >>  > >>  >         
 >>  > >>  >         
 >>  > >>  >      
 >>  > >>  >    
 >>  > >>  >
 >>  > >>  >   
 >>  > >>  >
 >>  > >>  >  
 >>  > >>  >
 >>  > >>  >    
 >>  > >>  >      
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >        
 >>  > >>  >      
 >>  > >>  >    
 >>  > >>  >
 >>  > >>  >  
 >>  > >>  >
 >>  > >>  >
 >>  > >>  >
 >>  > >>  >--------------------------
 >>  > >>  >
 >>  > >>  >
 >>  > >>  >
 >>  > >>  > >-------- ?????????? ????? --------
 >>  > >>  > >??:  Brian West 
 >>  > >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  > >>  > >??: freeswitch-users at lists.freeswitch.org
 >>  > >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
 >>  > >>  >
 >>  > >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
 >>  > >>  >You have to process all 302's in the dialplan. 
 >>  > >>  >Set this on your sofia profile
 >>  > >>  >You can set these variables sip_redirect_profile,
 >>  > >>  >sip_redirect_context,
 >>  > >>  >sip_redirect_dialplan,
 >>  > >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
 >>  > >>  >sip_redirected_to,
 >>  > >>  >sip_redirect_contact_user_%d,
 >>  > >>  >sip_redirect_contact_host_%d,
 >>  > >>  >sip_redirect_contact_params_%d,
 >>  > >>  >sip_redirect_dialstring_%d,
 >>  > >>  >sip_redirect_dialstring,
 >>  > >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
 >>  > >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
 >>  > >>  > >
 >>  > >>  >
 >>  > >>  >_______________________________________________
 >>  > >>  >FreeSWITCH-users mailing list
 >>  > >>  >FreeSWITCH-users at lists.freeswitch.org
 >>  > >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  > >>  >http://www.freeswitch.org
 >>  > >>  >
 >>  > >>
 >>  > >> _______________________________________________
 >>  > >> FreeSWITCH-users mailing list
 >>  > >> FreeSWITCH-users at lists.freeswitch.org
 >>  > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  > >> http://www.freeswitch.org
 >>  > >>   
 >>  > >
 >>  > >_______________________________________________
 >>  > >FreeSWITCH-users mailing list
 >>  > >FreeSWITCH-users at lists.freeswitch.org
 >>  > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  > >http://www.freeswitch.org
 >>  > >
 >>  >
 >>  >_______________________________________________
 >>  >FreeSWITCH-users mailing list
 >>  >FreeSWITCH-users at lists.freeswitch.org
 >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >http://www.freeswitch.org
 >>  >
 >>
 >> _______________________________________________
 >> FreeSWITCH-users mailing list
 >> FreeSWITCH-users at lists.freeswitch.org
 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >> http://www.freeswitch.org
 >>   
 >
 >_______________________________________________
 >FreeSWITCH-users mailing list
 >FreeSWITCH-users at lists.freeswitch.org
 >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >http://www.freeswitch.org
 >



From steveu at coppice.org  Fri Aug 28 10:01:12 2009
From: steveu at coppice.org (Steve Underwood)
Date: Sat, 29 Aug 2009 01:01:12 +0800
Subject: [Freeswitch-users] mod_fax not working
In-Reply-To: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com>
References: <960738410908280215x1b53ebb3kb66cb14178fa44d7@mail.gmail.com>
Message-ID: <4A980D58.5020808@coppice.org>

Hi Mathieu,

On 08/28/2009 05:15 PM, Mathieu Parent wrote:
> Hi,
>
> I just started testing Freeswitch, this works well. This is powerfull
> and easy to administrate.
>
> I just have one problem and one request. The request is BOUNTY-5: IMAP
> integration of voicemail (I will try to propose a patch if this is
> within my competencies)
>
> The problem is with mod_fax not receiving any fax.
>
> Relevant extension is:
>      <extension name="fax2mail">
>          <condition field="destination_number" expression="^(9998)$">
>              <action application="set" data="fax_verbose=true"/>
>              <action application="set" data="fax_disable_v17=false"/>
>              <action application="set" data="fax_use_ecm=true"/>
>              <action application="answer" />
>              <action application="playback" data="silence_stream://2000"/>
>              <action application="rxfax" data="/tmp/${uuid}.rxfax.tiff"/>
>              <action application="hangup"/>
>          </condition>
>      </extension>
>
> Test case is a 1-page fax coming from a Minolta printer connected to a
> Cisco VG248 (via RJ11). The VG248 port is registred on a Cisco Call
> Manager communicating with FS via a SIP trunk.
>
> Log is attached:
> - first try from 11:04:05 to 11:04:31
> - 2nd try from 11:04:57 to 11:05:22
> - 3rd try from 11:05:48 to 11:06:08
>
> Do you need more information?
>
> Regards
>
> Mathieu Parent
>    
The log shows the same thing happening every time. A bad CRC from the 
far end, followed by a good DCS frame followed by what seems to be 
rubbish. I think I'd need an audio log from one of these calls to figure 
out any more.

Steve




From msc at freeswitch.org  Fri Aug 28 10:24:27 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 10:24:27 -0700
Subject: [Freeswitch-users] newbie alert. Help with dialplan
In-Reply-To: <f9c510070908271116n33c3f35di44eea1790f62bfc8@mail.gmail.com>
References: <f9c510070908271116n33c3f35di44eea1790f62bfc8@mail.gmail.com>
Message-ID: <87f2f3b90908281024i1ecc6238o3bcab80e9464797e@mail.gmail.com>

Julian,

It might be better if you gave us the run-down on what your current Asterisk
extension does. Most likely there is an elegant way of doing it in
FreeSWITCH. Part of the unlearing of the Asterisk way is coming to grips
with the fact that the FS dialplan is not a programming language in and of
itself. While it's convenient to have GotoIf's all over the place, in the
long run that's inefficient.

Let's start from the top-down instead of the bottom-up. Discussion your
application in general terms, i.e., non-Asterisk-specific language. Tell us
what it does, not how it does it, and then we can think it through using FS
concepts.

-MC

On Thu, Aug 27, 2009 at 11:16 AM, Julian Lyndon-Smith <asterisk at dotr.com>wrote:

> I am a long-time asterisk user (2005), so I come from an unfortunate
> position of having to unlearn an awful lot of stuff in order to make
> freeswitch do the things I want ;(
>
> I *think* i've got my head around using mod_curl_xml (?) to read all
> the config stuff from my webserver. I *think* I've got my head around
> setting up the sip clients etc etc ...
>
> However, where I am really struggling is the dialplan. For the life of
> me I simply cannot seem to grasp the fs way - that's no disrespect to
> the fs way, but perhaps the failure of this old brain to change !
>
>  have pastebinned an example of show 1234546 at inboundq at
> http://www.pastebin.ca/1544890. If possible, would someone be able to
> show me how to convert this dialplan to the fs way ? If I could be
> given a little foot-up I'm sure that I would be able to convert the
> rest of the dialplan !
>
> Thanks in advance, and please go easy !
>
> Julian
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Fri Aug 28 10:43:35 2009
From: brian at freeswitch.org (Brian West)
Date: Fri, 28 Aug 2009 12:43:35 -0500
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
In-Reply-To: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>
References: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>
Message-ID: <86BDA956-125F-423E-931F-7FFC0F679C59@freeswitch.org>

Now you have to actually catch this in your dialplan on FreeSWITCH and  
execute the bridge application to the dialstring provided

/b

On Aug 28, 2009, at 11:42 AM, Hristo Benev wrote:

> variable_sip_redirect_dialstring_0: [sofia/internal/ 
> sip:France at PEER_01]
> variable_sip_redirect_dialstring: [sofia/internal/sip:France at PEER_01]




From paveraware at hotmail.com  Fri Aug 28 11:14:57 2009
From: paveraware at hotmail.com (Christensen Tom)
Date: Fri, 28 Aug 2009 18:14:57 +0000
Subject: [Freeswitch-users] Hello, and stuff.
Message-ID: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>


As a background, I ran an asterisk consulting company for about 3 years that I gave up on 2 years ago after repeatedly failing to achieve any sort of stability on any sort install over about 30 phones, I gave up.

Maybe that was wrong, I am open to the possibility that I just didn't know enough and I was building things wrong, but I worked inside the asterisk code (which I feel is a hopeless mess), I implemented a few small custom features, anyway... 

I'm coming back into the VoIP space now, and I'm wondering what sort of issues can I expect in trying to pick up and learn freeswitch?  From what I've read on the website, it appears to have a much more sane architecture.  I've used Cisco, Broadsoft, and asterisk in the past.  By far the least stable and worst general call quality was asterisk.  I constantly contended with strange call quality issues in asterisk, lots of echo (even with hardware echo cancellation cards), lots of jitter, lots of call break up (even on small systems with 10-20 users, using QoS on the network, and in general doing everything I could to prioritize voice over anything else).

When I used Cisco call manager and broadsoft, the voice quality issues were basically non-existant, as long as the network was running QoS echo, stutter, calls breaking up, just didn't happen.  So, I guess my question is, does freeswitch show a marked improvement over asterisk in this department?  As long as you configure QoS and have hardware echo cancellation does it actually work reliably?

Thanks for any additional information about freeswitch you can provide as well.  I am a software developer primarily by trade, but I do lots of consulting type work in the SME space and I've had a couple projects thrown to me that require some integration with a phone system, and I just can't in good conscience recommend asterisk anymore.


-Tom
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From msc at freeswitch.org  Fri Aug 28 11:30:06 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 11:30:06 -0700
Subject: [Freeswitch-users] Group Call
In-Reply-To: <5ff3e3a70908251015ue88b8b9p7a5e091d15a6b5ae@mail.gmail.com>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
	<5ff3e3a70908251015ue88b8b9p7a5e091d15a6b5ae@mail.gmail.com>
Message-ID: <87f2f3b90908281130g1e33668fs44b817f9c622641c@mail.gmail.com>

On Tue, Aug 25, 2009 at 10:15 AM, Shawn L. Djernes <sdjernes at gmail.com>wrote:

> Woops,
> That line was an attempt to try something from an older diaplan example.
>  Copied and pasted from a file on my computer not the server.  The actual
> line is:
>
> <action application="bridge"
> data="{ignore_early_media=true}${group_call(sdjernes+A@${domain_name})}"/>
>
> The +A is to ring all phones in the group at the same time.
>
>
Can you confirm what you see on the CLI when you do this:
group_call sdjernes+A at ewr.djernes.net <sdjernes%2BA at ewr.djernes.net>

Let's make sure the group is properly defined, etc. before doing anything
else.
-MC
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From rs at runsolutions.com  Fri Aug 28 11:52:25 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Fri, 28 Aug 2009 20:52:25 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Message-ID: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>

Hello List,

I have read the current thread about scalability and I would need some  
advice about a callcenter setup:

First where I come from:
I have lot's of problems with an asterisk solution. I have regular  
crash's and lock-ups, with downgrading and other stuff i got it  
somewhat stable, but have nevertheless regular hickups. I am desperate  
and want to get rid of asterisk and I hope that freeSwitch will  
provide me with a more stable solution.


Our Setup (really nothing special):
* 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3  
GIG of RAM, Xircom analog switch connected to mobile stations for 4  
different providers, Digium 4port cards TP400<something>
* 8 queues
* ~60 agents (which logon, logoff, pause, unpause), not more than 40  
concurrently online
* ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
* Music on Hold in the call-queues
* No special announcement
* Transfers between calls in queues and different agents as well as  
non agents (i mention this because we have transfer related chrashes  
in asterisk)

The current Problems:
* Lockups with different causes (ranging from calls not terminated to  
heavy thread locking through the AMI interface)
* Crashes and library aborts (pthread, libc, crashes related to music  
on hold, app_queue, transfers)

We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are  
now back to 1.4.21.2 (stock debian) as anything beyond that is for  
whatever reason highly unstable for our szenario. Maybe we should have  
been segmenting the box into one asterisk dedicted to talking to the  
hardware, one especial for queue/sip handling, i do not know. (all  
issues are well documented in issues.asterisk.org, but it seems to be  
very, very difficult to get to the bottom of them as they exist since  
1.4.23 as it seems and are open until know and not fixable since month.)


Now, I really would appreciate some success-stories on how you guys  
managed to get a stable pbx system with freeSWITCH in regard of HA and  
scalability:

* How to segment freeSWITCH? Or is it stable enough to handle all in  
one for such a szenario as outlined above?
* What would be the best strategie for High Availability / Failover?
	-> I read in the WIKI (featurelist) that Livemigration of calls from  
one box to another should be possible?
	-> I was thinking about using memcached for storing all state  
information so another freeswitch box can takeover calls from the  
first box if it dies, is this possible? If so, how?
	-> Is there anotherway to somehow configure freeSWITCH that in the  
event of a crash i do not loose the current established calls?

Basically I just want a stable PBX where I do not have to fear every  
day it will core-dump or abort or Lock up. Is freeSWITCH mature enough  
so i can sleep at night for at least 3 month without a crash?



Thank you for your Time and help in advance, and I am more than  
willing to take all the information gathered here and create a wiki  
page to help other people with the same questions/problems.

best
Ray


-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares




From msc at freeswitch.org  Fri Aug 28 11:54:25 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 11:54:25 -0700
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
Message-ID: <87f2f3b90908281154s2a1496b4l2e3aa06545649633@mail.gmail.com>

Tom,
Welcome! Sadly, your experience is not unique...

On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom <paveraware at hotmail.com>wrote:

>  As a background, I ran an asterisk consulting company for about 3 years
> that I gave up on 2 years ago after repeatedly failing to achieve any sort
> of stability on any sort install over about 30 phones, I gave up.
>

The consensus I've seen is that the larger the install, the more likely one
is to have inexplicable issues.

>
>
> Maybe that was wrong, I am open to the possibility that I just didn't know
> enough and I was building things wrong, but I worked inside the asterisk
> code (which I feel is a hopeless mess), I implemented a few small custom
> features, anyway...
>

Any software that openly admits that a function is "pure nastiness" but
doesn't change it from version 1.0, 1.2, 1.4, or 1.6 has questionable
leadership IMHO. (grep the Asterisk source tree for "nastiness" and you'll
see it.)

>
>
> I'm coming back into the VoIP space now, and I'm wondering what sort of
> issues can I expect in trying to pick up and learn freeswitch?  From what
> I've read on the website, it appears to have a much more sane architecture.
> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
> stable and worst general call quality was asterisk.  I constantly contended
> with strange call quality issues in asterisk, lots of echo (even with
> hardware echo cancellation cards), lots of jitter, lots of call break up
> (even on small systems with 10-20 users, using QoS on the network, and in
> general doing everything I could to prioritize voice over anything else).
>

Again, your experience isn't unique...

>
> When I used Cisco call manager and broadsoft, the voice quality issues were
> basically non-existant, as long as the network was running QoS echo,
> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
> does freeswitch show a marked improvement over asterisk in this department?
> As long as you configure QoS and have hardware echo cancellation does it
> actually work reliably?
>

We receive lots of reports that FreeSWITCH is a vast improvement over not
only Asterisk but proprietary solutions as well. The FS architecture is, as
you mentioned, not insane. It is well thought out and therefore highly
flexible, extensible, and scalable. I'm not aware of anything - OSS or
proprietary - that can match FS in these three areas.

>
> Thanks for any additional information about freeswitch you can provide as
> well.  I am a software developer primarily by trade, but I do lots of
> consulting type work in the SME space and I've had a couple projects thrown
> to me that require some integration with a phone system, and I just can't in
> good conscience recommend asterisk anymore.
>

Are you comfortable with the lack of a super slick GUI? :) Some GUIs are in
development but the power users are quite happy with doing the emacs (or
vim) shuffle with the XML config files. Furthermore, the ways that FS allows
you to connect and control are fantastic: mod_xml_curl for dynamic
configurations, event-socket for external control (think of it like AMI not
sucking and being turbo-charged), mod_xml_rpc for RPC goodness... Anyway,
the list is impressive.

I can honestly say that every week we get new people looking at FreeSWITCH
and saying, "Wow, this is incredible." I can definitely, in good conscience,
recommend you investigate FS more deeply. I'm confident you'll be happy with
the return on your investment.

Hope it all works out for you! Join us in #freeswitch on irc.freenode.net if
you want to chat in real-time.
-Michael
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From gmaruzz at celliax.org  Fri Aug 28 11:55:40 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 20:55:40 +0200
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
Message-ID: <7b197bef0908281155w92eff9dp1ffcb379ae4c8438@mail.gmail.com>

Hi Tom,

best is to try it for yourself, you cannot expect from the FS mailing
list an answer like: "you know, fs is nor a marked improvement on
anything, we just like to spend time together" :-)

-gm


On Fri, Aug 28, 2009 at 8:14 PM, Christensen Tom<paveraware at hotmail.com> wrote:
> As a background, I ran an asterisk consulting company for about 3 years that
> I gave up on 2 years ago after repeatedly failing to achieve any sort of
> stability on any sort install over about 30 phones, I gave up.
>
> Maybe that was wrong, I am open to the possibility that I just didn't know
> enough and I was building things wrong, but I worked inside the asterisk
> code (which I feel is a hopeless mess), I implemented a few small custom
> features, anyway...
>
> I'm coming back into the VoIP space now, and I'm wondering what sort of
> issues can I expect in trying to pick up and learn freeswitch?? From what
> I've read on the website, it appears to have a much more sane architecture.
> I've used Cisco, Broadsoft, and asterisk in the past.? By far the least
> stable and worst general call quality was asterisk.? I constantly contended
> with strange call quality issues in asterisk, lots of echo (even with
> hardware echo cancellation cards), lots of jitter, lots of call break up
> (even on small systems with 10-20 users, using QoS on the network, and in
> general doing everything I could to prioritize voice over anything else).
>
> When I used Cisco call manager and broadsoft, the voice quality issues were
> basically non-existant, as long as the network was running QoS echo,
> stutter, calls breaking up, just didn't happen.? So, I guess my question is,
> does freeswitch show a marked improvement over asterisk in this department?
> As long as you configure QoS and have hardware echo cancellation does it
> actually work reliably?
>
> Thanks for any additional information about freeswitch you can provide as
> well.? I am a software developer primarily by trade, but I do lots of
> consulting type work in the SME space and I've had a couple projects thrown
> to me that require some integration with a phone system, and I just can't in
> good conscience recommend asterisk anymore.
>
>
> -Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Fri Aug 28 12:04:22 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 12:04:22 -0700
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <7b197bef0908281155w92eff9dp1ffcb379ae4c8438@mail.gmail.com>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
	<7b197bef0908281155w92eff9dp1ffcb379ae4c8438@mail.gmail.com>
Message-ID: <87f2f3b90908281204g47e9f3cw3b5fa9c2c96d0e5e@mail.gmail.com>

On Fri, Aug 28, 2009 at 11:55 AM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> Hi Tom,
>
> best is to try it for yourself, you cannot expect from the FS mailing
> list an answer like: "you know, fs is nor a marked improvement on
> anything, we just like to spend time together" :-)
>

Although, in reality, some of us do actually enjoy spending time together.
:P
-MC
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From gmaruzz at celliax.org  Fri Aug 28 12:09:05 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 21:09:05 +0200
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <87f2f3b90908281204g47e9f3cw3b5fa9c2c96d0e5e@mail.gmail.com>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
	<7b197bef0908281155w92eff9dp1ffcb379ae4c8438@mail.gmail.com> 
	<87f2f3b90908281204g47e9f3cw3b5fa9c2c96d0e5e@mail.gmail.com>
Message-ID: <7b197bef0908281209w63b05fefhb85926f2eecbbdee@mail.gmail.com>

On Fri, Aug 28, 2009 at 9:04 PM, Michael Collins<msc at freeswitch.org> wrote:
>
>
> On Fri, Aug 28, 2009 at 11:55 AM, Giovanni Maruzzelli <gmaruzz at celliax.org>
> wrote:
>>
>> Hi Tom,
>>
>> best is to try it for yourself, you cannot expect from the FS mailing
>> list an answer like: "you know, fs is nor a marked improvement on
>> anything, we just like to spend time together" :-)
>
> Although, in reality, some of us do actually enjoy spending time together.
> :P
> -MC

meh too! ;-)



From anthony.minessale at gmail.com  Fri Aug 28 12:11:04 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 14:11:04 -0500
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <87f2f3b90908281154s2a1496b4l2e3aa06545649633@mail.gmail.com>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
	<87f2f3b90908281154s2a1496b4l2e3aa06545649633@mail.gmail.com>
Message-ID: <191c3a030908281211x3af65360q236d7b6a6564c055@mail.gmail.com>

http://www.freeswitch.org/node/117

That's essentially the story of why I wrote FS.


On Fri, Aug 28, 2009 at 1:54 PM, Michael Collins <msc at freeswitch.org> wrote:

> Tom,
> Welcome! Sadly, your experience is not unique...
>
> On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom <paveraware at hotmail.com>wrote:
>
>>  As a background, I ran an asterisk consulting company for about 3 years
>> that I gave up on 2 years ago after repeatedly failing to achieve any sort
>> of stability on any sort install over about 30 phones, I gave up.
>>
>
> The consensus I've seen is that the larger the install, the more likely one
> is to have inexplicable issues.
>
>>
>>
>> Maybe that was wrong, I am open to the possibility that I just didn't know
>> enough and I was building things wrong, but I worked inside the asterisk
>> code (which I feel is a hopeless mess), I implemented a few small custom
>> features, anyway...
>>
>
> Any software that openly admits that a function is "pure nastiness" but
> doesn't change it from version 1.0, 1.2, 1.4, or 1.6 has questionable
> leadership IMHO. (grep the Asterisk source tree for "nastiness" and you'll
> see it.)
>
>>
>>
>> I'm coming back into the VoIP space now, and I'm wondering what sort of
>> issues can I expect in trying to pick up and learn freeswitch?  From what
>> I've read on the website, it appears to have a much more sane architecture.
>> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
>> stable and worst general call quality was asterisk.  I constantly contended
>> with strange call quality issues in asterisk, lots of echo (even with
>> hardware echo cancellation cards), lots of jitter, lots of call break up
>> (even on small systems with 10-20 users, using QoS on the network, and in
>> general doing everything I could to prioritize voice over anything else).
>>
>
> Again, your experience isn't unique...
>
>>
>> When I used Cisco call manager and broadsoft, the voice quality issues
>> were basically non-existant, as long as the network was running QoS echo,
>> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
>> does freeswitch show a marked improvement over asterisk in this department?
>> As long as you configure QoS and have hardware echo cancellation does it
>> actually work reliably?
>>
>
> We receive lots of reports that FreeSWITCH is a vast improvement over not
> only Asterisk but proprietary solutions as well. The FS architecture is, as
> you mentioned, not insane. It is well thought out and therefore highly
> flexible, extensible, and scalable. I'm not aware of anything - OSS or
> proprietary - that can match FS in these three areas.
>
>>
>> Thanks for any additional information about freeswitch you can provide as
>> well.  I am a software developer primarily by trade, but I do lots of
>> consulting type work in the SME space and I've had a couple projects thrown
>> to me that require some integration with a phone system, and I just can't in
>> good conscience recommend asterisk anymore.
>>
>
> Are you comfortable with the lack of a super slick GUI? :) Some GUIs are in
> development but the power users are quite happy with doing the emacs (or
> vim) shuffle with the XML config files. Furthermore, the ways that FS allows
> you to connect and control are fantastic: mod_xml_curl for dynamic
> configurations, event-socket for external control (think of it like AMI not
> sucking and being turbo-charged), mod_xml_rpc for RPC goodness... Anyway,
> the list is impressive.
>
> I can honestly say that every week we get new people looking at FreeSWITCH
> and saying, "Wow, this is incredible." I can definitely, in good conscience,
> recommend you investigate FS more deeply. I'm confident you'll be happy with
> the return on your investment.
>
> Hope it all works out for you! Join us in #freeswitch on irc.freenode.netif you want to chat in real-time.
> -Michael
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From gmaruzz at celliax.org  Fri Aug 28 12:11:38 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 21:11:38 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
Message-ID: <7b197bef0908281211u51c1ce43q34f321378816b40@mail.gmail.com>

Usually you don't need to worry about stability issues with FS.

For scalability, peoples tend to use openser or some other sip
loadbalancer in fron of fs, but you probably would not need that.

Live migration of calls is not yet possible, tough.

-giovanni



On Fri, Aug 28, 2009 at 8:52 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
> Hello List,
>
> I have read the current thread about scalability and I would need some
> advice about a callcenter setup:
>
> First where I come from:
> I have lot's of problems with an asterisk solution. I have regular
> crash's and lock-ups, with downgrading and other stuff i got it
> somewhat stable, but have nevertheless regular hickups. I am desperate
> and want to get rid of asterisk and I hope that freeSwitch will
> provide me with a more stable solution.
>
>
> Our Setup (really nothing special):
> * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3
> GIG of RAM, Xircom analog switch connected to mobile stations for 4
> different providers, Digium 4port cards TP400<something>
> * 8 queues
> * ~60 agents (which logon, logoff, pause, unpause), not more than 40
> concurrently online
> * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
> * Music on Hold in the call-queues
> * No special announcement
> * Transfers between calls in queues and different agents as well as
> non agents (i mention this because we have transfer related chrashes
> in asterisk)
>
> The current Problems:
> * Lockups with different causes (ranging from calls not terminated to
> heavy thread locking through the AMI interface)
> * Crashes and library aborts (pthread, libc, crashes related to music
> on hold, app_queue, transfers)
>
> We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are
> now back to 1.4.21.2 (stock debian) as anything beyond that is for
> whatever reason highly unstable for our szenario. Maybe we should have
> been segmenting the box into one asterisk dedicted to talking to the
> hardware, one especial for queue/sip handling, i do not know. (all
> issues are well documented in issues.asterisk.org, but it seems to be
> very, very difficult to get to the bottom of them as they exist since
> 1.4.23 as it seems and are open until know and not fixable since month.)
>
>
> Now, I really would appreciate some success-stories on how you guys
> managed to get a stable pbx system with freeSWITCH in regard of HA and
> scalability:
>
> * How to segment freeSWITCH? Or is it stable enough to handle all in
> one for such a szenario as outlined above?
> * What would be the best strategie for High Availability / Failover?
> ? ? ? ?-> I read in the WIKI (featurelist) that Livemigration of calls from
> one box to another should be possible?
> ? ? ? ?-> I was thinking about using memcached for storing all state
> information so another freeswitch box can takeover calls from the
> first box if it dies, is this possible? If so, how?
> ? ? ? ?-> Is there anotherway to somehow configure freeSWITCH that in the
> event of a crash i do not loose the current established calls?
>
> Basically I just want a stable PBX where I do not have to fear every
> day it will core-dump or abort or Lock up. Is freeSWITCH mature enough
> so i can sleep at night for at least 3 month without a crash?
>
>
>
> Thank you for your Time and help in advance, and I am more than
> willing to take all the information gathered here and create a wiki
> page to help other people with the same questions/problems.
>
> best
> Ray
>
>
> --
> Raimund Sacherer
> -
> RunSolutions
> ? ? Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 - ?Palma de Mallorca
> Baleares
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From tayeb.meftah at gmail.com  Fri Aug 28 13:29:38 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Fri, 28 Aug 2009 20:29:38 +0000
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
 robustness patch
In-Reply-To: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com>
Message-ID: <4A983E32.2050003@gmail.com>

hello gmaruzz,
thank you for your good works!
Skypiax work Wel now, except for a small bug, but i'm not sur if you see 
it  or no:
if i load the module manualy, for the first load crach and don't detect 
the active runing skype client
but if i do load mod_skypiax aguin will load and work perfectly
thanks
Giovanni Maruzzelli wrote:
> Hi FreeSWITCHers,
>
> all the users of mod_skypiax are kindly requested to test the svn trunk 14519.
>
> It contains a lot of changes meant to add stability and robustness,
> toward a production environment.
>
> Let me know how your feelings, and please add to the Jira any possible
> bug/issue/etc.
>
> Thanks to you all,
>
> -giovanni
>
>
>
> Sincerely,
>
> Giovanni Maruzzelli
>
> Cell : +39-347-2665618
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>   



__________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com





From testa at voicetechnology.com.br  Fri Aug 28 12:33:51 2009
From: testa at voicetechnology.com.br (Fernando Testa)
Date: Fri, 28 Aug 2009 16:33:51 -0300
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
Message-ID: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>

Hi folks,What linux distro would you suggest to deploy FS with django:
Centos 5.3 or Ubuntu LTS?

Thanks
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From anthony.minessale at gmail.com  Fri Aug 28 12:37:08 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 14:37:08 -0500
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
Message-ID: <191c3a030908281237l29570d6fr6e591402d4a22f95@mail.gmail.com>

please use CentOS 5


On Fri, Aug 28, 2009 at 2:33 PM, Fernando Testa <
testa at voicetechnology.com.br> wrote:

> Hi folks,What linux distro would you suggest to deploy FS with django:
> Centos 5.3 or Ubuntu LTS?
>
> Thanks
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From msc at freeswitch.org  Fri Aug 28 12:37:17 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 12:37:17 -0700
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <7b197bef0908281211u51c1ce43q34f321378816b40@mail.gmail.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<7b197bef0908281211u51c1ce43q34f321378816b40@mail.gmail.com>
Message-ID: <87f2f3b90908281237ye0571e0m9116da183092b154@mail.gmail.com>

On Fri, Aug 28, 2009 at 12:11 PM, Giovanni Maruzzelli
<gmaruzz at celliax.org>wrote:

> Usually you don't need to worry about stability issues with FS.
>
> For scalability, peoples tend to use openser or some other sip
> loadbalancer in fron of fs, but you probably would not need that.
>
> Live migration of calls is not yet possible, tough.
>

Brian West has done some testing with live migrations but I don't know where
he left off. Brian, were you using OpenVZ? I forget... In any case, FS
allows you to try to do this with the hope that it will actually work in a
production environment.

As for the other things - yes, FS can work with the TDM card and the queues,
etc. If you are in a position to install FS on a sandbox machine for testing
then that would be your best bet. I recommend diving in, which is probably
what you did when you first started learning Asterisk...

Have fun!
-MC
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From msc at freeswitch.org  Fri Aug 28 12:37:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 12:37:54 -0700
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
Message-ID: <87f2f3b90908281237k2fb599c2ue36d20b8807cf31d@mail.gmail.com>

On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa <
testa at voicetechnology.com.br> wrote:

> Hi folks,What linux distro would you suggest to deploy FS with django:
> Centos 5.3 or Ubuntu LTS?
>
CentOS 5.3 FTW!
-MC
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From gmaruzz at celliax.org  Fri Aug 28 12:41:05 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 21:41:05 +0200
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness 	patch
In-Reply-To: <4A983E32.2050003@gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com> 
	<4A983E32.2050003@gmail.com>
Message-ID: <7b197bef0908281241v4d7bb74an437d6bb177dcc655@mail.gmail.com>

On Fri, Aug 28, 2009 at 10:29 PM, Meftah Tayeb<tayeb.meftah at gmail.com> wrote:
> if i load the module manualy, for the first load crach and don't detect
> the active runing skype client
> but if i do load mod_skypiax aguin will load and work perfectly

Meftah,

I've not seen that bug, it works first time for me.

Anyway, I'll double check it, and I open A jira for you ;-)

-giovanni



> thanks
> Giovanni Maruzzelli wrote:
>> Hi FreeSWITCHers,
>>
>> all the users of mod_skypiax are kindly requested to test the svn trunk 14519.
>>
>> It contains a lot of changes meant to add stability and robustness,
>> toward a production environment.
>>
>> Let me know how your feelings, and please add to the Jira any possible
>> bug/issue/etc.
>>
>> Thanks to you all,
>>
>> -giovanni
>>
>>
>>
>> Sincerely,
>>
>> Giovanni Maruzzelli
>>
>> Cell : +39-347-2665618
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>>
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From gmaruzz at celliax.org  Fri Aug 28 12:44:32 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 21:44:32 +0200
Subject: [Freeswitch-users] Skypiax, Skype endpoint and trunk,
	robustness 	patch
In-Reply-To: <7b197bef0908281241v4d7bb74an437d6bb177dcc655@mail.gmail.com>
References: <7b197bef0908141043i619e9032n1fda698d6d9bfc77@mail.gmail.com> 
	<4A983E32.2050003@gmail.com>
	<7b197bef0908281241v4d7bb74an437d6bb177dcc655@mail.gmail.com>
Message-ID: <7b197bef0908281244s4a6bff51m4f4634715f136c6c@mail.gmail.com>

http://jira.freeswitch.org/browse/MODSKYPIAX-49


On Fri, Aug 28, 2009 at 9:41 PM, Giovanni Maruzzelli<gmaruzz at celliax.org> wrote:
> On Fri, Aug 28, 2009 at 10:29 PM, Meftah Tayeb<tayeb.meftah at gmail.com> wrote:
>> if i load the module manualy, for the first load crach and don't detect
>> the active runing skype client
>> but if i do load mod_skypiax aguin will load and work perfectly
>
> Meftah,
>
> I've not seen that bug, it works first time for me.
>
> Anyway, I'll double check it, and I open A jira for you ;-)
>
> -giovanni
>
>
>
>> thanks
>> Giovanni Maruzzelli wrote:
>>> Hi FreeSWITCHers,
>>>
>>> all the users of mod_skypiax are kindly requested to test the svn trunk 14519.
>>>
>>> It contains a lot of changes meant to add stability and robustness,
>>> toward a production environment.
>>>
>>> Let me know how your feelings, and please add to the Jira any possible
>>> bug/issue/etc.
>>>
>>> Thanks to you all,
>>>
>>> -giovanni
>>>
>>>
>>>
>>> Sincerely,
>>>
>>> Giovanni Maruzzelli
>>>
>>> Cell : +39-347-2665618
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________
>>>
>>> The message was checked by ESET NOD32 Antivirus.
>>>
>>> http://www.eset.com
>>>
>>>
>>>
>>
>>
>>
>> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________
>>
>> The message was checked by ESET NOD32 Antivirus.
>>
>> http://www.eset.com
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>



From aep.lists at it46.se  Fri Aug 28 12:50:00 2009
From: aep.lists at it46.se (Alberto Escudero)
Date: Fri, 28 Aug 2009 21:50:00 +0200
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <87f2f3b90908281237k2fb599c2ue36d20b8807cf31d@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
	<87f2f3b90908281237k2fb599c2ue36d20b8807cf31d@mail.gmail.com>
Message-ID: <d36c422fa99228a711fb7eba47c7e94f.squirrel@correo.nodo50.org>

Debian :)
-- 
Stopping junk mailers is good for the environment

> On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa <
> testa at voicetechnology.com.br> wrote:
>
>> Hi folks,What linux distro would you suggest to deploy FS with django:
>> Centos 5.3 or Ubuntu LTS?
>>
> CentOS 5.3 FTW!
> -MC
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>





From paveraware at hotmail.com  Fri Aug 28 12:54:43 2009
From: paveraware at hotmail.com (Christensen Tom)
Date: Fri, 28 Aug 2009 19:54:43 +0000
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <87f2f3b90908281154s2a1496b4l2e3aa06545649633@mail.gmail.com>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
	<87f2f3b90908281154s2a1496b4l2e3aa06545649633@mail.gmail.com>
Message-ID: <BAY120-W71D36BC5F54F0DBEAE097DFF50@phx.gbl>


I am totally fine without a slick GUI interface.  The first 2 years of asterisk stuff I did was all in on the CLI in <editor of your choice> (I use vim most of the time, but not for religious reasons...).
Anyway, thanks for the info, I'll be setting up a freeswitch system this weekend expect to see me on IRC and here..
Thanks!
-Tom
Date: Fri, 28 Aug 2009 11:54:25 -0700
From: msc at freeswitch.org
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Hello, and stuff.

Tom,
Welcome! Sadly, your experience is not unique...

On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom <paveraware at hotmail.com> wrote:







As a background, I ran an asterisk consulting company for about 3 years that I gave up on 2 years ago after repeatedly failing to achieve any sort of stability on any sort install over about 30 phones, I gave up.


The consensus I've seen is that the larger the install, the more likely one is to have inexplicable issues. 




Maybe that was wrong, I am open to the possibility that I just didn't know enough and I was building things wrong, but I worked inside the asterisk code (which I feel is a hopeless mess), I implemented a few small custom features, anyway... 


Any software that openly admits that a function is "pure nastiness" but doesn't change it from version 1.0, 1.2, 1.4, or 1.6 has questionable leadership IMHO. (grep the Asterisk source tree for "nastiness" and you'll see it.) 




I'm coming back into the VoIP space now, and I'm wondering what sort of issues can I expect in trying to pick up and learn freeswitch?  From what I've read on the website, it appears to have a much more sane architecture.  I've used Cisco, Broadsoft, and asterisk in the past.  By far the least stable and worst general call quality was asterisk.  I constantly contended with strange call quality issues in asterisk, lots of echo (even with hardware echo cancellation cards), lots of jitter, lots of call break up (even on small systems with 10-20 users, using QoS on the network, and in general doing everything I could to prioritize voice over anything else).



Again, your experience isn't unique... 

When I used Cisco call manager and broadsoft, the voice quality issues were basically non-existant, as long as the network was running QoS echo, stutter, calls breaking up, just didn't happen.  So, I guess my question is, does freeswitch show a marked improvement over asterisk in this department?  As long as you configure QoS and have hardware echo cancellation does it actually work reliably?



We receive lots of reports that FreeSWITCH is a vast improvement over not only Asterisk but proprietary solutions as well. The FS architecture is, as you mentioned, not insane. It is well thought out and therefore highly flexible, extensible, and scalable. I'm not aware of anything - OSS or proprietary - that can match FS in these three areas. 



Thanks for any additional information about freeswitch you can provide as well.  I am a software developer primarily by trade, but I do lots of consulting type work in the SME space and I've had a couple projects thrown to me that require some integration with a phone system, and I just can't in good conscience recommend asterisk anymore.



Are you comfortable with the lack of a super slick GUI? :) Some GUIs are in development but the power users are quite happy with doing the emacs (or vim) shuffle with the XML config files. Furthermore, the ways that FS allows you to connect and control are fantastic: mod_xml_curl for dynamic configurations, event-socket for external control (think of it like AMI not sucking and being turbo-charged), mod_xml_rpc for RPC goodness... Anyway, the list is impressive.



I can honestly say that every week we get new people looking at FreeSWITCH and saying, "Wow, this is incredible." I can definitely, in good conscience, recommend you investigate FS more deeply. I'm confident you'll be happy with the return on your investment.



Hope it all works out for you! Join us in #freeswitch on irc.freenode.net if you want to chat in real-time.
-Michael


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From gmaruzz at celliax.org  Fri Aug 28 12:58:42 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 21:58:42 +0200
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <BAY120-W71D36BC5F54F0DBEAE097DFF50@phx.gbl>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
	<87f2f3b90908281154s2a1496b4l2e3aa06545649633@mail.gmail.com> 
	<BAY120-W71D36BC5F54F0DBEAE097DFF50@phx.gbl>
Message-ID: <7b197bef0908281258i2b71b724hd0bb19ca6b69bdac@mail.gmail.com>

Welcome on board Tom! And sorry for being witty before ;-)
-gm


On Fri, Aug 28, 2009 at 9:54 PM, Christensen Tom<paveraware at hotmail.com> wrote:
> I am totally fine without a slick GUI interface. ?The first 2 years of
> asterisk stuff I did was all in on the CLI in <editor of your choice> (I use
> vim most of the time, but not for religious reasons...).
> Anyway, thanks for the info, I'll be setting up a freeswitch system this
> weekend expect to see me on IRC and here..
> Thanks!
> -Tom
> ________________________________
> Date: Fri, 28 Aug 2009 11:54:25 -0700
> From: msc at freeswitch.org
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Hello, and stuff.
>
> Tom,
> Welcome! Sadly, your experience is not unique...
>
> On Fri, Aug 28, 2009 at 11:14 AM, Christensen Tom <paveraware at hotmail.com>
> wrote:
>
> As a background, I ran an asterisk consulting company for about 3 years that
> I gave up on 2 years ago after repeatedly failing to achieve any sort of
> stability on any sort install over about 30 phones, I gave up.
>
> The consensus I've seen is that the larger the install, the more likely one
> is to have inexplicable issues.
>
>
> Maybe that was wrong, I am open to the possibility that I just didn't know
> enough and I was building things wrong, but I worked inside the asterisk
> code (which I feel is a hopeless mess), I implemented a few small custom
> features, anyway...
>
> Any software that openly admits that a function is "pure nastiness" but
> doesn't change it from version 1.0, 1.2, 1.4, or 1.6 has questionable
> leadership IMHO. (grep the Asterisk source tree for "nastiness" and you'll
> see it.)
>
>
> I'm coming back into the VoIP space now, and I'm wondering what sort of
> issues can I expect in trying to pick up and learn freeswitch?? From what
> I've read on the website, it appears to have a much more sane architecture.
> I've used Cisco, Broadsoft, and asterisk in the past.? By far the least
> stable and worst general call quality was asterisk.? I constantly contended
> with strange call quality issues in asterisk, lots of echo (even with
> hardware echo cancellation cards), lots of jitter, lots of call break up
> (even on small systems with 10-20 users, using QoS on the network, and in
> general doing everything I could to prioritize voice over anything else).
>
> Again, your experience isn't unique...
>
> When I used Cisco call manager and broadsoft, the voice quality issues were
> basically non-existant, as long as the network was running QoS echo,
> stutter, calls breaking up, just didn't happen.? So, I guess my question is,
> does freeswitch show a marked improvement over asterisk in this department?
> As long as you configure QoS and have hardware echo cancellation does it
> actually work reliably?
>
> We receive lots of reports that FreeSWITCH is a vast improvement over not
> only Asterisk but proprietary solutions as well. The FS architecture is, as
> you mentioned, not insane. It is well thought out and therefore highly
> flexible, extensible, and scalable. I'm not aware of anything - OSS or
> proprietary - that can match FS in these three areas.
>
> Thanks for any additional information about freeswitch you can provide as
> well.? I am a software developer primarily by trade, but I do lots of
> consulting type work in the SME space and I've had a couple projects thrown
> to me that require some integration with a phone system, and I just can't in
> good conscience recommend asterisk anymore.
>
> Are you comfortable with the lack of a super slick GUI? :) Some GUIs are in
> development but the power users are quite happy with doing the emacs (or
> vim) shuffle with the XML config files. Furthermore, the ways that FS allows
> you to connect and control are fantastic: mod_xml_curl for dynamic
> configurations, event-socket for external control (think of it like AMI not
> sucking and being turbo-charged), mod_xml_rpc for RPC goodness... Anyway,
> the list is impressive.
>
> I can honestly say that every week we get new people looking at FreeSWITCH
> and saying, "Wow, this is incredible." I can definitely, in good conscience,
> recommend you investigate FS more deeply. I'm confident you'll be happy with
> the return on your investment.
>
> Hope it all works out for you! Join us in #freeswitch on irc.freenode.net if
> you want to chat in real-time.
> -Michael
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From michal.bielicki at halo2.pl  Fri Aug 28 13:00:44 2009
From: michal.bielicki at halo2.pl (Michal Bielicki)
Date: Fri, 28 Aug 2009 22:00:44 +0200
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <d36c422fa99228a711fb7eba47c7e94f.squirrel@correo.nodo50.org>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
	<87f2f3b90908281237k2fb599c2ue36d20b8807cf31d@mail.gmail.com>
	<d36c422fa99228a711fb7eba47c7e94f.squirrel@correo.nodo50.org>
Message-ID: <CFCEE02A-A1EE-41E3-AC3D-C57F35990111@halo2.pl>

OpenSolaris :P

Am 28.08.2009 um 21:50 schrieb Alberto Escudero:

> Debian :)
> --  
> Stopping junk mailers is good for the environment
>
>> On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa <
>> testa at voicetechnology.com.br> wrote:
>>
>>> Hi folks,What linux distro would you suggest to deploy FS with  
>>> django:
>>> Centos 5.3 or Ubuntu LTS?
>>>
>> CentOS 5.3 FTW!
>> -MC
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

Michal Bielicki
Leiter der Niederlassung
HaloKwadrat Sp. z o.o.
Niederlassung Kleinmachnow
Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
Ust.Id.: DE261885536
Geschaeftsfuehrer: Aleksander Wiercinski
Meiereifeld 2b, 14532 Kleinmachnow
t. +49 33203 263220
f. +49 33203 263229 sip. info at halokwadrat.de
e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
Hauptgesch?ftsstelle:
Halo Kwadrat Sp. z o.o.
ul. Polna 46/14
00-644 Warszawa, Polen
EIngetragen im HRB Warszawa, KRS 0000153539

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From asterisk at dotr.com  Fri Aug 28 13:02:29 2009
From: asterisk at dotr.com (Julian Lyndon-Smith)
Date: Fri, 28 Aug 2009 21:02:29 +0100
Subject: [Freeswitch-users] newbie alert. Help with dialplan
In-Reply-To: <87f2f3b90908281024i1ecc6238o3bcab80e9464797e@mail.gmail.com>
References: <f9c510070908271116n33c3f35di44eea1790f62bfc8@mail.gmail.com>
	<87f2f3b90908281024i1ecc6238o3bcab80e9464797e@mail.gmail.com>
Message-ID: <f9c510070908281302h6870a515h8294b04fb5ea1607@mail.gmail.com>

Thanks Michael. What the extension currently does is :

A) Add 1 to the current list of Queue calls (we have a blended system
where I need to dynamically allocate agents from outbound to inbound)
This counter is maintained in the phone system
B) call my application via a web call (CURL) to tell it that a new
Queue call has come in. This web call returns various bits of
information regarding the queue (message to play to the caller,
message to play to the agent, is the queue open etc)
C) if this fails (bad queue) then drop out
D) start recording, play the welcome message to the caller
E) Ask them if they want to participate in an automated questionnaire
at the end of the call.
F) If they agree, play a thank you message
G) If the call is out of hours, go to voicemail
H) Send a message to the application that a call is about to enter the queue
I) Join the queue
J) if the call is dropped out of the queue because no-one answered,
record that status
K) Stop recording

If they have answered yes to the questionnaire, then redirect the call
to an IVR when the agent hangs up

When the agent hangs up, or when the caller hangs up, then a jabber
message is sent to the agent, and a web service is called to indicate
the end of the call


2009/8/28 Michael Collins <msc at freeswitch.org>:
> Julian,
>
> It might be better if you gave us the run-down on what your current Asterisk
> extension does. Most likely there is an elegant way of doing it in
> FreeSWITCH. Part of the unlearing of the Asterisk way is coming to grips
> with the fact that the FS dialplan is not a programming language in and of
> itself. While it's convenient to have GotoIf's all over the place, in the
> long run that's inefficient.
>
> Let's start from the top-down instead of the bottom-up. Discussion your
> application in general terms, i.e., non-Asterisk-specific language. Tell us
> what it does, not how it does it, and then we can think it through using FS
> concepts.
>
> -MC
>
> On Thu, Aug 27, 2009 at 11:16 AM, Julian Lyndon-Smith <asterisk at dotr.com>
> wrote:
>>
>> I am a long-time asterisk user (2005), so I come from an unfortunate
>> position of having to unlearn an awful lot of stuff in order to make
>> freeswitch do the things I want ;(
>>
>> I *think* i've got my head around using mod_curl_xml (?) to read all
>> the config stuff from my webserver. I *think* I've got my head around
>> setting up the sip clients etc etc ...
>>
>> However, where I am really struggling is the dialplan. For the life of
>> me I simply cannot seem to grasp the fs way - that's no disrespect to
>> the fs way, but perhaps the failure of this old brain to change !
>>
>> ?have pastebinned an example of show 1234546 at inboundq at
>> http://www.pastebin.ca/1544890. If possible, would someone be able to
>> show me how to convert this dialplan to the fs way ? If I could be
>> given a little foot-up I'm sure that I would be able to convert the
>> rest of the dialplan !
>>
>> Thanks in advance, and please go easy !
>>
>> Julian
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From gmaruzz at celliax.org  Fri Aug 28 13:02:15 2009
From: gmaruzz at celliax.org (Giovanni Maruzzelli)
Date: Fri, 28 Aug 2009 22:02:15 +0200
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <CFCEE02A-A1EE-41E3-AC3D-C57F35990111@halo2.pl>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com> 
	<87f2f3b90908281237k2fb599c2ue36d20b8807cf31d@mail.gmail.com> 
	<d36c422fa99228a711fb7eba47c7e94f.squirrel@correo.nodo50.org> 
	<CFCEE02A-A1EE-41E3-AC3D-C57F35990111@halo2.pl>
Message-ID: <7b197bef0908281302n43fa2c8aoc6d713f2681fedd5@mail.gmail.com>

Vista!


On Fri, Aug 28, 2009 at 10:00 PM, Michal
Bielicki<michal.bielicki at halo2.pl> wrote:
> OpenSolaris :P
>
> Am 28.08.2009 um 21:50 schrieb Alberto Escudero:
>
>> Debian :)
>> -- Stopping junk mailers is good for the environment
>>
>>> On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa <
>>> testa at voicetechnology.com.br> wrote:
>>>
>>>> Hi folks,What linux distro would you suggest to deploy FS with django:
>>>> Centos 5.3 or Ubuntu LTS?
>>>>
>>> CentOS 5.3 FTW!
>>> -MC
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> Michal Bielicki
> Leiter der Niederlassung
> HaloKwadrat Sp. z o.o.
> Niederlassung Kleinmachnow
> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
> Ust.Id.: DE261885536
> Geschaeftsfuehrer: Aleksander Wiercinski
> Meiereifeld 2b, 14532 Kleinmachnow
> t. +49 33203 263220
> f. +49 33203 263229 sip. info at halokwadrat.de
> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
> Hauptgesch?ftsstelle:
> Halo Kwadrat Sp. z o.o.
> ul. Polna 46/14
> 00-644 Warszawa, Polen
> EIngetragen im HRB Warszawa, KRS 0000153539
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From edpimentl at gmail.com  Fri Aug 28 13:07:00 2009
From: edpimentl at gmail.com (EdPimentl)
Date: Fri, 28 Aug 2009 16:07:00 -0400
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
Message-ID: <9dc4a1670908281307o30723ddr210b5c6efc87230c@mail.gmail.com>

Centos 5.3 / Solaris ..... of course
And I am looking at http://moblin.org/  for tiny embedded system.

Best regards,
-E
Gpro.ws
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From lfurrea at gmail.com  Fri Aug 28 13:09:37 2009
From: lfurrea at gmail.com (Luis F Urrea)
Date: Fri, 28 Aug 2009 14:09:37 -0600
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <9dc4a1670908281307o30723ddr210b5c6efc87230c@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
	<9dc4a1670908281307o30723ddr210b5c6efc87230c@mail.gmail.com>
Message-ID: <f120f4c80908281309o28c3eca3h5e4f7efc821e1adf@mail.gmail.com>

Gladly using FreeBSD 64 bit for a >35 ext PBX

On Fri, Aug 28, 2009 at 2:07 PM, EdPimentl <edpimentl at gmail.com> wrote:

> Centos 5.3 / Solaris ..... of course
> And I am looking at http://moblin.org/  for tiny embedded system.
>
> Best regards,
> -E
> Gpro.ws
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Aug 28 13:09:06 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 15:09:06 -0500
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <9dc4a1670908281307o30723ddr210b5c6efc87230c@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
	<9dc4a1670908281307o30723ddr210b5c6efc87230c@mail.gmail.com>
Message-ID: <191c3a030908281309ma63a591tbc77294f98424a77@mail.gmail.com>

no seriously, use CentOS
None of these other guys have to support FS ;)

If you plan on doing it all without asking for help, then you are welcome to
any OS that FS compiles on. =D



On Fri, Aug 28, 2009 at 3:07 PM, EdPimentl <edpimentl at gmail.com> wrote:

> Centos 5.3 / Solaris ..... of course
> And I am looking at http://moblin.org/  for tiny embedded system.
>
> Best regards,
> -E
> Gpro.ws
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From pjintheusa at gmail.com  Fri Aug 28 13:13:12 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 28 Aug 2009 16:13:12 -0400
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
	<191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
Message-ID: <367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>

thanks for the reply.

Tried this and exactly the same result. The prompt is slowed - as if
the person is drunk.

I do notice that the originating call SDP shows:

a=rtpmap:0 PCMU/8000

but the play tries to use:

EXECUTE sofia/external/6093693828
playback(prompts/press-1-to-accept-call-from.wav)
2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
Activated L16 at 8000hz 1 channels 20ms


Should the codec be PCMU at 8000hz as opposed to L16 at 8000hz? If so of
there a way of forcing this codec for play.

Thanks


Phillip


Full code & logs
http://pastebin.freeswitch.org/10147
http://pastebin.freeswitch.org/10150



On Thu, Aug 27, 2009 at 4:30 PM, Anthony
Minessale<anthony.minessale at gmail.com> wrote:
> try wedging {ignore_early_media=true} before the first [ in your dial string
> and eliminate the code waiting for answer.
>
>
> On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <pjintheusa at gmail.com> wrote:
>>
>> Hi there,
>>
>> I know there are other ways of doing this. I am just trying get to
>> know have fun with the FreeSWITCH API.
>>
>> I am using originate and park a call:
>>
>> fsApi.Execute("originate",
>>
>> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
>>
>> That works great, the phone rings. I want to play something to this
>> called party when they pick up.
>>
>> ?while (!blegSession.answered())
>> ?{
>> ? ? ? ?Log.WriteLine(LogLevel.Alert,
>> "Inside::CallReturns:!Session.answered::Loop");
>> ? ? ? ?blegSession.sleep(500, 1);
>> ?}
>>
>> string promptFile = "prompts/whisper.wav";
>> blegSession.StreamFile(promptFile, 0);
>>
>>
>> This works - but the audio is choppy and slow.
>>
>> Is there something I need to do to that parked call before streaming that
>> file?
>>
>>
>> Thanks for any input.
>>
>> Phil
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From anthony.minessale at gmail.com  Fri Aug 28 13:33:47 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 15:33:47 -0500
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
	<191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
	<367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>
Message-ID: <191c3a030908281333w39f001bcx7b9914f7bdda4f8f@mail.gmail.com>

the L16 codec is used because the wav files is raw PCM audio
which is being encoded to PCMU

Have you tried doing this to a locally registered phone?



On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> thanks for the reply.
>
> Tried this and exactly the same result. The prompt is slowed - as if
> the person is drunk.
>
> I do notice that the originating call SDP shows:
>
> a=rtpmap:0 PCMU/8000
>
> but the play tries to use:
>
> EXECUTE sofia/external/6093693828
> playback(prompts/press-1-to-accept-call-from.wav)
> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
> Activated L16 at 8000hz 1 channels 20ms
>
>
> Should the codec be PCMU at 8000hz as opposed to L16 at 8000hz? If so of
> there a way of forcing this codec for play.
>
> Thanks
>
>
> Phillip
>
>
> Full code & logs
> http://pastebin.freeswitch.org/10147
> http://pastebin.freeswitch.org/10150
>
>
>
> On Thu, Aug 27, 2009 at 4:30 PM, Anthony
> Minessale<anthony.minessale at gmail.com> wrote:
> > try wedging {ignore_early_media=true} before the first [ in your dial
> string
> > and eliminate the code waiting for answer.
> >
> >
> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <pjintheusa at gmail.com>
> wrote:
> >>
> >> Hi there,
> >>
> >> I know there are other ways of doing this. I am just trying get to
> >> know have fun with the FreeSWITCH API.
> >>
> >> I am using originate and park a call:
> >>
> >> fsApi.Execute("originate",
> >>
> >>
> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
> >> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
> >>
> >> That works great, the phone rings. I want to play something to this
> >> called party when they pick up.
> >>
> >>  while (!blegSession.answered())
> >>  {
> >>        Log.WriteLine(LogLevel.Alert,
> >> "Inside::CallReturns:!Session.answered::Loop");
> >>        blegSession.sleep(500, 1);
> >>  }
> >>
> >> string promptFile = "prompts/whisper.wav";
> >> blegSession.StreamFile(promptFile, 0);
> >>
> >>
> >> This works - but the audio is choppy and slow.
> >>
> >> Is there something I need to do to that parked call before streaming
> that
> >> file?
> >>
> >>
> >> Thanks for any input.
> >>
> >> Phil
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From diego.viola at gmail.com  Fri Aug 28 13:34:41 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 28 Aug 2009 20:34:41 +0000
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
Message-ID: <86a32abc0908281334v4033e714jd695d19d6957eb28@mail.gmail.com>

Hi Christensen,

Welcome, you made a good choice on FreeSWITCH, and FS is much better at
those things than Asterisk.

Good luck, we are here to help you and tell us your experience later :).

Diego

On Fri, Aug 28, 2009 at 6:14 PM, Christensen Tom <paveraware at hotmail.com>wrote:

>  As a background, I ran an asterisk consulting company for about 3 years
> that I gave up on 2 years ago after repeatedly failing to achieve any sort
> of stability on any sort install over about 30 phones, I gave up.
>
> Maybe that was wrong, I am open to the possibility that I just didn't know
> enough and I was building things wrong, but I worked inside the asterisk
> code (which I feel is a hopeless mess), I implemented a few small custom
> features, anyway...
>
> I'm coming back into the VoIP space now, and I'm wondering what sort of
> issues can I expect in trying to pick up and learn freeswitch?  From what
> I've read on the website, it appears to have a much more sane architecture.
> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
> stable and worst general call quality was asterisk.  I constantly contended
> with strange call quality issues in asterisk, lots of echo (even with
> hardware echo cancellation cards), lots of jitter, lots of call break up
> (even on small systems with 10-20 users, using QoS on the network, and in
> general doing everything I could to prioritize voice over anything else).
>
> When I used Cisco call manager and broadsoft, the voice quality issues were
> basically non-existant, as long as the network was running QoS echo,
> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
> does freeswitch show a marked improvement over asterisk in this department?
> As long as you configure QoS and have hardware echo cancellation does it
> actually work reliably?
>
> Thanks for any additional information about freeswitch you can provide as
> well.  I am a software developer primarily by trade, but I do lots of
> consulting type work in the SME space and I've had a couple projects thrown
> to me that require some integration with a phone system, and I just can't in
> good conscience recommend asterisk anymore.
>
>
> -Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From anthony.minessale at gmail.com  Fri Aug 28 13:40:40 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 15:40:40 -0500
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <191c3a030908281333w39f001bcx7b9914f7bdda4f8f@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
	<191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
	<367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>
	<191c3a030908281333w39f001bcx7b9914f7bdda4f8f@mail.gmail.com>
Message-ID: <191c3a030908281340m83d3c55q96f1257341250b86@mail.gmail.com>

nevermind,

I didn't look closely at your code.

how did you end up with blegSession ?
Are you running this script from the CLI or via the application interface.







On Fri, Aug 28, 2009 at 3:33 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> the L16 codec is used because the wav files is raw PCM audio
> which is being encoded to PCMU
>
> Have you tried doing this to a locally registered phone?
>
>
>
>
> On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> thanks for the reply.
>>
>> Tried this and exactly the same result. The prompt is slowed - as if
>> the person is drunk.
>>
>> I do notice that the originating call SDP shows:
>>
>> a=rtpmap:0 PCMU/8000
>>
>> but the play tries to use:
>>
>> EXECUTE sofia/external/6093693828
>> playback(prompts/press-1-to-accept-call-from.wav)
>> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
>> Activated L16 at 8000hz 1 channels 20ms
>>
>>
>> Should the codec be PCMU at 8000hz as opposed to L16 at 8000hz? If so of
>> there a way of forcing this codec for play.
>>
>> Thanks
>>
>>
>> Phillip
>>
>>
>> Full code & logs
>> http://pastebin.freeswitch.org/10147
>> http://pastebin.freeswitch.org/10150
>>
>>
>>
>> On Thu, Aug 27, 2009 at 4:30 PM, Anthony
>> Minessale<anthony.minessale at gmail.com> wrote:
>> > try wedging {ignore_early_media=true} before the first [ in your dial
>> string
>> > and eliminate the code waiting for answer.
>> >
>> >
>> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <pjintheusa at gmail.com>
>> wrote:
>> >>
>> >> Hi there,
>> >>
>> >> I know there are other ways of doing this. I am just trying get to
>> >> know have fun with the FreeSWITCH API.
>> >>
>> >> I am using originate and park a call:
>> >>
>> >> fsApi.Execute("originate",
>> >>
>> >>
>> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>> >> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
>> >>
>> >> That works great, the phone rings. I want to play something to this
>> >> called party when they pick up.
>> >>
>> >>  while (!blegSession.answered())
>> >>  {
>> >>        Log.WriteLine(LogLevel.Alert,
>> >> "Inside::CallReturns:!Session.answered::Loop");
>> >>        blegSession.sleep(500, 1);
>> >>  }
>> >>
>> >> string promptFile = "prompts/whisper.wav";
>> >> blegSession.StreamFile(promptFile, 0);
>> >>
>> >>
>> >> This works - but the audio is choppy and slow.
>> >>
>> >> Is there something I need to do to that parked call before streaming
>> that
>> >> file?
>> >>
>> >>
>> >> Thanks for any input.
>> >>
>> >> Phil
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>



-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From pjintheusa at gmail.com  Fri Aug 28 13:53:16 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Fri, 28 Aug 2009 16:53:16 -0400
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <191c3a030908281340m83d3c55q96f1257341250b86@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
	<191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
	<367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>
	<191c3a030908281333w39f001bcx7b9914f7bdda4f8f@mail.gmail.com>
	<191c3a030908281340m83d3c55q96f1257341250b86@mail.gmail.com>
Message-ID: <367751820908281353l69b26639va13b3d63321b89e5@mail.gmail.com>

The dialplan is is kicking off the C# script:

<condition field="destination_number" expression="^(12125553666)$">
    * <action application="managed" data="MyFSProjects.ProcessCall"/>*
</condition>

I launch a thread and specify the uuid to originate:

fsApi.Execute("originate",
string.Format("{{ignore_early_media=true,absolute_codec_string='PCMU'}}[*
origination_uuid={0}*,origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
&park", *call.Uuid*, call.OutgoingCallerID, call.NumberToDial));

in the callback i try and play the wav file:

FreeSWITCH.Native.ManagedSession blegSession = new
FreeSWITCH.Native.ManagedSession(callID);

if (blegSession.Ready() && blegSession.mediaReady())
{
      blegSession.Answer();
      blegSession.Execute("playback",
"prompts/press-1-to-accept-call-from.wav");
}

I go on to bridge the leg a and b - that bits works fine.



On Fri, Aug 28, 2009 at 4:40 PM, Anthony Minessale<
anthony.minessale at gmail.com> wrote:
> nevermind,
>
> I didn't look closely at your code.
>
> how did you end up with blegSession ?
> Are you running this script from the CLI or via the application interface.
>
>
>
>
>
>
>
> On Fri, Aug 28, 2009 at 3:33 PM, Anthony Minessale
> <anthony.minessale at gmail.com> wrote:
>>
>> the L16 codec is used because the wav files is raw PCM audio
>> which is being encoded to PCMU
>>
>> Have you tried doing this to a locally registered phone?
>>
>>
>>
>> On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones <pjintheusa at gmail.com>
>> wrote:
>>>
>>> thanks for the reply.
>>>
>>> Tried this and exactly the same result. The prompt is slowed - as if
>>> the person is drunk.
>>>
>>> I do notice that the originating call SDP shows:
>>>
>>> a=rtpmap:0 PCMU/8000
>>>
>>> but the play tries to use:
>>>
>>> EXECUTE sofia/external/6093693828
>>> playback(prompts/press-1-to-accept-call-from.wav)
>>> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
>>> Activated L16 at 8000hz 1 channels 20ms
>>>
>>>
>>> Should the codec be PCMU at 8000hz as opposed to L16 at 8000hz? If so of
>>> there a way of forcing this codec for play.
>>>
>>> Thanks
>>>
>>>
>>> Phillip
>>>
>>>
>>> Full code & logs
>>> http://pastebin.freeswitch.org/10147
>>> http://pastebin.freeswitch.org/10150
>>>
>>>
>>>
>>> On Thu, Aug 27, 2009 at 4:30 PM, Anthony
>>> Minessale<anthony.minessale at gmail.com> wrote:
>>> > try wedging {ignore_early_media=true} before the first [ in your dial
>>> > string
>>> > and eliminate the code waiting for answer.
>>> >
>>> >
>>> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <pjintheusa at gmail.com>
>>> > wrote:
>>> >>
>>> >> Hi there,
>>> >>
>>> >> I know there are other ways of doing this. I am just trying get to
>>> >> know have fun with the FreeSWITCH API.
>>> >>
>>> >> I am using originate and park a call:
>>> >>
>>> >> fsApi.Execute("originate",
>>> >>
>>> >>
>>> >>
string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>>> >> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
>>> >>
>>> >> That works great, the phone rings. I want to play something to this
>>> >> called party when they pick up.
>>> >>
>>> >>  while (!blegSession.answered())
>>> >>  {
>>> >>        Log.WriteLine(LogLevel.Alert,
>>> >> "Inside::CallReturns:!Session.answered::Loop");
>>> >>        blegSession.sleep(500, 1);
>>> >>  }
>>> >>
>>> >> string promptFile = "prompts/whisper.wav";
>>> >> blegSession.StreamFile(promptFile, 0);
>>> >>
>>> >>
>>> >> This works - but the audio is choppy and slow.
>>> >>
>>> >> Is there something I need to do to that parked call before streaming
>>> >> that
>>> >> file?
>>> >>
>>> >>
>>> >> Thanks for any input.
>>> >>
>>> >> Phil
>>> >>
>>> >> _______________________________________________
>>> >> FreeSWITCH-users mailing list
>>> >> FreeSWITCH-users at lists.freeswitch.org
>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >>
>>> >> UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> >> http://www.freeswitch.org
>>> >
>>> >
>>> >
>>> > --
>>> > Anthony Minessale II
>>> >
>>> > FreeSWITCH http://www.freeswitch.org/
>>> > ClueCon http://www.cluecon.com/
>>> > Twitter: http://twitter.com/FreeSWITCH_wire
>>> >
>>> > AIM: anthm
>>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>> > IRC: irc.freenode.net #freeswitch
>>> >
>>> > FreeSWITCH Developer Conference
>>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>>> > iax:guest at conference.freeswitch.org/888
>>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>>> > pstn:213-799-1400
>>> >
>>> > _______________________________________________
>>> > FreeSWITCH-users mailing list
>>> > FreeSWITCH-users at lists.freeswitch.org
>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> >
>>> > UNSUBSCRIBE:
http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> > http://www.freeswitch.org
>>> >
>>> >
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
>> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:213-799-1400
>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From dave at 3c.co.uk  Fri Aug 28 14:01:53 2009
From: dave at 3c.co.uk (David Knell)
Date: Sat, 29 Aug 2009 00:01:53 +0300
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
Message-ID: <1251493313.4438.8.camel@dk-d820>

Hi Raimund,

One FreeSWITCH box will be quite enough to handle the call volumes that
you're talking about, and it ought to be much more stable than the
Asterisk solution which you've outlined below.

It's probably best to forget about live failover without calls dropping
- this isn't something that's supported, and there'd be a lot of work to
do to develop code to keep two boxed in sync.  Once you get used to a
stable solution - i.e. something which doesn't crash - then live
failover, HA, etc., will seem somewhat irrelevant.

I recently did some work on an FS-based call center solution - drop me a
note if I could be of any help with yours.

Cheers --

Dave

> Hello List,
> 
> I have read the current thread about scalability and I would need some  
> advice about a callcenter setup:
> 
> First where I come from:
> I have lot's of problems with an asterisk solution. I have regular  
> crash's and lock-ups, with downgrading and other stuff i got it  
> somewhat stable, but have nevertheless regular hickups. I am desperate  
> and want to get rid of asterisk and I hope that freeSwitch will  
> provide me with a more stable solution.
> 
> 
> Our Setup (really nothing special):
> * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3  
> GIG of RAM, Xircom analog switch connected to mobile stations for 4  
> different providers, Digium 4port cards TP400<something>
> * 8 queues
> * ~60 agents (which logon, logoff, pause, unpause), not more than 40  
> concurrently online
> * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
> * Music on Hold in the call-queues
> * No special announcement
> * Transfers between calls in queues and different agents as well as  
> non agents (i mention this because we have transfer related chrashes  
> in asterisk)
> 
> The current Problems:
> * Lockups with different causes (ranging from calls not terminated to  
> heavy thread locking through the AMI interface)
> * Crashes and library aborts (pthread, libc, crashes related to music  
> on hold, app_queue, transfers)
> 
> We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are  
> now back to 1.4.21.2 (stock debian) as anything beyond that is for  
> whatever reason highly unstable for our szenario. Maybe we should have  
> been segmenting the box into one asterisk dedicted to talking to the  
> hardware, one especial for queue/sip handling, i do not know. (all  
> issues are well documented in issues.asterisk.org, but it seems to be  
> very, very difficult to get to the bottom of them as they exist since  
> 1.4.23 as it seems and are open until know and not fixable since month.)
> 
> 
> Now, I really would appreciate some success-stories on how you guys  
> managed to get a stable pbx system with freeSWITCH in regard of HA and  
> scalability:
> 
> * How to segment freeSWITCH? Or is it stable enough to handle all in  
> one for such a szenario as outlined above?
> * What would be the best strategie for High Availability / Failover?
> 	-> I read in the WIKI (featurelist) that Livemigration of calls from  
> one box to another should be possible?
> 	-> I was thinking about using memcached for storing all state  
> information so another freeswitch box can takeover calls from the  
> first box if it dies, is this possible? If so, how?
> 	-> Is there anotherway to somehow configure freeSWITCH that in the  
> event of a crash i do not loose the current established calls?
> 
> Basically I just want a stable PBX where I do not have to fear every  
> day it will core-dump or abort or Lock up. Is freeSWITCH mature enough  
> so i can sleep at night for at least 3 month without a crash?
> 
> 
> 
> Thank you for your Time and help in advance, and I am more than  
> willing to take all the information gathered here and create a wiki  
> page to help other people with the same questions/problems.
> 
> best
> Ray
> 
> 
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk




From anthony.minessale at gmail.com  Fri Aug 28 14:12:14 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Fri, 28 Aug 2009 16:12:14 -0500
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <367751820908281353l69b26639va13b3d63321b89e5@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
	<191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
	<367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>
	<191c3a030908281333w39f001bcx7b9914f7bdda4f8f@mail.gmail.com>
	<191c3a030908281340m83d3c55q96f1257341250b86@mail.gmail.com>
	<367751820908281353l69b26639va13b3d63321b89e5@mail.gmail.com>
Message-ID: <191c3a030908281412h51b75088s8c40bd0a53688928@mail.gmail.com>

Taking over the session while it's parked like that from your code and
asking it to play a file is making it do 2 things at once.
The session's thread is already busy in the park loop so you would have to
change it's state to something passive like soft_execute so
it's thread was not doing anything or send it a message so the park loop
would pick it up by using the uuid_broadcast app::arg

there are variables you can set that begin group_confirm_* that you can set
to do this all without even using a script.

http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_variables




On Fri, Aug 28, 2009 at 3:53 PM, Phillip Jones <pjintheusa at gmail.com> wrote:

> The dialplan is is kicking off the C# script:
>
> <condition field="destination_number" expression="^(12125553666)$">
>     * <action application="managed" data="MyFSProjects.ProcessCall"/>*
> </condition>
>
> I launch a thread and specify the uuid to originate:
>
> fsApi.Execute("originate",
> string.Format("{{ignore_early_media=true,absolute_codec_string='PCMU'}}[*
> origination_uuid={0}*,origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
> &park", *call.Uuid*, call.OutgoingCallerID, call.NumberToDial));
>
> in the callback i try and play the wav file:
>
> FreeSWITCH.Native.ManagedSession blegSession = new
> FreeSWITCH.Native.ManagedSession(callID);
>
> if (blegSession.Ready() && blegSession.mediaReady())
> {
>       blegSession.Answer();
>       blegSession.Execute("playback",
> "prompts/press-1-to-accept-call-from.wav");
> }
>
> I go on to bridge the leg a and b - that bits works fine.
>
>
>
> On Fri, Aug 28, 2009 at 4:40 PM, Anthony Minessale<
> anthony.minessale at gmail.com> wrote:
> > nevermind,
> >
> > I didn't look closely at your code.
> >
> > how did you end up with blegSession ?
> > Are you running this script from the CLI or via the application
> interface.
> >
> >
> >
> >
> >
> >
> >
> > On Fri, Aug 28, 2009 at 3:33 PM, Anthony Minessale
> > <anthony.minessale at gmail.com> wrote:
> >>
> >> the L16 codec is used because the wav files is raw PCM audio
> >> which is being encoded to PCMU
> >>
> >> Have you tried doing this to a locally registered phone?
> >>
> >>
> >>
> >> On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones <pjintheusa at gmail.com>
> >> wrote:
> >>>
> >>> thanks for the reply.
> >>>
> >>> Tried this and exactly the same result. The prompt is slowed - as if
> >>> the person is drunk.
> >>>
> >>> I do notice that the originating call SDP shows:
> >>>
> >>> a=rtpmap:0 PCMU/8000
> >>>
> >>> but the play tries to use:
> >>>
> >>> EXECUTE sofia/external/6093693828
> >>> playback(prompts/press-1-to-accept-call-from.wav)
> >>> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
> >>> Activated L16 at 8000hz 1 channels 20ms
> >>>
> >>>
> >>> Should the codec be PCMU at 8000hz as opposed to L16 at 8000hz? If so of
> >>> there a way of forcing this codec for play.
> >>>
> >>> Thanks
> >>>
> >>>
> >>> Phillip
> >>>
> >>>
> >>> Full code & logs
> >>> http://pastebin.freeswitch.org/10147
> >>> http://pastebin.freeswitch.org/10150
> >>>
> >>>
> >>>
> >>> On Thu, Aug 27, 2009 at 4:30 PM, Anthony
> >>> Minessale<anthony.minessale at gmail.com> wrote:
> >>> > try wedging {ignore_early_media=true} before the first [ in your dial
> >>> > string
> >>> > and eliminate the code waiting for answer.
> >>> >
> >>> >
> >>> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <pjintheusa at gmail.com
> >
> >>> > wrote:
> >>> >>
> >>> >> Hi there,
> >>> >>
> >>> >> I know there are other ways of doing this. I am just trying get to
> >>> >> know have fun with the FreeSWITCH API.
> >>> >>
> >>> >> I am using originate and park a call:
> >>> >>
> >>> >> fsApi.Execute("originate",
> >>> >>
> >>> >>
> >>> >>
> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
> >>> >> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
> >>> >>
> >>> >> That works great, the phone rings. I want to play something to this
> >>> >> called party when they pick up.
> >>> >>
> >>> >>  while (!blegSession.answered())
> >>> >>  {
> >>> >>        Log.WriteLine(LogLevel.Alert,
> >>> >> "Inside::CallReturns:!Session.answered::Loop");
> >>> >>        blegSession.sleep(500, 1);
> >>> >>  }
> >>> >>
> >>> >> string promptFile = "prompts/whisper.wav";
> >>> >> blegSession.StreamFile(promptFile, 0);
> >>> >>
> >>> >>
> >>> >> This works - but the audio is choppy and slow.
> >>> >>
> >>> >> Is there something I need to do to that parked call before streaming
> >>> >> that
> >>> >> file?
> >>> >>
> >>> >>
> >>> >> Thanks for any input.
> >>> >>
> >>> >> Phil
> >>> >>
> >>> >> _______________________________________________
> >>> >> FreeSWITCH-users mailing list
> >>> >> FreeSWITCH-users at lists.freeswitch.org
> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >>
> >>> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> >> http://www.freeswitch.org
> >>> >
> >>> >
> >>> >
> >>> > --
> >>> > Anthony Minessale II
> >>> >
> >>> > FreeSWITCH http://www.freeswitch.org/
> >>> > ClueCon http://www.cluecon.com/
> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
> >>> >
> >>> > AIM: anthm
> >>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >>> > IRC: irc.freenode.net #freeswitch
> >>> >
> >>> > FreeSWITCH Developer Conference
> >>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
> >>> > iax:guest at conference.freeswitch.org/888
> >>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >>> > pstn:213-799-1400
> >>> >
> >>> > _______________________________________________
> >>> > FreeSWITCH-users mailing list
> >>> > FreeSWITCH-users at lists.freeswitch.org
> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> >
> >>> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> > http://www.freeswitch.org
> >>> >
> >>> >
> >>>
> >>> _______________________________________________
> >>> FreeSWITCH-users mailing list
> >>> FreeSWITCH-users at lists.freeswitch.org
> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >>> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >>> http://www.freeswitch.org
> >>
> >>
> >>
> >> --
> >> Anthony Minessale II
> >>
> >> FreeSWITCH http://www.freeswitch.org/
> >> ClueCon http://www.cluecon.com/
> >> Twitter: http://twitter.com/FreeSWITCH_wire
> >>
> >> AIM: anthm
> >> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> >> IRC: irc.freenode.net #freeswitch
> >>
> >> FreeSWITCH Developer Conference
> >> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> >> iax:guest at conference.freeswitch.org/888
> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> >> pstn:213-799-1400
> >
> >
> >
> > --
> > Anthony Minessale II
> >
> > FreeSWITCH http://www.freeswitch.org/
> > ClueCon http://www.cluecon.com/
> > Twitter: http://twitter.com/FreeSWITCH_wire
> >
> > AIM: anthm
> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> > IRC: irc.freenode.net #freeswitch
> >
> > FreeSWITCH Developer Conference
> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> > iax:guest at conference.freeswitch.org/888
> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> > pstn:213-799-1400
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888
googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400
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From oseslija at gmail.com  Fri Aug 28 15:03:12 2009
From: oseslija at gmail.com (Ognjen Seslija)
Date: Sat, 29 Aug 2009 00:03:12 +0200
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
Message-ID: <4468a6770908281503gd7c003ewf44fd353f78c21e4@mail.gmail.com>

I currently use FS for a managed PBX services for many companies. It's so
configurable and extensible, I don't think I would change it for anything. I
have Cisco CME and SCCP phones in my office all interconnected to FS doing
logic and thinking. I also have Asterisks as PSTN PRI gateways. FS does all
the job for me in the world, the other two are just used as protocol
gateways. I wouldn't change FS for anything else now, after a year and the
half of playing with it. I fully recommend it.

Regards,
Ognjen

P.S: To be fair, the echo on the hw ec cards is not really Asterisk's fault.



On Fri, Aug 28, 2009 at 8:14 PM, Christensen Tom <paveraware at hotmail.com>wrote:

>  As a background, I ran an asterisk consulting company for about 3 years
> that I gave up on 2 years ago after repeatedly failing to achieve any sort
> of stability on any sort install over about 30 phones, I gave up.
>
> Maybe that was wrong, I am open to the possibility that I just didn't know
> enough and I was building things wrong, but I worked inside the asterisk
> code (which I feel is a hopeless mess), I implemented a few small custom
> features, anyway...
>
> I'm coming back into the VoIP space now, and I'm wondering what sort of
> issues can I expect in trying to pick up and learn freeswitch?  From what
> I've read on the website, it appears to have a much more sane architecture.
> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
> stable and worst general call quality was asterisk.  I constantly contended
> with strange call quality issues in asterisk, lots of echo (even with
> hardware echo cancellation cards), lots of jitter, lots of call break up
> (even on small systems with 10-20 users, using QoS on the network, and in
> general doing everything I could to prioritize voice over anything else).
>
> When I used Cisco call manager and broadsoft, the voice quality issues were
> basically non-existant, as long as the network was running QoS echo,
> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
> does freeswitch show a marked improvement over asterisk in this department?
> As long as you configure QoS and have hardware echo cancellation does it
> actually work reliably?
>
> Thanks for any additional information about freeswitch you can provide as
> well.  I am a software developer primarily by trade, but I do lots of
> consulting type work in the SME space and I've had a couple projects thrown
> to me that require some integration with a phone system, and I just can't in
> good conscience recommend asterisk anymore.
>
>
> -Tom
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From asannucci at gmail.com  Fri Aug 28 15:05:17 2009
From: asannucci at gmail.com (bakko)
Date: Sat, 29 Aug 2009 00:05:17 +0200
Subject: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS
In-Reply-To: <7b197bef0908281302n43fa2c8aoc6d713f2681fedd5@mail.gmail.com>
References: <9cb0e15e0908281233j28d399a7iacbbc0f1d241c68b@mail.gmail.com>
	<87f2f3b90908281237k2fb599c2ue36d20b8807cf31d@mail.gmail.com>
	<d36c422fa99228a711fb7eba47c7e94f.squirrel@correo.nodo50.org>
	<CFCEE02A-A1EE-41E3-AC3D-C57F35990111@halo2.pl>
	<7b197bef0908281302n43fa2c8aoc6d713f2681fedd5@mail.gmail.com>
Message-ID: <5D224FBE6FC943AC826B02B5DB4E2166@voztovoice>

windows 95 :P

----- Original Message ----- 
From: "Giovanni Maruzzelli" <gmaruzz at celliax.org>
To: <freeswitch-users at lists.freeswitch.org>
Sent: Friday, August 28, 2009 10:02 PM
Subject: Re: [Freeswitch-users] Centos 5.3 vs Ubuntu LTS


Vista!


On Fri, Aug 28, 2009 at 10:00 PM, Michal
Bielicki<michal.bielicki at halo2.pl> wrote:
> OpenSolaris :P
>
> Am 28.08.2009 um 21:50 schrieb Alberto Escudero:
>
>> Debian :)
>> -- Stopping junk mailers is good for the environment
>>
>>> On Fri, Aug 28, 2009 at 12:33 PM, Fernando Testa <
>>> testa at voicetechnology.com.br> wrote:
>>>
>>>> Hi folks,What linux distro would you suggest to deploy FS with django:
>>>> Centos 5.3 or Ubuntu LTS?
>>>>
>>> CentOS 5.3 FTW!
>>> -MC
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
> Michal Bielicki
> Leiter der Niederlassung
> HaloKwadrat Sp. z o.o.
> Niederlassung Kleinmachnow
> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
> Ust.Id.: DE261885536
> Geschaeftsfuehrer: Aleksander Wiercinski
> Meiereifeld 2b, 14532 Kleinmachnow
> t. +49 33203 263220
> f. +49 33203 263229 sip. info at halokwadrat.de
> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
> Hauptgesch?ftsstelle:
> Halo Kwadrat Sp. z o.o.
> ul. Polna 46/14
> 00-644 Warszawa, Polen
> EIngetragen im HRB Warszawa, KRS 0000153539
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org 




From diego.viola at gmail.com  Fri Aug 28 15:16:56 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Fri, 28 Aug 2009 22:16:56 +0000
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <4468a6770908281503gd7c003ewf44fd353f78c21e4@mail.gmail.com>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
	<4468a6770908281503gd7c003ewf44fd353f78c21e4@mail.gmail.com>
Message-ID: <86a32abc0908281516kef890fg681dd0f916c101d@mail.gmail.com>

You guys should put your experiences here :).

http://wiki.freeswitch.org/wiki/Testimonials

Diego

On Fri, Aug 28, 2009 at 10:03 PM, Ognjen Seslija <oseslija at gmail.com> wrote:

> I currently use FS for a managed PBX services for many companies. It's so
> configurable and extensible, I don't think I would change it for anything. I
> have Cisco CME and SCCP phones in my office all interconnected to FS doing
> logic and thinking. I also have Asterisks as PSTN PRI gateways. FS does all
> the job for me in the world, the other two are just used as protocol
> gateways. I wouldn't change FS for anything else now, after a year and the
> half of playing with it. I fully recommend it.
>
> Regards,
> Ognjen
>
> P.S: To be fair, the echo on the hw ec cards is not really Asterisk's
> fault.
>
>
>
> On Fri, Aug 28, 2009 at 8:14 PM, Christensen Tom <paveraware at hotmail.com>wrote:
>
>>  As a background, I ran an asterisk consulting company for about 3 years
>> that I gave up on 2 years ago after repeatedly failing to achieve any sort
>> of stability on any sort install over about 30 phones, I gave up.
>>
>> Maybe that was wrong, I am open to the possibility that I just didn't know
>> enough and I was building things wrong, but I worked inside the asterisk
>> code (which I feel is a hopeless mess), I implemented a few small custom
>> features, anyway...
>>
>> I'm coming back into the VoIP space now, and I'm wondering what sort of
>> issues can I expect in trying to pick up and learn freeswitch?  >From what
>> I've read on the website, it appears to have a much more sane architecture.
>> I've used Cisco, Broadsoft, and asterisk in the past.  By far the least
>> stable and worst general call quality was asterisk.  I constantly contended
>> with strange call quality issues in asterisk, lots of echo (even with
>> hardware echo cancellation cards), lots of jitter, lots of call break up
>> (even on small systems with 10-20 users, using QoS on the network, and in
>> general doing everything I could to prioritize voice over anything else).
>>
>> When I used Cisco call manager and broadsoft, the voice quality issues
>> were basically non-existant, as long as the network was running QoS echo,
>> stutter, calls breaking up, just didn't happen.  So, I guess my question is,
>> does freeswitch show a marked improvement over asterisk in this department?
>> As long as you configure QoS and have hardware echo cancellation does it
>> actually work reliably?
>>
>> Thanks for any additional information about freeswitch you can provide as
>> well.  I am a software developer primarily by trade, but I do lots of
>> consulting type work in the SME space and I've had a couple projects thrown
>> to me that require some integration with a phone system, and I just can't in
>> good conscience recommend asterisk anymore.
>>
>>
>> -Tom
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tayeb.meftah at gmail.com  Fri Aug 28 17:17:19 2009
From: tayeb.meftah at gmail.com (Meftah Tayeb)
Date: Sat, 29 Aug 2009 00:17:19 +0000
Subject: [Freeswitch-users] Hello, and stuff.
In-Reply-To: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
References: <BAY120-W4805E0DD8A1DF8C181658FDFF50@phx.gbl>
Message-ID: <4A98738F.2000001@gmail.com>

hello,
freeswitch is Awesome and perfect!
if you know XML, freeswitch use XML for there configuration file, no 
same to asterisk that trying to create a configuration language!
this unic featur let you integrate freeswitch with any Application that 
support and Parce XML including Web 2.0
freeswitch also support protocol transcoding, including SIP, H.323, 
Iax2, Skype and GTalk!
Real Time Feature: freeswitch support XML_Curl that let you configure 
freeswitch remotly from a web server that fetch data from a DB and 
return it to freeswitch in XML format
XMLRPC let you execute remote api command using your web browser or your 
XMLRPC application
about Event Socket, this let you build any type of application that talk 
to freeswitch or freeswitch talk to it in inbound and outbound mode
good luk!
Christensen Tom wrote:
> As a background, I ran an asterisk consulting company for about 3 
> years that I gave up on 2 years ago after repeatedly failing to 
> achieve any sort of stability on any sort install over about 30 
> phones, I gave up.
>
> Maybe that was wrong, I am open to the possibility that I just didn't 
> know enough and I was building things wrong, but I worked inside the 
> asterisk code (which I feel is a hopeless mess), I implemented a few 
> small custom features, anyway...
>
> I'm coming back into the VoIP space now, and I'm wondering what sort 
> of issues can I expect in trying to pick up and learn freeswitch?  
> From what I've read on the website, it appears to have a much more 
> sane architecture.  I've used Cisco, Broadsoft, and asterisk in the 
> past.  By far the least stable and worst general call quality was 
> asterisk.  I constantly contended with strange call quality issues in 
> asterisk, lots of echo (even with hardware echo cancellation cards), 
> lots of jitter, lots of call break up (even on small systems with 
> 10-20 users, using QoS on the network, and in general doing everything 
> I could to prioritize voice over anything else).
>
> When I used Cisco call manager and broadsoft, the voice quality issues 
> were basically non-existant, as long as the network was running QoS 
> echo, stutter, calls breaking up, just didn't happen.  So, I guess my 
> question is, does freeswitch show a marked improvement over asterisk 
> in this department?  As long as you configure QoS and have hardware 
> echo cancellation does it actually work reliably?
>
> Thanks for any additional information about freeswitch you can provide 
> as well.  I am a software developer primarily by trade, but I do lots 
> of consulting type work in the SME space and I've had a couple 
> projects thrown to me that require some integration with a phone 
> system, and I just can't in good conscience recommend asterisk anymore.
>
>
> -Tom
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus 
> signature database 4378 (20090828) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
> ------------------------------------------------------------------------
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> __________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________
>
> The message was checked by ESET NOD32 Antivirus.
>
> http://www.eset.com
>
>   





__________ Information from ESET NOD32 Antivirus, version of virus signature database 4378 (20090828) __________

The message was checked by ESET NOD32 Antivirus.

http://www.eset.com

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From tomabroad at gmail.com  Fri Aug 28 16:35:46 2009
From: tomabroad at gmail.com (tom)
Date: Fri, 28 Aug 2009 19:35:46 -0400
Subject: [Freeswitch-users] alternative to java-asterisk
Message-ID: <6f7c60c40908281635i2618a23p65a879302ff113f4@mail.gmail.com>

hi,

with the java-asterisk bridge its possible to to AMI-stuff. is thi as well
possible with freeswitch? if so, are there any tutorials, docu etc. out
there?
thx
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From msc at freeswitch.org  Fri Aug 28 16:53:25 2009
From: msc at freeswitch.org (Michael Collins)
Date: Fri, 28 Aug 2009 16:53:25 -0700
Subject: [Freeswitch-users] alternative to java-asterisk
In-Reply-To: <6f7c60c40908281635i2618a23p65a879302ff113f4@mail.gmail.com>
References: <6f7c60c40908281635i2618a23p65a879302ff113f4@mail.gmail.com>
Message-ID: <87f2f3b90908281653s6d7c8f72i43feb8eb739f46db@mail.gmail.com>

On Fri, Aug 28, 2009 at 4:35 PM, tom <tomabroad at gmail.com> wrote:

> hi,
>
> with the java-asterisk bridge its possible to to AMI-stuff. is thi as well
> possible with freeswitch? if so, are there any tutorials, docu etc. out
> there?
> thx


The AMI in Asterisk is analogous to the event socket in FreeSWITCH. (The
event socket is extremely powerful.) The FS developers have created ESL -
the event socket library, which abstracts away some of the mundane parts of
creating a socket connection to FS. I believe they added provisional support
for Java. We don't have a tutorial per se on the ESL, especially for Java.
However, we sure could use an adventurous Java programmer to roll up his
sleeves and help test & document all this stuff. :)

Check out src/libs/esl in the FS source tree and the event socket stuff on
the wiki:
http://wiki.freeswitch.org/wiki/Event_Socket
http://wiki.freeswitch.org/wiki/Event_Socket_Library

Have fun!
-MC
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From sdjernes at gmail.com  Fri Aug 28 18:35:02 2009
From: sdjernes at gmail.com (Shawn L. Djernes)
Date: Fri, 28 Aug 2009 20:35:02 -0500
Subject: [Freeswitch-users] Group Call
In-Reply-To: <87f2f3b90908281130g1e33668fs44b817f9c622641c@mail.gmail.com>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
	<5ff3e3a70908251015ue88b8b9p7a5e091d15a6b5ae@mail.gmail.com>
	<87f2f3b90908281130g1e33668fs44b817f9c622641c@mail.gmail.com>
Message-ID: <5ff3e3a70908281835g3983c7d0ma8716629b45b5f78@mail.gmail.com>

Here is a copy of the data from the console at log 7
freeswitch at internal> group_call
sdjernes+A at ewr.djernes.net<sdjernes%2BA at ewr.djernes.net>
-ERR no reply

freeswitch at internal> group_call sdjernes at ewr.djernes.net
error/NO_ROUTE_DESTINATION
freeswitch at internal> group_call
sdjernes+F at ewr.djernes.net<sdjernes%2BF at ewr.djernes.net>
-ERR no reply

freeswitch at internal> group_call
sdjernes+A at freeswitch.djernes.net<sdjernes%2BA at freeswitch.djernes.net>
-ERR no reply

freeswitch at internal>

On Fri, Aug 28, 2009 at 1:30 PM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Tue, Aug 25, 2009 at 10:15 AM, Shawn L. Djernes <sdjernes at gmail.com>wrote:
>
>> Woops,
>> That line was an attempt to try something from an older diaplan example.
>>  Copied and pasted from a file on my computer not the server.  The actual
>> line is:
>>
>> <action application="bridge"
>> data="{ignore_early_media=true}${group_call(sdjernes+A@
>> ${domain_name})}"/>
>>
>> The +A is to ring all phones in the group at the same time.
>>
>>
> Can you confirm what you see on the CLI when you do this:
> group_call sdjernes+A at ewr.djernes.net <sdjernes%2BA at ewr.djernes.net>
>
> Let's make sure the group is properly defined, etc. before doing anything
> else.
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>


-- 
Shawn L. Djernes
SD Consulting
shawn at djernes.org | sdjernes at gmail.com
MSN: wizardwlf at hotmail.com
402.345.7734 | 402.350.6973 Cell
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From diego.viola at gmail.com  Fri Aug 28 19:34:32 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 29 Aug 2009 02:34:32 +0000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
Message-ID: <86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>

Yes, FreeSWITCH is a system that you can trust 100%. I have switched my
Asterisk servers to FreeSWITCH and have peace now.

If I were you I would get rid of Asterisk and use FreeSWITCH, FS will handle
all what you want very well.

And I agree with David, fail-over is kinda irrelevant since the FS doesn't
crash like Asterisk does.

Diego


On Fri, Aug 28, 2009 at 6:52 PM, Raimund Sacherer <rs at runsolutions.com>wrote:

> Hello List,
>
> I have read the current thread about scalability and I would need some
> advice about a callcenter setup:
>
> First where I come from:
> I have lot's of problems with an asterisk solution. I have regular
> crash's and lock-ups, with downgrading and other stuff i got it
> somewhat stable, but have nevertheless regular hickups. I am desperate
> and want to get rid of asterisk and I hope that freeSwitch will
> provide me with a more stable solution.
>
>
> Our Setup (really nothing special):
> * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3
> GIG of RAM, Xircom analog switch connected to mobile stations for 4
> different providers, Digium 4port cards TP400<something>
> * 8 queues
> * ~60 agents (which logon, logoff, pause, unpause), not more than 40
> concurrently online
> * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
> * Music on Hold in the call-queues
> * No special announcement
> * Transfers between calls in queues and different agents as well as
> non agents (i mention this because we have transfer related chrashes
> in asterisk)
>
> The current Problems:
> * Lockups with different causes (ranging from calls not terminated to
> heavy thread locking through the AMI interface)
> * Crashes and library aborts (pthread, libc, crashes related to music
> on hold, app_queue, transfers)
>
> We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are
> now back to 1.4.21.2 (stock debian) as anything beyond that is for
> whatever reason highly unstable for our szenario. Maybe we should have
> been segmenting the box into one asterisk dedicted to talking to the
> hardware, one especial for queue/sip handling, i do not know. (all
> issues are well documented in issues.asterisk.org, but it seems to be
> very, very difficult to get to the bottom of them as they exist since
> 1.4.23 as it seems and are open until know and not fixable since month.)
>
>
> Now, I really would appreciate some success-stories on how you guys
> managed to get a stable pbx system with freeSWITCH in regard of HA and
> scalability:
>
> * How to segment freeSWITCH? Or is it stable enough to handle all in
> one for such a szenario as outlined above?
> * What would be the best strategie for High Availability / Failover?
>        -> I read in the WIKI (featurelist) that Livemigration of calls from
> one box to another should be possible?
>        -> I was thinking about using memcached for storing all state
> information so another freeswitch box can takeover calls from the
> first box if it dies, is this possible? If so, how?
>        -> Is there anotherway to somehow configure freeSWITCH that in the
> event of a crash i do not loose the current established calls?
>
> Basically I just want a stable PBX where I do not have to fear every
> day it will core-dump or abort or Lock up. Is freeSWITCH mature enough
> so i can sleep at night for at least 3 month without a crash?
>
>
>
> Thank you for your Time and help in advance, and I am more than
> willing to take all the information gathered here and create a wiki
> page to help other people with the same questions/problems.
>
> best
> Ray
>
>
> --
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From diego.viola at gmail.com  Fri Aug 28 19:37:12 2009
From: diego.viola at gmail.com (Diego Viola)
Date: Sat, 29 Aug 2009 02:37:12 +0000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <1251493313.4438.8.camel@dk-d820>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<1251493313.4438.8.camel@dk-d820>
Message-ID: <86a32abc0908281937g4eea9d83x667bd3455f7a45bc@mail.gmail.com>

Hi David,

What have you used on FS for call center, mod_fifo?

Can you describe your experience with that, I'm currently interested in call
center + FS scenario.

Diego

On Fri, Aug 28, 2009 at 9:01 PM, David Knell <dave at 3c.co.uk> wrote:

> Hi Raimund,
>
> One FreeSWITCH box will be quite enough to handle the call volumes that
> you're talking about, and it ought to be much more stable than the
> Asterisk solution which you've outlined below.
>
> It's probably best to forget about live failover without calls dropping
> - this isn't something that's supported, and there'd be a lot of work to
> do to develop code to keep two boxed in sync.  Once you get used to a
> stable solution - i.e. something which doesn't crash - then live
> failover, HA, etc., will seem somewhat irrelevant.
>
> I recently did some work on an FS-based call center solution - drop me a
> note if I could be of any help with yours.
>
> Cheers --
>
> Dave
>
> > Hello List,
> >
> > I have read the current thread about scalability and I would need some
> > advice about a callcenter setup:
> >
> > First where I come from:
> > I have lot's of problems with an asterisk solution. I have regular
> > crash's and lock-ups, with downgrading and other stuff i got it
> > somewhat stable, but have nevertheless regular hickups. I am desperate
> > and want to get rid of asterisk and I hope that freeSwitch will
> > provide me with a more stable solution.
> >
> >
> > Our Setup (really nothing special):
> > * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack server, 3
> > GIG of RAM, Xircom analog switch connected to mobile stations for 4
> > different providers, Digium 4port cards TP400<something>
> > * 8 queues
> > * ~60 agents (which logon, logoff, pause, unpause), not more than 40
> > concurrently online
> > * ~ 7K - 9K calls (well, CDR entries) a day (not that much for a bpx)
> > * Music on Hold in the call-queues
> > * No special announcement
> > * Transfers between calls in queues and different agents as well as
> > non agents (i mention this because we have transfer related chrashes
> > in asterisk)
> >
> > The current Problems:
> > * Lockups with different causes (ranging from calls not terminated to
> > heavy thread locking through the AMI interface)
> > * Crashes and library aborts (pthread, libc, crashes related to music
> > on hold, app_queue, transfers)
> >
> > We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5, 1.4.26 and are
> > now back to 1.4.21.2 (stock debian) as anything beyond that is for
> > whatever reason highly unstable for our szenario. Maybe we should have
> > been segmenting the box into one asterisk dedicted to talking to the
> > hardware, one especial for queue/sip handling, i do not know. (all
> > issues are well documented in issues.asterisk.org, but it seems to be
> > very, very difficult to get to the bottom of them as they exist since
> > 1.4.23 as it seems and are open until know and not fixable since month.)
> >
> >
> > Now, I really would appreciate some success-stories on how you guys
> > managed to get a stable pbx system with freeSWITCH in regard of HA and
> > scalability:
> >
> > * How to segment freeSWITCH? Or is it stable enough to handle all in
> > one for such a szenario as outlined above?
> > * What would be the best strategie for High Availability / Failover?
> >       -> I read in the WIKI (featurelist) that Livemigration of calls
> from
> > one box to another should be possible?
> >       -> I was thinking about using memcached for storing all state
> > information so another freeswitch box can takeover calls from the
> > first box if it dies, is this possible? If so, how?
> >       -> Is there anotherway to somehow configure freeSWITCH that in the
> > event of a crash i do not loose the current established calls?
> >
> > Basically I just want a stable PBX where I do not have to fear every
> > day it will core-dump or abort or Lock up. Is freeSWITCH mature enough
> > so i can sleep at night for at least 3 month without a crash?
> >
> >
> >
> > Thank you for your Time and help in advance, and I am more than
> > willing to take all the information gathered here and create a wiki
> > page to help other people with the same questions/problems.
> >
> > best
> > Ray
> >
> >
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From dave at 3c.co.uk  Fri Aug 28 20:49:17 2009
From: dave at 3c.co.uk (David Knell)
Date: Sat, 29 Aug 2009 06:49:17 +0300
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <86a32abc0908281937g4eea9d83x667bd3455f7a45bc@mail.gmail.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<1251493313.4438.8.camel@dk-d820>
	<86a32abc0908281937g4eea9d83x667bd3455f7a45bc@mail.gmail.com>
Message-ID: <1251517757.4438.25.camel@dk-d820>

Hi Diego,

We didn't use mod_fifo; we built our own queues using an application
hanging off the event socket.  This was partly down to typical
programmer hubris, and partly to allow us to do things that mod_fifo
might not without either (a) trawling through mod_fifo, or (b) pestering
Anthony.

As a brief outline:
- operators each live in their own conference
- supervisors can add themselves to these conferences for monitoring or
coaching
- callers sit in queues for the various skills that the operators have.

When an operator is free, look at each queue for the skills that the
operator has, and choose the caller who's been waiting longest.

I used a one-thread-per-call model, so there's a little bit of thought
needed to get the access to shared structures right but, essentially,
given the low frequency of access to them and the fact that nothing was
time-critical at the sub-second level, I just wrapped all of the
accesses in a single global mutex, which was easy and is pretty
foolproof.  The latter is important, given that it was me writing the
stuff ;-)

Cheers --

Dave

> Hi David,
> 
> What have you used on FS for call center, mod_fifo?
> 
> Can you describe your experience with that, I'm currently interested
> in call center + FS scenario.
> 
> Diego
> 
> On Fri, Aug 28, 2009 at 9:01 PM, David Knell <dave at 3c.co.uk> wrote:
>         Hi Raimund,
>         
>         One FreeSWITCH box will be quite enough to handle the call
>         volumes that
>         you're talking about, and it ought to be much more stable than
>         the
>         Asterisk solution which you've outlined below.
>         
>         It's probably best to forget about live failover without calls
>         dropping
>         - this isn't something that's supported, and there'd be a lot
>         of work to
>         do to develop code to keep two boxed in sync.  Once you get
>         used to a
>         stable solution - i.e. something which doesn't crash - then
>         live
>         failover, HA, etc., will seem somewhat irrelevant.
>         
>         I recently did some work on an FS-based call center solution -
>         drop me a
>         note if I could be of any help with yours.
>         
>         Cheers --
>         
>         Dave
>         
>         
>         > Hello List,
>         >
>         > I have read the current thread about scalability and I would
>         need some
>         > advice about a callcenter setup:
>         >
>         > First where I come from:
>         > I have lot's of problems with an asterisk solution. I have
>         regular
>         > crash's and lock-ups, with downgrading and other stuff i got
>         it
>         > somewhat stable, but have nevertheless regular hickups. I am
>         desperate
>         > and want to get rid of asterisk and I hope that freeSwitch
>         will
>         > provide me with a more stable solution.
>         >
>         >
>         > Our Setup (really nothing special):
>         > * 1 Asterisk box, New IBM Hardware (3 month old), 2 HE rack
>         server, 3
>         > GIG of RAM, Xircom analog switch connected to mobile
>         stations for 4
>         > different providers, Digium 4port cards TP400<something>
>         > * 8 queues
>         > * ~60 agents (which logon, logoff, pause, unpause), not more
>         than 40
>         > concurrently online
>         > * ~ 7K - 9K calls (well, CDR entries) a day (not that much
>         for a bpx)
>         > * Music on Hold in the call-queues
>         > * No special announcement
>         > * Transfers between calls in queues and different agents as
>         well as
>         > non agents (i mention this because we have transfer related
>         chrashes
>         > in asterisk)
>         >
>         > The current Problems:
>         > * Lockups with different causes (ranging from calls not
>         terminated to
>         > heavy thread locking through the AMI interface)
>         > * Crashes and library aborts (pthread, libc, crashes related
>         to music
>         > on hold, app_queue, transfers)
>         >
>         > We used Asterisk: 1.4.23, 1.4.24, 1.4.26rc3, 1.4.26rc5,
>         1.4.26 and are
>         > now back to 1.4.21.2 (stock debian) as anything beyond that
>         is for
>         > whatever reason highly unstable for our szenario. Maybe we
>         should have
>         > been segmenting the box into one asterisk dedicted to
>         talking to the
>         > hardware, one especial for queue/sip handling, i do not
>         know. (all
>         > issues are well documented in issues.asterisk.org, but it
>         seems to be
>         > very, very difficult to get to the bottom of them as they
>         exist since
>         > 1.4.23 as it seems and are open until know and not fixable
>         since month.)
>         >
>         >
>         > Now, I really would appreciate some success-stories on how
>         you guys
>         > managed to get a stable pbx system with freeSWITCH in regard
>         of HA and
>         > scalability:
>         >
>         > * How to segment freeSWITCH? Or is it stable enough to
>         handle all in
>         > one for such a szenario as outlined above?
>         > * What would be the best strategie for High Availability /
>         Failover?
>         >       -> I read in the WIKI (featurelist) that Livemigration
>         of calls from
>         > one box to another should be possible?
>         >       -> I was thinking about using memcached for storing
>         all state
>         > information so another freeswitch box can takeover calls
>         from the
>         > first box if it dies, is this possible? If so, how?
>         >       -> Is there anotherway to somehow configure freeSWITCH
>         that in the
>         > event of a crash i do not loose the current established
>         calls?
>         >
>         > Basically I just want a stable PBX where I do not have to
>         fear every
>         > day it will core-dump or abort or Lock up. Is freeSWITCH
>         mature enough
>         > so i can sleep at night for at least 3 month without a
>         crash?
>         >
>         >
>         >
>         > Thank you for your Time and help in advance, and I am more
>         than
>         > willing to take all the information gathered here and create
>         a wiki
>         > page to help other people with the same questions/problems.
>         >
>         > best
>         > Ray
>         >
>         >
>         --
>         
>         David Knell, Director, 3C Limited
>         T: +44 20 3298 2000
>         E: dave at 3c.co.uk
>         W: http://www.3c.co.uk
>         
>         
>         
>         _______________________________________________
>         FreeSWITCH-users mailing list
>         FreeSWITCH-users at lists.freeswitch.org
>         http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>         UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>         http://www.freeswitch.org
>         
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk




From nagalenoj at gmail.com  Fri Aug 28 21:26:44 2009
From: nagalenoj at gmail.com (Nagalenoj)
Date: Fri, 28 Aug 2009 21:26:44 -0700 (PDT)
Subject: [Freeswitch-users] How can it be done with freeswitch?
In-Reply-To: <87f2f3b90908280907w4812a8dbr7a97385f9afdd376@mail.gmail.com>
References: <25184845.post@talk.nabble.com>
	<87f2f3b90908280907w4812a8dbr7a97385f9afdd376@mail.gmail.com>
Message-ID: <25199956.post@talk.nabble.com>


Thanks for your help. Is it the only possible way.?
Else, will it be done with the functions listed in the below url,
http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes

I was trying to do with these classes and functions.


mercutioviz wrote:
> 
> Nagalenoj,
> 
> This is definitely possible to do with FreeSWITCH. However, it is not
> exactly simple. You have some learning to do. My recommendation is to
> learn
> about event socket. Most likely you'll need to use outbound event socket.
> (Read about event socket and ESL, the event socket library, on the wiki.)
> 
> I also recommend that you look at some of the example scripts in
> src/libs/esl/perl/ which show some of the ways to handle calls with a
> script. You've already done a good job of breaking down the steps, now
> experiment with handling each of these steps with a script. We don't have
> extensive Perl script examples which means we would love to see your code
> when you have it all figured out! :)
> 
> -MC
> 
> On Thu, Aug 27, 2009 at 11:44 PM, Nagalenoj <nagalenoj at gmail.com> wrote:
> 
>>
>> Dear friends,
>>    I am using freeswitch for the last one month and trying to write perl
>> scripts to do some tasks.
>> I am trying a new thing, which is something that I don't how to do it and
>> I
>> don't whether it can be done with freeswitch.   my requirement is as
>> follows,
>>             A calls B[This extension has to execute a script].
>>             The script has to do the following,
>>              * Cut the call and callback to the number.
>>              * Get a phone number[DTMF] from A.
>>              * Dial the number(C) and bridge A and C.
>>              * When C cuts the call, the control has to comeback to
>> script.
>>              * Ask another number to A to dial out, get another phone
>> number[DTMF] from A.
>>              * Dial the number(D) and bridge A and D.
>>
>> Please, help me to do this. Is there any other documentation to read
>> about
>> the perl freeswitch functions.
>> --
>> View this message in context:
>> http://www.nabble.com/How-can-it-be-done-with-freeswitch--tp25184845p25184845.html
>> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 

-- 
View this message in context: http://www.nabble.com/How-can-it-be-done-with-freeswitch--tp25184845p25199956.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From kawarod at laposte.net  Fri Aug 28 22:43:24 2009
From: kawarod at laposte.net (rod)
Date: Sat, 29 Aug 2009 09:43:24 +0400
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
In-Reply-To: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>
References: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>
Message-ID: <4A98BFFC.9060003@laposte.net>

Ok,

I found what's happening. I probably did some change on the wiki to 
start reflect the new configuration, without having time enough to check 
the configuration. There is a mistake in your dialplan configuration. 
You should put this instead:

wrong line:

<action application="set" data="ROUTE_GW=${sip_redirect_contact_user_0}"/>

please correct with:
<action application="set" data="ROUTE_GW=${sip_redirect_contact_host_0}"/>


This should match PEER_01 in dialplan instead of trying matching 
France at PEER_01.

Let me know if this is right now.

rod

Hristo Benev a ?crit :
>  Hello Rod,
>
> I did the change.
>
> Here is extract of console:
> -----------------------------------------------------------------------------------------
> variable_continue_on_fail: [true]
> variable_sip_h_X-ROUTE: [LOOKUP]
> variable_export_vars: [sip_h_X-ROUTE]
> variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20]
> variable_sip_redirect_contact_0: [sip:France at PEER_01]
> variable_sip_redirected_to: [sip:France at PEER_01]
> variable_sip_redirect_contact_user_0: [France] <-------------------------------------------
> variable_sip_redirect_contact_host_0: [PEER_01]
> variable_sip_redirect_dialstring_0: [sofia/internal/sip:France at PEER_01]
> variable_sip_redirect_dialstring: [sofia/internal/sip:France at PEER_01]
> variable_proto_specific_hangup_cause: [sip:503]
> variable_sip_hangup_phrase: [DNS Error]
> variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE]
> variable_ROUTE_GW: [France]
> variable_AREA: [France]
> variable_current_application: [info]
> ------------------------------------------------------------------------------------------
> I have different value it is actually the description field as shown here:
>
> --------------------------------
>  /opt/kamailio/sbin/kamctl cr show
> cr carrier names
> +----+---------+
> | id | carrier |
> +----+---------+
> |  1 | default |
> +----+---------+
> cr domain names
> +----+---------+
> | id | domain  |
> +----+---------+
> |  1 | default |
> +----+---------+
> cr routes
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description |
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> |  1 |       1 |      1 | 1000        |     0 |    0 |    1 |     0 | PEER_01      |                |                | France      |
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> -----------------------------------------
>
> And here is what I have in kamailio:
> -------------------------------------------------------------------------------------
> ####### Routing Logic ########
>
>
> # main request routing logic
>
> route{
>
>         if (!mf_process_maxfwd_header("10")) {
>                 sl_send_reply("483","Too Many Hops");
>                 exit;
>         }
>
>         t_check_trans();
>
>         if ($rU==NULL) {
>                 # request with no Username in RURI
>                 sl_send_reply("484","Address Incomplete");
>                 exit;
>         }
>
>         # LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP
>         if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){
>                 if(!cr_route("default", "default", "$rU", "$rU", "call_id","$avp(s:route_desc)")){
>                      #xlog("ROUTING FAILED: no route found for $rU");
>                      sl_send_reply("604", "Unable to route this call");
>                      exit;
>                 } else {
>                      xlog("LOOKUP FOUND: $rd $avp(s:route_desc)");
>                      avp_pushto("$ru/username", "$avp(s:route_desc)");
>                      sl_send_reply("302", "$rd");
>                      exit;
>                 }
>         }
> }
> ----------------------------------------------------------------------------------
>
> Another question...
> In that part of FreeSwitch dialplan.xml
>
> -----------------------------------
>   <context name="ROUTING">
>
>     <extension name="PEER_01">
>       <condition field="${sip_h_X-ROUTE}" expression="PEER_01">
>         <action application="set" data="hangup_after_bridge=true"/>
>
> -----------------------------------
>
> X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the regex is true.
>
>
> As for thanks - for sure by default they are also for the developers of both apps.
> I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning so it will be nice to have something working to start with.
>
>
>
>  >-------- ?????????? ????? --------
>  >??:  rod 
>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >??: freeswitch-users at lists.freeswitch.org
>  >????????? ??: ?????, 2009, ?????? 28 09:54:00 EEST
>
>  >Hello,
>  >
>  >the trace seems good.
>  >If you check the answer from Kamailio, you'll see that Kamailio answers 
>  >with "302 PEER_01".
>  >
>  >As Michael Collins stated before, you can get the variable containing 
>  >"PEER_01", then this variable is stored in a custom variable.
>  >
>  >In your dialplan, may you please add:
>  >
>  >, just before the transfer line, eg:
>  >
>  > 
>  >         
>  >         
>  >         
>  >         
>  >         
>  >         
>  >	 
>  >         
>  >      
>  >
>  >
>  >Using application Info, you'll see on the console (or CDR) the list of 
>  >variables used for this call.  You should see the content of 
>  >"${sip_redirect_contact_user_0}" that should contain  the value 
>  >"PEER_01". Please check this and let me know.
>  >
>  >For the new configuration file, no problem for sharing. But as I wrote 
>  >on the wiki page, I worked on this setup cause LCR module was not 
>  >available when I start working on FS, nor I'm a good programmer to write 
>  >a server side HTTP script (used by xml_curl) that could scale to my needs.
>  >
>  >I enhanced a bit this configuration with support for fallback routing, 
>  >almost realtime graph (every minutes using www.cacti.net and some 
>  >functions in FS like limit_hash) of number of concurrent calls per AREA, 
>  >PEER...
>  >
>  >Thanks for "the good tutorial", but don't forget the dev team who did 
>  >this great product ;-)
>  >
>  >rod.
>  >
>  >
>  >Hristo Benev a ?crit :
>  >>  I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).
>  >>
>  >> If you need additional info I'll provide it.
>  >>
>  >> Here is trace:
>  >>
>  >> ngrep -d any -nn -i '1000' port 5062 -W byline
>  >> interface: any
>  >> filter: (ip or ip6) and ( port 5062 )
>  >> match: 1000
>  >> #
>  >> U 10.10.10.10:5090 -> 127.0.0.1:5062
>  >> INVITE sip:1000 at 127.0.0.1:5062 SIP/2.0.
>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
>  >> Max-Forwards: 69.
>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
>  >> To: .
>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
>  >> CSeq: 119574771 INVITE.
>  >> Contact: .
>  >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
>  >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
>  >> Supported: timer, precondition, path, replaces.
>  >> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
>  >> Content-Type: application/sdp.
>  >> Content-Disposition: session.
>  >> Content-Length: 429.
>  >> X-ROUTE: LOOKUP.
>  >> Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off.
>  >> .
>  >> v=0.
>  >> o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
>  >> s=FreeSWITCH.
>  >> c=IN IP4 10.10.10.10.
>  >> t=0 0.
>  >> m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
>  >> a=rtpmap:0 PCMU/8000.
>  >> a=rtpmap:115 G7221/32000.
>  >> a=fmtp:115 bitrate=48000.
>  >> a=rtpmap:107 G7221/16000.
>  >> a=fmtp:107 bitrate=32000.
>  >> a=rtpmap:9 G722/8000.
>  >> a=rtpmap:8 PCMA/8000.
>  >> a=rtpmap:3 GSM/8000.
>  >> a=rtpmap:101 telephone-event/8000.
>  >> a=fmtp:101 0-16.
>  >> a=rtpmap:13 CN/8000.
>  >> a=ptime:20.
>  >>
>  >> #
>  >> U 127.0.0.1:5062 -> 10.10.10.10:5090
>  >> SIP/2.0 302 PEER_01.
>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
>  >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
>  >> CSeq: 119574771 INVITE.
>  >> Contact: sip:France at PEER_01.
>  >> Server: Kamailio (1.5.2-notls (i386/linux)).
>  >> Content-Length: 0.
>  >> .
>  >>
>  >> #
>  >> U 10.10.10.10:5090 -> 127.0.0.1:5062
>  >> ACK sip:1000 at 127.0.0.1:5062 SIP/2.0.
>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
>  >> Max-Forwards: 69.
>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
>  >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
>  >> CSeq: 119574771 ACK.
>  >> Content-Length: 0.
>  >> .
>  >>
>  >>
>  >>
>  >>
>  >>
>  >>  >-------- ?????????? ????? --------
>  >>  >??:  Hristo Benev 
>  >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  >??: freeswitch-users at lists.freeswitch.org
>  >>  >????????? ??: ?????????, 2009, ?????? 27 14:58:53 EEST
>  >>
>  >>  > 
>  >>  >Bojnour,
>  >>  >
>  >>  >I'll send a trace ASAP.
>  >>  >
>  >>  >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
>  >>  >
>  >>  >I'm really interested to see the updates of the project.
>  >>  >
>  >>  >Thank you for the good tutorial.
>  >>  >
>  >>  >Hristo
>  >>  >
>  >>  >
>  >>  > >-------- ?????????? ????? --------
>  >>  > >??:  rod 
>  >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  > >??: freeswitch-users at lists.freeswitch.org
>  >>  > >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST
>  >>  >
>  >>  > >Hi Hristo,
>  >>  > >
>  >>  > >I'm the author of this setup and wiki page. I did a lot of modifications 
>  >>  > >on this setup (alternative routing if failure essentially) but don't 
>  >>  > >have too much time to update the wiki.
>  >>  > >
>  >>  > >May you please send me an ngrep trace when you call 1000:
>  >>  > >
>  >>  > >ngrep -d any -nn -i '1000' port 5060 -W byline 
>  >>  > >
>  >>  > >I will check what's happening.
>  >>  > >Do you have an entry for 1000 in your mysql database ?
>  >>  > >
>  >>  > >regards,
>  >>  > >rod
>  >>  > >
>  >>  > >Hristo Benev a ?crit :
>  >>  > >>  
>  >>  > >> It seems that the problem is on kamailio configuration.
>  >>  > >> Will ask on their list.
>  >>  > >>
>  >>  > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
>  >>  > >>
>  >>  > >> I have it setup as friend in asterisk, but still ???
>  >>  > >>
>  >>  > >> Any ideas?
>  >>  > >>
>  >>  > >> Thanks,
>  >>  > >>
>  >>  > >>
>  >>  > >>  >-------- ?????????? ????? --------
>  >>  > >>  >??:  Hristo Benev 
>  >>  > >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  > >>  >??: freeswitch-users at lists.freeswitch.org
>  >>  > >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
>  >>  > >>
>  >>  > >>  > I think that the problem is here:
>  >>  > >>  >-------------------------
>  >>  > >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
>  >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
>  >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
>  >>  > >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
>  >>  > >>  >--------------------------
>  >>  > >>  >
>  >>  > >>  >Actually Regex FAIL
>  >>  > >>  >
>  >>  > >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
>  >>  > >>  >Here is my  default.xml:
>  >>  > >>  >----------------
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >  
>  >>  > >>  >
>  >>  > >>  >     
>  >>  > >>  >      
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >      
>  >>  > >>  >    
>  >>  > >>  >
>  >>  > >>  >   
>  >>  > >>  >
>  >>  > >>  >  
>  >>  > >>  >
>  >>  > >>  >    
>  >>  > >>  >      
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >      
>  >>  > >>  >    
>  >>  > >>  >
>  >>  > >>  >  
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >--------------------------
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  > >-------- ?????????? ????? --------
>  >>  > >>  > >??:  Brian West 
>  >>  > >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  > >>  > >??: freeswitch-users at lists.freeswitch.org
>  >>  > >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
>  >>  > >>  >
>  >>  > >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
>  >>  > >>  >You have to process all 302's in the dialplan. 
>  >>  > >>  >Set this on your sofia profile
>  >>  > >>  >You can set these variables sip_redirect_profile,
>  >>  > >>  >sip_redirect_context,
>  >>  > >>  >sip_redirect_dialplan,
>  >>  > >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
>  >>  > >>  >sip_redirected_to,
>  >>  > >>  >sip_redirect_contact_user_%d,
>  >>  > >>  >sip_redirect_contact_host_%d,
>  >>  > >>  >sip_redirect_contact_params_%d,
>  >>  > >>  >sip_redirect_dialstring_%d,
>  >>  > >>  >sip_redirect_dialstring,
>  >>  > >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
>  >>  > >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
>  >>  > >>  > >
>  >>  > >>  >
>  >>  > >>  >_______________________________________________
>  >>  > >>  >FreeSWITCH-users mailing list
>  >>  > >>  >FreeSWITCH-users at lists.freeswitch.org
>  >>  > >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  > >>  >http://www.freeswitch.org
>  >>  > >>  >
>  >>  > >>
>  >>  > >> _______________________________________________
>  >>  > >> FreeSWITCH-users mailing list
>  >>  > >> FreeSWITCH-users at lists.freeswitch.org
>  >>  > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  > >> http://www.freeswitch.org
>  >>  > >>   
>  >>  > >
>  >>  > >_______________________________________________
>  >>  > >FreeSWITCH-users mailing list
>  >>  > >FreeSWITCH-users at lists.freeswitch.org
>  >>  > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  > >http://www.freeswitch.org
>  >>  > >
>  >>  >
>  >>  >_______________________________________________
>  >>  >FreeSWITCH-users mailing list
>  >>  >FreeSWITCH-users at lists.freeswitch.org
>  >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  >http://www.freeswitch.org
>  >>  >
>  >>
>  >> _______________________________________________
>  >> FreeSWITCH-users mailing list
>  >> FreeSWITCH-users at lists.freeswitch.org
>  >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >> http://www.freeswitch.org
>  >>   
>  >
>  >_______________________________________________
>  >FreeSWITCH-users mailing list
>  >FreeSWITCH-users at lists.freeswitch.org
>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >http://www.freeswitch.org
>  >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From kawarod at laposte.net  Fri Aug 28 22:56:22 2009
From: kawarod at laposte.net (rod)
Date: Sat, 29 Aug 2009 09:56:22 +0400
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
In-Reply-To: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>
References: <324473687.13970.1251477757337.JavaMail.apache@mail23.abv.bg>
Message-ID: <4A98C306.1060801@laposte.net>

you are right for the regex. This is part of an old setup, correct with:

<extension name="PEER_01">
      <condition field="${ROUTE_GW}" expression="PEER_01">
	<action application="set" data="hangup_after_bridge=true"/>



Hristo Benev a ?crit :
>  Hello Rod,
>
> I did the change.
>
> Here is extract of console:
> -----------------------------------------------------------------------------------------
> variable_continue_on_fail: [true]
> variable_sip_h_X-ROUTE: [LOOKUP]
> variable_export_vars: [sip_h_X-ROUTE]
> variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20]
> variable_sip_redirect_contact_0: [sip:France at PEER_01]
> variable_sip_redirected_to: [sip:France at PEER_01]
> variable_sip_redirect_contact_user_0: [France] <-------------------------------------------
> variable_sip_redirect_contact_host_0: [PEER_01]
> variable_sip_redirect_dialstring_0: [sofia/internal/sip:France at PEER_01]
> variable_sip_redirect_dialstring: [sofia/internal/sip:France at PEER_01]
> variable_proto_specific_hangup_cause: [sip:503]
> variable_sip_hangup_phrase: [DNS Error]
> variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE]
> variable_ROUTE_GW: [France]
> variable_AREA: [France]
> variable_current_application: [info]
> ------------------------------------------------------------------------------------------
> I have different value it is actually the description field as shown here:
>
> --------------------------------
>  /opt/kamailio/sbin/kamctl cr show
> cr carrier names
> +----+---------+
> | id | carrier |
> +----+---------+
> |  1 | default |
> +----+---------+
> cr domain names
> +----+---------+
> | id | domain  |
> +----+---------+
> |  1 | default |
> +----+---------+
> cr routes
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description |
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> |  1 |       1 |      1 | 1000        |     0 |    0 |    1 |     0 | PEER_01      |                |                | France      |
> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
> -----------------------------------------
>
> And here is what I have in kamailio:
> -------------------------------------------------------------------------------------
> ####### Routing Logic ########
>
>
> # main request routing logic
>
> route{
>
>         if (!mf_process_maxfwd_header("10")) {
>                 sl_send_reply("483","Too Many Hops");
>                 exit;
>         }
>
>         t_check_trans();
>
>         if ($rU==NULL) {
>                 # request with no Username in RURI
>                 sl_send_reply("484","Address Incomplete");
>                 exit;
>         }
>
>         # LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP
>         if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){
>                 if(!cr_route("default", "default", "$rU", "$rU", "call_id","$avp(s:route_desc)")){
>                      #xlog("ROUTING FAILED: no route found for $rU");
>                      sl_send_reply("604", "Unable to route this call");
>                      exit;
>                 } else {
>                      xlog("LOOKUP FOUND: $rd $avp(s:route_desc)");
>                      avp_pushto("$ru/username", "$avp(s:route_desc)");
>                      sl_send_reply("302", "$rd");
>                      exit;
>                 }
>         }
> }
> ----------------------------------------------------------------------------------
>
> Another question...
> In that part of FreeSwitch dialplan.xml
>
> -----------------------------------
>   <context name="ROUTING">
>
>     <extension name="PEER_01">
>       <condition field="${sip_h_X-ROUTE}" expression="PEER_01">
>         <action application="set" data="hangup_after_bridge=true"/>
>
> -----------------------------------
>
> X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the regex is true.
>
>
> As for thanks - for sure by default they are also for the developers of both apps.
> I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning so it will be nice to have something working to start with.
>
>
>
>  >-------- ?????????? ????? --------
>  >??:  rod 
>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >??: freeswitch-users at lists.freeswitch.org
>  >????????? ??: ?????, 2009, ?????? 28 09:54:00 EEST
>
>  >Hello,
>  >
>  >the trace seems good.
>  >If you check the answer from Kamailio, you'll see that Kamailio answers 
>  >with "302 PEER_01".
>  >
>  >As Michael Collins stated before, you can get the variable containing 
>  >"PEER_01", then this variable is stored in a custom variable.
>  >
>  >In your dialplan, may you please add:
>  >
>  >, just before the transfer line, eg:
>  >
>  > 
>  >         
>  >         
>  >         
>  >         
>  >         
>  >         
>  >	 
>  >         
>  >      
>  >
>  >
>  >Using application Info, you'll see on the console (or CDR) the list of 
>  >variables used for this call.  You should see the content of 
>  >"${sip_redirect_contact_user_0}" that should contain  the value 
>  >"PEER_01". Please check this and let me know.
>  >
>  >For the new configuration file, no problem for sharing. But as I wrote 
>  >on the wiki page, I worked on this setup cause LCR module was not 
>  >available when I start working on FS, nor I'm a good programmer to write 
>  >a server side HTTP script (used by xml_curl) that could scale to my needs.
>  >
>  >I enhanced a bit this configuration with support for fallback routing, 
>  >almost realtime graph (every minutes using www.cacti.net and some 
>  >functions in FS like limit_hash) of number of concurrent calls per AREA, 
>  >PEER...
>  >
>  >Thanks for "the good tutorial", but don't forget the dev team who did 
>  >this great product ;-)
>  >
>  >rod.
>  >
>  >
>  >Hristo Benev a ?crit :
>  >>  I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).
>  >>
>  >> If you need additional info I'll provide it.
>  >>
>  >> Here is trace:
>  >>
>  >> ngrep -d any -nn -i '1000' port 5062 -W byline
>  >> interface: any
>  >> filter: (ip or ip6) and ( port 5062 )
>  >> match: 1000
>  >> #
>  >> U 10.10.10.10:5090 -> 127.0.0.1:5062
>  >> INVITE sip:1000 at 127.0.0.1:5062 SIP/2.0.
>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
>  >> Max-Forwards: 69.
>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
>  >> To: .
>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
>  >> CSeq: 119574771 INVITE.
>  >> Contact: .
>  >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
>  >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
>  >> Supported: timer, precondition, path, replaces.
>  >> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
>  >> Content-Type: application/sdp.
>  >> Content-Disposition: session.
>  >> Content-Length: 429.
>  >> X-ROUTE: LOOKUP.
>  >> Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off.
>  >> .
>  >> v=0.
>  >> o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
>  >> s=FreeSWITCH.
>  >> c=IN IP4 10.10.10.10.
>  >> t=0 0.
>  >> m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
>  >> a=rtpmap:0 PCMU/8000.
>  >> a=rtpmap:115 G7221/32000.
>  >> a=fmtp:115 bitrate=48000.
>  >> a=rtpmap:107 G7221/16000.
>  >> a=fmtp:107 bitrate=32000.
>  >> a=rtpmap:9 G722/8000.
>  >> a=rtpmap:8 PCMA/8000.
>  >> a=rtpmap:3 GSM/8000.
>  >> a=rtpmap:101 telephone-event/8000.
>  >> a=fmtp:101 0-16.
>  >> a=rtpmap:13 CN/8000.
>  >> a=ptime:20.
>  >>
>  >> #
>  >> U 127.0.0.1:5062 -> 10.10.10.10:5090
>  >> SIP/2.0 302 PEER_01.
>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
>  >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
>  >> CSeq: 119574771 INVITE.
>  >> Contact: sip:France at PEER_01.
>  >> Server: Kamailio (1.5.2-notls (i386/linux)).
>  >> Content-Length: 0.
>  >> .
>  >>
>  >> #
>  >> U 10.10.10.10:5090 -> 127.0.0.1:5062
>  >> ACK sip:1000 at 127.0.0.1:5062 SIP/2.0.
>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
>  >> Max-Forwards: 69.
>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
>  >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
>  >> CSeq: 119574771 ACK.
>  >> Content-Length: 0.
>  >> .
>  >>
>  >>
>  >>
>  >>
>  >>
>  >>  >-------- ?????????? ????? --------
>  >>  >??:  Hristo Benev 
>  >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  >??: freeswitch-users at lists.freeswitch.org
>  >>  >????????? ??: ?????????, 2009, ?????? 27 14:58:53 EEST
>  >>
>  >>  > 
>  >>  >Bojnour,
>  >>  >
>  >>  >I'll send a trace ASAP.
>  >>  >
>  >>  >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
>  >>  >
>  >>  >I'm really interested to see the updates of the project.
>  >>  >
>  >>  >Thank you for the good tutorial.
>  >>  >
>  >>  >Hristo
>  >>  >
>  >>  >
>  >>  > >-------- ?????????? ????? --------
>  >>  > >??:  rod 
>  >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  > >??: freeswitch-users at lists.freeswitch.org
>  >>  > >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST
>  >>  >
>  >>  > >Hi Hristo,
>  >>  > >
>  >>  > >I'm the author of this setup and wiki page. I did a lot of modifications 
>  >>  > >on this setup (alternative routing if failure essentially) but don't 
>  >>  > >have too much time to update the wiki.
>  >>  > >
>  >>  > >May you please send me an ngrep trace when you call 1000:
>  >>  > >
>  >>  > >ngrep -d any -nn -i '1000' port 5060 -W byline 
>  >>  > >
>  >>  > >I will check what's happening.
>  >>  > >Do you have an entry for 1000 in your mysql database ?
>  >>  > >
>  >>  > >regards,
>  >>  > >rod
>  >>  > >
>  >>  > >Hristo Benev a ?crit :
>  >>  > >>  
>  >>  > >> It seems that the problem is on kamailio configuration.
>  >>  > >> Will ask on their list.
>  >>  > >>
>  >>  > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
>  >>  > >>
>  >>  > >> I have it setup as friend in asterisk, but still ???
>  >>  > >>
>  >>  > >> Any ideas?
>  >>  > >>
>  >>  > >> Thanks,
>  >>  > >>
>  >>  > >>
>  >>  > >>  >-------- ?????????? ????? --------
>  >>  > >>  >??:  Hristo Benev 
>  >>  > >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  > >>  >??: freeswitch-users at lists.freeswitch.org
>  >>  > >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
>  >>  > >>
>  >>  > >>  > I think that the problem is here:
>  >>  > >>  >-------------------------
>  >>  > >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
>  >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
>  >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
>  >>  > >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
>  >>  > >>  >--------------------------
>  >>  > >>  >
>  >>  > >>  >Actually Regex FAIL
>  >>  > >>  >
>  >>  > >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
>  >>  > >>  >Here is my  default.xml:
>  >>  > >>  >----------------
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >  
>  >>  > >>  >
>  >>  > >>  >     
>  >>  > >>  >      
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >         
>  >>  > >>  >      
>  >>  > >>  >    
>  >>  > >>  >
>  >>  > >>  >   
>  >>  > >>  >
>  >>  > >>  >  
>  >>  > >>  >
>  >>  > >>  >    
>  >>  > >>  >      
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >        
>  >>  > >>  >      
>  >>  > >>  >    
>  >>  > >>  >
>  >>  > >>  >  
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >--------------------------
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  >
>  >>  > >>  > >-------- ?????????? ????? --------
>  >>  > >>  > >??:  Brian West 
>  >>  > >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
>  >>  > >>  > >??: freeswitch-users at lists.freeswitch.org
>  >>  > >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
>  >>  > >>  >
>  >>  > >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
>  >>  > >>  >You have to process all 302's in the dialplan. 
>  >>  > >>  >Set this on your sofia profile
>  >>  > >>  >You can set these variables sip_redirect_profile,
>  >>  > >>  >sip_redirect_context,
>  >>  > >>  >sip_redirect_dialplan,
>  >>  > >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
>  >>  > >>  >sip_redirected_to,
>  >>  > >>  >sip_redirect_contact_user_%d,
>  >>  > >>  >sip_redirect_contact_host_%d,
>  >>  > >>  >sip_redirect_contact_params_%d,
>  >>  > >>  >sip_redirect_dialstring_%d,
>  >>  > >>  >sip_redirect_dialstring,
>  >>  > >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
>  >>  > >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
>  >>  > >>  > >
>  >>  > >>  >
>  >>  > >>  >_______________________________________________
>  >>  > >>  >FreeSWITCH-users mailing list
>  >>  > >>  >FreeSWITCH-users at lists.freeswitch.org
>  >>  > >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  > >>  >http://www.freeswitch.org
>  >>  > >>  >
>  >>  > >>
>  >>  > >> _______________________________________________
>  >>  > >> FreeSWITCH-users mailing list
>  >>  > >> FreeSWITCH-users at lists.freeswitch.org
>  >>  > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  > >> http://www.freeswitch.org
>  >>  > >>   
>  >>  > >
>  >>  > >_______________________________________________
>  >>  > >FreeSWITCH-users mailing list
>  >>  > >FreeSWITCH-users at lists.freeswitch.org
>  >>  > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  > >http://www.freeswitch.org
>  >>  > >
>  >>  >
>  >>  >_______________________________________________
>  >>  >FreeSWITCH-users mailing list
>  >>  >FreeSWITCH-users at lists.freeswitch.org
>  >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >>  >http://www.freeswitch.org
>  >>  >
>  >>
>  >> _______________________________________________
>  >> FreeSWITCH-users mailing list
>  >> FreeSWITCH-users at lists.freeswitch.org
>  >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >> http://www.freeswitch.org
>  >>   
>  >
>  >_______________________________________________
>  >FreeSWITCH-users mailing list
>  >FreeSWITCH-users at lists.freeswitch.org
>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>  >http://www.freeswitch.org
>  >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>   



From thangappan143 at gmail.com  Fri Aug 28 23:45:28 2009
From: thangappan143 at gmail.com (Thangappan.M)
Date: Sat, 29 Aug 2009 12:15:28 +0530
Subject: [Freeswitch-users] FreeSWITCH environmental variables
Message-ID: <7aa29e790908282345r511dcc82kd767063b93d370dd@mail.gmail.com>

Dear all,

        In the case of asterisk PBX. I can get all the information about the
call from the environmental variable itself like Caller id ,called id,
channel no. But in freeswitch where all the environmental variables are
resides? How can I access all the variables?

Please Help me?

-- 
Regards,
Thangappan.M
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From jaybinks at gmail.com  Sat Aug 29 01:17:31 2009
From: jaybinks at gmail.com (Jay Binks)
Date: Sat, 29 Aug 2009 18:17:31 +1000
Subject: [Freeswitch-users] FreeSWITCH environmental variables
In-Reply-To: <7aa29e790908282345r511dcc82kd767063b93d370dd@mail.gmail.com>
References: <7aa29e790908282345r511dcc82kd767063b93d370dd@mail.gmail.com>
Message-ID: <82795EF6-1698-4DA5-A3F3-23A6CF47CDAC@gmail.com>

Check out INFO
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info

Throw that in your dialplan the look at your logs...  You should find  
what your after..

Jay



On 29/08/2009, at 16:45, "Thangappan.M" <thangappan143 at gmail.com> wrote:

> Dear all,
>
>         In the case of asterisk PBX. I can get all the information  
> about the call from the environmental variable itself like Caller  
> id ,called id, channel no. But in freeswitch where all the  
> environmental variables are resides? How can I access all the  
> variables?
>
> Please Help me?
>
> -- 
> Regards,
> Thangappan.M
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From m.krivushin at imarto.net  Sat Aug 29 01:21:16 2009
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Sat, 29 Aug 2009 15:21:16 +0700
Subject: [Freeswitch-users] FreeSWITCH environmental variables
In-Reply-To: <7aa29e790908282345r511dcc82kd767063b93d370dd@mail.gmail.com>
References: <7aa29e790908282345r511dcc82kd767063b93d370dd@mail.gmail.com>
Message-ID: <5be734a50908290121ya04c76n2a0bb3c50083edda@mail.gmail.com>

Try to read docuemntation first:

http://wiki.freeswitch.org/wiki/Channel_Variables

2009/8/29 Thangappan.M <thangappan143 at gmail.com>

> Dear all,
>
>         In the case of asterisk PBX. I can get all the information about
> the call from the environmental variable itself like Caller id ,called id,
> channel no. But in freeswitch where all the environmental variables are
> resides? How can I access all the variables?
>
> Please Help me?
>
> --
> Regards,
> Thangappan.M
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From paveraware at hotmail.com  Sat Aug 29 02:28:23 2009
From: paveraware at hotmail.com (Christensen Tom)
Date: Sat, 29 Aug 2009 09:28:23 +0000
Subject: [Freeswitch-users] Success! and an issue.
Message-ID: <BAY120-W6EC265B637176ED41128BDFF40@phx.gbl>


well, just made my first extension to extension call using freeswitch, so that was pretty painless.  Huge fan of the config layout at least (having the extensions defined each in their own file is going to be so much nicer than asterisks huge jumble).

However, I did run into one issue, not sure if its somehow related to the phones I'm using (softphone of unknown quality (weephone) on iphones)... I'm using the default config, when I dial 4000 for voicemail, it works, however at apparently random times during the menu navigation I get call waiting beeps and I have an incoming call from 4000.  If I decline or leave this call unanswered, my call into voicemail gets disconnected.  If I "accept" this call, it apparently disconnects immediately (IE I only have 1 active call), however I can then continue navigating the menu, etc... Very strange to me, I will be setting up a couple of polycom 501s on this system tomorrow and will report if I see similar issues.
Anyway, so far I'm pleased with freeswitch, I've got a list of features I need to support before I can deploy it, so I'll be bugging people (sorry), but so far thanks!
-Tom


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From steve.kurzeja at gmail.com  Sat Aug 29 02:58:37 2009
From: steve.kurzeja at gmail.com (Steve Kurzeja)
Date: Sat, 29 Aug 2009 21:58:37 +1200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
Message-ID: <5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>

On Sat, Aug 29, 2009 at 2:34 PM, Diego Viola <diego.viola at gmail.com> wrote:

> Yes, FreeSWITCH is a system that you can trust 100%. I have switched my
> Asterisk servers to FreeSWITCH and have peace now.
>
> If I were you I would get rid of Asterisk and use FreeSWITCH, FS will
> handle all what you want very well.
>
> And I agree with David, fail-over is kinda irrelevant since the FS doesn't
> crash like Asterisk does.
>
>
>
You still have hardware failures and fail-over is also useful for hit-less
maintenance on boxes.

I'd be interested to know how Brian West was approaching his live migration
work.

Steve
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From rs at runsolutions.com  Sat Aug 29 06:17:11 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Sat, 29 Aug 2009 15:17:11 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
Message-ID: <71FD2763-32C5-49CA-8E4D-7C0E301D2187@runsolutions.com>

Oh yeah, that would be so helpfull for my situation, as my client  
*demands* now a solution where he can press a big red button and all  
fails over to another box. Hi es totally scared because of the Lockups  
in Asterisk which under specific situations including AMI, Automated  
Call Setup, and murphy led to a lockup of the entire machine, no  
console was working anymore, only cold-reset could do it.

So, IF there is the possibility for life-takeover, / failover etc. I  
would love to here how has been done.

I am very experienced with openvz and use for about two years now only  
openvz virtualization servers for anything because of live-migration  
etc. But as I am new in this company we could not adopt this until now.

So Please Ken, if you can, describe what need's to be done to get a  
failover / takeover working (an outline would be enough)

Thanks in Advance

-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 29, 2009, at 11:58 AM, Steve Kurzeja wrote:

> On Sat, Aug 29, 2009 at 2:34 PM, Diego Viola <diego.viola at gmail.com>  
> wrote:
> Yes, FreeSWITCH is a system that you can trust 100%. I have switched  
> my Asterisk servers to FreeSWITCH and have peace now.
>
> If I were you I would get rid of Asterisk and use FreeSWITCH, FS  
> will handle all what you want very well.
>
> And I agree with David, fail-over is kinda irrelevant since the FS  
> doesn't crash like Asterisk does.
>
>
>
> You still have hardware failures and fail-over is also useful for  
> hit-less maintenance on boxes.
>
> I'd be interested to know how Brian West was approaching his live  
> migration work.
>
> Steve
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From tomabroad at gmail.com  Sat Aug 29 06:46:27 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 09:46:27 -0400
Subject: [Freeswitch-users] newbie questions
Message-ID: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>

hi, totally new to freeswitch, but not to asterisk. just installed it, made
samples but do not have 1000-1019 as extensions. so i thought i create:
<include>
  <user id="1000" mailbox="1000">
    <params>
      <param name="password" value="1234"/>
      <param name="vm-password" value="1000"/>
    </params>
    <variables>
      <variable name="accountcode" value="1000"/>
      <variable name="user_context" value="default"/>
      <variable name="effective_caller_id_name" value="Extension 1000"/>
      <variable name="effective_caller_id_number" value="1000"/>
    </variables>
  </user>
</include>
 and save it as 1000.xml in the internal-sip folder. cli tells me now:
2009-08-29 09:43:55.858101 [WARNING] sofia_reg.c:1771 Can't find user [
1000 at 192.168.1.4]
You must define a domain called '192.168.1.4' in your directory and add a
user with the id="1000" attribute
and you must configure your device to use the proper domain in it's
authentication credentials.

what do i have to do now to get it
a) internally running
b) talk to an external cllient?


thx
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From rupa at rupa.com  Sat Aug 29 07:07:15 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Sat, 29 Aug 2009 09:07:15 -0500
Subject: [Freeswitch-users] mod_limit and memcache
In-Reply-To: <b39273530908270742i1e018ed2x953203e4eee7bb2f@mail.gmail.com>
References: <a44688a0908270649y6cde3c26pb739c7d144e1b86b@mail.gmail.com>
	<b39273530908270742i1e018ed2x953203e4eee7bb2f@mail.gmail.com>
Message-ID: <b39273530908290707y625223y6a993d54cb8ce88b@mail.gmail.com>

Ok, it was a bit more work than expected, but I have a first cut at
mod_limit using memcache checked in.  Test if you can.  I've not put
docs on the wiki 'cause some of the implementation / setup details are
going to change with regards to mod_memcache.   Oh, and mod_limit's
api may be changing to make it more pluggable.

Basically,

Setup mod_memcache (on wiki)

It uses the same user facing api as limit_hash and friends except it
is limit_memcache and limit_memcache_usage.  so follow mod_limit's
docs with suitable substitutions.


On Thu, Aug 27, 2009 at 9:42 AM, Rupa Schomaker<rupa at rupa.com> wrote:
> limit and memcache haven't been introduced to each other yet -- it is
> on my (semi-long) list of things to do.
>
> If you want it you can:
>
> 1) do it yourself and submit the patches
> 2) open a jira and hope someone does it
> 3) open a jira + bounty and someone will probably do it
>
> It will get done eventually, just hasn't been a itch for ME to scratch yet.
>
>
> To do it:
>
> 1) I need to make it possible to call inc/dec methods of mod_memcache
> to support an expiration time.
>
> 2) mod_limit.c - use the hash limit as a guide
>
> Initial pitfalls:
>
> hash limits concurrent access/modification of the hash and by
> implication limit_hash_item_t (hash data) by using a mutex. ?We can't
> mutex across FS instances.
>
> So perhaps split up limit_has_item_t and spread it across multiple
> keys. ?So instead of one key marked as realm_id, we could have
> realm_id_total_usage realm_id_rate_usage and realm_id_last_check.
>
> This does mean that rate_usage and total_usage can inc/dec independent
> of each other, but I think the logic will still be "ok" *IF* we
> remember to decrement earlier incremented items in the event a later
> item is failed. ? (so, say we increment rate but fail on total we need
> to remember to decrement rate so that we have no net effect on the
> counters)
>
> Alternatively, we could use CAS support and pull the limit_hash_item_t
> item from memcache, twiddle it and then try to put it back only if the
> check info is the same (no one else has changed the entry). ?If the
> entry has changed, pull the new version, do the limit logic, and try
> again. ?Loop that a few times until you succeed or give up. ?Problem
> is that CAS needs to be explicitly turned on in memcache (some distros
> compile with it off), is relatively new in memcache (hint: may have
> issues) and has some performance/memory downsides though by how much
> I'm not sure.
>
> Thoughts?
>
> On Thu, Aug 27, 2009 at 8:49 AM, Woody Dickson<woodydickson at gmail.com> wrote:
>> Hello,
>>
>> I read something that talks about using memcache for mod_limit before.?? Is
>> it something that is available now?
>>
>> If I have multiple instances of freeswitch that need to share the same limit
>> status, it there any existing solution?
>>
>> If no existing solution is available, what is the best way to go about
>> modifying mod_limit to accomplish limiting for multiple freeswitch servers
>> together?
>>
>> Thanks,
>> Woody
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
>
> --
> -Rupa
>



-- 
-Rupa



From max.bridgewater at gmail.com  Sat Aug 29 07:07:57 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sat, 29 Aug 2009 10:07:57 -0400
Subject: [Freeswitch-users] Disabling Core Dumping
Message-ID: <c55d46950908290707u16fc5d40m3c23f9a9ca450738@mail.gmail.com>

Hi,

Whenever i stop freeswich, it creates a core dump. How can i disable that?

thanks,
Max.
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From rs at runsolutions.com  Sat Aug 29 07:33:15 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Sat, 29 Aug 2009 16:33:15 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <71FD2763-32C5-49CA-8E4D-7C0E301D2187@runsolutions.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
	<71FD2763-32C5-49CA-8E4D-7C0E301D2187@runsolutions.com>
Message-ID: <136791EF-26D2-4DC7-91F6-2D468122F06C@runsolutions.com>

Thinking about it, maybe we can create a solution, if some of us work  
together:

My strength are in virtualization, linux, development, databases,  
integration, etc.
What I do not now much about is how SIP (and everything else for that  
matter in the Voice world) works under the hood, and how it's  
implemented in FS.

I know that the state information for a call has to be stored and  
retrieved somewhere and somehow, only I do not know that part. What I  
know is that it hast to be do-able to store all the stream information  
(ip's, port's, current state's, etc.) in a very fast database (e.g. my  
idea would be memcached) so another FS could just take this  
information and take over the call, maybe you loose a second of voice,  
maybe you loose the recorded call file or a part of it, but that  
should be it. (SipFoundry has a boxed opensource PBX, which, of course  
is not flexible like FreeSWITCH or Asterisk, but has Call Live  
Migration and Call Live Failover integrated!).

What I want is for my company to be able to sell a 99.99 uptime PBX  
(we do mostly call-center related stuff), which can scale well, and  
can grow with the company without lot's of hassles, my Dream would be:

To begin with:

One Hardware Node with the essential hardware (digium cards for  
example).
On this node are OpenVZ virtualized containers:
[VirtCnt1: FS which only talks to the Hardware and forwards  
everything] = Could be replaced with hardware media gateway, etc.
[VirtCnt2: FS which handles the PBX] \___ Loadbalanced, with odbc or  
xml, Failover, Livetakeover
[VirtCnt3: FS which handles the PBX] /
[VirtCnt4: Database for state information] (maybe something as  
resource-friendly as memcached? ressource heavvy database?)

With this we can achieve all this:

Problem with VirtCnt2 (e.g. crash, lock, ...)
* VirtCnt3 can take over.
-> You are free without stress to investigate the problem, you can  
debug and analyze whyle the machine is still running
-> you can also create a machine-state-dump of the virtual container,  
dump the container as well, copy the data to your lab and restore the  
machine up the state which it was running with the problem, so you can  
liveinvestigate it in the lab (some prerequirements given, but easy  
doable)
-> just think about the possibility of better bugreports because  
someone can take the time to read out all the data with GDB to  
investigate the proper cause of a machine Lock!

You want to upgrade to a new FreeSWITCH version?
* Take VirtCnt2 out of the LoadBalancing Scheme,
* Stop it, Clone it,
* Upgrade FreeSWITCH in the cloned Container
* Start the cloned container
* if there's something wrong, stop it and restart the original VirtCnt2
-> No problem at all, you can Test on the Live Hardware, with part of  
the Live users (maybe a low-volume queue) to be sure everything works  
out fine before you activate the full loadbalance

Server on it's own can't handle the load
* Buy new machine
* Setup Hardware Node
* Livemigrate VirtCnt3 (no downtime)

Now the first Server with the VrtCnt1 and VirtCnt2 as well has to much  
load
* Buy new machine
* Setup Hardware Node
* Livemigrate VirtCnt2 (no downtime)
-> Now you have a 3 server solution (1 mediaprox, 2 loadbalanced /  
failover PBXes) out of the first box you bought, without headaches,  
because the system was built for it from the beginning!

The Database drains to much?
* Buy new machine
* Setup Hardware Node
* Livemigrate database VirtCnt4 (no downtime)

You want to upgrade Hardware/Kernel in Hardware node 1?
* Livemigrate VirtCnt2 to a hotstandby machine, or to the other PBX  
machine, upgrade the hardware, Re-Livemigrate the containers. (no  
downtime)
* OR just break the loadbalancing, wait until all current calls are  
teared down correctly, upgrade machine, reenable the loadbalancer

You want an exact copy of the first server for Hardware HA?
* Buy new machine
* Setup Hardware node
* Buy hardware PRI switchover box
* Clone VirtCnt1 - VirtCnt4 to the new machine
* Make basic failover configuration


-> the sky's the limit, as the saying goes ...


So, I can do all the openvz stuff and the integration with database /  
memcached / heartbeat / whatever is needed here, someone there to be  
willing to work with me on this on the FreeSWITCH side? or at least  
provide me with the necessary information about what's needed / how to  
talk / what states from FreeSWITCH?

I know this seems very ambitious but if this could be made in a rather  
relativly easy to setup package, with good documentation, it would be  
a boost for FreeSWITCH, i am sure, because after all this is what  
everyone is grown accustomed to from good old phone companys and the  
good old pbx's: carrier grade uptimes ...

Thanks for everyone reading up until here,
all the best,

Ray



-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 29, 2009, at 3:17 PM, Raimund Sacherer wrote:

> Oh yeah, that would be so helpfull for my situation, as my client  
> *demands* now a solution where he can press a big red button and all  
> fails over to another box. Hi es totally scared because of the  
> Lockups in Asterisk which under specific situations including AMI,  
> Automated Call Setup, and murphy led to a lockup of the entire  
> machine, no console was working anymore, only cold-reset could do it.
>
> So, IF there is the possibility for life-takeover, / failover etc. I  
> would love to here how has been done.
>
> I am very experienced with openvz and use for about two years now  
> only openvz virtualization servers for anything because of live- 
> migration etc. But as I am new in this company we could not adopt  
> this until now.
>
> So Please Ken, if you can, describe what need's to be done to get a  
> failover / takeover working (an outline would be enough)
>
> Thanks in Advance
>
> -- 
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
> On Aug 29, 2009, at 11:58 AM, Steve Kurzeja wrote:
>
>> On Sat, Aug 29, 2009 at 2:34 PM, Diego Viola  
>> <diego.viola at gmail.com> wrote:
>> Yes, FreeSWITCH is a system that you can trust 100%. I have  
>> switched my Asterisk servers to FreeSWITCH and have peace now.
>>
>> If I were you I would get rid of Asterisk and use FreeSWITCH, FS  
>> will handle all what you want very well.
>>
>> And I agree with David, fail-over is kinda irrelevant since the FS  
>> doesn't crash like Asterisk does.
>>
>>
>>
>> You still have hardware failures and fail-over is also useful for  
>> hit-less maintenance on boxes.
>>
>> I'd be interested to know how Brian West was approaching his live  
>> migration work.
>>
>> Steve
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sat Aug 29 08:09:26 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:09:26 -0500
Subject: [Freeswitch-users] Disabling Core Dumping
In-Reply-To: <c55d46950908290707u16fc5d40m3c23f9a9ca450738@mail.gmail.com>
References: <c55d46950908290707u16fc5d40m3c23f9a9ca450738@mail.gmail.com>
Message-ID: <6EA6BE3E-AE31-4628-A204-76B385F6FC20@freeswitch.org>

report the issue on jira.... If its dumping a core something is wrong  
and we need to fix it.

/b

On Aug 29, 2009, at 9:07 AM, Max Bridgewater wrote:

> Hi,
>
> Whenever i stop freeswich, it creates a core dump. How can i disable  
> that?
>
> thanks,
> Max.
> ________




From brian at freeswitch.org  Sat Aug 29 08:12:35 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:12:35 -0500
Subject: [Freeswitch-users] Group Call
In-Reply-To: <ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
Message-ID: <81148697-292E-4E9C-A26A-D1536E9D8A8E@freeswitch.org>

Don't confuse this with the actual group/ endpoint.

/b

On Aug 25, 2009, at 11:41 AM, Jo?o Mesquita wrote:

> According to wiki:
> group,[insert|delete|call]:<group name>:<url>,group [insert|delete| 
> call
>




From brian at freeswitch.org  Sat Aug 29 08:14:37 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:14:37 -0500
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
Message-ID: <6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>

they don't go in sip_profiles... its in conf/directory/default/ and  
they are all there already.  Maybe your configs were not properly  
installed?

/b

On Aug 29, 2009, at 8:46 AM, tom wrote:

>
> what do i have to do now to get it
> a) internally running
> b) talk to an external cllient?




From brian at freeswitch.org  Sat Aug 29 08:15:35 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:15:35 -0500
Subject: [Freeswitch-users] Success! and an issue.
In-Reply-To: <BAY120-W6EC265B637176ED41128BDFF40@phx.gbl>
References: <BAY120-W6EC265B637176ED41128BDFF40@phx.gbl>
Message-ID: <92598075-4C54-4EED-80D0-3263DE495C68@freeswitch.org>

Can you get a sip trace and console trace because this is highly  
unlikely.

/b

On Aug 29, 2009, at 4:28 AM, Christensen Tom wrote:

> However, I did run into one issue, not sure if its somehow related  
> to the phones I'm using (softphone of unknown quality (weephone) on  
> iphones)... I'm using the default config, when I dial 4000 for  
> voicemail, it works, however at apparently random times during the  
> menu navigation I get call waiting beeps and I have an incoming call  
> from 4000.  If I decline or leave this call unanswered, my call into  
> voicemail gets disconnected.  If I "accept" this call, it apparently  
> disconnects immediately (IE I only have 1 active call), however I  
> can then continue navigating the menu, etc... Very strange to me, I  
> will be setting up a couple of polycom 501s on this system tomorrow  
> and will report if I see similar issues.

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From brian at freeswitch.org  Sat Aug 29 08:16:13 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:16:13 -0500
Subject: [Freeswitch-users] FreeSWITCH environmental variables
In-Reply-To: <82795EF6-1698-4DA5-A3F3-23A6CF47CDAC@gmail.com>
References: <7aa29e790908282345r511dcc82kd767063b93d370dd@mail.gmail.com>
	<82795EF6-1698-4DA5-A3F3-23A6CF47CDAC@gmail.com>
Message-ID: <F94A3DA5-7481-43B1-9011-076FCC8188A0@freeswitch.org>

You can also uuid_dump the uuid to see the info also.

/b

On Aug 29, 2009, at 3:17 AM, Jay Binks wrote:

> Check out INFO
> http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_info
>
> Throw that in your dialplan the look at your logs...  You should find
> what your after..
>
> Jay

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From tomabroad at gmail.com  Sat Aug 29 08:19:03 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 11:19:03 -0400
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
Message-ID: <6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>

yes u are right, i was mistaken. everything in there by default...still the
same domain error msg. what am i supposed to do?
tjx
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From brian at freeswitch.org  Sat Aug 29 08:19:32 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:19:32 -0500
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
Message-ID: <A9DC8F8B-B0D3-4AC3-A67B-88F5CFCFDB3A@freeswitch.org>

I was able to do this using OpenVZ, You can get away with it on  
smaller instances... like if you're doing one instance per company but  
don't expect live migration to work as well on large instances with  
thousands of calls up at once. You need a fast network, fast disks and  
to follow the howto on the wiki.

/b

On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote:

> You still have hardware failures and fail-over is also useful for  
> hit-less maintenance on boxes.
>
> I'd be interested to know how Brian West was approaching his live  
> migration work.
>
> Steve




From brian at freeswitch.org  Sat Aug 29 08:22:52 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:22:52 -0500
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
Message-ID: <153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>

Are you on the latest configs?  This message is unlikely in the  
default configs.  Please double check that you have the internal.xml  
from the default install it forces the register domain to match the IP  
so you won't get this error.

/b

On Aug 29, 2009, at 10:19 AM, tom wrote:

> yes u are right, i was mistaken. everything in there by  
> default...still the same domain error msg. what am i supposed to do?
> tjx
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From tomabroad at gmail.com  Sat Aug 29 08:28:32 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 11:28:32 -0400
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
	<153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
Message-ID: <6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>

mh, a search for "domain=" doesnt brin anything up ininternal.xml... im
total new to this...sorry
do i ahev to add this?

//yes im using the 1.0.4 tar




On Sat, Aug 29, 2009 at 11:22 AM, Brian West <brian at freeswitch.org> wrote:

> Are you on the latest configs?  This message is unlikely in the
> default configs.  Please double check that you have the internal.xml
> from the default install it forces the register domain to match the IP
> so you won't get this error.
>
> /b
>
> On Aug 29, 2009, at 10:19 AM, tom wrote:
>
> > yes u are right, i was mistaken. everything in there by
> > default...still the same domain error msg. what am i supposed to do?
> > tjx
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From brian at freeswitch.org  Sat Aug 29 08:33:28 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 10:33:28 -0500
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
	<153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
	<6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>
Message-ID: <98E01B85-5134-48FA-9200-C01D96E48AA5@freeswitch.org>

If you don't have these

     <!--all inbound reg will look in this domain for the users -->
     <param name="force-register-domain" value="$${domain}"/>
     <!--all inbound reg will stored in the db using this domain -->
     <param name="force-register-db-domain" value="$${domain}"/>

Then you need to wipe your conf dir and make samples again please.

/b

On Aug 29, 2009, at 10:28 AM, tom wrote:

>
> mh, a search for "domain=" doesnt brin anything up ininternal.xml...  
> im total new to this...sorry
> do i ahev to add this?
>
> //yes im using the 1.0.4 tar
>
>
>
>
> On Sat, Aug 29, 2009 at 11:22 AM, Brian West <brian at freeswitch.org>  
> wrote:
> Are you on the latest configs?  This message is unlikely in the
> default configs.  Please double check that you have the internal.xml
> from the default install it forces the register domain to match the IP
> so you won't get this error.
>
> /b
>
> On Aug 29, 2009, at 10:19 AM, tom wrote:
>
> > yes u are right, i was mistaken. everything in there by
> > default...still the same domain error msg. what am i supposed to do?
> > tjx
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From tomabroad at gmail.com  Sat Aug 29 08:40:04 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 11:40:04 -0400
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <98E01B85-5134-48FA-9200-C01D96E48AA5@freeswitch.org>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
	<153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
	<6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>
	<98E01B85-5134-48FA-9200-C01D96E48AA5@freeswitch.org>
Message-ID: <6f7c60c40908290840w466c37d6v28dcc0f38f632cce@mail.gmail.com>

ok, what u sent is fine, i have that. do i have to do somethign in that
section:?
  <domains>
    <!-- indicator to parse the directory for domains with parse="true" to
get gateways-->
    <domain name="$${domain}" parse="true"/>
    <!-- indicator to parse the directory for domains with parse="true" to
get gateways and alias every domain to this profile -->
    <!--<domain name="all" alias="true" parse="true"/>-->
    <domain name="all" alias="true" parse="false"/>
  </domains>


?thx


On Sat, Aug 29, 2009 at 11:33 AM, Brian West <brian at freeswitch.org> wrote:

> If you don't have these
>     <!--all inbound reg will look in this domain for the users -->
>
>
>     <param name="force-register-domain" value="$${domain}"/>
>
>
>     <!--all inbound reg will stored in the db using this domain -->
>
>
>     <param name="force-register-db-domain" value="$${domain}"/>
>
> Then you need to wipe your conf dir and make samples again please.
>
> /b
>
> On Aug 29, 2009, at 10:28 AM, tom wrote:
>
>
> mh, a search for "domain=" doesnt brin anything up ininternal.xml... im
> total new to this...sorry
> do i ahev to add this?
>
> //yes im using the 1.0.4 tar
>
>
>
>
> On Sat, Aug 29, 2009 at 11:22 AM, Brian West <brian at freeswitch.org> wrote:
>
>> Are you on the latest configs?  This message is unlikely in the
>> default configs.  Please double check that you have the internal.xml
>> from the default install it forces the register domain to match the IP
>> so you won't get this error.
>>
>> /b
>>
>> On Aug 29, 2009, at 10:19 AM, tom wrote:
>>
>> > yes u are right, i was mistaken. everything in there by
>> > default...still the same domain error msg. what am i supposed to do?
>> > tjx
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Sat Aug 29 08:56:13 2009
From: msc at freeswitch.org (Michael S Collins)
Date: Sat, 29 Aug 2009 08:56:13 -0700
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <6f7c60c40908290840w466c37d6v28dcc0f38f632cce@mail.gmail.com>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
	<153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
	<6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>
	<98E01B85-5134-48FA-9200-C01D96E48AA5@freeswitch.org>
	<6f7c60c40908290840w466c37d6v28dcc0f38f632cce@mail.gmail.com>
Message-ID: <B549B159-D098-4155-A0A7-F5CCDF4BFE0A@freeswitch.org>

Tom,

You should not have to do anything other than make samples. If you are  
having these kinds of problems then something went very wrong during  
the install. Try starting from scratch just to be sure. Also, what  
distro are you running?
-MC

Sent from my iPhone

On Aug 29, 2009, at 8:40 AM, tom <tomabroad at gmail.com> wrote:

>
> ok, what u sent is fine, i have that. do i have to do somethign in  
> that section:?
>   <domains>
>     <!-- indicator to parse the directory for domains with  
> parse="true" to get gateways-->
>     <domain name="$${domain}" parse="true"/>
>     <!-- indicator to parse the directory for domains with  
> parse="true" to get gateways and alias every domain to this profile  
> -->
>     <!--<domain name="all" alias="true" parse="true"/>-->
>     <domain name="all" alias="true" parse="false"/>
>   </domains>
>
>
> ?thx
>
>
> On Sat, Aug 29, 2009 at 11:33 AM, Brian West <brian at freeswitch.org>  
> wrote:
> If you don't have these
>
>     <!--all inbound reg will look in this domain for the users -->
>     <param name="force-register-domain" value="$${domain}"/>
>     <!--all inbound reg will stored in the db using this domain -->
>     <param name="force-register-db-domain" value="$${domain}"/>
>
> Then you need to wipe your conf dir and make samples again please.
>
> /b
>
> On Aug 29, 2009, at 10:28 AM, tom wrote:
>
>>
>> mh, a search for "domain=" doesnt brin anything up  
>> ininternal.xml... im total new to this...sorry
>> do i ahev to add this?
>>
>> //yes im using the 1.0.4 tar
>>
>>
>>
>>
>> On Sat, Aug 29, 2009 at 11:22 AM, Brian West <brian at freeswitch.org>  
>> wrote:
>> Are you on the latest configs?  This message is unlikely in the
>> default configs.  Please double check that you have the internal.xml
>> from the default install it forces the register domain to match the  
>> IP
>> so you won't get this error.
>>
>> /b
>>
>> On Aug 29, 2009, at 10:19 AM, tom wrote:
>>
>> > yes u are right, i was mistaken. everything in there by
>> > default...still the same domain error msg. what am i supposed to  
>> do?
>> > tjx
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
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From anthony.minessale at gmail.com  Sat Aug 29 09:01:46 2009
From: anthony.minessale at gmail.com (Anthony Minessale)
Date: Sat, 29 Aug 2009 11:01:46 -0500
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <A9DC8F8B-B0D3-4AC3-A67B-88F5CFCFDB3A@freeswitch.org>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
	<A9DC8F8B-B0D3-4AC3-A67B-88F5CFCFDB3A@freeswitch.org>
Message-ID: <191c3a030908290901w56b4fd26h467d75ec58dbc380@mail.gmail.com>

We have previously estimated the development of live fail over (after a box
dies where live migration is no longer possible) to exceed 100k in
development costs.

It requires several additions to the sofia sip library, freeswitch and a
dependancy on some other code we would have to implement to manage it.

It may or may not be worth it to raise that kind of funding just to avoid an
occasional disaster.

Then there is a matter of securing the time of the developers necessary to
carry out the implementation.

On Aug 29, 2009 10:19 AM, "Brian West" <brian at freeswitch.org> wrote:

I was able to do this using OpenVZ, You can get away with it on
smaller instances... like if you're doing one instance per company but
don't expect live migration to work as well on large instances with
thousands of calls up at once. You need a fast network, fast disks and
to follow the howto on the wiki.

/b

On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: > You still have hardware
failures and fail-over...

_______________________________________________ FreeSWITCH-users mailing
list FreeSWITCH-users at lists...
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From e.schmidbauer at gmail.com  Sat Aug 29 09:05:42 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Sat, 29 Aug 2009 12:05:42 -0400
Subject: [Freeswitch-users] external sip profile
Message-ID: <2cef777b0908290905jea80cech40b8b75895f28218@mail.gmail.com>

I am using a fresh install of freeswitch trunk and I am unable to
register a phone on port 5080. I set the the registrar of the phone to
server.host.com:5080 but when i do a sofia status profile external,
there are no registrations. when i do a sofia status profile internal,
it shows the phone registered to port 5060. Is this a bug in
freeswitch or is there some configuration I need to change?



From brian at freeswitch.org  Sat Aug 29 09:14:45 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 11:14:45 -0500
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <6f7c60c40908290840w466c37d6v28dcc0f38f632cce@mail.gmail.com>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
	<153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
	<6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>
	<98E01B85-5134-48FA-9200-C01D96E48AA5@freeswitch.org>
	<6f7c60c40908290840w466c37d6v28dcc0f38f632cce@mail.gmail.com>
Message-ID: <7400B5BC-EC99-464E-B97F-D3DCD3CE2180@freeswitch.org>

No you shouldn't have to touch ANYTHING in the default configs at all  
to get 1000 with password 1234 to register... I recommend you:

cd /usr/local/freeswitch
mv conf conf.old
then do make samples again.

/b

On Aug 29, 2009, at 10:40 AM, tom wrote:

>
> ok, what u sent is fine, i have that. do i have to do somethign in  
> that section:?
>   <domains>
>     <!-- indicator to parse the directory for domains with  
> parse="true" to get gateways-->
>     <domain name="$${domain}" parse="true"/>
>     <!-- indicator to parse the directory for domains with  
> parse="true" to get gateways and alias every domain to this profile  
> -->
>     <!--<domain name="all" alias="true" parse="true"/>-->
>     <domain name="all" alias="true" parse="false"/>
>   </domains>
>
>
> ?thx




From brian at freeswitch.org  Sat Aug 29 09:16:05 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 11:16:05 -0500
Subject: [Freeswitch-users] external sip profile
In-Reply-To: <2cef777b0908290905jea80cech40b8b75895f28218@mail.gmail.com>
References: <2cef777b0908290905jea80cech40b8b75895f28218@mail.gmail.com>
Message-ID: <53656C07-1D94-45EB-88D0-81A6C59058A9@freeswitch.org>

No, enable sip trace I suspect your config is wrong in the end point.

/b

On Aug 29, 2009, at 11:05 AM, e schmidbauer wrote:

> I am using a fresh install of freeswitch trunk and I am unable to
> register a phone on port 5080. I set the the registrar of the phone to
> server.host.com:5080 but when i do a sofia status profile external,
> there are no registrations. when i do a sofia status profile internal,
> it shows the phone registered to port 5060. Is this a bug in
> freeswitch or is there some configuration I need to change?




From e.schmidbauer at gmail.com  Sat Aug 29 10:20:42 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Sat, 29 Aug 2009 13:20:42 -0400
Subject: [Freeswitch-users] external sip profile
In-Reply-To: <53656C07-1D94-45EB-88D0-81A6C59058A9@freeswitch.org>
References: <2cef777b0908290905jea80cech40b8b75895f28218@mail.gmail.com>
	<53656C07-1D94-45EB-88D0-81A6C59058A9@freeswitch.org>
Message-ID: <2cef777b0908291020jdbeac97q3160606ffc6d6a72@mail.gmail.com>

You were correct. The client (ekiga) i was using forces the connection
on port 5060.

On Sat, Aug 29, 2009 at 12:16 PM, Brian West<brian at freeswitch.org> wrote:
> No, enable sip trace I suspect your config is wrong in the end point.
>
> /b
>
> On Aug 29, 2009, at 11:05 AM, e schmidbauer wrote:
>
>> I am using a fresh install of freeswitch trunk and I am unable to
>> register a phone on port 5080. I set the the registrar of the phone to
>> server.host.com:5080 but when i do a sofia status profile external,
>> there are no registrations. when i do a sofia status profile internal,
>> it shows the phone registered to port 5060. Is this a bug in
>> freeswitch or is there some configuration I need to change?
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



From mike at jerris.com  Sat Aug 29 10:45:08 2009
From: mike at jerris.com (Michael Jerris)
Date: Sat, 29 Aug 2009 13:45:08 -0400
Subject: [Freeswitch-users] Success! and an issue.
In-Reply-To: <92598075-4C54-4EED-80D0-3263DE495C68@freeswitch.org>
References: <BAY120-W6EC265B637176ED41128BDFF40@phx.gbl>
	<92598075-4C54-4EED-80D0-3263DE495C68@freeswitch.org>
Message-ID: <2FF2B6E7-7E01-449B-87A4-F06CD65B4AF7@jerris.com>

This sounds to me like a phone that doesn't know that a re-invite is  
not a new call and session timers are turned on.  A sip trace will  
show you if this is the case, and if it is, that phone is indeed badly  
broken.

Mike


On Aug 29, 2009, at 11:15 AM, Brian West wrote:

> Can you get a sip trace and console trace because this is highly  
> unlikely.
>
> /b
>
> On Aug 29, 2009, at 4:28 AM, Christensen Tom wrote:
>
>> However, I did run into one issue, not sure if its somehow related  
>> to the phones I'm using (softphone of unknown quality (weephone) on  
>> iphones)... I'm using the default config, when I dial 4000 for  
>> voicemail, it works, however at apparently random times during the  
>> menu navigation I get call waiting beeps and I have an incoming  
>> call from 4000.  If I decline or leave this call unanswered, my  
>> call into voicemail gets disconnected.  If I "accept" this call, it  
>> apparently disconnects immediately (IE I only have 1 active call),  
>> however I can then continue navigating the menu, etc... Very  
>> strange to me, I will be setting up a couple of polycom 501s on  
>> this system tomorrow and will report if I see similar issues.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From dave at 3c.co.uk  Sat Aug 29 11:05:57 2009
From: dave at 3c.co.uk (David Knell)
Date: Sat, 29 Aug 2009 21:05:57 +0300
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <136791EF-26D2-4DC7-91F6-2D468122F06C@runsolutions.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
	<71FD2763-32C5-49CA-8E4D-7C0E301D2187@runsolutions.com>
	<136791EF-26D2-4DC7-91F6-2D468122F06C@runsolutions.com>
Message-ID: <1251569157.5146.22.camel@dk-d820>

Hi Raimund,

The difficult bit in all of this is having calls seamlessly transferred
from one box to another when the first box dies.  There's a lot of state
associated with a call, not all of which is easy to replicate across.

99.99% uptime implies an average of no more than 15 seconds downtime
during a 40 hour business week (or about one unscheduled reboot every
couple of months), which is easily achievable using FreeSWITCH (and, in
my experience, Asterisk) on standard hardware.

People agonize about their four or five nines too much, in my opinion.
Folk are used to their phones crashing, needing rebooting and dropping
calls - we've cellphones to thank for that - and a half-decent VoIP
solution will knock the spots off your favourite mobile carrier for
reliability.  Plus there's all the external factors - 99.999% uptime on
your PRI means that someone can only drive a digger through the cable
once every 273 years, assuming that it takes a day to fix, and no
telco's going to give you that in an SLA.  And your power can't go off.
And so on.

Lastly, I'm afraid that virtualization and VoIP don't play well
together, at least not if you want to achieve a sensible density.  The
large number of small packets being moved around the network interfaces
- both physical and virtual - will quickly chew up your CPU.

Cheers --

Dave


> Thinking about it, maybe we can create a solution, if some of us work
> together:
> 
> 
> My strength are in virtualization, linux, development, databases,
> integration, etc.
> What I do not now much about is how SIP (and everything else for that
> matter in the Voice world) works under the hood, and how it's
> implemented in FS.
> 
> 
> I know that the state information for a call has to be stored and
> retrieved somewhere and somehow, only I do not know that part. What I
> know is that it hast to be do-able to store all the stream information
> (ip's, port's, current state's, etc.) in a very fast database (e.g. my
> idea would be memcached) so another FS could just take this
> information and take over the call, maybe you loose a second of voice,
> maybe you loose the recorded call file or a part of it, but that
> should be it. (SipFoundry has a boxed opensource PBX, which, of course
> is not flexible like FreeSWITCH or Asterisk, but has Call Live
> Migration and Call Live Failover integrated!).
> 
> 
> What I want is for my company to be able to sell a 99.99 uptime PBX
> (we do mostly call-center related stuff), which can scale well, and
> can grow with the company without lot's of hassles, my Dream would be:
> 
> 
> To begin with: 
> 
> 
> One Hardware Node with the essential hardware (digium cards for
> example).
> On this node are OpenVZ virtualized containers:
> [VirtCnt1: FS which only talks to the Hardware and forwards
> everything] = Could be replaced with hardware media gateway, etc.
> [VirtCnt2: FS which handles the PBX] \___ Loadbalanced, with odbc or
> xml, Failover, Livetakeover
> [VirtCnt3: FS which handles the PBX] /
> [VirtCnt4: Database for state information] (maybe something as
> resource-friendly as memcached? ressource heavvy database?)
> 
> 
> With this we can achieve all this:
> 
> 
> Problem with VirtCnt2 (e.g. crash, lock, ...)
> * VirtCnt3 can take over. 
> -> You are free without stress to investigate the problem, you can
> debug and analyze whyle the machine is still running
> -> you can also create a machine-state-dump of the virtual container,
> dump the container as well, copy the data to your lab and restore the
> machine up the state which it was running with the problem, so you can
> liveinvestigate it in the lab (some prerequirements given, but easy
> doable)
> -> just think about the possibility of better bugreports because
> someone can take the time to read out all the data with GDB to
> investigate the proper cause of a machine Lock!
> 
> 
> You want to upgrade to a new FreeSWITCH version?
> * Take VirtCnt2 out of the LoadBalancing Scheme,
> * Stop it, Clone it, 
> * Upgrade FreeSWITCH in the cloned Container
> * Start the cloned container
> * if there's something wrong, stop it and restart the original
> VirtCnt2
> -> No problem at all, you can Test on the Live Hardware, with part of
> the Live users (maybe a low-volume queue) to be sure everything works
> out fine before you activate the full loadbalance
> 
> 
> Server on it's own can't handle the load
> * Buy new machine
> * Setup Hardware Node
> * Livemigrate VirtCnt3 (no downtime) 
> 
> 
> Now the first Server with the VrtCnt1 and VirtCnt2 as well has to much
> load
> * Buy new machine
> * Setup Hardware Node
> * Livemigrate VirtCnt2 (no downtime)
> -> Now you have a 3 server solution (1 mediaprox, 2 loadbalanced /
> failover PBXes) out of the first box you bought, without headaches,
> because the system was built for it from the beginning!
> 
> 
> The Database drains to much?
> * Buy new machine
> * Setup Hardware Node
> * Livemigrate database VirtCnt4 (no downtime)
> 
> 
> You want to upgrade Hardware/Kernel in Hardware node 1?
> * Livemigrate VirtCnt2 to a hotstandby machine, or to the other PBX
> machine, upgrade the hardware, Re-Livemigrate the containers. (no
> downtime)
> * OR just break the loadbalancing, wait until all current calls are
> teared down correctly, upgrade machine, reenable the loadbalancer
> 
> 
> You want an exact copy of the first server for Hardware HA?
> * Buy new machine
> * Setup Hardware node
> * Buy hardware PRI switchover box
> * Clone VirtCnt1 - VirtCnt4 to the new machine
> * Make basic failover configuration 
> 
> 
> 
> 
> -> the sky's the limit, as the saying goes ...
> 
> 
> 
> 
> So, I can do all the openvz stuff and the integration with database /
> memcached / heartbeat / whatever is needed here, someone there to be
> willing to work with me on this on the FreeSWITCH side? or at least
> provide me with the necessary information about what's needed / how to
> talk / what states from FreeSWITCH?
> 
> 
> I know this seems very ambitious but if this could be made in a rather
> relativly easy to setup package, with good documentation, it would be
> a boost for FreeSWITCH, i am sure, because after all this is what
> everyone is grown accustomed to from good old phone companys and the
> good old pbx's: carrier grade uptimes ...
> 
> 
> Thanks for everyone reading up until here,
> all the best,
> 
> 
> Ray
> 
> 
> 
> 
> 
> 
> -- 
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
> 
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
> 
> On Aug 29, 2009, at 3:17 PM, Raimund Sacherer wrote:
> 
> > Oh yeah, that would be so helpfull for my situation, as my client
> > *demands* now a solution where he can press a big red button and all
> > fails over to another box. Hi es totally scared because of the
> > Lockups in Asterisk which under specific situations including AMI,
> > Automated Call Setup, and murphy led to a lockup of the entire
> > machine, no console was working anymore, only cold-reset could do
> > it.
> > 
> > 
> > So, IF there is the possibility for life-takeover, / failover etc. I
> > would love to here how has been done.
> > 
> > 
> > I am very experienced with openvz and use for about two years now
> > only openvz virtualization servers for anything because of
> > live-migration etc. But as I am new in this company we could not
> > adopt this until now.
> > 
> > 
> > So Please Ken, if you can, describe what need's to be done to get a
> > failover / takeover working (an outline would be enough)
> > 
> > 
> > Thanks in Advance
> > 
> > -- 
> > Raimund Sacherer
> > -
> > RunSolutions
> >     Open Source It Consulting
> > -
> > 
> > Parc Bit - Centro Empresarial Son Espanyol
> > Edificio Estel - Local 3D
> > 07121 -  Palma de Mallorca
> > Baleares
> > 
> > On Aug 29, 2009, at 11:58 AM, Steve Kurzeja wrote:
> > 
> > > On Sat, Aug 29, 2009 at 2:34 PM, Diego Viola
> > > <diego.viola at gmail.com> wrote:
> > >         Yes, FreeSWITCH is a system that you can trust 100%. I
> > >         have switched my Asterisk servers to FreeSWITCH and have
> > >         peace now.
> > >         
> > >         If I were you I would get rid of Asterisk and use
> > >         FreeSWITCH, FS will handle all what you want very well.
> > >         
> > >         And I agree with David, fail-over is kinda irrelevant
> > >         since the FS doesn't crash like Asterisk does.
> > >         
> > >         
> > >  
> > > You still have hardware failures and fail-over is also useful for
> > > hit-less maintenance on boxes.
> > > 
> > > I'd be interested to know how Brian West was approaching his live
> > > migration work. 
> > > 
> > > Steve
> > > _______________________________________________
> > > FreeSWITCH-users mailing list
> > > FreeSWITCH-users at lists.freeswitch.org
> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > > http://www.freeswitch.org
> > 
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk




From erik at erikwickstrom.com  Sat Aug 29 11:21:03 2009
From: erik at erikwickstrom.com (Erik Wickstrom)
Date: Sat, 29 Aug 2009 11:21:03 -0700
Subject: [Freeswitch-users] Transferring to another extension is VERY slow.
Message-ID: <3d381e170908291121k3bd64ecnacf4f7941962e2ed@mail.gmail.com>

Hi,

I'm having some trouble debugging why the transfer command is so slow
in my case.  I have a lua IVR setup which responds to user selections
by transferring the call to another dialplan ext.  Pretty basic stuff.

The problem is, it takes 30-40 seconds for the the transfer to
actually start after the key press.  After about 30 seconds of
silence, "hold music" finally starts and the other extension finally
answers shortly thereafter.

I've tried transferring to endpoint extensions, the 5000 sample IVR,
9999 hold music -- in all cases there is this long pause.

Does anyone know what could be causing this?

Thanks!
Erik



From rs at runsolutions.com  Sat Aug 29 11:33:21 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Sat, 29 Aug 2009 20:33:21 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <191c3a030908290901w56b4fd26h467d75ec58dbc380@mail.gmail.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
	<A9DC8F8B-B0D3-4AC3-A67B-88F5CFCFDB3A@freeswitch.org>
	<191c3a030908290901w56b4fd26h467d75ec58dbc380@mail.gmail.com>
Message-ID: <261B13B1-CBE5-4AE4-9ACB-3AE19D556B8A@runsolutions.com>

Hmm, so basically 100 interested companys which each chip in 1000  
bucks :-)

sounds like lots of manpower, but on the other hand, I do not know the  
issues regarding the SIP protocoll, but, basically, isn't it *just* to  
tell another FS box to listen on port x for voicetraffic, forward it  
to ip on port y?

ok, i understand there's a lot going on under the hood, i guess it  
would mean to setup a call, but take care to not really set up the  
call, just the internal state ...

hmm, could it theoretically be done with the event system? ok, i guess  
I have to dive further into the internals to fully understand the scope.


But a live migration, where the box is available, be possible right  
now? Would be a step i would like to implement just to be able to do  
work on a hardware node if necesary without interrupting the service ...


-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 29, 2009, at 6:01 PM, Anthony Minessale wrote:

> We have previously estimated the development of live fail over  
> (after a box dies where live migration is no longer possible) to  
> exceed 100k in development costs.
>
> It requires several additions to the sofia sip library, freeswitch  
> and a dependancy on some other code we would have to implement to  
> manage it.
>
> It may or may not be worth it to raise that kind of funding just to  
> avoid an occasional disaster.
>
> Then there is a matter of securing the time of the developers  
> necessary to carry out the implementation.
>
>
>> On Aug 29, 2009 10:19 AM, "Brian West" <brian at freeswitch.org> wrote:
>>
>> I was able to do this using OpenVZ, You can get away with it on
>> smaller instances... like if you're doing one instance per company  
>> but
>> don't expect live migration to work as well on large instances with
>> thousands of calls up at once. You need a fast network, fast disks  
>> and
>> to follow the howto on the wiki.
>>
>> /b
>> On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: > You still have  
>> hardware failures and fail-over...
>>
>> _______________________________________________ FreeSWITCH-users  
>> mailing list FreeSWITCH-users at lists...
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From mitul at enterux.com  Sat Aug 29 11:36:24 2009
From: mitul at enterux.com (Mitul Limbani)
Date: Sun, 30 Aug 2009 00:06:24 +0530
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <1251569157.5146.22.camel@dk-d820>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
	<71FD2763-32C5-49CA-8E4D-7C0E301D2187@runsolutions.com>
	<136791EF-26D2-4DC7-91F6-2D468122F06C@runsolutions.com>
	<1251569157.5146.22.camel@dk-d820>
Message-ID: <416B910B-6058-4387-9563-3491B091D8B4@enterux.com>

On PRI circuits failover with ha is possible using a red tone  
appliance, but I m sure it's going to be almost impossible to transfer  
the running calls during the event of fs box going down.

Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com

On 29-Aug-2009, at 11:35 PM, David Knell <dave at 3c.co.uk> wrote:

> Hi Raimund,
>
> The difficult bit in all of this is having calls seamlessly  
> transferred
> from one box to another when the first box dies.  There's a lot of  
> state
> associated with a call, not all of which is easy to replicate across.
>
> 99.99% uptime implies an average of no more than 15 seconds downtime
> during a 40 hour business week (or about one unscheduled reboot every
> couple of months), which is easily achievable using FreeSWITCH (and,  
> in
> my experience, Asterisk) on standard hardware.
>
> People agonize about their four or five nines too much, in my opinion.
> Folk are used to their phones crashing, needing rebooting and dropping
> calls - we've cellphones to thank for that - and a half-decent VoIP
> solution will knock the spots off your favourite mobile carrier for
> reliability.  Plus there's all the external factors - 99.999% uptime  
> on
> your PRI means that someone can only drive a digger through the cable
> once every 273 years, assuming that it takes a day to fix, and no
> telco's going to give you that in an SLA.  And your power can't go  
> off.
> And so on.
>
> Lastly, I'm afraid that virtualization and VoIP don't play well
> together, at least not if you want to achieve a sensible density.  The
> large number of small packets being moved around the network  
> interfaces
> - both physical and virtual - will quickly chew up your CPU.
>
> Cheers --
>
> Dave
>
>
>> Thinking about it, maybe we can create a solution, if some of us work
>> together:
>>
>>
>> My strength are in virtualization, linux, development, databases,
>> integration, etc.
>> What I do not now much about is how SIP (and everything else for that
>> matter in the Voice world) works under the hood, and how it's
>> implemented in FS.
>>
>>
>> I know that the state information for a call has to be stored and
>> retrieved somewhere and somehow, only I do not know that part. What I
>> know is that it hast to be do-able to store all the stream  
>> information
>> (ip's, port's, current state's, etc.) in a very fast database (e.g.  
>> my
>> idea would be memcached) so another FS could just take this
>> information and take over the call, maybe you loose a second of  
>> voice,
>> maybe you loose the recorded call file or a part of it, but that
>> should be it. (SipFoundry has a boxed opensource PBX, which, of  
>> course
>> is not flexible like FreeSWITCH or Asterisk, but has Call Live
>> Migration and Call Live Failover integrated!).
>>
>>
>> What I want is for my company to be able to sell a 99.99 uptime PBX
>> (we do mostly call-center related stuff), which can scale well, and
>> can grow with the company without lot's of hassles, my Dream would  
>> be:
>>
>>
>> To begin with:
>>
>>
>> One Hardware Node with the essential hardware (digium cards for
>> example).
>> On this node are OpenVZ virtualized containers:
>> [VirtCnt1: FS which only talks to the Hardware and forwards
>> everything] = Could be replaced with hardware media gateway, etc.
>> [VirtCnt2: FS which handles the PBX] \___ Loadbalanced, with odbc or
>> xml, Failover, Livetakeover
>> [VirtCnt3: FS which handles the PBX] /
>> [VirtCnt4: Database for state information] (maybe something as
>> resource-friendly as memcached? ressource heavvy database?)
>>
>>
>> With this we can achieve all this:
>>
>>
>> Problem with VirtCnt2 (e.g. crash, lock, ...)
>> * VirtCnt3 can take over.
>> -> You are free without stress to investigate the problem, you can
>> debug and analyze whyle the machine is still running
>> -> you can also create a machine-state-dump of the virtual container,
>> dump the container as well, copy the data to your lab and restore the
>> machine up the state which it was running with the problem, so you  
>> can
>> liveinvestigate it in the lab (some prerequirements given, but easy
>> doable)
>> -> just think about the possibility of better bugreports because
>> someone can take the time to read out all the data with GDB to
>> investigate the proper cause of a machine Lock!
>>
>>
>> You want to upgrade to a new FreeSWITCH version?
>> * Take VirtCnt2 out of the LoadBalancing Scheme,
>> * Stop it, Clone it,
>> * Upgrade FreeSWITCH in the cloned Container
>> * Start the cloned container
>> * if there's something wrong, stop it and restart the original
>> VirtCnt2
>> -> No problem at all, you can Test on the Live Hardware, with part of
>> the Live users (maybe a low-volume queue) to be sure everything works
>> out fine before you activate the full loadbalance
>>
>>
>> Server on it's own can't handle the load
>> * Buy new machine
>> * Setup Hardware Node
>> * Livemigrate VirtCnt3 (no downtime)
>>
>>
>> Now the first Server with the VrtCnt1 and VirtCnt2 as well has to  
>> much
>> load
>> * Buy new machine
>> * Setup Hardware Node
>> * Livemigrate VirtCnt2 (no downtime)
>> -> Now you have a 3 server solution (1 mediaprox, 2 loadbalanced /
>> failover PBXes) out of the first box you bought, without headaches,
>> because the system was built for it from the beginning!
>>
>>
>> The Database drains to much?
>> * Buy new machine
>> * Setup Hardware Node
>> * Livemigrate database VirtCnt4 (no downtime)
>>
>>
>> You want to upgrade Hardware/Kernel in Hardware node 1?
>> * Livemigrate VirtCnt2 to a hotstandby machine, or to the other PBX
>> machine, upgrade the hardware, Re-Livemigrate the containers. (no
>> downtime)
>> * OR just break the loadbalancing, wait until all current calls are
>> teared down correctly, upgrade machine, reenable the loadbalancer
>>
>>
>> You want an exact copy of the first server for Hardware HA?
>> * Buy new machine
>> * Setup Hardware node
>> * Buy hardware PRI switchover box
>> * Clone VirtCnt1 - VirtCnt4 to the new machine
>> * Make basic failover configuration
>>
>>
>>
>>
>> -> the sky's the limit, as the saying goes ...
>>
>>
>>
>>
>> So, I can do all the openvz stuff and the integration with database /
>> memcached / heartbeat / whatever is needed here, someone there to be
>> willing to work with me on this on the FreeSWITCH side? or at least
>> provide me with the necessary information about what's needed / how  
>> to
>> talk / what states from FreeSWITCH?
>>
>>
>> I know this seems very ambitious but if this could be made in a  
>> rather
>> relativly easy to setup package, with good documentation, it would be
>> a boost for FreeSWITCH, i am sure, because after all this is what
>> everyone is grown accustomed to from good old phone companys and the
>> good old pbx's: carrier grade uptimes ...
>>
>>
>> Thanks for everyone reading up until here,
>> all the best,
>>
>>
>> Ray
>>
>>
>>
>>
>>
>>
>> -- 
>> Raimund Sacherer
>> -
>> RunSolutions
>>    Open Source It Consulting
>> -
>>
>> Parc Bit - Centro Empresarial Son Espanyol
>> Edificio Estel - Local 3D
>> 07121 -  Palma de Mallorca
>> Baleares
>>
>> On Aug 29, 2009, at 3:17 PM, Raimund Sacherer wrote:
>>
>>> Oh yeah, that would be so helpfull for my situation, as my client
>>> *demands* now a solution where he can press a big red button and all
>>> fails over to another box. Hi es totally scared because of the
>>> Lockups in Asterisk which under specific situations including AMI,
>>> Automated Call Setup, and murphy led to a lockup of the entire
>>> machine, no console was working anymore, only cold-reset could do
>>> it.
>>>
>>>
>>> So, IF there is the possibility for life-takeover, / failover etc. I
>>> would love to here how has been done.
>>>
>>>
>>> I am very experienced with openvz and use for about two years now
>>> only openvz virtualization servers for anything because of
>>> live-migration etc. But as I am new in this company we could not
>>> adopt this until now.
>>>
>>>
>>> So Please Ken, if you can, describe what need's to be done to get a
>>> failover / takeover working (an outline would be enough)
>>>
>>>
>>> Thanks in Advance
>>>
>>> -- 
>>> Raimund Sacherer
>>> -
>>> RunSolutions
>>>    Open Source It Consulting
>>> -
>>>
>>> Parc Bit - Centro Empresarial Son Espanyol
>>> Edificio Estel - Local 3D
>>> 07121 -  Palma de Mallorca
>>> Baleares
>>>
>>> On Aug 29, 2009, at 11:58 AM, Steve Kurzeja wrote:
>>>
>>>> On Sat, Aug 29, 2009 at 2:34 PM, Diego Viola
>>>> <diego.viola at gmail.com> wrote:
>>>>        Yes, FreeSWITCH is a system that you can trust 100%. I
>>>>        have switched my Asterisk servers to FreeSWITCH and have
>>>>        peace now.
>>>>
>>>>        If I were you I would get rid of Asterisk and use
>>>>        FreeSWITCH, FS will handle all what you want very well.
>>>>
>>>>        And I agree with David, fail-over is kinda irrelevant
>>>>        since the FS doesn't crash like Asterisk does.
>>>>
>>>>
>>>>
>>>> You still have hardware failures and fail-over is also useful for
>>>> hit-less maintenance on boxes.
>>>>
>>>> I'd be interested to know how Brian West was approaching his live
>>>> migration work.
>>>>
>>>> Steve
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
> -- 
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org



From krice at freeswitch.org  Sat Aug 29 11:44:26 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sat, 29 Aug 2009 13:44:26 -0500
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <261B13B1-CBE5-4AE4-9ACB-3AE19D556B8A@runsolutions.com>
Message-ID: <C6BEE13A.5E372%krice@freeswitch.org>

It is not possible to do a live migration at all right now... There is no
way to move a call and have all the states required for that call to
magically re-appear on a different instance. This will require a fair bit of
work to get there. 

It is possible to configure fs via some back end DB magic to share
configurations ie: n+1 or ?warm standby? style fault tolerance but you are
going to loose the calls that are up when the failure occurs.


From: Raimund Sacherer <rs at runsolutions.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Sat, 29 Aug 2009 20:33:21 +0200
To: <freeswitch-users at lists.freeswitch.org>
Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

Hmm, so basically 100 interested companys which each chip in 1000 bucks :-)

sounds like lots of manpower, but on the other hand, I do not know the
issues regarding the SIP protocoll, but, basically, isn't it *just* to tell
another FS box to listen on port x for voicetraffic, forward it to ip on
port y?

ok, i understand there's a lot going on under the hood, i guess it would
mean to setup a call, but take care to not really set up the call, just the
internal state ...

hmm, could it theoretically be done with the event system? ok, i guess I
have to dive further into the internals to fully understand the scope.


But a live migration, where the box is available, be possible right now?
Would be a step i would like to implement just to be able to do work on a
hardware node if necesary without interrupting the service ...


-- 
Raimund Sacherer
-
RunSolutions
    Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 29, 2009, at 6:01 PM, Anthony Minessale wrote:

> 
> We have previously estimated the development of live fail over (after a box
> dies where live migration is no longer possible) to exceed 100k in development
> costs.
> 
> It requires several additions to the sofia sip library, freeswitch and a
> dependancy on some other code we would have to implement to manage it.
> 
> It may or may not be worth it to raise that kind of funding just to avoid an
> occasional disaster.
> 
> Then there is a matter of securing the time of the developers necessary to
> carry out the implementation.
> 
> 
>> On Aug 29, 2009 10:19 AM, "Brian West" <brian at freeswitch.org> wrote:
>> 
>> I was able to do this using OpenVZ, You can get away with it on
>> smaller instances... like if you're doing one instance per company but
>> don't expect live migration to work as well on large instances with
>> thousands of calls up at once. You need a fast network, fast disks and
>> to follow the howto on the wiki.
>> 
>> /b
>> 
>> On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: > You still have hardware
>> failures and fail-over...
>> 
>> _______________________________________________ FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists...
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org

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From mgg at giagnocavo.net  Sat Aug 29 12:39:52 2009
From: mgg at giagnocavo.net (Michael Giagnocavo)
Date: Sat, 29 Aug 2009 15:39:52 -0400
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <261B13B1-CBE5-4AE4-9ACB-3AE19D556B8A@runsolutions.com>
References: <D3DADB48-0D39-4964-9997-23D7B84F0D53@runsolutions.com>
	<86a32abc0908281934q4481d81r52e152cce60841c2@mail.gmail.com>
	<5f7152000908290258w34a9cb0h48e0cca51b4deb6e@mail.gmail.com>
	<A9DC8F8B-B0D3-4AC3-A67B-88F5CFCFDB3A@freeswitch.org>
	<191c3a030908290901w56b4fd26h467d75ec58dbc380@mail.gmail.com>
	<261B13B1-CBE5-4AE4-9ACB-3AE19D556B8A@runsolutions.com>
Message-ID: <6E8D2069C08AA84A83D336E996AE4C6702D7DF5127@mse17be1.mse17.exchange.ms>

Also please remember this means every _application_ also must be "failover" aware. So, if you're using a little calling card or billing app to control your routing, that app will have to explicitly enroll its state as well as restore itself.

Live migration is possible via OpenVZ. OVZ does not have the same overhead of other virtualization techniques such as Hyper-V, ESX, Xen, etc. So you can do media without much worry.  OVZ can work with other virtualization, too; you can spin it up inside of a VMware machine.

But this is *live* migration of the container. It's "just" pausing your container, copying the whole memory, and resuming it on another node. This is not "dead" migration or failover after calls break. However, you could try running a very frequent sync in OVZ, so that if there was a node failure, you could resume from a previously saved state. But, I highly doubt performance would be good enough for this to work successfully on any serious deployment.

-Michael

From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Raimund Sacherer
Sent: Saturday, August 29, 2009 12:33 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing

Hmm, so basically 100 interested companys which each chip in 1000 bucks :-)

sounds like lots of manpower, but on the other hand, I do not know the issues regarding the SIP protocoll, but, basically, isn't it *just* to tell another FS box to listen on port x for voicetraffic, forward it to ip on port y?

ok, i understand there's a lot going on under the hood, i guess it would mean to setup a call, but take care to not really set up the call, just the internal state ...

hmm, could it theoretically be done with the event system? ok, i guess I have to dive further into the internals to fully understand the scope.


But a live migration, where the box is available, be possible right now? Would be a step i would like to implement just to be able to do work on a hardware node if necesary without interrupting the service ...


--
Raimund Sacherer
-
RunSolutions
    Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 29, 2009, at 6:01 PM, Anthony Minessale wrote:



We have previously estimated the development of live fail over (after a box dies where live migration is no longer possible) to exceed 100k in development costs.

It requires several additions to the sofia sip library, freeswitch and a dependancy on some other code we would have to implement to manage it.

It may or may not be worth it to raise that kind of funding just to avoid an occasional disaster.

Then there is a matter of securing the time of the developers necessary to carry out the implementation.

On Aug 29, 2009 10:19 AM, "Brian West" <brian at freeswitch.org<mailto:brian at freeswitch.org>> wrote:

I was able to do this using OpenVZ, You can get away with it on
smaller instances... like if you're doing one instance per company but
don't expect live migration to work as well on large instances with
thousands of calls up at once. You need a fast network, fast disks and
to follow the howto on the wiki.

/b

On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: > You still have hardware failures and fail-over...

_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists...

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org<mailto:FreeSWITCH-users at lists.freeswitch.org>
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From mitul at enterux.com  Sat Aug 29 12:56:16 2009
From: mitul at enterux.com (Mitul Limbani)
Date: Sun, 30 Aug 2009 01:26:16 +0530
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <C6BEE13A.5E372%krice@freeswitch.org>
References: <C6BEE13A.5E372%krice@freeswitch.org>
Message-ID: <A6E1A4F9-F0D5-4645-A72D-26365CF6FC7B@enterux.com>

Is itnpossible to have a db cluster know the state of each and every  
call and then use Heartbeat on this db +
fs cluster so that clients see only one ip where as internally all fs  
boxes refer db for call states, db again is under replication.

This in the thioery can be written, but I am sure if we think bit more  
on this direction the problem seem to be getting addressed.

Other guys also chip in their 2 cents, we just need 50 of em to make a  
full dollar.

Thanks & Regards,
Mitul Limbani,
Founder & CEO,
Enterux Solutions Pvt. Ltd.,
The Enterprise Linux Company (r),
http://www.enterux.com
http://www.entVoice.com

On 30-Aug-2009, at 12:14 AM, Ken Rice <krice at freeswitch.org> wrote:

> It is not possible to do a live migration at all right now... There  
> is no way to move a call and have all the states required for that  
> call to magically re-appear on a different instance. This will  
> require a fair bit of work to get there.
>
> It is possible to configure fs via some back end DB magic to share  
> configurations ie: n+1 or ?warm standby? style fault tolerance  
> but you are going to loose the calls that are up when the failure oc 
> curs.
>
> From: Raimund Sacherer <rs at runsolutions.com>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Sat, 29 Aug 2009 20:33:21 +0200
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
>
> Hmm, so basically 100 interested companys which each chip in 1000  
> bucks :-)
>
> sounds like lots of manpower, but on the other hand, I do not know  
> the issues regarding the SIP protocoll, but, basically, isn't it  
> *just* to tell another FS box to listen on port x for voicetraffic,  
> forward it to ip on port y?
>
> ok, i understand there's a lot going on under the hood, i guess it  
> would mean to setup a call, but take care to not really set up the  
> call, just the internal state ...
>
> hmm, could it theoretically be done with the event system? ok, i  
> guess I have to dive further into the internals to fully understand  
> the scope.
>
>
> But a live migration, where the box is available, be possible right  
> now? Would be a step i would like to implement just to be able to do  
> work on a hardware node if necesary without interrupting the  
> service ...
>
>
> -- 
> Raimund Sacherer
> -
> RunSolutions
>     Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 -  Palma de Mallorca
> Baleares
>
> On Aug 29, 2009, at 6:01 PM, Anthony Minessale wrote:
>
>
> We have previously estimated the development of live fail over  
> (after a box dies where live migration is no longer possible) to  
> exceed 100k in development costs.
>
> It requires several additions to the sofia sip library, freeswitch  
> and a dependancy on some other code we would have to implement to  
> manage it.
>
> It may or may not be worth it to raise that kind of funding just to  
> avoid an occasional disaster.
>
> Then there is a matter of securing the time of the developers  
> necessary to carry out the implementation.
>
>
> On Aug 29, 2009 10:19 AM, "Brian West" <brian at freeswitch.org> wrote:
>
> I was able to do this using OpenVZ, You can get away with it on
> smaller instances... like if you're doing one instance per company but
> don't expect live migration to work as well on large instances with
> thousands of calls up at once. You need a fast network, fast disks and
> to follow the howto on the wiki.
>
> /b
>
> On Aug 29, 2009, at 4:58 AM, Steve Kurzeja wrote: > You still have  
> hardware failures and fail-over...
>
> _______________________________________________ FreeSWITCH-users  
> mailing list FreeSWITCH-users at lists...
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
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From brian at freeswitch.org  Sat Aug 29 13:04:25 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 15:04:25 -0500
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <A6E1A4F9-F0D5-4645-A72D-26365CF6FC7B@enterux.com>
References: <C6BEE13A.5E372%krice@freeswitch.org>
	<A6E1A4F9-F0D5-4645-A72D-26365CF6FC7B@enterux.com>
Message-ID: <B55A1F03-DC5E-4F32-99C8-3AB9D11B2C8A@freeswitch.org>

The sip stack needs to be modified to spin that data up into the state  
machine so that it can take over calls once the fail over takes  
place... its not an easy task.

/b

On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:

> Is itnpossible to have a db cluster know the state of each and every  
> call and then use Heartbeat on this db +
> fs cluster so that clients see only one ip where as internally all  
> fs boxes refer db for call states, db again is under replication.
>
> This in the thioery can be written, but I am sure if we think bit  
> more on this direction the problem seem to be getting addressed.
>
> Other guys also chip in their 2 cents, we just need 50 of em to make  
> a full dollar.
>
> Thanks & Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions Pvt. Ltd.,
> The Enterprise Linux Company (r),
> http://www.enterux.com
> http://www.entVoice.com

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From pete at privateconnect.com  Sat Aug 29 14:25:31 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Sat, 29 Aug 2009 14:25:31 -0700
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Message-ID: <20090829142531.2ad02225396a31c9de30536f2e338977.911e38f47a.wbe@email04.secureserver.net>

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From pete at privateconnect.com  Sat Aug 29 14:52:29 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Sat, 29 Aug 2009 14:52:29 -0700
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Message-ID: <20090829145229.2ad02225396a31c9de30536f2e338977.c53fcdc891.wbe@email04.secureserver.net>

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From sdjernes at gmail.com  Sat Aug 29 14:55:41 2009
From: sdjernes at gmail.com (Shawn L. Djernes)
Date: Sat, 29 Aug 2009 16:55:41 -0500
Subject: [Freeswitch-users] Group Call
In-Reply-To: <81148697-292E-4E9C-A26A-D1536E9D8A8E@freeswitch.org>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
	<81148697-292E-4E9C-A26A-D1536E9D8A8E@freeswitch.org>
Message-ID: <5ff3e3a70908291455n61f34479t29f83b60528aadc0@mail.gmail.com>

This seems to be a problem in the 1.0.4 tarball release.  A friend with SVN
14305 the group_call from fs_cli works.
Shawn

On Sat, Aug 29, 2009 at 10:12 AM, Brian West <brian at freeswitch.org> wrote:

> Don't confuse this with the actual group/ endpoint.
>
> /b
>
> On Aug 25, 2009, at 11:41 AM, Jo?o Mesquita wrote:
>
> > According to wiki:
> > group,[insert|delete|call]:<group name>:<url>,group [insert|delete|
> > call
> >
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shawn L. Djernes
SD Consulting
shawn at djernes.org | sdjernes at gmail.com
MSN: wizardwlf at hotmail.com
402.345.7734 | 402.350.6973 Cell
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From brian at freeswitch.org  Sat Aug 29 17:14:08 2009
From: brian at freeswitch.org (Brian West)
Date: Sat, 29 Aug 2009 19:14:08 -0500
Subject: [Freeswitch-users] Group Call
In-Reply-To: <5ff3e3a70908291455n61f34479t29f83b60528aadc0@mail.gmail.com>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
	<81148697-292E-4E9C-A26A-D1536E9D8A8E@freeswitch.org>
	<5ff3e3a70908291455n61f34479t29f83b60528aadc0@mail.gmail.com>
Message-ID: <DE1C323C-ADE0-4575-993F-6D6BB6649AA6@freeswitch.org>

Group call works for me on mine... I suspect your config is not  
correct... tar up your directory and mail it to me off list if you can  
please.

freeswitch at default> version
FreeSWITCH Version 1.0.trunk (14665M)

freeswitch at default> group_call default
[presence_id=1000 at example.com]sofia/internal/sip:1000 at 192.168.1.31,[presence_id=1006 at example.com 
]sofia/internal/sip:gw+11 at 192.168.1.112:6060;transport=udp,[presence_id=1007 at example.com 
]sofia/internal/sip:1007 at 192.168.1.22, 
[presence_id=1015 at example.com]sofia/internal/sip:1015 at 192.168.1.101:5061;fs_nat=yes;fs_path=sip%3A1015%4080.93.247.26%3A1100
freeswitch at default> group_call sales
[presence_id=1000 at example.com]sofia/internal/sip:1000 at 192.168.1.31,[presence_id=1004 at example.com 
]sofia/internal/sip:gw+15 at 12.237.205.134:6060;transport=udp
freeswitch at default> group_call billing
[presence_id=1006 at example.com]sofia/internal/sip:gw+11 at 192.168.1.112:6060;transport=udp 
,[presence_id=1007 at example.com]sofia/internal/sip:1007 at 192.168.1.22
freeswitch at default>


Thanks,
/b

On Aug 29, 2009, at 4:55 PM, Shawn L. Djernes wrote:

> This seems to be a problem in the 1.0.4 tarball release.  A friend  
> with SVN 14305 the group_call from fs_cli works.
>
> Shawn




From tomabroad at gmail.com  Sat Aug 29 17:33:22 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 20:33:22 -0400
Subject: [Freeswitch-users] newbie questions
In-Reply-To: <7400B5BC-EC99-464E-B97F-D3DCD3CE2180@freeswitch.org>
References: <6f7c60c40908290646m42a47ac2v1d1755f6343a8b95@mail.gmail.com>
	<6DD54DEF-9B67-4EAE-937A-FB8543EC4CA7@freeswitch.org>
	<6f7c60c40908290819r55598a6fpd9ab49f4e4f23ba@mail.gmail.com>
	<153EF6B0-586E-4655-A0EB-76D7F03D0B3D@freeswitch.org>
	<6f7c60c40908290828g349920bbw7d4e6231407385fd@mail.gmail.com>
	<98E01B85-5134-48FA-9200-C01D96E48AA5@freeswitch.org>
	<6f7c60c40908290840w466c37d6v28dcc0f38f632cce@mail.gmail.com>
	<7400B5BC-EC99-464E-B97F-D3DCD3CE2180@freeswitch.org>
Message-ID: <6f7c60c40908291733t527068e4y92310ba83569e3fc@mail.gmail.com>

hi and thx for all ur help! i basically re-svned & installed it & internally
is fine now. playtime ;-)
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From tomabroad at gmail.com  Sat Aug 29 17:34:09 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 20:34:09 -0400
Subject: [Freeswitch-users] sipdroid
Message-ID: <6f7c60c40908291734k555b06f5hf321e6e2a514f5c2@mail.gmail.com>

hi,

sipdroid was working with asterisk, anyone running it with FS?

thx
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From tomabroad at gmail.com  Sat Aug 29 17:49:52 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 20:49:52 -0400
Subject: [Freeswitch-users] FS - external users
Message-ID: <6f7c60c40908291749sbb3e63do3011f8e251e2332a@mail.gmail.com>

hi,

as newbie here a simple one:

based on the std-installation, what do i need to do to have a user connected
from the outside. lets assume i want to leave 1000-1020 as they are, i make
a copy of 1020, call it 1021 and then?
should i leave it where it is? do i have to move the file? and what about he
conetxt?

thx
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From max.bridgewater at gmail.com  Sat Aug 29 18:30:10 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sat, 29 Aug 2009 21:30:10 -0400
Subject: [Freeswitch-users] What is the difference between preAnswer and
	ring_ready?
Message-ID: <c55d46950908291830w6b83a6a7h17a6faf67732d58a@mail.gmail.com>

Hi,
Assuming an inbound call, I have trouble understanding what the supposed
difference between the following two set of instructions is:

session.execute("ring_ready");
session.set("set", "ringback=/home/ring.wav");

and

session.preAnswer();
session.streamFile("/home/ring.wav")

In practice, however, the first scenario doesn't work for me although the
example here suggests otherwise:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready
Any idea?

Thanks,
Max.
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From jason at jasonjgw.net  Sat Aug 29 18:50:30 2009
From: jason at jasonjgw.net (Jason White)
Date: Sun, 30 Aug 2009 11:50:30 +1000
Subject: [Freeswitch-users] What is the difference between preAnswer and
 ring_ready?
In-Reply-To: <c55d46950908291830w6b83a6a7h17a6faf67732d58a@mail.gmail.com>
References: <c55d46950908291830w6b83a6a7h17a6faf67732d58a@mail.gmail.com>
Message-ID: <20090830015030.GA3163@jdc.jasonjgw.net>

Max Bridgewater <max.bridgewater at gmail.com> wrote:
> Hi,
> Assuming an inbound call, I have trouble understanding what the supposed
> difference between the following two set of instructions is:
> 
> session.execute("ring_ready");
> session.set("set", "ringback=/home/ring.wav");

Ring_ready sends a SIP "ringing" message and will use your device's default
ringback sound - at least, that has been my experience. I should remember the
message number in the SIP protocol, but I don't (the SIP experts here will
undoubtedly have the details at the ready).
> 
> and
> 
> session.preAnswer();
> session.streamFile("/home/ring.wav")

I think that will play the file as early media.




From tomabroad at gmail.com  Sat Aug 29 19:37:47 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 22:37:47 -0400
Subject: [Freeswitch-users] liverpie as gui-proxy?
Message-ID: <6f7c60c40908291937h57c632d5l84b6ec5ac1286c48@mail.gmail.com>

hi,

i was actually looking for something completely else, but reading wiki  led
me to the tought:
if mod_event can basically do all kinds of FS-things, why not having a
standalone ruby-on-rails as as gui which interacts with the evnet-socket via
liverpie?

im a total newbie to FS, just writing down my thoughts...


thx for ur replies
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From tomabroad at gmail.com  Sat Aug 29 19:55:14 2009
From: tomabroad at gmail.com (tom)
Date: Sat, 29 Aug 2009 22:55:14 -0400
Subject: [Freeswitch-users] FS: possible to create conference + password on
	the fly?
Message-ID: <6f7c60c40908291955i6417d51emfbade3aefc9cdf52@mail.gmail.com>

at best, from an external application, language is not important. as far i
have seen, could liverpie (aka the mod_event) do that? thx
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From sdjernes at gmail.com  Sat Aug 29 19:57:32 2009
From: sdjernes at gmail.com (Shawn L. Djernes)
Date: Sat, 29 Aug 2009 21:57:32 -0500
Subject: [Freeswitch-users] Group Call
In-Reply-To: <DE1C323C-ADE0-4575-993F-6D6BB6649AA6@freeswitch.org>
References: <5ff3e3a70908250730n28a3f257w967af7b513d38b4c@mail.gmail.com>
	<ebbc2d550908250941m25b122c5pc66063b4b50a8cef@mail.gmail.com>
	<81148697-292E-4E9C-A26A-D1536E9D8A8E@freeswitch.org>
	<5ff3e3a70908291455n61f34479t29f83b60528aadc0@mail.gmail.com>
	<DE1C323C-ADE0-4575-993F-6D6BB6649AA6@freeswitch.org>
Message-ID: <5ff3e3a70908291957u3b63721bjf60a37b939fced9b@mail.gmail.com>

The two systems in question are Debian with Freeswitch build using the
dpkg-buildpackage.

One of the systems has been customized but the other has a plain build and
install on it, so it is not the customization.

Is there a debian / ubuntu user on the list that can test and tell me if
this works for you.  The 14305 system is Fedora build with RPM so something
may be broken in the build system.

Thanks,
Shawn

On Sat, Aug 29, 2009 at 7:14 PM, Brian West <brian at freeswitch.org> wrote:

> Group call works for me on mine... I suspect your config is not
> correct... tar up your directory and mail it to me off list if you can
> please.
>
> freeswitch at default> version
> FreeSWITCH Version 1.0.trunk (14665M)
>
> freeswitch at default> group_call default
> [presence_id=1000 at example.com]sofia/internal/sip:1000 at 192.168.1.31<sip%3A1000 at 192.168.1.31>
> ,[presence_id=1006 at example.com
> ]sofia/internal/sip:gw+11 at 192.168.1.112:6060;transport=udp,[presence_id=
> 1007 at example.com
> ]sofia/internal/sip:1007 at 192.168.1.22 <sip%3A1007 at 192.168.1.22>,
> [presence_id=1015 at example.com]sofia/internal/sip:1015 at 192.168.1.101:5061
> ;fs_nat=yes;fs_path=sip%3A1015%4080.93.247.26%3A1100
> freeswitch at default> group_call sales
> [presence_id=1000 at example.com]sofia/internal/sip:1000 at 192.168.1.31<sip%3A1000 at 192.168.1.31>
> ,[presence_id=1004 at example.com
> ]sofia/internal/sip:gw+15 at 12.237.205.134:6060;transport=udp
> freeswitch at default> group_call billing
> [presence_id=1006 at example.com]sofia/internal/sip:gw+11 at 192.168.1.112:6060
> ;transport=udp
> ,[presence_id=1007 at example.com]sofia/internal/sip:1007 at 192.168.1.22<sip%3A1007 at 192.168.1.22>
> freeswitch at default>
>
>
> Thanks,
> /b
>
> On Aug 29, 2009, at 4:55 PM, Shawn L. Djernes wrote:
>
> > This seems to be a problem in the 1.0.4 tarball release.  A friend
> > with SVN 14305 the group_call from fs_cli works.
> >
> > Shawn
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Shawn L. Djernes
SD Consulting
shawn at djernes.org | sdjernes at gmail.com
MSN: wizardwlf at hotmail.com
402.345.7734 | 402.350.6973 Cell
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From steveu at coppice.org  Sat Aug 29 20:17:01 2009
From: steveu at coppice.org (Steve Underwood)
Date: Sun, 30 Aug 2009 11:17:01 +0800
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <20090829145229.2ad02225396a31c9de30536f2e338977.c53fcdc891.wbe@email04.secureserver.net>
References: <20090829145229.2ad02225396a31c9de30536f2e338977.c53fcdc891.wbe@email04.secureserver.net>
Message-ID: <4A99EF2D.3010704@coppice.org>

This sounds like so many "redundancy" projects that will probably offer 
nothing in the real world.

On 08/30/2009 05:52 AM, Pete Mueller wrote:
> I guess I should also mention that Xen is a side-project.
>
> When considering this issue for an existing production systems, we 
> chose to put as much HA into hardware as we can.  We are not concerned 
> with FS crashing, as so far we've never seen that happen (except when 
> our module caused it :)  So for each of our systems:
> - We have dual NIC cards (onboad NIC + PCI card) both bridged together 
> in case one fails
NICs hardly ever fail. Its the wiring which is the vulnerable area. How 
independent can you make the two wiring paths, when they come from the 
same box?
> - We have redundant power supplies.
Even with a good UPS, power fails more often than a high quality power 
supply. Just how independent are the two power sources feeding your two 
power supplies? Do you have two completely independent UPS sets? Do you 
have spacially diverse wiring from them?
> - We use Mirrored Solid State Disks for local storage (far better MTBF 
> than HDD, a lot faster too)
My experience so far is that SSD reliability is very poor, with entire 
drives disappearing, rather than just getting the odd bad sector. I 
guess to balance this, hard disk drive reliability seems to have 
plummeted in the last year or so, after several good years.
> - All but OS and speed-critical data is stored on a NAS device
NAS == more wiring. More wiring == more vulnerabilities. Are you sure 
your setup is a win? NAS tends to help keep the data secure, but it 
isn't good for reliable access to that data.
> - We have redundant DBs with Memcache infront for speed
>
> At the same time we chose to use COTS hardware (SuperMicro 
> chassis/MoBo) rather than the big-boys like IBM or Dell.  This kept 
> the overall cost per machine low.  Initially some were concerned that 
> not having a name like IBM on our servers would be concerning to some 
> potential clients.  The solution was to pay a company to deisgn and 
> build a custom face plate for the SuperMicro boxes.  Which oddly looks 
> more impressive to clients that a rack full of IBM faceplates.  It was 
> suprisingly low cost for the faceplates too.
Some years ago we made an entire custom chassis for off the shelf 
boards. The quotes for fabricating that in small numbers were all over 
the place, but we ended with a good quality chassis at low cost. Most 
off the shelf rack mount enclosures are really pricy, so it isn't that 
hard to match their price with a custom build. We ended up with  a 
better design (at least for our purposes) that cost us no more. It can 
really make your stuff stand out.

A simple respray of the front panel can achieve a distinctive look at 
low cost too. :-)
>
> For scalability, OpenSIPS was our choice.  There's a very nice 
> tutorial on their website on how to configure Load Balancing.

Regards,
Steve




From marc at kasteris.com  Sat Aug 29 20:24:46 2009
From: marc at kasteris.com (Marc Orenberg)
Date: Sat, 29 Aug 2009 20:24:46 -0700 (PDT)
Subject: [Freeswitch-users] prompt playback volume using mod_shout
Message-ID: <158584.63541.qm@web50801.mail.re2.yahoo.com>

Hi, 

I'm finding that when I record a caller's voice using mod_shout in MP3 format, the recording is too soft when I play it back.
I see there's a "volume" setting in shout.conf.xml, but it says, "Dont change these unless you are insane".
When I insanely try to increase this value, I don't here any difference in volume.
Is this the right knob, or is there some other way to do this?

Thanks,
Marc
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From mrene_lists at avgs.ca  Sat Aug 29 20:54:29 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sat, 29 Aug 2009 23:54:29 -0400
Subject: [Freeswitch-users] What is the difference between preAnswer and
	ring_ready?
In-Reply-To: <20090830015030.GA3163@jdc.jasonjgw.net>
References: <c55d46950908291830w6b83a6a7h17a6faf67732d58a@mail.gmail.com>
	<20090830015030.GA3163@jdc.jasonjgw.net>
Message-ID: <6B5963B0-C53A-4005-A14F-DD3140F29560@avgs.ca>

Hi,

In SIP, ring_ready will send a 180 ringing, pre answer will send a 183  
with a session description, so that some audio can be exchanged while  
the call isnt answered yet (used a lot for inband error messages etc).

Regards,

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 29-Aug-09, at 9:50 PM, Jason White wrote:

> Max Bridgewater <max.bridgewater at gmail.com> wrote:
>> Hi,
>> Assuming an inbound call, I have trouble understanding what the  
>> supposed
>> difference between the following two set of instructions is:
>>
>> session.execute("ring_ready");
>> session.set("set", "ringback=/home/ring.wav");
>
> Ring_ready sends a SIP "ringing" message and will use your device's  
> default
> ringback sound - at least, that has been my experience. I should  
> remember the
> message number in the SIP protocol, but I don't (the SIP experts  
> here will
> undoubtedly have the details at the ready).
>>
>> and
>>
>> session.preAnswer();
>> session.streamFile("/home/ring.wav")
>
> I think that will play the file as early media.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org




From dave at 3c.co.uk  Sat Aug 29 21:07:43 2009
From: dave at 3c.co.uk (David Knell)
Date: Sun, 30 Aug 2009 07:07:43 +0300
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <4A99EF2D.3010704@coppice.org>
References: <20090829145229.2ad02225396a31c9de30536f2e338977.c53fcdc891.wbe@email04.secureserver.net>
	<4A99EF2D.3010704@coppice.org>
Message-ID: <1251605263.5146.40.camel@dk-d820>

On Sun, 2009-08-30 at 11:17 +0800, Steve Underwood wrote:
> This sounds like so many "redundancy" projects that will probably offer 
> nothing in the real world.
> 
> On 08/30/2009 05:52 AM, Pete Mueller wrote:
> > I guess I should also mention that Xen is a side-project.
> >
> > When considering this issue for an existing production systems, we 
> > chose to put as much HA into hardware as we can.  We are not concerned 
> > with FS crashing, as so far we've never seen that happen (except when 
> > our module caused it :)  So for each of our systems:
> > - We have dual NIC cards (onboad NIC + PCI card) both bridged together 
> > in case one fails
> NICs hardly ever fail. Its the wiring which is the vulnerable area. How 
> independent can you make the two wiring paths, when they come from the 
> same box?

This is one area where you can do quite well.  A simple setup:
two machines (1, 2), two NICs (A, B) in each, two switches (S1, S2)
- wire up 1A <-> S1 <-> 2A, 1B <-> S2 <-> 2B
- run OSPF across the links
allows you to unplug any cable or any switch without interrupting
communications for more than a second or two if the OSPF timers are
suitably set.

This generalises nicely - we used to run two machines as web servers,
each advertising the same IP address via OSPF to the routers via a setup
like the one above.  Unplug any one thing, and the whole still worked.

Three complete power outages in the data center we were in in 18 months,
one of which took out a number of power supplies, neatly illustrated
Steve's point: our real-world reliability was determined elsewhere.

--Dave

-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk




From pete at privateconnect.com  Sat Aug 29 22:06:20 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Sat, 29 Aug 2009 22:06:20 -0700
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Message-ID: <20090829220620.2ad02225396a31c9de30536f2e338977.7a28828e7b.wbe@email04.secureserver.net>

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From m.krivushin at imarto.net  Sat Aug 29 22:06:41 2009
From: m.krivushin at imarto.net (Mikhail Krivushin)
Date: Sun, 30 Aug 2009 12:06:41 +0700
Subject: [Freeswitch-users] FS: possible to create conference + password
	on the fly?
In-Reply-To: <6f7c60c40908291955i6417d51emfbade3aefc9cdf52@mail.gmail.com>
References: <6f7c60c40908291955i6417d51emfbade3aefc9cdf52@mail.gmail.com>
Message-ID: <5be734a50908292206m3a6cafa0l6c02b17bba3aff01@mail.gmail.com>

I think you need to read documentation around a week, try to do something
with FS. And then most of our questions go out.

Yes, you can do that - dynamic conference with pass.

2009/8/30 tom <tomabroad at gmail.com>

> at best, from an external application, language is not important. as far i
> have seen, could liverpie (aka the mod_event) do that? thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From jason at jasonjgw.net  Sat Aug 29 22:34:45 2009
From: jason at jasonjgw.net (Jason White)
Date: Sun, 30 Aug 2009 15:34:45 +1000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <20090829220620.2ad02225396a31c9de30536f2e338977.7a28828e7b.wbe@email04.secureserver.net>
References: <20090829220620.2ad02225396a31c9de30536f2e338977.7a28828e7b.wbe@email04.secureserver.net>
Message-ID: <20090830053445.GA15076@jdc.jasonjgw.net>

Pete Mueller <pete at privateconnect.com> wrote:
>    There is a need for ensuring that calls do not drop, but we must balance
>    that with the cost of making the system redundant.? We took some small,
>    inexpensive measures, to improve our odds, but we could spend a lot more,
>    for basically nothing more than giving some client a warm fuzzy.

I think this is one area where, as indicated earlier in the thread, a lot of
development effort would be needed to obtain that extra degree of reliability.

>From a broader perspective, the question is whether, over the next decade or
two, VoIP can compete with the PSTN in reliability. My (limited) understanding
is that PSTN equipment typically achieves 99.99999% uptime, and if VoIP
systems are going to play in that arena, it would be desirable for
free/open-source software to do so.

If FreeSWITCH itself is working correctly, all you need is a hardware failure
or a kernel panic or a network outage to drop that up-time substantially, not
to mention dropping the calls as well, which I've never experienced as a user
of the PSTN due to equipment at the telephone exchange.

I have, however, experienced some rather low-quality PSTN calls over
international lines, which have the added disadvantage of being expensive to
use.




From jan.kubr at gmail.com  Sat Aug 29 23:28:43 2009
From: jan.kubr at gmail.com (Jan Kubr)
Date: Sun, 30 Aug 2009 08:28:43 +0200
Subject: [Freeswitch-users] liverpie as gui-proxy?
In-Reply-To: <6f7c60c40908291937h57c632d5l84b6ec5ac1286c48@mail.gmail.com>
References: <6f7c60c40908291937h57c632d5l84b6ec5ac1286c48@mail.gmail.com>
Message-ID: <698401620908292328w22ce398do3936d09beb1a1fb2@mail.gmail.com>

You can use this setup to control calls, but I believe not to change
configuration. Dunno what you're after.

On Sunday, August 30, 2009, tom <tomabroad at gmail.com> wrote:
> hi,
>
> i was actually looking for something completely else, but reading wiki? led me to the tought:
> if mod_event can basically do all kinds of FS-things, why not having a standalone ruby-on-rails as as gui which interacts with the evnet-socket via liverpie?
>
> im a total newbie to FS, just writing down my thoughts...
>
>
> thx for ur replies
>



From rs at runsolutions.com  Sun Aug 30 06:19:54 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Sun, 30 Aug 2009 15:19:54 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <B55A1F03-DC5E-4F32-99C8-3AB9D11B2C8A@freeswitch.org>
References: <C6BEE13A.5E372%krice@freeswitch.org>
	<A6E1A4F9-F0D5-4645-A72D-26365CF6FC7B@enterux.com>
	<B55A1F03-DC5E-4F32-99C8-3AB9D11B2C8A@freeswitch.org>
Message-ID: <FC48B8FE-9919-40EC-A878-5F06410186F2@runsolutions.com>

So, the first way would be to have a look on the sip stack, which is,  
in fact, sofia ..., well, sound's like a nice, fun, long-going hobby  
project to me :-)

I will definitly at least look at the sip code to check if it is  
something i would myself willingly give over to ...

But I *really* do want  a setup where it does not matter if one  
specific FS instance is UP or NOT ...


-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 29, 2009, at 10:04 PM, Brian West wrote:

> The sip stack needs to be modified to spin that data up into the  
> state machine so that it can take over calls once the fail over  
> takes place... its not an easy task.
>
> /b
>
> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>
>> Is itnpossible to have a db cluster know the state of each and  
>> every call and then use Heartbeat on this db +
>> fs cluster so that clients see only one ip where as internally all  
>> fs boxes refer db for call states, db again is under replication.
>>
>> This in the thioery can be written, but I am sure if we think bit  
>> more on this direction the problem seem to be getting addressed.
>>
>> Other guys also chip in their 2 cents, we just need 50 of em to  
>> make a full dollar.
>>
>> Thanks & Regards,
>> Mitul Limbani,
>> Founder & CEO,
>> Enterux Solutions Pvt. Ltd.,
>> The Enterprise Linux Company (r),
>> http://www.enterux.com
>> http://www.entVoice.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From max.bridgewater at gmail.com  Sun Aug 30 08:25:45 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 30 Aug 2009 11:25:45 -0400
Subject: [Freeswitch-users] Caller Authentication
Message-ID: <c55d46950908300825v6a298150y406931f286891ed@mail.gmail.com>

Guys,

This question might be a bit off-topic, so please bear with me as i could
hardly find a better place to solicit some help.
I suspect caller authentication can be challenging in the broader context of
VoIP/SIP. Now, if we restrict the set of calls to those coming from the GSM
network, is there a way to tell if the Caller Id is coming from the right
source?  It doesn't need to be a bulletproofed solution. For instance, if i
can already say that a given call is certainly coming from a GSM network
(and that the call metadata weren't tampered with), i could maybe rely on
the GSM authentication mechanism.

Any idea?
Max.
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From erwin.davis at gmail.com  Sat Aug 29 15:01:17 2009
From: erwin.davis at gmail.com (Erwin Huang)
Date: Sat, 29 Aug 2009 18:01:17 -0400
Subject: [Freeswitch-users] freeswitch compile error
Message-ID: <ec9a78d50908291501k111ee428la67ab73345bb1a9f@mail.gmail.com>

Hi, I am new to freeswitch. I downloaded 1.04 source tarball and I got error
during the compiling.
Below is the error message, please advise what was going wrong. Thanks,

mkdir .libs
Compiling mod_cepstral.c ...
mod_cepstral.c:41:19: error: swift.h: No such file or directory
mod_cepstral.c:53: error: expected '=', ',', ';', 'asm' or '__attribute__'
before '*' token
mod_cepstral.c:57: error: expected specifier-qualifier-list before
'swift_background_t'
mod_cepstral.c:69: error: expected '=', ',', ';', 'asm' or '__attribute__'
before 'write_audio'
mod_cepstral.c: In function 'cepstral_speech_open':
mod_cepstral.c:122: error: 'cepstral_t' has no member named 'audio_buffer'
mod_cepstral.c:128: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c:131: error: 'cepstral_t' has no member named 'params'
mod_cepstral.c:132: error: 'cepstral_t' has no member named 'params'
mod_cepstral.c:134: error: 'cepstral_t' has no member named 'params'
mod_cepstral.c:138: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:138: error: 'engine' undeclared (first use in this function)
mod_cepstral.c:138: error: (Each undeclared identifier is reported only once
mod_cepstral.c:138: error: for each function it appears in.)
mod_cepstral.c:138: error: 'cepstral_t' has no member named 'params'
mod_cepstral.c:144: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:151: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:151: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:157: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:157: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:162: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:169: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:169: error: 'write_audio' undeclared (first use in this
function)
mod_cepstral.c:169: error: 'SWIFT_EVENT_AUDIO' undeclared (first use in this
function)
mod_cepstral.c: In function 'cepstral_speech_close':
mod_cepstral.c:187: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:188: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:189: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:189: error: 'SWIFT_ASYNC_ANY' undeclared (first use in this
function)
mod_cepstral.c:189: error: 'SWIFT_EVENT_NOW' undeclared (first use in this
function)
mod_cepstral.c:191: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:192: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:195: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:198: error: 'cepstral_t' has no member named 'audio_buffer'
mod_cepstral.c: In function 'cepstral_speech_feed_tts':
mod_cepstral.c:213: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:214: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:216: error: 'cepstral_t' has no member named 'tts_stream'
mod_cepstral.c:226: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:226: error: 'cepstral_t' has no member named 'tts_stream'
mod_cepstral.c:234: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:234: error: 'cepstral_t' has no member named 'tts_stream'
mod_cepstral.c: In function 'cepstral_speech_flush_tts':
mod_cepstral.c:249: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:250: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:251: error: 'cepstral_t' has no member named 'audio_buffer'
mod_cepstral.c:252: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c:253: error: 'cepstral_t' has no member named 'audio_buffer'
mod_cepstral.c:254: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c:256: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:256: error: 'SWIFT_ASYNC_ANY' undeclared (first use in this
function)
mod_cepstral.c:256: error: 'SWIFT_EVENT_NOW' undeclared (first use in this
function)
mod_cepstral.c: In function 'cepstral_speech_read_tts':
mod_cepstral.c:270: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:271: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:272: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:272: error: 'cepstral_t' has no member named 'tts_stream'
mod_cepstral.c:274: error: 'SWIFT_STATUS_RUNNING' undeclared (first use in
this function)
mod_cepstral.c:275: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:279: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c:280: error: 'cepstral_t' has no member named 'audio_buffer'
mod_cepstral.c:281: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c:284: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:292: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:306: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c:307: error: 'cepstral_t' has no member named 'audio_buffer'
mod_cepstral.c:318: error: 'cepstral_t' has no member named 'audio_lock'
mod_cepstral.c: In function 'cepstral_text_param_tts':
mod_cepstral.c:337: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:337: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:338: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:338: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:339: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:339: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:342: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:347: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:355: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:355: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:357: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:357: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:362: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c:362: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:364: error: 'cepstral_t' has no member named 'done'
mod_cepstral.c:364: error: 'cepstral_t' has no member named 'done_gen'
mod_cepstral.c:368: error: 'cepstral_t' has no member named 'voice'
mod_cepstral.c:378: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c: In function 'cepstral_numeric_param_tts':
mod_cepstral.c:388: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c: In function 'cepstral_float_param_tts':
mod_cepstral.c:400: error: 'cepstral_t' has no member named 'port'
mod_cepstral.c: In function 'mod_cepstral_load':
mod_cepstral.c:409: error: 'engine' undeclared (first use in this function)
make[5]: *** [mod_cepstral.lo] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_cepstral-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
[root at proxy1 freeswitch]#
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From nandy1925 at gmail.com  Sat Aug 29 18:53:22 2009
From: nandy1925 at gmail.com (Nandy Dagondon)
Date: Sun, 30 Aug 2009 09:53:22 +0800
Subject: [Freeswitch-users] FS - external users
In-Reply-To: <6f7c60c40908291749sbb3e63do3011f8e251e2332a@mail.gmail.com>
References: <6f7c60c40908291749sbb3e63do3011f8e251e2332a@mail.gmail.com>
Message-ID: <7d0bfd8c0908291853q564bcf87tb566046988deb0d2@mail.gmail.com>

no need to modify anything on the directory entries - unless you add more
extension numbers. FS has a uPNP feature already. if your router has uPNP,
the external phone only needs the ff:
1. public IP of the FS (or hostname)
2. username
3. password

Items 2 & 3 are found in the directory entries.


On Sun, Aug 30, 2009 at 8:49 AM, tom <tomabroad at gmail.com> wrote:

> hi,
>
> as newbie here a simple one:
>
> based on the std-installation, what do i need to do to have a user
> connected from the outside. lets assume i want to leave 1000-1020 as they
> are, i make a copy of 1020, call it 1021 and then?
> should i leave it where it is? do i have to move the file? and what about
> he conetxt?
>
> thx
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From brian at freeswitch.org  Sun Aug 30 09:13:14 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 11:13:14 -0500
Subject: [Freeswitch-users] freeswitch compile error
In-Reply-To: <ec9a78d50908291501k111ee428la67ab73345bb1a9f@mail.gmail.com>
References: <ec9a78d50908291501k111ee428la67ab73345bb1a9f@mail.gmail.com>
Message-ID: <61A2778C-A1CA-4FD9-A2E5-6766C76CBCCF@freeswitch.org>

Unless you have installed the Cepstral SDK you can't compile  
mod_cepstral.  Please visit www.cepstral.com to purchase voices and  
the SDK is included.

/b

On Aug 29, 2009, at 5:01 PM, Erwin Huang wrote:

> mkdir .libs
> Compiling mod_cepstral.c ...
> mod_cepstral.c:41:19: error: swift.h: No such file or directory




From dave at 3c.co.uk  Sun Aug 30 09:25:17 2009
From: dave at 3c.co.uk (David Knell)
Date: Sun, 30 Aug 2009 19:25:17 +0300
Subject: [Freeswitch-users] Caller Authentication
In-Reply-To: <c55d46950908300825v6a298150y406931f286891ed@mail.gmail.com>
References: <c55d46950908300825v6a298150y406931f286891ed@mail.gmail.com>
Message-ID: <1251649517.6737.10.camel@dk-d820>

Hi Max,

I think that the answer is generally no - for example, I'm trivially
easily able to place a call to my GSM phone with any CLI that I choose.
If you can work around that, then consider the case where I divert my
GSM phone to you - the call to you will come from the GSM network, but I
can still set an arbitrary CLI.

Cheers --

Dave

> Guys,
> 
> This question might be a bit off-topic, so please bear with me as i
> could hardly find a better place to solicit some help.
> I suspect caller authentication can be challenging in the broader
> context of VoIP/SIP. Now, if we restrict the set of calls to those
> coming from the GSM network, is there a way to tell if the Caller Id
> is coming from the right source?  It doesn't need to be a
> bulletproofed solution. For instance, if i can already say that a
> given call is certainly coming from a GSM network (and that the call
> metadata weren't tampered with), i could maybe rely on the GSM
> authentication mechanism. 
> 
> Any idea?
> Max.
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk




From tomabroad at gmail.com  Sun Aug 30 09:25:47 2009
From: tomabroad at gmail.com (tom)
Date: Sun, 30 Aug 2009 12:25:47 -0400
Subject: [Freeswitch-users] FS - external users
In-Reply-To: <7d0bfd8c0908291853q564bcf87tb566046988deb0d2@mail.gmail.com>
References: <6f7c60c40908291749sbb3e63do3011f8e251e2332a@mail.gmail.com>
	<7d0bfd8c0908291853q564bcf87tb566046988deb0d2@mail.gmail.com>
Message-ID: <6f7c60c40908300925p5c4e62abk898c3025eea3725a@mail.gmail.com>

thx, all working now!

On Sat, Aug 29, 2009 at 9:53 PM, Nandy Dagondon <nandy1925 at gmail.com> wrote:

> no need to modify anything on the directory entries - unless you add more
> extension numbers. FS has a uPNP feature already. if your router has uPNP,
> the external phone only needs the ff:
> 1. public IP of the FS (or hostname)
> 2. username
> 3. password
>
> Items 2 & 3 are found in the directory entries.
>
>
> On Sun, Aug 30, 2009 at 8:49 AM, tom <tomabroad at gmail.com> wrote:
>
>> hi,
>>
>> as newbie here a simple one:
>>
>> based on the std-installation, what do i need to do to have a user
>> connected from the outside. lets assume i want to leave 1000-1020 as they
>> are, i make a copy of 1020, call it 1021 and then?
>> should i leave it where it is? do i have to move the file? and what about
>> he conetxt?
>>
>> thx
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From aep.lists at it46.se  Sun Aug 30 09:59:17 2009
From: aep.lists at it46.se (Alberto Escudero)
Date: Sun, 30 Aug 2009 18:59:17 +0200
Subject: [Freeswitch-users] freeswitch compile error
In-Reply-To: <61A2778C-A1CA-4FD9-A2E5-6766C76CBCCF@freeswitch.org>
References: <ec9a78d50908291501k111ee428la67ab73345bb1a9f@mail.gmail.com>
	<61A2778C-A1CA-4FD9-A2E5-6766C76CBCCF@freeswitch.org>
Message-ID: <1c14317d05b7457e797b3e94b0db02dc.squirrel@correo.nodo50.org>

You can always download it and test it for a while!. It will insert some
nice commercials when running TTS. I ended up buying a license, Allison
convinced me :) And if you plan to use TTS via PSTN or GSM, purchase a 8
Khz voice.

/aep
-- 
Stopping junk mailers is good for the environment

> Unless you have installed the Cepstral SDK you can't compile
> mod_cepstral.  Please visit www.cepstral.com to purchase voices and
> the SDK is included.
>
> /b
>
> On Aug 29, 2009, at 5:01 PM, Erwin Huang wrote:
>
>> mkdir .libs
>> Compiling mod_cepstral.c ...
>> mod_cepstral.c:41:19: error: swift.h: No such file or directory
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>





From max.bridgewater at gmail.com  Sun Aug 30 10:47:58 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 30 Aug 2009 13:47:58 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
Message-ID: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>

Hi,

I was trying to build freeswitch with cepstral enabled. Here is the procdure
i am following:
http://wiki.freeswitch.org/wiki/Mod_cepstral#Can_I_use_a_16khz_.22desktop_voice.22.3F

Unfortunately, the build fails as show in the log message hereunder.  In a
previous thread, Brian says that the SDK is needed. Yet on the above wiki
page, it is said that the SDK is not needed.
Any suggestion?

Max.

Compiling mod_cepstral.c ...
Creating mod_cepstral.so...
/usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when searching
for -lswift
/usr/bin/ld: cannot find -lswift
collect2: ld returned 1 exit status
gcc -w -I/home/installs/freeswitch/src/include
-I/home/installs/freeswitch/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -I/include
-I/opt/swift/include -shared -o .libs/mod_cepstral.so -shared -Wl,-x
.libs/mod_cepstral.o  -L/lib -L/opt/swift/lib -lswift -lm
/home/installs/freeswitch/.libs/libfreeswitch.so
-L/home/installs/freeswitch/libs/apr-util/xml/expat/lib
/home/installs/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/home/installs/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt
-lpthread -L/home/installs/freeswitch/libs/srtp -L/usr/kerberos/lib64 -lssl
-lcrypto -ldl -lz -lncurses   -Wl,--rpath -Wl,/usr/local/freeswitch/lib
-Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_cepstral.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_cepstral-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
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From brian at freeswitch.org  Sun Aug 30 10:58:16 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 12:58:16 -0500
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
Message-ID: <77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>

You still have to have the voices installed... the SDK comes with each  
voice for Linux ... for windows you have to have the SDK...

/b

On Aug 30, 2009, at 12:47 PM, Max Bridgewater wrote:

> Hi,
>
> I was trying to build freeswitch with cepstral enabled. Here is the  
> procdure i am following:
> http://wiki.freeswitch.org/wiki/Mod_cepstral#Can_I_use_a_16khz_.22desktop_voice.22.3F
>
> Unfortunately, the build fails as show in the log message  
> hereunder.  In a previous thread, Brian says that the SDK is needed.  
> Yet on the above wiki page, it is said that the SDK is not needed.
> Any suggestion?
>
> Max.

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From max.bridgewater at gmail.com  Sun Aug 30 11:09:31 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 30 Aug 2009 14:09:31 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
	<77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
Message-ID: <c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>

OK, it's bad and good news. It's good news because it's reassuring that i
don't need a separate SDK. Its bad news because i already installed a voice
(Jean-Pierre) following the procedure describe on the wiki. I will now
delete it and install the Allison voice to see if there is any difference.
Will let ya know.

On Sun, Aug 30, 2009 at 1:58 PM, Brian West <brian at freeswitch.org> wrote:

> You still have to have the voices installed... the SDK comes with each
> voice for Linux ... for windows you have to have the SDK...
> /b
>
> On Aug 30, 2009, at 12:47 PM, Max Bridgewater wrote:
>
> Hi,
>
> I was trying to build freeswitch with cepstral enabled. Here is the
> procdure i am following:
>
> http://wiki.freeswitch.org/wiki/Mod_cepstral#Can_I_use_a_16khz_.22desktop_voice.22.3F
>
> Unfortunately, the build fails as show in the log message hereunder.  In a
> previous thread, Brian says that the SDK is needed. Yet on the above wiki
> page, it is said that the SDK is not needed.
> Any suggestion?
>
> Max.
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From max.bridgewater at gmail.com  Sun Aug 30 11:33:24 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 30 Aug 2009 14:33:24 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
	<77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
	<c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
Message-ID: <c55d46950908301133r2888ebffq8bfb93bbaf9e24a4@mail.gmail.com>

Even more bad news. I downloaded the Allison voice and have the same error
message. Any idea what this means?


mkdir .libs
Compiling mod_cepstral.c ...
Creating mod_cepstral.so...
/usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when searching
for -lswift
/usr/bin/ld: cannot find -lswift
collect2: ld returned 1 exit status
gcc -w -I/home/installs/freeswitch/src/include
-I/home/installs/freeswitch/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -I/opt/swift/include
-I/opt/swift/include -shared -o .libs/mod_cepstral.so -shared -Wl,-x
.libs/mod_cepstral.o  -L/opt/swift/lib -lswift -lm
/home/installs/freeswitch/.libs/libfreeswitch.so
-L/home/installs/freeswitch/libs/apr-util/xml/expat/lib
/home/installs/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/home/installs/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt
-lpthread -L/home/installs/freeswitch/libs/srtp -L/usr/kerberos/lib64 -lssl
-lcrypto -ldl -lz -lncurses   -Wl,--rpath -Wl,/usr/local/freeswitch/lib
-Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_cepstral.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_cepstral-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
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From brian at freeswitch.org  Sun Aug 30 11:36:38 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 13:36:38 -0500
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
	<77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
	<c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
Message-ID: <2ABAA784-4B60-4AB0-932C-616BE3102904@freeswitch.org>

You also failed to read this:

http://wiki.freeswitch.org/wiki/Mod_cepstral#libswift.so. 
4:_cannot_open_shared_object_file:_No_such_file_or_directory

You have to add the directory to your ld.so.conf so it can find -lswift.

/b

On Aug 30, 2009, at 1:09 PM, Max Bridgewater wrote:

> OK, it's bad and good news. It's good news because it's reassuring  
> that i don't need a separate SDK. Its bad news because i already  
> installed a voice (Jean-Pierre) following the procedure describe on  
> the wiki. I will now delete it and install the Allison voice to see  
> if there is any difference. Will let ya know.
>
> On Sun, Aug 30, 2009 at 1:58 PM, Brian West <brian at freeswitch.org>  
> wrote:
> You still have to have the voices installed... the SDK comes with  
> each voice for Linux ... for windows you have to have the SDK...
>
> /b
>
> On Aug 30, 2009, at 12:47 PM, Max Bridgewater wrote:
>
>> Hi,
>>
>> I was trying to build freeswitch with cepstral enabled. Here is the  
>> procdure i am following:
>> http://wiki.freeswitch.org/wiki/Mod_cepstral#Can_I_use_a_16khz_.22desktop_voice.22.3F
>>
>> Unfortunately, the build fails as show in the log message  
>> hereunder.  In a previous thread, Brian says that the SDK is  
>> needed. Yet on the above wiki page, it is said that the SDK is not  
>> needed.
>> Any suggestion?
>>
>> Max.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org

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From brian at freeswitch.org  Sun Aug 30 11:46:56 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 13:46:56 -0500
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <c55d46950908301133r2888ebffq8bfb93bbaf9e24a4@mail.gmail.com>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
	<77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
	<c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
	<c55d46950908301133r2888ebffq8bfb93bbaf9e24a4@mail.gmail.com>
Message-ID: <42B40E01-ACBC-4303-8D91-53D2DD1C97A0@freeswitch.org>

Yes please add /opt/swift/lib to /etc/ld.so.conf as per the FAQ.

/b

On Aug 30, 2009, at 1:33 PM, Max Bridgewater wrote:

> /usr/bin/ld: cannot find -lswift




From max.bridgewater at gmail.com  Sun Aug 30 11:59:36 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 30 Aug 2009 14:59:36 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <2ABAA784-4B60-4AB0-932C-616BE3102904@freeswitch.org>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
	<77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
	<c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
	<2ABAA784-4B60-4AB0-932C-616BE3102904@freeswitch.org>
Message-ID: <c55d46950908301159j46d06904k2f2a97ecb9915650@mail.gmail.com>

Please bear with me. Here is the content of my  /etc/ld.so.conf:

include ld.so.conf.d/*.conf
/opt/swift/lib

And when i do "ldconfig -p | grep swift" , i get the following:

   libswift.so.5 (libc6) => /opt/swift/lib/libswift.so.5
   libswift.so (libc6) => /opt/swift/lib/libswift.so
   libceplex_us.so.5 (libc6) => /opt/swift/lib/libceplex_us.so.5
   libceplex_us.so (libc6) => /opt/swift/lib/libceplex_us.so
   libceplang_en.so.5 (libc6) => /opt/swift/lib/libceplang_en.so.5
   libceplang_en.so (libc6) => /opt/swift/lib/libceplang_en.so

So, can it be that Freeswitch is looking for libswift.so.4 instead of
libswift.so.5 (as the first line of the wiki page is actually saying)? If
so, where do i get this 4.x version? I can't find it in on cepstral site.

Max.
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From pbd at suspiria.net  Sun Aug 30 12:00:38 2009
From: pbd at suspiria.net (Public Dump)
Date: Sun, 30 Aug 2009 21:00:38 +0200
Subject: [Freeswitch-users] delay buildup in conference
References: <13C421883438EB42B9E2C30069FD4AB77004E8C0A2@crushinator.central.local>
	<191c3a030908261156t13c23cc9xada4b727abbe1b78@mail.gmail.com>
	<13C421883438EB42B9E2C30069FD4AB77004E8C0A8@crushinator.central.local>
	<191c3a030908261345t6141e55r11f1d057cc628cf4@mail.gmail.com> 
Message-ID: <13C421883438EB42B9E2C30069FD4AB77004E8C0F7@crushinator.central.local>

I have tested the suggested setting:

<settings>

      <param name="rtp-autoflush-during-bridge" value="true"/>



But the problem remains. L



Von: Public Dump
Gesendet: Mittwoch, 26. August 2009 23:37
An: 'freeswitch-users at lists.freeswitch.org'
Betreff: AW: [Freeswitch-users] delay buildup in conference



I am quite sure that it's not a problem with the network connection, since the delay builds up gradually over time, while the host is reachable without any problems.



I have added the suggested config lines and will check the results.



By the way, can I remap the hash key, so that is does not end the conference ? I've tried remapping it, but freeswitch seems to ignore the setting (the console shows: hangup mapped to '*' ... but # still ends the call).

It is irritating for users, since most users try to enter the conference pin the #-key at the end.



Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 22:45
An: freeswitch-users at lists.freeswitch.org
Betreff: Re: [Freeswitch-users] delay buildup in conference



There is no bug, it's all dependent on your network conditions.
I spend literally 8-12 hours a day on a conference and there is no delay.

The important param is
<param name="rtp-autoflush-during-bridge" value="true"/>

in the sofia profile in question.

if you have delay with that in place, then it's probably not FS

On Wed, Aug 26, 2009 at 3:30 PM, Public Dump <pbd at suspiria.net<mailto:pbd at suspiria.net>> wrote:

Hi,



I am on a quite recent version (i assume):



FreeSWITCH Version 1.0.trunk (14461)



Should the bug be fixed in this revision ? What config settings would a have to check to limit delay (even at the cost of reduction in quality).



Thanks



Von: freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org> [mailto:freeswitch-users-bounces at lists.freeswitch.org<mailto:freeswitch-users-bounces at lists.freeswitch.org>] Im Auftrag von Anthony Minessale
Gesendet: Mittwoch, 26. August 2009 20:57
An: freeswitch-users at lists.freeswitch.org<mailto:freeswitch-users at lists.freeswitch.org>
Betreff: Re: [Freeswitch-users] delay buildup in conference



which revision are you on?
The defaults on the latest code and examples should be configured to minimize delay.
Some of the older revisions built up some delay issues from udp buffering when timers were not synced.

On Wed, Aug 26, 2009 at 12:28 PM, Public Dump <pbd at suspiria.net<mailto:pbd at suspiria.net>> wrote:

When running conferences with users dialed in from a PSTN gateway (SIP) and directly from remote SIP endpoints there is an ever longer buildup in delay, reaching up to multiple seconds. Is there any way to limit the delay ?



I am not 100% sure whether the delays is caused by the SIP jitter buffer of freeswitch or directly by the conference module.



Any advice?



Thanks


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--
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FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

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MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
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sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400


_______________________________________________
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--
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com<mailto:MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<mailto:PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net<http://irc.freenode.net> #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org<mailto:sip%3A888 at conference.freeswitch.org>
iax:guest at conference.freeswitch.org/888<http://iax:guest at conference.freeswitch.org/888>
googletalk:conf+888 at conference.freeswitch.org<mailto:googletalk%3Aconf%2B888 at conference.freeswitch.org>
pstn:213-799-1400

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From brian at freeswitch.org  Sun Aug 30 12:10:40 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 14:10:40 -0500
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <c55d46950908301159j46d06904k2f2a97ecb9915650@mail.gmail.com>
References: <c55d46950908301047p58876105v2e5607f1ee468289@mail.gmail.com>
	<77AD885A-8536-4F24-92F8-920C71074E2F@freeswitch.org>
	<c55d46950908301109g9c8a276i2761fae63bbf2c@mail.gmail.com>
	<2ABAA784-4B60-4AB0-932C-616BE3102904@freeswitch.org>
	<c55d46950908301159j46d06904k2f2a97ecb9915650@mail.gmail.com>
Message-ID: <83278230-3C0C-40B1-85A6-1CF2ED19DBF6@freeswitch.org>

make sure you run ldconfig after you add the line then try to compile  
it again.


/b

On Aug 30, 2009, at 1:59 PM, Max Bridgewater wrote:

> Please bear with me. Here is the content of my  /etc/ld.so.conf:
>
> include ld.so.conf.d/*.conf
> /opt/swift/lib




From pete at privateconnect.com  Sun Aug 30 12:33:32 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Sun, 30 Aug 2009 12:33:32 -0700
Subject: [Freeswitch-users]
	=?utf-8?q?Troubles_build_with_mod=5Fcepstral?=
Message-ID: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>

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From mrene_lists at avgs.ca  Sun Aug 30 12:35:41 2009
From: mrene_lists at avgs.ca (Mathieu Rene)
Date: Sun, 30 Aug 2009 15:35:41 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
References: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
Message-ID: <7CF0A496-7F1A-4CC7-A1F5-88C17BD990DF@avgs.ca>

... skipping incompatible /opt/swift/lib/libswift.so when searching  
for -lswift thats a sign you're trying to link a 64 bits binary with a  
32 bits one.

Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mrene at avgs.ca




On 30-Aug-09, at 3:33 PM, Pete Mueller wrote:

>  skipping incompatible /opt/swift/lib/libswift.so when searching for  
> -lswift

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From brian at freeswitch.org  Sun Aug 30 13:51:41 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 15:51:41 -0500
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
References: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
Message-ID: <7B458C2D-8DC2-40B9-8EF1-B7FC934A5F0F@freeswitch.org>

http://downloads.cepstral.com/cepstral/x86-64-linux/Cepstral_Allison_x86-64-linux_5.1.0.tar.gz

Make sure you download the 64bit version.

/b



On Aug 30, 2009, at 2:33 PM, Pete Mueller wrote:

>  I don't think Cepstral supports 64-bit, as the SDK is 32-bit.




From mike at jerris.com  Sun Aug 30 14:45:21 2009
From: mike at jerris.com (Michael Jerris)
Date: Sun, 30 Aug 2009 17:45:21 -0400
Subject: [Freeswitch-users] Transferring to another extension is VERY
	slow.
In-Reply-To: <3d381e170908291121k3bd64ecnacf4f7941962e2ed@mail.gmail.com>
References: <3d381e170908291121k3bd64ecnacf4f7941962e2ed@mail.gmail.com>
Message-ID: <93769c20908301445l37bf8d65j6d00523f75eb3002@mail.gmail.com>

Try reviewing the debug log to see why it is delayed.  You may want to look
at the inline sip trace at all to see if anything is amiss there as well.
Mike


On Sat, Aug 29, 2009 at 2:21 PM, Erik Wickstrom <erik at erikwickstrom.com>wrote:

> Hi,
>
> I'm having some trouble debugging why the transfer command is so slow
> in my case.  I have a lua IVR setup which responds to user selections
> by transferring the call to another dialplan ext.  Pretty basic stuff.
>
> The problem is, it takes 30-40 seconds for the the transfer to
> actually start after the key press.  After about 30 seconds of
> silence, "hold music" finally starts and the other extension finally
> answers shortly thereafter.
>
> I've tried transferring to endpoint extensions, the 5000 sample IVR,
> 9999 hold music -- in all cases there is this long pause.
>
> Does anyone know what could be causing this?
>
> Thanks!
> Erik
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Sun Aug 30 14:49:13 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Sun, 30 Aug 2009 17:49:13 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <7B458C2D-8DC2-40B9-8EF1-B7FC934A5F0F@freeswitch.org>
References: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
	<7B458C2D-8DC2-40B9-8EF1-B7FC934A5F0F@freeswitch.org>
Message-ID: <c55d46950908301449u6bb3d43cifd92a353cca6cae5@mail.gmail.com>

Nop, the 64bit  cepstral doesn't work either:

Creating mod_cepstral.so...
/usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when searching
for -lswift
/usr/bin/ld: cannot find -lswift
collect2: ld returned 1 exit status
gcc -w -I/home/installs/freeswitch/src/include
-I/home/installs/freeswitch/libs/libteletone/src -fPIC -Werror
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb
-g -O2 -Wall -std=c99 -pedantic -D_GNU_SOURCE -I/opt/swift/include
-I/opt/swift/include -shared -o .libs/mod_cepstral.so -shared -Wl,-x
.libs/mod_cepstral.o  -L/opt/swift/lib -lswift -lm
/home/installs/freeswitch/.libs/libfreeswitch.so
-L/home/installs/freeswitch/libs/apr-util/xml/expat/lib
/home/installs/freeswitch/libs/apr-util/xml/expat/lib/.libs/libexpat.a
/home/installs/freeswitch/libs/apr/.libs/libapr-1.a -luuid -lrt -lcrypt
-lpthread -L/home/installs/freeswitch/libs/srtp -L/usr/kerberos/lib64 -lssl
-lcrypto -ldl -lz -lncurses   -Wl,--rpath -Wl,/usr/local/freeswitch/lib
-Wl,--rpath -Wl,/usr/local/freeswitch/mod
make[5]: *** [mod_cepstral.so] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_cepstral-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
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From brian at freeswitch.org  Sun Aug 30 14:53:11 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 16:53:11 -0500
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <c55d46950908301449u6bb3d43cifd92a353cca6cae5@mail.gmail.com>
References: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
	<7B458C2D-8DC2-40B9-8EF1-B7FC934A5F0F@freeswitch.org>
	<c55d46950908301449u6bb3d43cifd92a353cca6cae5@mail.gmail.com>
Message-ID: <399999D5-EF3E-444F-8E06-05A0006DD053@freeswitch.org>

Make sure you rm -rf /opt/swift then reinstall

/b

On Aug 30, 2009, at 4:49 PM, Max Bridgewater wrote:

> Nop, the 64bit  cepstral doesn't work either:
>
> Creating mod_cepstral.so...
> /usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when  
> searching for -lswift
> /usr/bin/ld: cannot find -lswift
> collect2: ld returned 1 exit status




From Prometheus001 at gmx.net  Sun Aug 30 15:25:01 2009
From: Prometheus001 at gmx.net (Peter P GMX)
Date: Mon, 31 Aug 2009 00:25:01 +0200
Subject: [Freeswitch-users] sofia_reg_external in odbc?
Message-ID: <4A9AFC3D.9060208@gmx.net>

Hello,

is there a chance to have sofia_reg_external in odbc/mysql instead of
sqlite?
In a B2BUA environment we have thousand of external registrations during
a migration phase, and it would be good to have easy external control
over the registered gateways (like in fs_internal. sip_registrations).

Best regards
Peter




From brian at freeswitch.org  Sun Aug 30 16:43:50 2009
From: brian at freeswitch.org (Brian West)
Date: Sun, 30 Aug 2009 18:43:50 -0500
Subject: [Freeswitch-users] sofia_reg_external in odbc?
In-Reply-To: <4A9AFC3D.9060208@gmx.net>
References: <4A9AFC3D.9060208@gmx.net>
Message-ID: <F939B13C-83EB-44D2-928A-24728F2A6DDF@freeswitch.org>

<param name="odbc-dsn" value="dsn:user:pass"/>

On the profile.

/b


On Aug 30, 2009, at 5:25 PM, Peter P GMX wrote:

> Hello,
>
> is there a chance to have sofia_reg_external in odbc/mysql instead of
> sqlite?
> In a B2BUA environment we have thousand of external registrations  
> during
> a migration phase, and it would be good to have easy external control
> over the registered gateways (like in fs_internal. sip_registrations).
>
> Best regards
> Peter




From jim at evolutiontel.net  Sun Aug 30 19:54:50 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 31 Aug 2009 12:54:50 +1000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <FC48B8FE-9919-40EC-A878-5F06410186F2@runsolutions.com>
References: <C6BEE13A.5E372%krice@freeswitch.org>
	<A6E1A4F9-F0D5-4645-A72D-26365CF6FC7B@enterux.com>
	<B55A1F03-DC5E-4F32-99C8-3AB9D11B2C8A@freeswitch.org>
	<FC48B8FE-9919-40EC-A878-5F06410186F2@runsolutions.com>
Message-ID: <ddad80610908301954i37ecdf96i78b6cb3dc9a2c633@mail.gmail.com>

High Availability is always an interesting problem.  PSTN equipment
generally overcomes it through duplication of messaging and hardware.
Obviously at an E1 (T1 for the US folks) or Subscribers Line Level
this is not possible.  This duplication comes at a cost and lowers the
calls per minute a softswitch can handle for the amount of hardware
purchased.  Like everything it is a trade-off between cost and
reliability.

Per Dave's comment, duplication on a network level can be achieved
easily through duplicating network switches and using OSPF, VRRP or
network bonding to ensure continuity if a path is compromised.  From
experience this is exactly what the vendors do and it does work.
During failover the calls experience a minut breakup of voice quality
but they still hold up ok.

Conceptually in a pure SIP installation of Freeswitch, hot standby or
active standby could be achieved through duplication of messages
(INVITE, 200OK and BYE for calls) between 2 boxes and then using VRRP
to change the IP address when the box falls over.  Going this way
allows call state to be as up to date as possible.  Obviously this
would require logic added to sofia to transfer the RTP port
information and UUID information between the FS instances (Easily
achieved using additional SIP headers).  It would also require that
sofia does not forward duplicated messages to gateways or user agents.
 CDR information would also need to be generated correctly (My
experience of Radius with Opensips is that it generates start and stop
messages on INVITE/200OK then BYE respectively so this could be an
easy solution to the CDR issue).

Regards,



On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
> So, the first way would be to have a look on the sip stack, which is, in
> fact, sofia ..., well, sound's like a nice, fun, long-going hobby project to
> me :-)
> I will definitly at least look at the sip code to check if it is something i
> would myself willingly give over to ...
> But I *really* do want ?a setup where it does not matter if one specific FS
> instance is UP or NOT ...
>
> --
> Raimund Sacherer
> -
> RunSolutions
> ????Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 - ?Palma de Mallorca
> Baleares
> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>
> The sip stack needs to be modified to spin that data up into the state
> machine so that it can take over calls once the fail over takes place... its
> not an easy task.
> /b
> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>
> Is itnpossible to have a db cluster know the state of each and every call
> and then use Heartbeat on this db +
> fs cluster so that clients see only one ip where as internally all fs boxes
> refer db for call states, db again is under replication.
> This in the thioery can be written, but I am sure if we think bit more on
> this direction the problem seem to be getting addressed.
> Other guys also chip in their 2 cents, we just need 50 of em to make a full
> dollar.
>
> Thanks & Regards,
> Mitul Limbani,
> Founder & CEO,
> Enterux Solutions Pvt. Ltd.,
> The Enterprise Linux Company (r),
> http://www.enterux.com
> http://www.entVoice.com
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net



From krice at freeswitch.org  Sun Aug 30 20:23:40 2009
From: krice at freeswitch.org (Ken Rice)
Date: Sun, 30 Aug 2009 22:23:40 -0500
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <ddad80610908301954i37ecdf96i78b6cb3dc9a2c633@mail.gmail.com>
Message-ID: <C6C0AC6C.5E412%krice@freeswitch.org>

Your forgetting that with RTP theres the duplication of RTP that may be
required along with a mechanism that keeps media timers working properly...

Where many things don't support RTP timers they sure come in handy on a
regular basis especially dealing with several of the other software based
platforms out there that just love to have calls end but never want to send
a BYE


> From: Jim Burke <jim at evolutiontel.net>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 31 Aug 2009 12:54:50 +1000
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
> 
> Conceptually in a pure SIP installation of Freeswitch, hot standby or
> active standby could be achieved through duplication of messages
> (INVITE, 200OK and BYE for calls) between 2 boxes and then using VRRP
> to change the IP address when the box falls over.  Going this way
> allows call state to be as up to date as possible.  Obviously this
> would require logic added to sofia to transfer the RTP port
> information and UUID information between the FS instances (Easily
> achieved using additional SIP headers).  It would also require that
> sofia does not forward duplicated messages to gateways or user agents.
>  CDR information would also need to be generated correctly (My
> experience of Radius with Opensips is that it generates start and stop
> messages on INVITE/200OK then BYE respectively so this could be an
> easy solution to the CDR issue).
> 
> Regards,
> 
> 
> 
> On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
>> So, the first way would be to have a look on the sip stack, which is, in
>> fact, sofia ..., well, sound's like a nice, fun, long-going hobby project to
>> me :-)
>> I will definitly at least look at the sip code to check if it is something i
>> would myself willingly give over to ...
>> But I *really* do want ?a setup where it does not matter if one specific FS
>> instance is UP or NOT ...
>> 
>> --
>> Raimund Sacherer
>> -
>> RunSolutions
>> ????Open Source It Consulting
>> -
>> 
>> Parc Bit - Centro Empresarial Son Espanyol
>> Edificio Estel - Local 3D
>> 07121 - ?Palma de Mallorca
>> Baleares
>> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>> 
>> The sip stack needs to be modified to spin that data up into the state
>> machine so that it can take over calls once the fail over takes place... its
>> not an easy task.
>> /b
>> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>> 
>> Is itnpossible to have a db cluster know the state of each and every call
>> and then use Heartbeat on this db +
>> fs cluster so that clients see only one ip where as internally all fs boxes
>> refer db for call states, db again is under replication.
>> This in the thioery can be written, but I am sure if we think bit more on
>> this direction the problem seem to be getting addressed.
>> Other guys also chip in their 2 cents, we just need 50 of em to make a full
>> dollar.
>> 
>> Thanks & Regards,
>> Mitul Limbani,
>> Founder & CEO,
>> Enterux Solutions Pvt. Ltd.,
>> The Enterprise Linux Company (r),
>> http://www.enterux.com
>> http://www.entVoice.com
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
>> 
> 
> 
> 
> -- 
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From jim at evolutiontel.net  Sun Aug 30 21:32:22 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Mon, 31 Aug 2009 14:32:22 +1000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <C6C0AC6C.5E412%krice@freeswitch.org>
References: <ddad80610908301954i37ecdf96i78b6cb3dc9a2c633@mail.gmail.com>
	<C6C0AC6C.5E412%krice@freeswitch.org>
Message-ID: <ddad80610908302132i4804be0eg3ea7a7ae2c8a1bfa@mail.gmail.com>

Hi Ken,

I don't disagree, the scenario you mentioned is just the tip of the
iceberg when it comes to getting something like this working smoothly.
 However to try and cover them all in a single post (assuming that I
could) would have resulted in something akin to War and Peace and
probably bored the pants off of most readers :)  For any HA solution
to move from a concept into reality, it would require some serious
brainstorming to identify all scenarios and how to handle them.

Given that many posts in this thread mentioned transfer of call state
at time of failure or some form of virtualisation, I wanted to provide
another solution that would provide real time call state updates
between FS boxes utilising mechanisms that are largely already
in-place and thus (hopefully) reducing software development time if
someone was willing to cough up the $$$$$.

Regards,

On Mon, Aug 31, 2009 at 1:23 PM, Ken Rice<krice at freeswitch.org> wrote:
> Your forgetting that with RTP theres the duplication of RTP that may be
> required along with a mechanism that keeps media timers working properly...
>
> Where many things don't support RTP timers they sure come in handy on a
> regular basis especially dealing with several of the other software based
> platforms out there that just love to have calls end but never want to send
> a BYE
>
>
>> From: Jim Burke <jim at evolutiontel.net>
>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>> Date: Mon, 31 Aug 2009 12:54:50 +1000
>> To: <freeswitch-users at lists.freeswitch.org>
>> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
>>
>> Conceptually in a pure SIP installation of Freeswitch, hot standby or
>> active standby could be achieved through duplication of messages
>> (INVITE, 200OK and BYE for calls) between 2 boxes and then using VRRP
>> to change the IP address when the box falls over. ?Going this way
>> allows call state to be as up to date as possible. ?Obviously this
>> would require logic added to sofia to transfer the RTP port
>> information and UUID information between the FS instances (Easily
>> achieved using additional SIP headers). ?It would also require that
>> sofia does not forward duplicated messages to gateways or user agents.
>> ?CDR information would also need to be generated correctly (My
>> experience of Radius with Opensips is that it generates start and stop
>> messages on INVITE/200OK then BYE respectively so this could be an
>> easy solution to the CDR issue).
>>
>> Regards,
>>
>>
>>
>> On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
>>> So, the first way would be to have a look on the sip stack, which is, in
>>> fact, sofia ..., well, sound's like a nice, fun, long-going hobby project to
>>> me :-)
>>> I will definitly at least look at the sip code to check if it is something i
>>> would myself willingly give over to ...
>>> But I *really* do want ?a setup where it does not matter if one specific FS
>>> instance is UP or NOT ...
>>>
>>> --
>>> Raimund Sacherer
>>> -
>>> RunSolutions
>>> ????Open Source It Consulting
>>> -
>>>
>>> Parc Bit - Centro Empresarial Son Espanyol
>>> Edificio Estel - Local 3D
>>> 07121 - ?Palma de Mallorca
>>> Baleares
>>> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>>>
>>> The sip stack needs to be modified to spin that data up into the state
>>> machine so that it can take over calls once the fail over takes place... its
>>> not an easy task.
>>> /b
>>> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>>>
>>> Is itnpossible to have a db cluster know the state of each and every call
>>> and then use Heartbeat on this db +
>>> fs cluster so that clients see only one ip where as internally all fs boxes
>>> refer db for call states, db again is under replication.
>>> This in the thioery can be written, but I am sure if we think bit more on
>>> this direction the problem seem to be getting addressed.
>>> Other guys also chip in their 2 cents, we just need 50 of em to make a full
>>> dollar.
>>>
>>> Thanks & Regards,
>>> Mitul Limbani,
>>> Founder & CEO,
>>> Enterux Solutions Pvt. Ltd.,
>>> The Enterprise Linux Company (r),
>>> http://www.enterux.com
>>> http://www.entVoice.com
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>>>
>>>
>>
>>
>>
>> --
>> Jim Burke
>> Director Evolutiontel.
>> http://www.evolutiontel.net
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net



From krice at freeswitch.org  Sun Aug 30 22:37:00 2009
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 31 Aug 2009 00:37:00 -0500
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <ddad80610908302132i4804be0eg3ea7a7ae2c8a1bfa@mail.gmail.com>
Message-ID: <C6C0CBAC.5E420%krice@freeswitch.org>

I was thinking more like something the size of war and peace but written
with the style of an RFC (and about as interesting and an RFC)

I know there has already been some discussion on several fronts of atleast
getting the core and several other pieces to where they need to be for
stateful failover and I'm not sure if its been mentioned here, but sofia is
going to be a bit of work and estimates run in the 100K USD range. Now if we
could get a get a couple of corporate sponsors to help here it would be
great.

K


> From: Jim Burke <jim at evolutiontel.net>
> Reply-To: <freeswitch-users at lists.freeswitch.org>
> Date: Mon, 31 Aug 2009 14:32:22 +1000
> To: <freeswitch-users at lists.freeswitch.org>
> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
> 
> Hi Ken,
> 
> I don't disagree, the scenario you mentioned is just the tip of the
> iceberg when it comes to getting something like this working smoothly.
>  However to try and cover them all in a single post (assuming that I
> could) would have resulted in something akin to War and Peace and
> probably bored the pants off of most readers :)  For any HA solution
> to move from a concept into reality, it would require some serious
> brainstorming to identify all scenarios and how to handle them.
> 
> Given that many posts in this thread mentioned transfer of call state
> at time of failure or some form of virtualisation, I wanted to provide
> another solution that would provide real time call state updates
> between FS boxes utilising mechanisms that are largely already
> in-place and thus (hopefully) reducing software development time if
> someone was willing to cough up the $$$$$.
> 
> Regards,
> 
> On Mon, Aug 31, 2009 at 1:23 PM, Ken Rice<krice at freeswitch.org> wrote:
>> Your forgetting that with RTP theres the duplication of RTP that may be
>> required along with a mechanism that keeps media timers working properly...
>> 
>> Where many things don't support RTP timers they sure come in handy on a
>> regular basis especially dealing with several of the other software based
>> platforms out there that just love to have calls end but never want to send
>> a BYE
>> 
>> 
>>> From: Jim Burke <jim at evolutiontel.net>
>>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>>> Date: Mon, 31 Aug 2009 12:54:50 +1000
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
>>> 
>>> Conceptually in a pure SIP installation of Freeswitch, hot standby or
>>> active standby could be achieved through duplication of messages
>>> (INVITE, 200OK and BYE for calls) between 2 boxes and then using VRRP
>>> to change the IP address when the box falls over. ?Going this way
>>> allows call state to be as up to date as possible. ?Obviously this
>>> would require logic added to sofia to transfer the RTP port
>>> information and UUID information between the FS instances (Easily
>>> achieved using additional SIP headers). ?It would also require that
>>> sofia does not forward duplicated messages to gateways or user agents.
>>> ?CDR information would also need to be generated correctly (My
>>> experience of Radius with Opensips is that it generates start and stop
>>> messages on INVITE/200OK then BYE respectively so this could be an
>>> easy solution to the CDR issue).
>>> 
>>> Regards,
>>> 
>>> 
>>> 
>>> On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com>
>>> wrote:
>>>> So, the first way would be to have a look on the sip stack, which is, in
>>>> fact, sofia ..., well, sound's like a nice, fun, long-going hobby project
>>>> to
>>>> me :-)
>>>> I will definitly at least look at the sip code to check if it is something
>>>> i
>>>> would myself willingly give over to ...
>>>> But I *really* do want ?a setup where it does not matter if one specific FS
>>>> instance is UP or NOT ...
>>>> 
>>>> --
>>>> Raimund Sacherer
>>>> -
>>>> RunSolutions
>>>> ????Open Source It Consulting
>>>> -
>>>> 
>>>> Parc Bit - Centro Empresarial Son Espanyol
>>>> Edificio Estel - Local 3D
>>>> 07121 - ?Palma de Mallorca
>>>> Baleares
>>>> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>>>> 
>>>> The sip stack needs to be modified to spin that data up into the state
>>>> machine so that it can take over calls once the fail over takes place...
>>>> its
>>>> not an easy task.
>>>> /b
>>>> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>>>> 
>>>> Is itnpossible to have a db cluster know the state of each and every call
>>>> and then use Heartbeat on this db +
>>>> fs cluster so that clients see only one ip where as internally all fs boxes
>>>> refer db for call states, db again is under replication.
>>>> This in the thioery can be written, but I am sure if we think bit more on
>>>> this direction the problem seem to be getting addressed.
>>>> Other guys also chip in their 2 cents, we just need 50 of em to make a full
>>>> dollar.
>>>> 
>>>> Thanks & Regards,
>>>> Mitul Limbani,
>>>> Founder & CEO,
>>>> Enterux Solutions Pvt. Ltd.,
>>>> The Enterprise Linux Company (r),
>>>> http://www.enterux.com
>>>> http://www.entVoice.com
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>>> http://www.freeswitch.org
>>>> 
>>>> 
>>> 
>>> 
>>> 
>>> --
>>> Jim Burke
>>> Director Evolutiontel.
>>> http://www.evolutiontel.net
>>> 
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>>> http://www.freeswitch.org
>> 
>> 
>> 
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>> 
> 
> 
> 
> -- 
> Jim Burke
> Director Evolutiontel.
> http://www.evolutiontel.net
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org





From jason at jasonjgw.net  Sun Aug 30 22:57:44 2009
From: jason at jasonjgw.net (Jason White)
Date: Mon, 31 Aug 2009 15:57:44 +1000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <C6C0CBAC.5E420%krice@freeswitch.org>
References: <ddad80610908302132i4804be0eg3ea7a7ae2c8a1bfa@mail.gmail.com>
	<C6C0CBAC.5E420%krice@freeswitch.org>
Message-ID: <20090831055744.GA10477@jdc.jasonjgw.net>

Ken Rice <krice at freeswitch.org> wrote:
 
> I know there has already been some discussion on several fronts of atleast
> getting the core and several other pieces to where they need to be for
> stateful failover and I'm not sure if its been mentioned here, but sofia is
> going to be a bit of work and estimates run in the 100K USD range. Now if we
> could get a get a couple of corporate sponsors to help here it would be
> great.

I agree. Perhaps some of those corporations which are saving money over
proprietary solutions by using FreeSWITCH, and who also want the
high-availability functionality, would be ideal sponsors for the work.




From rs at runsolutions.com  Mon Aug 31 00:04:12 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Mon, 31 Aug 2009 09:04:12 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <20090831055744.GA10477@jdc.jasonjgw.net>
References: <ddad80610908302132i4804be0eg3ea7a7ae2c8a1bfa@mail.gmail.com>
	<C6C0CBAC.5E420%krice@freeswitch.org>
	<20090831055744.GA10477@jdc.jasonjgw.net>
Message-ID: <0F51835A-3E93-47D4-BBFF-623C25AC67D1@runsolutions.com>

Shouldn't be so hard, right? But there's human nature, even if the  
corporations get to save, like, 100k worth of licenses and hardware  
and other costs it is very hard to get them to spend a little back to  
the community which helped to achive that goal.

I think my first, very ambitous goal to make the sofia rewrite a  
personal long-term hobby is not going to be very realistic, because of  
my daytime job and my private life.

But I will try to persuade my employer to make at least a little  
contribution, as I really want this in FS, maybe we should

* make it a bounty project
* try to bring in some big sponsors (who are the biggest companys  
using FS anyway?)
* make some awareness in the voip scene??

seriously what else can be done to get the estimated 100k to become  
reality?

I really want this feature in FS! :-)

best,

-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 31, 2009, at 7:57 AM, Jason White wrote:

> Ken Rice <krice at freeswitch.org> wrote:
>
>> I know there has already been some discussion on several fronts of  
>> atleast
>> getting the core and several other pieces to where they need to be  
>> for
>> stateful failover and I'm not sure if its been mentioned here, but  
>> sofia is
>> going to be a bit of work and estimates run in the 100K USD range.  
>> Now if we
>> could get a get a couple of corporate sponsors to help here it  
>> would be
>> great.
>
> I agree. Perhaps some of those corporations which are saving money  
> over
> proprietary solutions by using FreeSWITCH, and who also want the
> high-availability functionality, would be ideal sponsors for the work.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From anatoliy at kounitskiy.com  Mon Aug 31 02:15:10 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Mon, 31 Aug 2009 12:15:10 +0300
Subject: [Freeswitch-users] Question about mod_limit
Message-ID: <1cd828b60908310215n60bdb242r3210a140b0811538@mail.gmail.com>

I have question about mod_limit: is there a reason to have difference
between the api applications and the dialplan? I mean when you use db
function from the cli you can use insert/select/delete but when you
use it from the dialplan you can only use insert/delete.

#define DB_USAGE "[insert|delete]/<realm>/<key>/<val>"
#define DB_DESC "save data"

SWITCH_STANDARD_APP(db_function)

but in the
SWITCH_STANDARD_API(db_api_function)
i have three options insert,delete and select

What i try to accomplish is to be able to insert values from the
dialplan and after that to be able to use them ( to select and use the
result) from another extension.

Thank you in advanced,
-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540



From pete at privateconnect.com  Mon Aug 31 03:33:43 2009
From: pete at privateconnect.com (Pete Mueller)
Date: Mon, 31 Aug 2009 03:33:43 -0700
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
Message-ID: <20090831033343.2ad02225396a31c9de30536f2e338977.a975af96c2.wbe@email04.secureserver.net>

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From erwin.davis at gmail.com  Mon Aug 31 04:31:52 2009
From: erwin.davis at gmail.com (Erwin Huang)
Date: Mon, 31 Aug 2009 07:31:52 -0400
Subject: [Freeswitch-users] some errors from FS startup
Message-ID: <ec9a78d50908310431g120729f8jc5a4302aa74b9840@mail.gmail.com>

Hi, I was able to build the FS successfully and make a couple of test calls.
But when I looked at the startup logs, I saw some errors. Please point out
how to fix them. Thanks,

Regards,

e

2009-08-31 06:59:55.54063 [ERR] mod_xml_curl.c:404 Binding has no url!
2009-08-31 06:59:55.54094 [CRIT] switch_loadable_module.c:871 Error Loading
modu
le /usr/local/freeswitch/mod/mod_xml_curl.so
**Module load routine returned an error**


2009-08-31 06:59:57.393133 [CONSOLE] mod_local_stream.c:142 Can't open
directory
: /usr/local/freeswitch/sounds/music/8000
2009-08-31 06:59:57.404228 [CONSOLE] mod_local_stream.c:142 Can't open
directory
: /usr/local/freeswitch/sounds/music/8000
2009-08-31 06:59:57.414708 [CONSOLE] mod_local_stream.c:142 Can't open
directory
: /usr/local/freeswitch/sounds/music/16000
2009-08-31 06:59:57.424568 [CONSOLE] mod_local_stream.c:142 Can't open
directory
: /usr/local/freeswitch/sounds/music/32000


// The errors below indicate the files donot exist. How to include these two
files into FS make build?

2009-08-31 06:59:57.477374 [CRIT] switch_loadable_module.c:871 Error Loading
mod
ule /usr/local/freeswitch/mod/mod_java.so
**/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No
such
 file or directory**


2009-08-31 06:59:57.485242 [CRIT] switch_loadable_module.c:871 Error Loading
mod
ule /usr/local/freeswitch/mod/mod_flite.so
**/usr/local/freeswitch/mod/mod_flite.so: cannot open shared object file: No
suc
h file or directory**
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From rs at runsolutions.com  Mon Aug 31 04:50:16 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Mon, 31 Aug 2009 13:50:16 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <20090831033343.2ad02225396a31c9de30536f2e338977.a975af96c2.wbe@email04.secureserver.net>
References: <20090831033343.2ad02225396a31c9de30536f2e338977.a975af96c2.wbe@email04.secureserver.net>
Message-ID: <237DA68E-B77C-489C-8CCB-35C241B7C8A8@runsolutions.com>

Hello Pete,

wow, i like it when you say getting the money is the easiest part!

I think if you are able to help procure the money we should have, as  
Jim Burke indicated, a rather good brainstorming session behind us.
I will create today a Wiki Page where we can define the Project, break  
up the development parts, generelly layout what has to be done and  
what will be the best and cleanest way to do it, as well define what  
and when to failover, etc.

If we have a pretty clear Idea on what will be the best way, you can  
get and fetch the money :-)

Seriously, i think we should discuss the possibility with Anthony how,  
if you/the community can come up with the money, this could be  
integrated into the development cycle!

Let's keep this rolling,
best

-- 
Raimund Sacherer
-
RunSolutions
     Open Source It Consulting
-

Parc Bit - Centro Empresarial Son Espanyol
Edificio Estel - Local 3D
07121 -  Palma de Mallorca
Baleares

On Aug 31, 2009, at 12:33 PM, Pete Mueller wrote:

> Actually, getting the 100K is probably the easiest part of the  
> battle.  The problem will come in with the strings attached to the  
> 100K.  I could persuade my company to give FS devs the money, but  
> there would be conditions, such as:
> - There would need to be a defined timeline, milestones, and release  
> date
> - That release cannot be too far in the future (no one will fund an  
> 18 month project any more)
> - If those dates are missed, penalties will be applied
> - There will be some of branding/marketing/PR agreement (turning the  
> payment of development into a marketing tool)
> - There will be all sorts of legal waivers the devs will have to  
> sign (indemnifications, perpetual licenses, etc)
>
> From previous OSS experience, the "baggage" that comes with a large  
> payment is usually undesirable to dev groups.  Which leave  
> collecting a lot of little payments from many places.  Anyone that  
> has worked for some time fund raising for an organization will know  
> how much of a logistical mess that is. Not to mention people that  
> commit to pay, then back out.
>
> One good way to get funding for a project like this is to target the  
> government or non-industry related corporate entities.  Both cases  
> you target groups that don't compete in your market space (or rely  
> on your competitors), but would benefit from the outcome of the  
> project.
>
> An example would be Dell.  Dell would not see FS as competition in  
> any way, but if the case was made that adding feature X would  
> increase sales of Dell computers in the telcom space, or that  
> Company X would buy Dells to run FS if this features existed, then  
> Dell could look at investing in the project as a marketing expense.
>
> There are down sides with this method. Devs need to walk a fine line  
> with the corporation, ensuring that their product does not conflict  
> with some corporate initiative or large client.  Many times you will  
> be partnering with VARS or SIs to present the proposal, as there has  
> to be a project to justify the work.  Also, expect a long sales  
> cycle (5+ months). And, some purists would argue that the app is not  
> FOSS once your get some large corp to fund all/part of it.  Other  
> are just content with getting the job done.
>
> I offer my help in this area if the devs are interested in exploring  
> this style of fund raising.
> -pete
>
> -------- Original Message --------
> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
> From: Raimund Sacherer <rs at runsolutions.com>
> Date: Mon, August 31, 2009 12:04 am
> To: freeswitch-users at lists.freeswitch.org
>
> Shouldn't be so hard, right? But there's human nature, even if the
> corporations get to save, like, 100k worth of licenses and hardware
> and other costs it is very hard to get them to spend a little back to
> the community which helped to achive that goal.
>
> I think my first, very ambitous goal to make the sofia rewrite a
> personal long-term hobby is not going to be very realistic, because of
> my daytime job and my private life.
>
> But I will try to persuade my employer to make at least a little
> contribution, as I really want this in FS, maybe we should
>
> * make it a bounty project
> * try to bring in some big sponsors (who are the biggest companys
> using FS anyway?)
> * make some awareness in the voip scene??
>
> seriously what else can be done to get the estimated 100k to become
> reality?
>
> I really want this feature in FS! :-)
>
> best,
>
> -- 
> Raimund Sacherer
> -
> RunSolutions
> Open Source It Consulting
> -
>
> Parc Bit - Centro Empresarial Son Espanyol
> Edificio Estel - Local 3D
> 07121 - Palma de Mallorca
> Baleares
>
> On Aug 31, 2009, at 7:57 AM, Jason White wrote:
>
> > Ken Rice <krice at freeswitch.org> wrote:
> >
> >> I know there has already been some discussion on several fronts of
> >> atleast
> >> getting the core and several other pieces to where they need to be
> >> for
> >> stateful failover and I'm not sure if its been mentioned here, but
> >> sofia is
> >> going to be a bit of work and estimates run in the 100K USD range.
> >> Now if we
> >> could get a get a couple of corporate sponsors to help here it
> >> would be
> >> great.
> >
> > I agree. Perhaps some of those corporations which are saving money
> > over
> > proprietary solutions by using FreeSWITCH, and who also want the
> > high-availability functionality, would be ideal sponsors for the  
> work.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
> > users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org

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From dave at 3c.co.uk  Mon Aug 31 05:22:23 2009
From: dave at 3c.co.uk (David Knell)
Date: Mon, 31 Aug 2009 15:22:23 +0300
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <C6C0AC6C.5E412%krice@freeswitch.org>
References: <C6C0AC6C.5E412%krice@freeswitch.org>
Message-ID: <1251721344.6737.32.camel@dk-d820>

A random thought.  It strikes me that adding support for live call
failover to FreeSWITCH might well be the wrong thing to do - it's
something which is probably not too difficult to achieve if starting
from the ground up, but which is going to be a right PITA to do
retrospectively.  And, once you've done switching, there's conferencing,
IVR stuff...

Here's an alternative approach.  Have two FS boxes (which I'll call FS1
and FS2 - you can't fault *my* imagination) and a front-end B2BUA/RTP
proxy with a bit of a difference.  The B2BUA sends incoming stuff to
both FS1 and FS2 as two separate call legs, and the RTP proxy forwards
inbound RTP to both FS1 and FS2.  Assuming that FS1 is the current
"master", it forwards RTP and call state transitions from FS1 back to
the caller.  If it detects that FS1 has gone away, then - as FS2 should
be in roughly the same state as FS1 - it flips to using FS2 as its
source.

The internal state of the B2BUA/RTP proxy should be relatively simple,
and therefore should be able to be duplicated to a hot standby.

I can see a collection of gotchas, such as registration (the B2BUA needs
access to the user database), and I haven't even begun to try to think
of how to deal with calls *from* the FS boxes.  Actually, that's not
true: I have, and they clearly need to go through the same sort of
mechanism in reverse, which is fine until I start trying to work out how
to pair up calls from FS1 and FS2 reliably, and then my head hurts.

But I think that this idea has merit.  Firstly, it abstracts the
failover mechanism away into one place; secondly, it is general - FS1
and FS2 could just as easily be Asterisk1 and Asterisk2.

And I'll do it for just $98,650..

--Dave

> Your forgetting that with RTP theres the duplication of RTP that may be
> required along with a mechanism that keeps media timers working properly...
> 
> Where many things don't support RTP timers they sure come in handy on a
> regular basis especially dealing with several of the other software based
> platforms out there that just love to have calls end but never want to send
> a BYE
> 
> 
> > From: Jim Burke <jim at evolutiontel.net>
> > Reply-To: <freeswitch-users at lists.freeswitch.org>
> > Date: Mon, 31 Aug 2009 12:54:50 +1000
> > To: <freeswitch-users at lists.freeswitch.org>
> > Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
> > 
> > Conceptually in a pure SIP installation of Freeswitch, hot standby or
> > active standby could be achieved through duplication of messages
> > (INVITE, 200OK and BYE for calls) between 2 boxes and then using VRRP
> > to change the IP address when the box falls over.  Going this way
> > allows call state to be as up to date as possible.  Obviously this
> > would require logic added to sofia to transfer the RTP port
> > information and UUID information between the FS instances (Easily
> > achieved using additional SIP headers).  It would also require that
> > sofia does not forward duplicated messages to gateways or user agents.
> >  CDR information would also need to be generated correctly (My
> > experience of Radius with Opensips is that it generates start and stop
> > messages on INVITE/200OK then BYE respectively so this could be an
> > easy solution to the CDR issue).
> > 
> > Regards,
> > 
> > 
> > 
> > On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
> >> So, the first way would be to have a look on the sip stack, which is, in
> >> fact, sofia ..., well, sound's like a nice, fun, long-going hobby project to
> >> me :-)
> >> I will definitly at least look at the sip code to check if it is something i
> >> would myself willingly give over to ...
> >> But I *really* do want  a setup where it does not matter if one specific FS
> >> instance is UP or NOT ...
> >> 
> >> --
> >> Raimund Sacherer
> >> -
> >> RunSolutions
> >>     Open Source It Consulting
> >> -
> >> 
> >> Parc Bit - Centro Empresarial Son Espanyol
> >> Edificio Estel - Local 3D
> >> 07121 -  Palma de Mallorca
> >> Baleares
> >> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
> >> 
> >> The sip stack needs to be modified to spin that data up into the state
> >> machine so that it can take over calls once the fail over takes place... its
> >> not an easy task.
> >> /b
> >> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
> >> 
> >> Is itnpossible to have a db cluster know the state of each and every call
> >> and then use Heartbeat on this db +
> >> fs cluster so that clients see only one ip where as internally all fs boxes
> >> refer db for call states, db again is under replication.
> >> This in the thioery can be written, but I am sure if we think bit more on
> >> this direction the problem seem to be getting addressed.
> >> Other guys also chip in their 2 cents, we just need 50 of em to make a full
> >> dollar.
> >> 
> >> Thanks & Regards,
> >> Mitul Limbani,
> >> Founder & CEO,
> >> Enterux Solutions Pvt. Ltd.,
> >> The Enterprise Linux Company (r),
> >> http://www.enterux.com
> >> http://www.entVoice.com
> >> 
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> 
> >> 
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >> 
> >> 
> > 
> > 
> > 
> > -- 
> > Jim Burke
> > Director Evolutiontel.
> > http://www.evolutiontel.net
> > 
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
W: http://www.3c.co.uk




From fraunhofer.lists.freeswitch-001 at traced.net  Mon Aug 31 05:22:19 2009
From: fraunhofer.lists.freeswitch-001 at traced.net (Benedikt Fraunhofer)
Date: Mon, 31 Aug 2009 14:22:19 +0200
Subject: [Freeswitch-users] memory leak
In-Reply-To: <86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
	<3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
	<20090828001050.GA8108@jdc.jasonjgw.net>
	<86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com>
Message-ID: <a37510170908310522n73753a45sa346e6a3f255fcb9@mail.gmail.com>

Hello *,

still no luck. Removing the default dialplan with all those
hash-inserts made the memory-usage-curve flatter but it oom-coredumped
stating:

  freeswitch: src/switch_core_memory.c:443: switch_core_perform_alloc:
Assertion `ptr != ((void *)0)' failed.

after eating up 3.2 gig ram+swap.

Any chance or idea to track that down or any hint where to look at?

The other two freeswitch-machines in this scenario play well, they're
dumb load-sinks. what makes this machine different is on one hand the
lua-scripts that are run on it and on the other hand it's the only
machine who gets it's commands via xml_rpc.

Cheers
  Beni.



From rs at runsolutions.com  Mon Aug 31 05:36:48 2009
From: rs at runsolutions.com (Raimund Sacherer)
Date: Mon, 31 Aug 2009 14:36:48 +0200
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <1251721344.6737.32.camel@dk-d820>
References: <C6C0AC6C.5E412%krice@freeswitch.org>
	<1251721344.6737.32.camel@dk-d820>
Message-ID: <C99339FA-1E87-4A23-9C25-FB7D564B83CB@runsolutions.com>

Hi David,

On Aug 31, 2009, at 2:22 PM, David Knell wrote:

> A random thought.  It strikes me that adding support for live call
> failover to FreeSWITCH might well be the wrong thing to do - it's
> something which is probably not too difficult to achieve if starting
> from the ground up, but which is going to be a right PITA to do
> retrospectively.  And, once you've done switching, there's  
> conferencing,
> IVR stuff...
>
> Here's an alternative approach.  Have two FS boxes (which I'll call  
> FS1
> and FS2 - you can't fault *my* imagination) and a front-end B2BUA/RTP
> proxy with a bit of a difference.  The B2BUA sends incoming stuff to
> both FS1 and FS2 as two separate call legs, and the RTP proxy forwards
> inbound RTP to both FS1 and FS2.  Assuming that FS1 is the current
> "master", it forwards RTP and call state transitions from FS1 back to
> the caller.  If it detects that FS1 has gone away, then - as FS2  
> should
> be in roughly the same state as FS1 - it flips to using FS2 as its
> source.
>
> The internal state of the B2BUA/RTP proxy should be relatively simple,
> and therefore should be able to be duplicated to a hot standby.
>
> I can see a collection of gotchas, such as registration (the B2BUA  
> needs
> access to the user database), and I haven't even begun to try to think
> of how to deal with calls *from* the FS boxes.  Actually, that's not
> true: I have, and they clearly need to go through the same sort of
> mechanism in reverse, which is fine until I start trying to work out  
> how
> to pair up calls from FS1 and FS2 reliably, and then my head hurts.
Yeah, sounds like a real challenge

>
> But I think that this idea has merit.  Firstly, it abstracts the
> failover mechanism away into one place; secondly, it is general - FS1
> and FS2 could just as easily be Asterisk1 and Asterisk2.
this, in theory, sound quite interesting ...
would be nice to have a proof of concept setup thought.

-> Sound's not quite THAT nice than a full-built-in solution to FS ...  
but if it works ...


>
> And I'll do it for just $98,650..
Haha, that's really funny :-)


>
> --Dave
>
>> Your forgetting that with RTP theres the duplication of RTP that  
>> may be
>> required along with a mechanism that keeps media timers working  
>> properly...
>>
>> Where many things don't support RTP timers they sure come in handy  
>> on a
>> regular basis especially dealing with several of the other software  
>> based
>> platforms out there that just love to have calls end but never want  
>> to send
>> a BYE
>>
>>
>>> From: Jim Burke <jim at evolutiontel.net>
>>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>>> Date: Mon, 31 Aug 2009 12:54:50 +1000
>>> To: <freeswitch-users at lists.freeswitch.org>
>>> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
>>>
>>> Conceptually in a pure SIP installation of Freeswitch, hot standby  
>>> or
>>> active standby could be achieved through duplication of messages
>>> (INVITE, 200OK and BYE for calls) between 2 boxes and then using  
>>> VRRP
>>> to change the IP address when the box falls over.  Going this way
>>> allows call state to be as up to date as possible.  Obviously this
>>> would require logic added to sofia to transfer the RTP port
>>> information and UUID information between the FS instances (Easily
>>> achieved using additional SIP headers).  It would also require that
>>> sofia does not forward duplicated messages to gateways or user  
>>> agents.
>>> CDR information would also need to be generated correctly (My
>>> experience of Radius with Opensips is that it generates start and  
>>> stop
>>> messages on INVITE/200OK then BYE respectively so this could be an
>>> easy solution to the CDR issue).
>>>
>>> Regards,
>>>
>>>
>>>
>>> On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com 
>>> > wrote:
>>>> So, the first way would be to have a look on the sip stack, which  
>>>> is, in
>>>> fact, sofia ..., well, sound's like a nice, fun, long-going hobby  
>>>> project to
>>>> me :-)
>>>> I will definitly at least look at the sip code to check if it is  
>>>> something i
>>>> would myself willingly give over to ...
>>>> But I *really* do want  a setup where it does not matter if one  
>>>> specific FS
>>>> instance is UP or NOT ...
>>>>
>>>> --
>>>> Raimund Sacherer
>>>> -
>>>> RunSolutions
>>>>    Open Source It Consulting
>>>> -
>>>>
>>>> Parc Bit - Centro Empresarial Son Espanyol
>>>> Edificio Estel - Local 3D
>>>> 07121 -  Palma de Mallorca
>>>> Baleares
>>>> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>>>>
>>>> The sip stack needs to be modified to spin that data up into the  
>>>> state
>>>> machine so that it can take over calls once the fail over takes  
>>>> place... its
>>>> not an easy task.
>>>> /b
>>>> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>>>>
>>>> Is itnpossible to have a db cluster know the state of each and  
>>>> every call
>>>> and then use Heartbeat on this db +
>>>> fs cluster so that clients see only one ip where as internally  
>>>> all fs boxes
>>>> refer db for call states, db again is under replication.
>>>> This in the thioery can be written, but I am sure if we think bit  
>>>> more on
>>>> this direction the problem seem to be getting addressed.
>>>> Other guys also chip in their 2 cents, we just need 50 of em to  
>>>> make a full
>>>> dollar.
>>>>
>>>> Thanks & Regards,
>>>> Mitul Limbani,
>>>> Founder & CEO,
>>>> Enterux Solutions Pvt. Ltd.,
>>>> The Enterprise Linux Company (r),
>>>> http://www.enterux.com
>>>> http://www.entVoice.com
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ 
>>>> freeswitch-users
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>
>>>
>>>
>>> -- 
>>> Jim Burke
>>> Director Evolutiontel.
>>> http://www.evolutiontel.net
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>>> users
>>> http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
>> users
>> http://www.freeswitch.org
>>
> -- 
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From enno.egbert at googlemail.com  Mon Aug 31 06:11:26 2009
From: enno.egbert at googlemail.com (NOx-WHV)
Date: Mon, 31 Aug 2009 06:11:26 -0700 (PDT)
Subject: [Freeswitch-users]  external Gateways
Message-ID: <25223286.post@talk.nabble.com>


Hello,

i use a sip gateway witch is only be able to allocate one sip channel.  How
can I bridge a call to a second gateway if the first gateway is in use. 

Now the dialplan look like this:

  <condition field="${toll_allow}" expression="domestic"/>
  <condition field="destination_number" expression="^04(\d+)$">
   <action application="set" data="effective_caller_id_number=300"/>
   <action application="set" data="sip_secure_media=false"/>
   <action application="bridge"
data="sofia/gateway/mygateway/04$1 at XXX.XXX.XXX.XXX"/>
  </condition> 

Thanks for help

 
-- 
View this message in context: http://www.nabble.com/external-Gateways-tp25223286p25223286.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.




From brian at freeswitch.org  Mon Aug 31 06:32:53 2009
From: brian at freeswitch.org (Brian West)
Date: Mon, 31 Aug 2009 08:32:53 -0500
Subject: [Freeswitch-users] memory leak
In-Reply-To: <a37510170908310522n73753a45sa346e6a3f255fcb9@mail.gmail.com>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
	<3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
	<20090828001050.GA8108@jdc.jasonjgw.net>
	<86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com>
	<a37510170908310522n73753a45sa346e6a3f255fcb9@mail.gmail.com>
Message-ID: <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org>

Use valgrind.

/b

On Aug 31, 2009, at 7:22 AM, Benedikt Fraunhofer wrote:

>
> The other two freeswitch-machines in this scenario play well, they're
> dumb load-sinks. what makes this machine different is on one hand the
> lua-scripts that are run on it and on the other hand it's the only
> machine who gets it's commands via xml_rpc.




From rupa at rupa.com  Mon Aug 31 07:07:41 2009
From: rupa at rupa.com (Rupa Schomaker)
Date: Mon, 31 Aug 2009 09:07:41 -0500
Subject: [Freeswitch-users] memory leak
In-Reply-To: <8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org>
References: <a37510170908270615i731e0f71geed9ebc3ef0d75cb@mail.gmail.com>
	<191c3a030908270738n2a80ed31y328cf8808ea63b53@mail.gmail.com>
	<a37510170908270922l3a36018asa318e998df40d1fe@mail.gmail.com>
	<3D2A253B-9420-4001-918F-ECFD10AF9C22@gmail.com>
	<20090828001050.GA8108@jdc.jasonjgw.net>
	<86a32abc0908280359h243db728te2ea54f4f0f44946@mail.gmail.com>
	<a37510170908310522n73753a45sa346e6a3f255fcb9@mail.gmail.com>
	<8BBA6B6E-1B8C-415B-AC8C-D93ACC6C6418@freeswitch.org>
Message-ID: <b39273530908310707k5b061aaaoc8fac345225ea3e6@mail.gmail.com>

Isn't there a known issue with lua+sql leaking memory on some platforms?

On Mon, Aug 31, 2009 at 8:32 AM, Brian West<brian at freeswitch.org> wrote:
> Use valgrind.
>
> /b
>
> On Aug 31, 2009, at 7:22 AM, Benedikt Fraunhofer wrote:
>
>>
>> The other two freeswitch-machines in this scenario play well, they're
>> dumb load-sinks. what makes this machine different is on one hand the
>> lua-scripts that are run on it and on the other hand it's the only
>> machine who gets it's commands via xml_rpc.

> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
-Rupa



From anatoliy at kounitskiy.com  Mon Aug 31 07:32:05 2009
From: anatoliy at kounitskiy.com (Anatoliy Kounitskiy)
Date: Mon, 31 Aug 2009 17:32:05 +0300
Subject: [Freeswitch-users] Question about mod_limit
In-Reply-To: <1cd828b60908310215n60bdb242r3210a140b0811538@mail.gmail.com>
References: <1cd828b60908310215n60bdb242r3210a140b0811538@mail.gmail.com>
Message-ID: <1cd828b60908310732p1c6db054kf3db1ea842e5a004@mail.gmail.com>

OK, I was able to answer one of the questions:
if I want to use the function db - i can insert and delete:
   <action application="db"
data="insert/test/${username}/${dialed_extension:3}"/>
   <action application="db"
data="delete/test/${username}/${dialed_extension:3}"/>
but if i want to select i have to use the API function db:
   <action application="set" data="test=${db(select/test/${username})}"/>

Probably there is a reason, that the dialplan application db can't do
select - someone input here?
I was looking something like this:
  <action application="db"
data="test=select/test/${username}/${dialed_extension:3}"/>
but at the end i found a solution. I'm just searching for a reason :)

Bear in mind that I'm newbie in the FreeSwitch :)


On Mon, Aug 31, 2009 at 12:15 PM, Anatoliy
Kounitskiy<anatoliy at kounitskiy.com> wrote:
> I have question about mod_limit: is there a reason to have difference
> between the api applications and the dialplan? I mean when you use db
> function from the cli you can use insert/select/delete but when you
> use it from the dialplan you can only use insert/delete.
>
> #define DB_USAGE "[insert|delete]/<realm>/<key>/<val>"
> #define DB_DESC "save data"
>
> SWITCH_STANDARD_APP(db_function)
>
> but in the
> SWITCH_STANDARD_API(db_api_function)
> i have three options insert,delete and select
>
> What i try to accomplish is to be able to insert values from the
> dialplan and after that to be able to use them ( to select and use the
> result) from another extension.
>
> Thank you in advanced,
> --
> Anatoliy Kounitskiy
> -------------------------
> E-mail: anatoliy at kounitskiy.com
> Mobile: +359898913540
>



-- 
Anatoliy Kounitskiy
-------------------------
E-mail: anatoliy at kounitskiy.com
Mobile: +359898913540



From email.list.subscriber at gmail.com  Mon Aug 31 07:46:43 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Mon, 31 Aug 2009 10:46:43 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4A92566C.3030200@blueface.ie>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>	<4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>
	<4A92566C.3030200@blueface.ie>
Message-ID: <4a9be20d.22ba720a.5469.ffffb6ad@mx.google.com>

Sorry for the late reply, I was out of the office for the past two
weeks.  I'm not sure if there were updates over the mailing list
(still catching up), but while I was out, one of our sys admins was
able to get this compiled and has been verified to be functional on
our Solaris 10/x86 server.

Vladimir

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Bruce McAlister
Sent: Monday, August 24, 2009 4:59 AM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

Hi Vladimir,

Did you get an update on this at all?

Thanks


vmorales wrote:
> Hi Michal,
> 
> Just checking in to see if you've been able to take a stab at this.
> 
> Thanks,
> Vladimir
> 
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Michal Bielicki
> Sent: Tuesday, August 11, 2009 5:33 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
> 
> I'll retst it later today and give you a link with instructions
> 
> Am 10.08.2009 um 20:14 schrieb vmorales:
> 
>> By "./compile" I was referring to "./configure"
>>
>> Vladimir
>>
>> -----Original Message-----
>> From: vmorales [mailto:email.list.subscriber at gmail.com]
>> Sent: Monday, August 10, 2009 11:49 AM
>> To: 'freeswitch-users at lists.freeswitch.org'
>> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> Thanks for the response(s):
>>
>> I ran the "./compile" script with a set PREFIX.  This took a few
>> attempts with errors before it was able to complete error-free, as
I
>> had to install libtool.
>>
>> Since then, I have tried running 'make', 'gmake', and
>> '/opt/gnu/bin/make', but each results with an error.  This is the
>> error when running 'make' or 'gmake':
>>
>> <snip>
>> make: Fatal error: Command failed for target `all-recursive'
>> Current working directory /home/vmorales/freeswitch-1.0.4
>> *** Error code 1
>> make: Fatal error: Command failed for target `all'
>> </snip>
>>
>>
>> This is the error when running '/opt/gnu/bin/make':
>>
>> <snip>
>> make[5]: *** [mod_amr.so] Error 1
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_amr-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> Making all in build
>> +-------- FreeSWITCH Build Complete -----------+
>> + FreeSWITCH has been successfully built.      +
>> + Install by running:                          +
>> +                                              +
>> +               /opt/gnu/bin/make install                   +
>> +----------------------------------------------+
>> make[1]: *** [all-recursive] Error 1
>> make: *** [all] Error 2
>> </snip>
>>
>>
>> I re-untar'd before each compile/make attempt.  Let me know if this
> is
>> something that I can resolve.
>>
>> Vladimir
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Michael Jerris
>> Sent: Saturday, August 08, 2009 12:37 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> This is not currently a supported platform, it only builds on 64
bit
>> right now I think on solaris.
>>
>> Mike
>>
>> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>>
>>> Hello,
>>>
>>> Does anyone have, or know where to get, a pre-compiled copy of
>>> FreeSwitch for Solaris 10/x86?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>> http://www.freeswitch.org
> 
> Michal Bielicki
> Leiter der Niederlassung
> HaloKwadrat Sp. z o.o.
> Niederlassung Kleinmachnow
> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
> Ust.Id.: DE261885536
> Geschaeftsfuehrer: Aleksander Wiercinski
> Meiereifeld 2b, 14532 Kleinmachnow
> t. +49 33203 263220
> f. +49 33203 263229 sip. info at halokwadrat.de
> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
> Hauptgesch?ftsstelle:
> Halo Kwadrat Sp. z o.o.
> ul. Polna 46/14
> 00-644 Warszawa, Polen
> EIngetragen im HRB Warszawa, KRS 0000153539
> 
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
> http://www.freeswitch.org
> 
> 

_______________________________________________
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rs
http://www.freeswitch.org




From kawarod at laposte.net  Mon Aug 31 08:00:59 2009
From: kawarod at laposte.net (rod)
Date: Mon, 31 Aug 2009 19:00:59 +0400
Subject: [Freeswitch-users] Set disable-transcoding in dialplan
Message-ID: <4A9BE5AB.4030304@laposte.net>

Hi all,

I'm wondering if I can do something like this:
    - in my internal profile, I have this because of some PEER using G729:
          - <param name="disable-transcoding" value="true"/>

But for a specific PEER, I'd like to activate transcoding:
          - for this PEER, only G711 is used
          - I'd like to transcode DTMF SIP INFO or RFC2833 to INBAND

So in my dialplan, I tried before bridging:

    - <action application="set" data="disable-transcoding=false"/>
    - <action application="start_dtmf_generate" data="true"/>

But I still see RFC2833 events between my FS and PEER and the DTMF are 
not working.

So 2 questions:
    - does application "start_dtmf_generate" requires transcoding
    - if yes, can I set the variable disable-transcoding in my dialplan

regards,
rod



From steveu at coppice.org  Mon Aug 31 08:09:01 2009
From: steveu at coppice.org (Steve Underwood)
Date: Mon, 31 Aug 2009 23:09:01 +0800
Subject: [Freeswitch-users] T.38 with V.34 support
Message-ID: <4A9BE78D.5040000@coppice.org>

Hi,

Does anyone have access to a system that supports T.38 FAXing at V.34 
rates. These things seem to be as rare as bugless software.

Steve




From gregt at cgicommunications.com  Mon Aug 31 08:22:08 2009
From: gregt at cgicommunications.com (Greg Thoen)
Date: Mon, 31 Aug 2009 11:22:08 -0400
Subject: [Freeswitch-users] Incorrect method of PHP call control?
Message-ID: <4E10B925-CEFD-4AF1-8527-282DCDEC0603@cgicommunications.com>

Hi. Before I go to far down this path, I wonder if what I intend to do  
is not a good practice.

I started using mod_xml_curl to use PHP on localhost to generate a  
dialplan dynamically, based on the Caller-Destination-Number variable  
that is posted. It prints out the XML that calls the javascript that  
then controls the call. For example,

$response = <<< XML
<?xml version="1.0" encoding="UTF-8" standalone="no"?>
<document type="freeswitch/xml">
   <section name="dialplan" description="example_curl_dialplan">
     <context name="public">
       <extension name="curl_test">
         <condition field="destination_number" expression="^(\+1|1?) 
(5844111)$">
           <action application="javascript" data="demos/stest-examp- 
st.js" />
         </condition>
       </extension>
     </context>
   </section>
</document>
XML;

Then I thought, that's silly to go back out to javascript to handle  
the actions, playing files, using pocketsphinx, etc. I should just  
stay in PHP, using esl.php to answer and handle the call.

Then I rethought, is that a good practice to take over the call  
control from freeswitch at that point, while it is in the xml-curl  
dialplan hunt?

Then I also thought, is it even possible to do some of the things I  
need to do from the php esl, like the equivalent of this javascript:
session.collectInput(onInputsml, "emptyobject", 7000);
--
Greg Thoen


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From foxb at abv.bg  Mon Aug 31 08:49:55 2009
From: foxb at abv.bg (Hristo Benev)
Date: Mon, 31 Aug 2009 18:49:55 +0300 (EEST)
Subject: [Freeswitch-users] freeswitch as SBC and kamailio - no route
Message-ID: <1590255305.13053.1251733804539.JavaMail.apache@mail21.abv.bg>

 Now it is correct.

Thank you for your time.



 >-------- ?????????? ????? --------
 >??:  rod 
 >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >??: freeswitch-users at lists.freeswitch.org
 >????????? ??: ??????, 2009, ?????? 29 08:43:24 EEST

 >Ok,
 >
 >I found what's happening. I probably did some change on the wiki to 
 >start reflect the new configuration, without having time enough to check 
 >the configuration. There is a mistake in your dialplan configuration. 
 >You should put this instead:
 >
 >wrong line:
 >
 >
 >
 >please correct with:
 >
 >
 >
 >This should match PEER_01 in dialplan instead of trying matching 
 >France at PEER_01.
 >
 >Let me know if this is right now.
 >
 >rod
 >
 >Hristo Benev a ?crit :
 >>  Hello Rod,
 >>
 >> I did the change.
 >>
 >> Here is extract of console:
 >> -----------------------------------------------------------------------------------------
 >> variable_continue_on_fail: [true]
 >> variable_sip_h_X-ROUTE: [LOOKUP]
 >> variable_export_vars: [sip_h_X-ROUTE]
 >> variable_signal_bond: [a66d089a-93ee-11de-ab7a-094dff0ede20]
 >> variable_sip_redirect_contact_0: [sip:France at PEER_01]
 >> variable_sip_redirected_to: [sip:France at PEER_01]
 >> variable_sip_redirect_contact_user_0: [France]  variable_sip_redirect_contact_host_0: [PEER_01]
 >> variable_sip_redirect_dialstring_0: [sofia/internal/sip:France at PEER_01]
 >> variable_sip_redirect_dialstring: [sofia/internal/sip:France at PEER_01]
 >> variable_proto_specific_hangup_cause: [sip:503]
 >> variable_sip_hangup_phrase: [DNS Error]
 >> variable_originate_disposition: [NORMAL_TEMPORARY_FAILURE]
 >> variable_ROUTE_GW: [France]
 >> variable_AREA: [France]
 >> variable_current_application: [info]
 >> ------------------------------------------------------------------------------------------
 >> I have different value it is actually the description field as shown here:
 >>
 >> --------------------------------
 >>  /opt/kamailio/sbin/kamctl cr show
 >> cr carrier names
 >> +----+---------+
 >> | id | carrier |
 >> +----+---------+
 >> |  1 | default |
 >> +----+---------+
 >> cr domain names
 >> +----+---------+
 >> | id | domain  |
 >> +----+---------+
 >> |  1 | default |
 >> +----+---------+
 >> cr routes
 >> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
 >> | id | carrier | domain | scan_prefix | flags | mask | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | description |
 >> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
 >> |  1 |       1 |      1 | 1000        |     0 |    0 |    1 |     0 | PEER_01      |                |                | France      |
 >> +----+---------+--------+-------------+-------+------+------+-------+--------------+----------------+----------------+-------------+
 >> -----------------------------------------
 >>
 >> And here is what I have in kamailio:
 >> -------------------------------------------------------------------------------------
 >> ####### Routing Logic ########
 >>
 >>
 >> # main request routing logic
 >>
 >> route{
 >>
 >>         if (!mf_process_maxfwd_header("10")) {
 >>                 sl_send_reply("483","Too Many Hops");
 >>                 exit;
 >>         }
 >>
 >>         t_check_trans();
 >>
 >>         if ($rU==NULL) {
 >>                 # request with no Username in RURI
 >>                 sl_send_reply("484","Address Incomplete");
 >>                 exit;
 >>         }
 >>
 >>         # LOOKUP ROUTE TABLE WHEN ASKED BY HEADER: X-ROUTE:LOOKUP
 >>         if (is_method("INVITE") && $hdr(X-ROUTE)=="LOOKUP"){
 >>                 if(!cr_route("default", "default", "$rU", "$rU", "call_id","$avp(s:route_desc)")){
 >>                      #xlog("ROUTING FAILED: no route found for $rU");
 >>                      sl_send_reply("604", "Unable to route this call");
 >>                      exit;
 >>                 } else {
 >>                      xlog("LOOKUP FOUND: $rd $avp(s:route_desc)");
 >>                      avp_pushto("$ru/username", "$avp(s:route_desc)");
 >>                      sl_send_reply("302", "$rd");
 >>                      exit;
 >>                 }
 >>         }
 >> }
 >> ----------------------------------------------------------------------------------
 >>
 >> Another question...
 >> In that part of FreeSwitch dialplan.xml
 >>
 >> -----------------------------------
 >>   
 >>
 >>     
 >>       
 >>         
 >>
 >> -----------------------------------
 >>
 >> X-ROUTE maybe should be replaced with ROUTE_GW otherwise there is no way the regex is true.
 >>
 >>
 >> As for thanks - for sure by default they are also for the developers of both apps.
 >> I'm new in freeSwitch and/or Kamailio for now I'm still testing and learning so it will be nice to have something working to start with.
 >>
 >>
 >>
 >>  >-------- ?????????? ????? --------
 >>  >??:  rod 
 >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >??: freeswitch-users at lists.freeswitch.org
 >>  >????????? ??: ?????, 2009, ?????? 28 09:54:00 EEST
 >>
 >>  >Hello,
 >>  >
 >>  >the trace seems good.
 >>  >If you check the answer from Kamailio, you'll see that Kamailio answers 
 >>  >with "302 PEER_01".
 >>  >
 >>  >As Michael Collins stated before, you can get the variable containing 
 >>  >"PEER_01", then this variable is stored in a custom variable.
 >>  >
 >>  >In your dialplan, may you please add:
 >>  >
 >>  >, just before the transfer line, eg:
 >>  >
 >>  > 
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >         
 >>  >	 
 >>  >         
 >>  >      
 >>  >
 >>  >
 >>  >Using application Info, you'll see on the console (or CDR) the list of 
 >>  >variables used for this call.  You should see the content of 
 >>  >"${sip_redirect_contact_user_0}" that should contain  the value 
 >>  >"PEER_01". Please check this and let me know.
 >>  >
 >>  >For the new configuration file, no problem for sharing. But as I wrote 
 >>  >on the wiki page, I worked on this setup cause LCR module was not 
 >>  >available when I start working on FS, nor I'm a good programmer to write 
 >>  >a server side HTTP script (used by xml_curl) that could scale to my needs.
 >>  >
 >>  >I enhanced a bit this configuration with support for fallback routing, 
 >>  >almost realtime graph (every minutes using www.cacti.net and some 
 >>  >functions in FS like limit_hash) of number of concurrent calls per AREA, 
 >>  >PEER...
 >>  >
 >>  >Thanks for "the good tutorial", but don't forget the dev team who did 
 >>  >this great product ;-)
 >>  >
 >>  >rod.
 >>  >
 >>  >
 >>  >Hristo Benev a ?crit :
 >>  >>  I assume you asked for port 5062 since I do not have any traffic on 5060 (I have one IP and my internal sip port is 5090 and external 5080).
 >>  >>
 >>  >> If you need additional info I'll provide it.
 >>  >>
 >>  >> Here is trace:
 >>  >>
 >>  >> ngrep -d any -nn -i '1000' port 5062 -W byline
 >>  >> interface: any
 >>  >> filter: (ip or ip6) and ( port 5062 )
 >>  >> match: 1000
 >>  >> #
 >>  >> U 10.10.10.10:5090 -> 127.0.0.1:5062
 >>  >> INVITE sip:1000 at 127.0.0.1:5062 SIP/2.0.
 >>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
 >>  >> Max-Forwards: 69.
 >>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
 >>  >> To: .
 >>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
 >>  >> CSeq: 119574771 INVITE.
 >>  >> Contact: .
 >>  >> User-Agent: FreeSWITCH-mod_sofia/1.0.4-exported.
 >>  >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH.
 >>  >> Supported: timer, precondition, path, replaces.
 >>  >> Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer.
 >>  >> Content-Type: application/sdp.
 >>  >> Content-Disposition: session.
 >>  >> Content-Length: 429.
 >>  >> X-ROUTE: LOOKUP.
 >>  >> Remote-Party-ID: "Extension 1001" ;party=calling;screen=yes;privacy=off.
 >>  >> .
 >>  >> v=0.
 >>  >> o=FreeSWITCH 1251355692 1251355693 IN IP4 10.10.10.10.
 >>  >> s=FreeSWITCH.
 >>  >> c=IN IP4 10.10.10.10.
 >>  >> t=0 0.
 >>  >> m=audio 30522 RTP/AVP 0 115 107 9 8 3 101 13.
 >>  >> a=rtpmap:0 PCMU/8000.
 >>  >> a=rtpmap:115 G7221/32000.
 >>  >> a=fmtp:115 bitrate=48000.
 >>  >> a=rtpmap:107 G7221/16000.
 >>  >> a=fmtp:107 bitrate=32000.
 >>  >> a=rtpmap:9 G722/8000.
 >>  >> a=rtpmap:8 PCMA/8000.
 >>  >> a=rtpmap:3 GSM/8000.
 >>  >> a=rtpmap:101 telephone-event/8000.
 >>  >> a=fmtp:101 0-16.
 >>  >> a=rtpmap:13 CN/8000.
 >>  >> a=ptime:20.
 >>  >>
 >>  >> #
 >>  >> U 127.0.0.1:5062 -> 10.10.10.10:5090
 >>  >> SIP/2.0 302 PEER_01.
 >>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport=5090;branch=z9hG4bKtrDyU1US5X5tj.
 >>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
 >>  >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
 >>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
 >>  >> CSeq: 119574771 INVITE.
 >>  >> Contact: sip:France at PEER_01.
 >>  >> Server: Kamailio (1.5.2-notls (i386/linux)).
 >>  >> Content-Length: 0.
 >>  >> .
 >>  >>
 >>  >> #
 >>  >> U 10.10.10.10:5090 -> 127.0.0.1:5062
 >>  >> ACK sip:1000 at 127.0.0.1:5062 SIP/2.0.
 >>  >> Via: SIP/2.0/UDP 10.10.10.10:5090;rport;branch=z9hG4bKtrDyU1US5X5tj.
 >>  >> Max-Forwards: 69.
 >>  >> From: "Extension 1001" ;tag=978g69jZaFpBD.
 >>  >> To: ;tag=458fb4012080e656b6742c09466dabcd.31c8.
 >>  >> Call-ID: 7f0eff2d-0dbf-122d-bfb0-612c8433bc7c.
 >>  >> CSeq: 119574771 ACK.
 >>  >> Content-Length: 0.
 >>  >> .
 >>  >>
 >>  >>
 >>  >>
 >>  >>
 >>  >>
 >>  >>  >-------- ?????????? ????? --------
 >>  >>  >??:  Hristo Benev 
 >>  >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >>  >??: freeswitch-users at lists.freeswitch.org
 >>  >>  >????????? ??: ?????????, 2009, ?????? 27 14:58:53 EEST
 >>  >>
 >>  >>  > 
 >>  >>  >Bojnour,
 >>  >>  >
 >>  >>  >I'll send a trace ASAP.
 >>  >>  >
 >>  >>  >What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config)
 >>  >>  >
 >>  >>  >I'm really interested to see the updates of the project.
 >>  >>  >
 >>  >>  >Thank you for the good tutorial.
 >>  >>  >
 >>  >>  >Hristo
 >>  >>  >
 >>  >>  >
 >>  >>  > >-------- ?????????? ????? --------
 >>  >>  > >??:  rod 
 >>  >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >>  > >??: freeswitch-users at lists.freeswitch.org
 >>  >>  > >????????? ??: ?????????, 2009, ?????? 27 09:41:05 EEST
 >>  >>  >
 >>  >>  > >Hi Hristo,
 >>  >>  > >
 >>  >>  > >I'm the author of this setup and wiki page. I did a lot of modifications 
 >>  >>  > >on this setup (alternative routing if failure essentially) but don't 
 >>  >>  > >have too much time to update the wiki.
 >>  >>  > >
 >>  >>  > >May you please send me an ngrep trace when you call 1000:
 >>  >>  > >
 >>  >>  > >ngrep -d any -nn -i '1000' port 5060 -W byline 
 >>  >>  > >
 >>  >>  > >I will check what's happening.
 >>  >>  > >Do you have an entry for 1000 in your mysql database ?
 >>  >>  > >
 >>  >>  > >regards,
 >>  >>  > >rod
 >>  >>  > >
 >>  >>  > >Hristo Benev a ?crit :
 >>  >>  > >>  
 >>  >>  > >> It seems that the problem is on kamailio configuration.
 >>  >>  > >> Will ask on their list.
 >>  >>  > >>
 >>  >>  > >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required.
 >>  >>  > >>
 >>  >>  > >> I have it setup as friend in asterisk, but still ???
 >>  >>  > >>
 >>  >>  > >> Any ideas?
 >>  >>  > >>
 >>  >>  > >> Thanks,
 >>  >>  > >>
 >>  >>  > >>
 >>  >>  > >>  >-------- ?????????? ????? --------
 >>  >>  > >>  >??:  Hristo Benev 
 >>  >>  > >>  >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >>  > >>  >??: freeswitch-users at lists.freeswitch.org
 >>  >>  > >>  >????????? ??: ?????, 2009, ?????? 26 22:02:13 EEST
 >>  >>  > >>
 >>  >>  > >>  > I think that the problem is here:
 >>  >>  > >>  >-------------------------
 >>  >>  > >>  >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING
 >>  >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 parsing [ROUTING->PEER_01] continue=false
 >>  >>  > >>  >Dialplan: sofia/internal/1001 at 209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false
 >>  >>  > >>  >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting
 >>  >>  > >>  >--------------------------
 >>  >>  > >>  >
 >>  >>  > >>  >Actually Regex FAIL
 >>  >>  > >>  >
 >>  >>  > >>  >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success?
 >>  >>  > >>  >Here is my  default.xml:
 >>  >>  > >>  >----------------
 >>  >>  > >>  >
 >>  >>  > >>  >
 >>  >>  > >>  >
 >>  >>  > >>  >  
 >>  >>  > >>  >
 >>  >>  > >>  >     
 >>  >>  > >>  >      
 >>  >>  > >>  >         
 >>  >>  > >>  >         
 >>  >>  > >>  >         
 >>  >>  > >>  >         
 >>  >>  > >>  >         
 >>  >>  > >>  >         
 >>  >>  > >>  >         
 >>  >>  > >>  >      
 >>  >>  > >>  >    
 >>  >>  > >>  >
 >>  >>  > >>  >   
 >>  >>  > >>  >
 >>  >>  > >>  >  
 >>  >>  > >>  >
 >>  >>  > >>  >    
 >>  >>  > >>  >      
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >        
 >>  >>  > >>  >      
 >>  >>  > >>  >    
 >>  >>  > >>  >
 >>  >>  > >>  >  
 >>  >>  > >>  >
 >>  >>  > >>  >
 >>  >>  > >>  >
 >>  >>  > >>  >--------------------------
 >>  >>  > >>  >
 >>  >>  > >>  >
 >>  >>  > >>  >
 >>  >>  > >>  > >-------- ?????????? ????? --------
 >>  >>  > >>  > >??:  Brian West 
 >>  >>  > >>  > >???????: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
 >>  >>  > >>  > >??: freeswitch-users at lists.freeswitch.org
 >>  >>  > >>  > >????????? ??: ?????, 2009, ?????? 26 19:47:37 EEST
 >>  >>  > >>  >
 >>  >>  > >>  > >We do not blindly follow 302's as that is a dangerous thing to do. 
 >>  >>  > >>  >You have to process all 302's in the dialplan. 
 >>  >>  > >>  >Set this on your sofia profile
 >>  >>  > >>  >You can set these variables sip_redirect_profile,
 >>  >>  > >>  >sip_redirect_context,
 >>  >>  > >>  >sip_redirect_dialplan,
 >>  >>  > >>  >When a redirect happens you get these variables - sip_redirect_contact_%d,
 >>  >>  > >>  >sip_redirected_to,
 >>  >>  > >>  >sip_redirect_contact_user_%d,
 >>  >>  > >>  >sip_redirect_contact_host_%d,
 >>  >>  > >>  >sip_redirect_contact_params_%d,
 >>  >>  > >>  >sip_redirect_dialstring_%d,
 >>  >>  > >>  >sip_redirect_dialstring,
 >>  >>  > >>  >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the
 >>  >>  > >>  >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems?
 >>  >>  > >>  > >
 >>  >>  > >>  >
 >>  >>  > >>  >_______________________________________________
 >>  >>  > >>  >FreeSWITCH-users mailing list
 >>  >>  > >>  >FreeSWITCH-users at lists.freeswitch.org
 >>  >>  > >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >>  > >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >>  > >>  >http://www.freeswitch.org
 >>  >>  > >>  >
 >>  >>  > >>
 >>  >>  > >> _______________________________________________
 >>  >>  > >> FreeSWITCH-users mailing list
 >>  >>  > >> FreeSWITCH-users at lists.freeswitch.org
 >>  >>  > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >>  > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >>  > >> http://www.freeswitch.org
 >>  >>  > >>   
 >>  >>  > >
 >>  >>  > >_______________________________________________
 >>  >>  > >FreeSWITCH-users mailing list
 >>  >>  > >FreeSWITCH-users at lists.freeswitch.org
 >>  >>  > >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >>  > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >>  > >http://www.freeswitch.org
 >>  >>  > >
 >>  >>  >
 >>  >>  >_______________________________________________
 >>  >>  >FreeSWITCH-users mailing list
 >>  >>  >FreeSWITCH-users at lists.freeswitch.org
 >>  >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >>  >http://www.freeswitch.org
 >>  >>  >
 >>  >>
 >>  >> _______________________________________________
 >>  >> FreeSWITCH-users mailing list
 >>  >> FreeSWITCH-users at lists.freeswitch.org
 >>  >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >> http://www.freeswitch.org
 >>  >>   
 >>  >
 >>  >_______________________________________________
 >>  >FreeSWITCH-users mailing list
 >>  >FreeSWITCH-users at lists.freeswitch.org
 >>  >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >>  >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >>  >http://www.freeswitch.org
 >>  >
 >>
 >> _______________________________________________
 >> FreeSWITCH-users mailing list
 >> FreeSWITCH-users at lists.freeswitch.org
 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >> http://www.freeswitch.org
 >>   
 >
 >_______________________________________________
 >FreeSWITCH-users mailing list
 >FreeSWITCH-users at lists.freeswitch.org
 >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
 >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
 >http://www.freeswitch.org
 >



From msc at freeswitch.org  Mon Aug 31 09:15:33 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:15:33 -0700
Subject: [Freeswitch-users] external Gateways
In-Reply-To: <25223286.post@talk.nabble.com>
References: <25223286.post@talk.nabble.com>
Message-ID: <87f2f3b90908310915o45bd1b8bv441d82cfce887f4d@mail.gmail.com>

On Mon, Aug 31, 2009 at 6:11 AM, NOx-WHV <enno.egbert at googlemail.com> wrote:

>
> Hello,
>
> i use a sip gateway witch is only be able to allocate one sip channel.  How
> can I bridge a call to a second gateway if the first gateway is in use.
>
> Now the dialplan look like this:
>
>  <condition field="${toll_allow}" expression="domestic"/>
>  <condition field="destination_number" expression="^04(\d+)$">
>   <action application="set" data="effective_caller_id_number=300"/>
>   <action application="set" data="sip_secure_media=false"/>
>   <action application="bridge"
> data="sofia/gateway/mygateway/04$1 at XXX.XXX.XXX.XXX"/>
>  </condition>
>
> Thanks for help
>
>
Try pipe-delimited list of dialstrings in your bridge:

<action application="bridge"
data="sofia/gateway/mygateway/04$1 at XXX.XXX.XXX.XXX
|sofia/gateway/mygw2/foo at bar|sofia/gateway/mygw3/foo at bar"/>

-MC

>
> --
> View this message in context:
> http://www.nabble.com/external-Gateways-tp25223286p25223286.html
> Sent from the Freeswitch-users mailing list archive at Nabble.com.
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Aug 31 09:17:22 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:17:22 -0700
Subject: [Freeswitch-users] liverpie as gui-proxy?
In-Reply-To: <6f7c60c40908291937h57c632d5l84b6ec5ac1286c48@mail.gmail.com>
References: <6f7c60c40908291937h57c632d5l84b6ec5ac1286c48@mail.gmail.com>
Message-ID: <87f2f3b90908310917g62b4ba2er1588314ecf25bd06@mail.gmail.com>

On Sat, Aug 29, 2009 at 7:37 PM, tom <tomabroad at gmail.com> wrote:

> hi,
>
> i was actually looking for something completely else, but reading wiki  led
> me to the tought:
> if mod_event can basically do all kinds of FS-things, why not having a
> standalone ruby-on-rails as as gui which interacts with the evnet-socket via
> liverpie?
>
>
Yes, this is totally possible, at least in theory. The actual execute, of
course, requires real work.
-MC

> im a total newbie to FS, just writing down my thoughts...
>
>
> thx for ur replies
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Aug 31 09:21:46 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:21:46 -0700
Subject: [Freeswitch-users] How can it be done with freeswitch?
In-Reply-To: <25199956.post@talk.nabble.com>
References: <25184845.post@talk.nabble.com>
	<87f2f3b90908280907w4812a8dbr7a97385f9afdd376@mail.gmail.com>
	<25199956.post@talk.nabble.com>
Message-ID: <87f2f3b90908310921h57685f14o1da57ec4ec6390cb@mail.gmail.com>

On Fri, Aug 28, 2009 at 9:26 PM, Nagalenoj <nagalenoj at gmail.com> wrote:

>
> Thanks for your help. Is it the only possible way.?
> Else, will it be done with the functions listed in the below url,
> http://wiki.freeswitch.org/wiki/Mod_perl_functions_and_classes
>
>
Yes, these classes and functions are the building blocks that you'll use to
construct your application. We have a few Perl examples you can use to get
you going. At this point you will probably just need to roll up your sleeves
and start hacking away at it. Definitely email or drop by #freeswitch on
irc.freenode.net if you have more questions.

-MC
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From msc at freeswitch.org  Mon Aug 31 09:25:01 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:25:01 -0700
Subject: [Freeswitch-users] What is the difference between preAnswer and
	ring_ready?
In-Reply-To: <6B5963B0-C53A-4005-A14F-DD3140F29560@avgs.ca>
References: <c55d46950908291830w6b83a6a7h17a6faf67732d58a@mail.gmail.com>
	<20090830015030.GA3163@jdc.jasonjgw.net>
	<6B5963B0-C53A-4005-A14F-DD3140F29560@avgs.ca>
Message-ID: <87f2f3b90908310925p2fa672a4n11d5bb3bef599f5b@mail.gmail.com>

Well said, Math. BTW, this info is available on the wiki:
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_ring_ready
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_pre_answer

-MC

On Sat, Aug 29, 2009 at 8:54 PM, Mathieu Rene <mrene_lists at avgs.ca> wrote:

> Hi,
>
> In SIP, ring_ready will send a 180 ringing, pre answer will send a 183
> with a session description, so that some audio can be exchanged while
> the call isnt answered yet (used a lot for inband error messages etc).
>
> Regards,
>
> Mathieu Rene
> Avant-Garde Solutions Inc
> Office: + 1 (514) 664-1044 x100
> Cell: +1 (514) 664-1044 x200
> mrene at avgs.ca
>
>
>
>
> On 29-Aug-09, at 9:50 PM, Jason White wrote:
>
> > Max Bridgewater <max.bridgewater at gmail.com> wrote:
> >> Hi,
> >> Assuming an inbound call, I have trouble understanding what the
> >> supposed
> >> difference between the following two set of instructions is:
> >>
> >> session.execute("ring_ready");
> >> session.set("set", "ringback=/home/ring.wav");
> >
> > Ring_ready sends a SIP "ringing" message and will use your device's
> > default
> > ringback sound - at least, that has been my experience. I should
> > remember the
> > message number in the SIP protocol, but I don't (the SIP experts
> > here will
> > undoubtedly have the details at the ready).
> >>
> >> and
> >>
> >> session.preAnswer();
> >> session.streamFile("/home/ring.wav")
> >
> > I think that will play the file as early media.
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From max.bridgewater at gmail.com  Mon Aug 31 09:31:59 2009
From: max.bridgewater at gmail.com (Max Bridgewater)
Date: Mon, 31 Aug 2009 12:31:59 -0400
Subject: [Freeswitch-users] Troubles build with mod_cepstral
In-Reply-To: <399999D5-EF3E-444F-8E06-05A0006DD053@freeswitch.org>
References: <20090830123332.2ad02225396a31c9de30536f2e338977.dced3325fe.wbe@email04.secureserver.net>
	<7B458C2D-8DC2-40B9-8EF1-B7FC934A5F0F@freeswitch.org>
	<c55d46950908301449u6bb3d43cifd92a353cca6cae5@mail.gmail.com>
	<399999D5-EF3E-444F-8E06-05A0006DD053@freeswitch.org>
Message-ID: <c55d46950908310931i56d4a272t937514c4ae253d7f@mail.gmail.com>

I thought i did that before trying. But i tried again and everything works.

Thanks for you unyielding patience.
Max.

On Sun, Aug 30, 2009 at 5:53 PM, Brian West <brian at freeswitch.org> wrote:

> Make sure you rm -rf /opt/swift then reinstall
>
> /b
>
> On Aug 30, 2009, at 4:49 PM, Max Bridgewater wrote:
>
> > Nop, the 64bit  cepstral doesn't work either:
> >
> > Creating mod_cepstral.so...
> > /usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when
> > searching for -lswift
> > /usr/bin/ld: cannot find -lswift
> > collect2: ld returned 1 exit status
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Aug 31 09:38:35 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:38:35 -0700
Subject: [Freeswitch-users] Set disable-transcoding in dialplan
In-Reply-To: <4A9BE5AB.4030304@laposte.net>
References: <4A9BE5AB.4030304@laposte.net>
Message-ID: <87f2f3b90908310938oa7e838dkd6e07579f224de37@mail.gmail.com>

Check out this page:
http://wiki.freeswitch.org/wiki/Codec_negotiation

Late negotiation will probably let you handle all the cases you need.
-MC

On Mon, Aug 31, 2009 at 8:00 AM, rod <kawarod at laposte.net> wrote:

> Hi all,
>
> I'm wondering if I can do something like this:
>    - in my internal profile, I have this because of some PEER using G729:
>          - <param name="disable-transcoding" value="true"/>
>
> But for a specific PEER, I'd like to activate transcoding:
>          - for this PEER, only G711 is used
>          - I'd like to transcode DTMF SIP INFO or RFC2833 to INBAND
>
> So in my dialplan, I tried before bridging:
>
>    - <action application="set" data="disable-transcoding=false"/>
>    - <action application="start_dtmf_generate" data="true"/>
>
> But I still see RFC2833 events between my FS and PEER and the DTMF are
> not working.
>
> So 2 questions:
>    - does application "start_dtmf_generate" requires transcoding
>    - if yes, can I set the variable disable-transcoding in my dialplan
>
> regards,
> rod
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Aug 31 09:45:00 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:45:00 -0700
Subject: [Freeswitch-users] some errors from FS startup
In-Reply-To: <ec9a78d50908310431g120729f8jc5a4302aa74b9840@mail.gmail.com>
References: <ec9a78d50908310431g120729f8jc5a4302aa74b9840@mail.gmail.com>
Message-ID: <87f2f3b90908310945i2f2118eam3fa4330a121f4090@mail.gmail.com>

Most likely you've enabled a few modules in
conf/autoload_configs/modules.conf.xml but you haven't built them in the
src/freeswitch.trunk/modules.conf file. Be sure to open modules.conf,
uncomment the modules that are throwing errors, like mod_xml_curl. Then run
"make install" again. For the music errors most likely you just need to
install the MOH files. Be sure to get the regular sound files too:

make cd-moh-install && make cd-sounds-install

-MC

On Mon, Aug 31, 2009 at 4:31 AM, Erwin Huang <erwin.davis at gmail.com> wrote:

> Hi, I was able to build the FS successfully and make a couple of test
> calls. But when I looked at the startup logs, I saw some errors. Please
> point out how to fix them. Thanks,
>
> Regards,
>
> e
>
> 2009-08-31 06:59:55.54063 [ERR] mod_xml_curl.c:404 Binding has no url!
> 2009-08-31 06:59:55.54094 [CRIT] switch_loadable_module.c:871 Error Loading
> modu
> le /usr/local/freeswitch/mod/mod_xml_curl.so
> **Module load routine returned an error**
>
>
> 2009-08-31 06:59:57.393133 [CONSOLE] mod_local_stream.c:142 Can't open
> directory
> : /usr/local/freeswitch/sounds/music/8000
> 2009-08-31 06:59:57.404228 [CONSOLE] mod_local_stream.c:142 Can't open
> directory
> : /usr/local/freeswitch/sounds/music/8000
> 2009-08-31 06:59:57.414708 [CONSOLE] mod_local_stream.c:142 Can't open
> directory
> : /usr/local/freeswitch/sounds/music/16000
> 2009-08-31 06:59:57.424568 [CONSOLE] mod_local_stream.c:142 Can't open
> directory
> : /usr/local/freeswitch/sounds/music/32000
>
>
> // The errors below indicate the files donot exist. How to include these
> two files into FS make build?
>
> 2009-08-31 06:59:57.477374 [CRIT] switch_loadable_module.c:871 Error
> Loading mod
> ule /usr/local/freeswitch/mod/mod_java.so
> **/usr/local/freeswitch/mod/mod_java.so: cannot open shared object file: No
> such
>  file or directory**
>
>
> 2009-08-31 06:59:57.485242 [CRIT] switch_loadable_module.c:871 Error
> Loading mod
> ule /usr/local/freeswitch/mod/mod_flite.so
> **/usr/local/freeswitch/mod/mod_flite.so: cannot open shared object file:
> No suc
> h file or directory**
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From bruce.mcalister at blueface.ie  Mon Aug 31 09:49:00 2009
From: bruce.mcalister at blueface.ie (Bruce McAlister)
Date: Mon, 31 Aug 2009 17:49:00 +0100
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4a9be20d.22ba720a.5469.ffffb6ad@mx.google.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>	<4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>	<4A92566C.3030200@blueface.ie>
	<4a9be20d.22ba720a.5469.ffffb6ad@mx.google.com>
Message-ID: <4A9BFEFC.2090800@blueface.ie>

Hi Vladimir,

Would you mind letting us know what your sysadmin did to achieve a 
successful build?

I am still having issues building freeswitch.

Thanks
Bruce

vmorales wrote:
> Sorry for the late reply, I was out of the office for the past two
> weeks.  I'm not sure if there were updates over the mailing list
> (still catching up), but while I was out, one of our sys admins was
> able to get this compiled and has been verified to be functional on
> our Solaris 10/x86 server.
> 
> Vladimir
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Bruce McAlister
> Sent: Monday, August 24, 2009 4:59 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
> 
> Hi Vladimir,
> 
> Did you get an update on this at all?
> 
> Thanks
> 
> 
> vmorales wrote:
>> Hi Michal,
>>
>> Just checking in to see if you've been able to take a stab at this.
>>
>> Thanks,
>> Vladimir
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Michal Bielicki
>> Sent: Tuesday, August 11, 2009 5:33 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> I'll retst it later today and give you a link with instructions
>>
>> Am 10.08.2009 um 20:14 schrieb vmorales:
>>
>>> By "./compile" I was referring to "./configure"
>>>
>>> Vladimir
>>>
>>> -----Original Message-----
>>> From: vmorales [mailto:email.list.subscriber at gmail.com]
>>> Sent: Monday, August 10, 2009 11:49 AM
>>> To: 'freeswitch-users at lists.freeswitch.org'
>>> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>>> 10/x86
>>>
>>> Thanks for the response(s):
>>>
>>> I ran the "./compile" script with a set PREFIX.  This took a few
>>> attempts with errors before it was able to complete error-free, as
> I
>>> had to install libtool.
>>>
>>> Since then, I have tried running 'make', 'gmake', and
>>> '/opt/gnu/bin/make', but each results with an error.  This is the
>>> error when running 'make' or 'gmake':
>>>
>>> <snip>
>>> make: Fatal error: Command failed for target `all-recursive'
>>> Current working directory /home/vmorales/freeswitch-1.0.4
>>> *** Error code 1
>>> make: Fatal error: Command failed for target `all'
>>> </snip>
>>>
>>>
>>> This is the error when running '/opt/gnu/bin/make':
>>>
>>> <snip>
>>> make[5]: *** [mod_amr.so] Error 1
>>> make[4]: *** [all] Error 1
>>> make[3]: *** [mod_amr-all] Error 1
>>> make[2]: *** [all-recursive] Error 1
>>> Making all in build
>>> +-------- FreeSWITCH Build Complete -----------+
>>> + FreeSWITCH has been successfully built.      +
>>> + Install by running:                          +
>>> +                                              +
>>> +               /opt/gnu/bin/make install                   +
>>> +----------------------------------------------+
>>> make[1]: *** [all-recursive] Error 1
>>> make: *** [all] Error 2
>>> </snip>
>>>
>>>
>>> I re-untar'd before each compile/make attempt.  Let me know if this
>> is
>>> something that I can resolve.
>>>
>>> Vladimir
>>>
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Michael Jerris
>>> Sent: Saturday, August 08, 2009 12:37 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>>> 10/x86
>>>
>>> This is not currently a supported platform, it only builds on 64
> bit
>>> right now I think on solaris.
>>>
>>> Mike
>>>
>>> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>>>
>>>> Hello,
>>>>
>>>> Does anyone have, or know where to get, a pre-compiled copy of
>>>> FreeSwitch for Solaris 10/x86?
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> rs
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>>> http://www.freeswitch.org
>> Michal Bielicki
>> Leiter der Niederlassung
>> HaloKwadrat Sp. z o.o.
>> Niederlassung Kleinmachnow
>> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
>> Ust.Id.: DE261885536
>> Geschaeftsfuehrer: Aleksander Wiercinski
>> Meiereifeld 2b, 14532 Kleinmachnow
>> t. +49 33203 263220
>> f. +49 33203 263229 sip. info at halokwadrat.de
>> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
>> Hauptgesch?ftsstelle:
>> Halo Kwadrat Sp. z o.o.
>> ul. Polna 46/14
>> 00-644 Warszawa, Polen
>> EIngetragen im HRB Warszawa, KRS 0000153539
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
> 
> 



From msc at freeswitch.org  Mon Aug 31 09:55:54 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 09:55:54 -0700
Subject: [Freeswitch-users] Incorrect method of PHP call control?
In-Reply-To: <4E10B925-CEFD-4AF1-8527-282DCDEC0603@cgicommunications.com>
References: <4E10B925-CEFD-4AF1-8527-282DCDEC0603@cgicommunications.com>
Message-ID: <87f2f3b90908310955u399f5c5v1214d58e6004e392@mail.gmail.com>

On Mon, Aug 31, 2009 at 8:22 AM, Greg Thoen <gregt at cgicommunications.com>wrote:

> Hi. Before I go to far down this path, I wonder if what I intend to do is
> not a good practice.
> I started using mod_xml_curl to use PHP on localhost to generate a dialplan
> dynamically, based on the Caller-Destination-Number variable that is posted.
> It prints out the XML that calls the javascript that then controls the call.
> For example,
>
> $response = <<< XML
> <?xml version="1.0" encoding="UTF-8" standalone="no"?>
> <document type="freeswitch/xml">
>   <section name="dialplan" description="example_curl_dialplan">
>     <context name="public">
>       <extension name="curl_test">
>         <condition field="destination_number"
> expression="^(\+1|1?)(5844111)$">
>           <action application="javascript" data="demos/stest-examp-st.js"
> />
>         </condition>
>       </extension>
>     </context>
>   </section>
> </document>
> XML;
>
> Then I thought, that's silly to go back out to javascript to handle the
> actions, playing files, using pocketsphinx, etc. I should just stay in PHP,
> using esl.php to answer and handle the call.
>
> Then I rethought, is that a good practice to take over the call control
> from freeswitch at that point, while it is in the xml-curl dialplan hunt?
>
> Then I also thought, is it even possible to do some of the things I need to
> do from the php esl, like the equivalent of this javascript:
> session.collectInput(onInputsml, "emptyobject", 7000);
>
> --
>
> Greg Thoen
>
>
Just remember that you're dealing with two somewhat related but still
distinctly separate entities: generating a dialplan and executing some sort
of call control from the dialplan. You need some sort of dialplan no matter
what, so the issue there is whether you need a dynamic one or not. If you're
just going to drop calls to an extension that opens an outbound socket to
your call control program then you may not need the dynamic dp generation
that mod_xml_curl gives you. You'll have to decide on static vs. dynamic
based on your needs. In either case, once the call is connected to your
socket you've got all sorts of control options. PHP has an ESL abstraction
just like the other languages so there shouldn't be any issue about PHP
lacking the ability to control calls.

I say start hacking away at it and see what happens. :) Definitely join us
in #freeswitch on irc.freenode.net if you want to discuss this more in
realtime.
-MC
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From email.list.subscriber at gmail.com  Mon Aug 31 10:16:13 2009
From: email.list.subscriber at gmail.com (vmorales)
Date: Mon, 31 Aug 2009 13:16:13 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4A9BFEFC.2090800@blueface.ie>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>	<4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>	<4A92566C.3030200@blueface.ie>	<4a9be20d.22ba720a.5469.ffffb6ad@mx.google.com>
	<4A9BFEFC.2090800@blueface.ie>
Message-ID: <4a9c0516.19bd720a.23b1.27df@mx.google.com>

Using Sun Studio and exporting the following flag:

	export CFLAGS="-lresolv"

Is how our sysadmin was able to get FS to build.

Vladimir

-----Original Message-----
From: freeswitch-users-bounces at lists.freeswitch.org
[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
Bruce McAlister
Sent: Monday, August 31, 2009 12:49 PM
To: freeswitch-users at lists.freeswitch.org
Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
10/x86

Hi Vladimir,

Would you mind letting us know what your sysadmin did to achieve a 
successful build?

I am still having issues building freeswitch.

Thanks
Bruce

vmorales wrote:
> Sorry for the late reply, I was out of the office for the past two
> weeks.  I'm not sure if there were updates over the mailing list
> (still catching up), but while I was out, one of our sys admins was
> able to get this compiled and has been verified to be functional on
> our Solaris 10/x86 server.
> 
> Vladimir
> 
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Bruce McAlister
> Sent: Monday, August 24, 2009 4:59 AM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
> 
> Hi Vladimir,
> 
> Did you get an update on this at all?
> 
> Thanks
> 
> 
> vmorales wrote:
>> Hi Michal,
>>
>> Just checking in to see if you've been able to take a stab at this.
>>
>> Thanks,
>> Vladimir
>>
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Michal Bielicki
>> Sent: Tuesday, August 11, 2009 5:33 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> I'll retst it later today and give you a link with instructions
>>
>> Am 10.08.2009 um 20:14 schrieb vmorales:
>>
>>> By "./compile" I was referring to "./configure"
>>>
>>> Vladimir
>>>
>>> -----Original Message-----
>>> From: vmorales [mailto:email.list.subscriber at gmail.com]
>>> Sent: Monday, August 10, 2009 11:49 AM
>>> To: 'freeswitch-users at lists.freeswitch.org'
>>> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for
Solaris
>>> 10/x86
>>>
>>> Thanks for the response(s):
>>>
>>> I ran the "./compile" script with a set PREFIX.  This took a few
>>> attempts with errors before it was able to complete error-free, as
> I
>>> had to install libtool.
>>>
>>> Since then, I have tried running 'make', 'gmake', and
>>> '/opt/gnu/bin/make', but each results with an error.  This is the
>>> error when running 'make' or 'gmake':
>>>
>>> <snip>
>>> make: Fatal error: Command failed for target `all-recursive'
>>> Current working directory /home/vmorales/freeswitch-1.0.4
>>> *** Error code 1
>>> make: Fatal error: Command failed for target `all'
>>> </snip>
>>>
>>>
>>> This is the error when running '/opt/gnu/bin/make':
>>>
>>> <snip>
>>> make[5]: *** [mod_amr.so] Error 1
>>> make[4]: *** [all] Error 1
>>> make[3]: *** [mod_amr-all] Error 1
>>> make[2]: *** [all-recursive] Error 1
>>> Making all in build
>>> +-------- FreeSWITCH Build Complete -----------+
>>> + FreeSWITCH has been successfully built.      +
>>> + Install by running:                          +
>>> +                                              +
>>> +               /opt/gnu/bin/make install                   +
>>> +----------------------------------------------+
>>> make[1]: *** [all-recursive] Error 1
>>> make: *** [all] Error 2
>>> </snip>
>>>
>>>
>>> I re-untar'd before each compile/make attempt.  Let me know if
this
>> is
>>> something that I can resolve.
>>>
>>> Vladimir
>>>
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
Of
>>> Michael Jerris
>>> Sent: Saturday, August 08, 2009 12:37 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for
Solaris
>>> 10/x86
>>>
>>> This is not currently a supported platform, it only builds on 64
> bit
>>> right now I think on solaris.
>>>
>>> Mike
>>>
>>> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>>>
>>>> Hello,
>>>>
>>>> Does anyone have, or know where to get, a pre-compiled copy of
>>>> FreeSwitch for Solaris 10/x86?
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> rs
>>> http://www.freeswitch.org
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>>> http://www.freeswitch.org
>> Michal Bielicki
>> Leiter der Niederlassung
>> HaloKwadrat Sp. z o.o.
>> Niederlassung Kleinmachnow
>> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
>> Ust.Id.: DE261885536
>> Geschaeftsfuehrer: Aleksander Wiercinski
>> Meiereifeld 2b, 14532 Kleinmachnow
>> t. +49 33203 263220
>> f. +49 33203 263229 sip. info at halokwadrat.de
>> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
>> Hauptgesch?ftsstelle:
>> Halo Kwadrat Sp. z o.o.
>> ul. Polna 46/14
>> 00-644 Warszawa, Polen
>> EIngetragen im HRB Warszawa, KRS 0000153539
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>> http://www.freeswitch.org
>>
>>
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
> 
> 
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
> http://www.freeswitch.org
> 
> 

_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
rs
http://www.freeswitch.org




From e.schmidbauer at gmail.com  Mon Aug 31 10:20:44 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Mon, 31 Aug 2009 13:20:44 -0400
Subject: [Freeswitch-users] remote endpoints
Message-ID: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>

I am unable to call a user outside of our local area network. the user
is registered on the external profile but there is no way to call the
phone. does anyone have any suggestions how to do this?



From msc at freeswitch.org  Mon Aug 31 10:42:21 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 10:42:21 -0700
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
Message-ID: <87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>

Could you give a few more details? For example, could you pastebin the
output of "sofia status profile external"?
-MC

On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer <e.schmidbauer at gmail.com>wrote:

> I am unable to call a user outside of our local area network. the user
> is registered on the external profile but there is no way to call the
> phone. does anyone have any suggestions how to do this?
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Aug 31 11:01:47 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 11:01:47 -0700
Subject: [Freeswitch-users] prompt playback volume using mod_shout
In-Reply-To: <158584.63541.qm@web50801.mail.re2.yahoo.com>
References: <158584.63541.qm@web50801.mail.re2.yahoo.com>
Message-ID: <87f2f3b90908311101m5a4343d9w963535d11e4e55ec@mail.gmail.com>

If I may ask, what "insane" values did you try? Just curious.
-MC

On Sat, Aug 29, 2009 at 8:24 PM, Marc Orenberg <marc at kasteris.com> wrote:

> Hi,
>
> I'm finding that when I record a caller's voice using mod_shout in MP3
> format, the recording is too soft when I play it back.
> I see there's a "volume" setting in shout.conf.xml, but it says, "Dont
> change these unless you are insane".
> When I insanely try to increase this value, I don't here any difference in
> volume.
> Is this the right knob, or is there some other way to do this?
>
> Thanks,
> Marc
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From marc at kasteris.com  Mon Aug 31 11:17:05 2009
From: marc at kasteris.com (Marc Orenberg)
Date: Mon, 31 Aug 2009 11:17:05 -0700 (PDT)
Subject: [Freeswitch-users] prompt playback volume using mod_shout
In-Reply-To: <87f2f3b90908311101m5a4343d9w963535d11e4e55ec@mail.gmail.com>
References: <158584.63541.qm@web50801.mail.re2.yahoo.com>
	<87f2f3b90908311101m5a4343d9w963535d11e4e55ec@mail.gmail.com>
Message-ID: <879878.54860.qm@web50812.mail.re2.yahoo.com>

Michael,

Thanks for the reply.
It was initially set to .1.
So I first tried 1, and then 5.
Then, after a few beers to further impair my judgement, I raised it to 10.
None of these had any effect.

Thanks,
Marc








________________________________
From: Michael Collins <msc at freeswitch.org>
To: freeswitch-users at lists.freeswitch.org
Sent: Monday, August 31, 2009 2:01:47 PM
Subject: Re: [Freeswitch-users] prompt playback volume using mod_shout

If I may ask, what "insane" values did you try? Just curious. 
-MC


On Sat, Aug 29, 2009 at 8:24 PM, Marc Orenberg <marc at kasteris.com> wrote:

Hi, 
>
>
>I'm finding that when I record a caller's voice using mod_shout in MP3 format, the recording is too soft when I play it back.
>I see there's a "volume" setting in shout.conf.xml, but it says, "Dont change these unless you are insane".
>When I insanely try to increase this value, I don't here any difference in volume.
>Is this the right knob, or is there some other way to do this?
>
>
>Thanks,
>Marc
>
>_______________________________________________
>>FreeSWITCH-users mailing list
>FreeSWITCH-users at lists.freeswitch.org
>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>http://www.freeswitch.org
>
>
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From msc at freeswitch.org  Mon Aug 31 11:39:47 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 11:39:47 -0700
Subject: [Freeswitch-users] prompt playback volume using mod_shout
In-Reply-To: <879878.54860.qm@web50812.mail.re2.yahoo.com>
References: <158584.63541.qm@web50801.mail.re2.yahoo.com>
	<87f2f3b90908311101m5a4343d9w963535d11e4e55ec@mail.gmail.com>
	<879878.54860.qm@web50812.mail.re2.yahoo.com>
Message-ID: <87f2f3b90908311139o6936fb5fpca9b23919553d180@mail.gmail.com>

Hmmm... in the libmpg123 docs it says that vol is a double. Try doing
smaller values like .01 and see if you can affect the volume that way.
-MC

On Mon, Aug 31, 2009 at 11:17 AM, Marc Orenberg <marc at kasteris.com> wrote:

> Michael,
>
> Thanks for the reply.
> It was initially set to .1.
> So I first tried 1, and then 5.
> Then, after a few beers to further impair my judgement, I raised it to 10.
> None of these had any effect.
>
> Thanks,
> Marc
>
>
>
>
>
>
> ------------------------------
> *From:* Michael Collins <msc at freeswitch.org>
> *To:* freeswitch-users at lists.freeswitch.org
> *Sent:* Monday, August 31, 2009 2:01:47 PM
> *Subject:* Re: [Freeswitch-users] prompt playback volume using mod_shout
>
> If I may ask, what "insane" values did you try? Just curious.
> -MC
>
> On Sat, Aug 29, 2009 at 8:24 PM, Marc Orenberg <marc at kasteris.com> wrote:
>
>> Hi,
>>
>> I'm finding that when I record a caller's voice using mod_shout in MP3
>> format, the recording is too soft when I play it back.
>> I see there's a "volume" setting in shout.conf.xml, but it says, "Dont
>> change these unless you are insane".
>> When I insanely try to increase this value, I don't here any difference in
>> volume.
>> Is this the right knob, or is there some other way to do this?
>>
>> Thanks,
>> Marc
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From e.schmidbauer at gmail.com  Mon Aug 31 11:54:52 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Mon, 31 Aug 2009 14:54:52 -0400
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
Message-ID: <2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>

=================================================================================================
Name                    external
Domain Name             N/A
DBName                  sofia_reg_external
Pres Hosts
Dialplan                XML
Context                 public
Challenge Realm         auto_to
RTP-IP                  192.168.0.125
Ext-RTP-IP              98.118.151.30
SIP-IP                  192.168.0.125
Ext-SIP-IP              98.118.151.30
URL                     sip:mod_sofia at 192.168.0.125:5080
BIND-URL                sip:mod_sofia at 192.168.0.125:5080
HOLD-MUSIC              local_stream://moh
OUTBOUND-PROXY          N/A
CODECS
CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h@20i,PCMU,PCMA,GSM
TEL-EVENT               101
DTMF-MODE               rfc2833
CNG                     13
SESSION-TO              0
MAX-DIALOG              0
NOMEDIA                 false
LATE-NEG                false
PROXY-MEDIA             false
AGGRESSIVENAT           false
STUN-ENABLED            true
STUN-AUTO-DISABLE       false
CALLS-IN                17
FAILED-CALLS-IN         11
CALLS-OUT               9
FAILED-CALLS-OUT        9

Registrations:
=================================================================================================
Call-ID:        MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
User:           1000 at server1.altpressonline.com
Contact:        "1000" <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
Agent:          X-Lite release 1103k stamp 53621
Status:         Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
Host:           server1.altpressonline.com
IP:             69.204.30.67
Port:           16006
Auth-User:      1000
Auth-Realm:     server1.altpressonline.com

=================================================================================================

sorry, i don't know how to login to pastebin

On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org> wrote:
> Could you give a few more details? For example, could you pastebin the
> output of "sofia status profile external"?
> -MC
>
> On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer <e.schmidbauer at gmail.com>
> wrote:
>>
>> I am unable to call a user outside of our local area network. the user
>> is registered on the external profile but there is no way to call the
>> phone. does anyone have any suggestions how to do this?
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From jmesquita at freeswitch.org  Mon Aug 31 12:01:01 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 31 Aug 2009 16:01:01 -0300
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
Message-ID: <ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>

Check the password dialog. It will tell you what the username/password is.

post the logs for a call as well, please.

Regards,

jmesquita

On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer <e.schmidbauer at gmail.com>wrote:

>
> =================================================================================================
> Name                    external
> Domain Name             N/A
> DBName                  sofia_reg_external
> Pres Hosts
> Dialplan                XML
> Context                 public
> Challenge Realm         auto_to
> RTP-IP                  192.168.0.125
> Ext-RTP-IP              98.118.151.30
> SIP-IP                  192.168.0.125
> Ext-SIP-IP              98.118.151.30
> URL                     sip:mod_sofia at 192.168.0.125:5080
> BIND-URL                sip:mod_sofia at 192.168.0.125:5080
> HOLD-MUSIC              local_stream://moh
> OUTBOUND-PROXY          N/A
> CODECS
> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h@20i,PCMU,PCMA,GSM
> TEL-EVENT               101
> DTMF-MODE               rfc2833
> CNG                     13
> SESSION-TO              0
> MAX-DIALOG              0
> NOMEDIA                 false
> LATE-NEG                false
> PROXY-MEDIA             false
> AGGRESSIVENAT           false
> STUN-ENABLED            true
> STUN-AUTO-DISABLE       false
> CALLS-IN                17
> FAILED-CALLS-IN         11
> CALLS-OUT               9
> FAILED-CALLS-OUT        9
>
> Registrations:
>
> =================================================================================================
> Call-ID:        MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
> User:           1000 at server1.altpressonline.com
> Contact:        "1000" <sip:1000 at 69.204.30.67:16006
> ;rinstance=50cbd6140e1ab991>
> Agent:          X-Lite release 1103k stamp 53621
> Status:         Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
> Host:           server1.altpressonline.com
> IP:             69.204.30.67
> Port:           16006
> Auth-User:      1000
> Auth-Realm:     server1.altpressonline.com
>
>
> =================================================================================================
>
> sorry, i don't know how to login to pastebin
>
> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org>
> wrote:
> > Could you give a few more details? For example, could you pastebin the
> > output of "sofia status profile external"?
> > -MC
> >
> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer <e.schmidbauer at gmail.com
> >
> > wrote:
> >>
> >> I am unable to call a user outside of our local area network. the user
> >> is registered on the external profile but there is no way to call the
> >> phone. does anyone have any suggestions how to do this?
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From e.schmidbauer at gmail.com  Mon Aug 31 12:07:33 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Mon, 31 Aug 2009 15:07:33 -0400
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
Message-ID: <2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>

thanks...heres the pastebin:
http://pastebin.freeswitch.org/10171

2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Check the password dialog. It will tell you what the username/password is.
>
> post the logs for a call as well, please.
>
> Regards,
>
> jmesquita
>
> On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer <e.schmidbauer at gmail.com>
> wrote:
>>
>>
>> =================================================================================================
>> Name ? ? ? ? ? ? ? ? ? ?external
>> Domain Name ? ? ? ? ? ? N/A
>> DBName ? ? ? ? ? ? ? ? ?sofia_reg_external
>> Pres Hosts
>> Dialplan ? ? ? ? ? ? ? ?XML
>> Context ? ? ? ? ? ? ? ? public
>> Challenge Realm ? ? ? ? auto_to
>> RTP-IP ? ? ? ? ? ? ? ? ?192.168.0.125
>> Ext-RTP-IP ? ? ? ? ? ? ?98.118.151.30
>> SIP-IP ? ? ? ? ? ? ? ? ?192.168.0.125
>> Ext-SIP-IP ? ? ? ? ? ? ?98.118.151.30
>> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.0.125:5080
>> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.0.125:5080
>> HOLD-MUSIC ? ? ? ? ? ? ?local_stream://moh
>> OUTBOUND-PROXY ? ? ? ? ?N/A
>> CODECS
>> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h@20i,PCMU,PCMA,GSM
>> TEL-EVENT ? ? ? ? ? ? ? 101
>> DTMF-MODE ? ? ? ? ? ? ? rfc2833
>> CNG ? ? ? ? ? ? ? ? ? ? 13
>> SESSION-TO ? ? ? ? ? ? ?0
>> MAX-DIALOG ? ? ? ? ? ? ?0
>> NOMEDIA ? ? ? ? ? ? ? ? false
>> LATE-NEG ? ? ? ? ? ? ? ?false
>> PROXY-MEDIA ? ? ? ? ? ? false
>> AGGRESSIVENAT ? ? ? ? ? false
>> STUN-ENABLED ? ? ? ? ? ?true
>> STUN-AUTO-DISABLE ? ? ? false
>> CALLS-IN ? ? ? ? ? ? ? ?17
>> FAILED-CALLS-IN ? ? ? ? 11
>> CALLS-OUT ? ? ? ? ? ? ? 9
>> FAILED-CALLS-OUT ? ? ? ?9
>>
>> Registrations:
>>
>> =================================================================================================
>> Call-ID: ? ? ? ?MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
>> User: ? ? ? ? ? 1000 at server1.altpressonline.com
>> Contact: ? ? ? ?"1000"
>> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
>> Agent: ? ? ? ? ?X-Lite release 1103k stamp 53621
>> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
>> Host: ? ? ? ? ? server1.altpressonline.com
>> IP: ? ? ? ? ? ? 69.204.30.67
>> Port: ? ? ? ? ? 16006
>> Auth-User: ? ? ?1000
>> Auth-Realm: ? ? server1.altpressonline.com
>>
>>
>> =================================================================================================
>>
>> sorry, i don't know how to login to pastebin
>>
>> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org>
>> wrote:
>> > Could you give a few more details? For example, could you pastebin the
>> > output of "sofia status profile external"?
>> > -MC
>> >
>> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
>> > <e.schmidbauer at gmail.com>
>> > wrote:
>> >>
>> >> I am unable to call a user outside of our local area network. the user
>> >> is registered on the external profile but there is no way to call the
>> >> phone. does anyone have any suggestions how to do this?
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From jmesquita at freeswitch.org  Mon Aug 31 12:30:11 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 31 Aug 2009 16:30:11 -0300
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
	<2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
Message-ID: <ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>

   1. 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel
   sofia/external/Anonymous at anonymous.invalid entering state [terminated][
   487]

The far end seems to be replying with 487 - Request Terminated...

Nothing wrong on FS, seems to be a problem with your endpoints. Can you
enable a sip trace?

jmesquita


On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <e.schmidbauer at gmail.com>wrote:

> thanks...heres the pastebin:
> http://pastebin.freeswitch.org/10171
>
> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > Check the password dialog. It will tell you what the username/password
> is.
> >
> > post the logs for a call as well, please.
> >
> > Regards,
> >
> > jmesquita
> >
> > On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer <e.schmidbauer at gmail.com>
> > wrote:
> >>
> >>
> >>
> =================================================================================================
> >> Name                    external
> >> Domain Name             N/A
> >> DBName                  sofia_reg_external
> >> Pres Hosts
> >> Dialplan                XML
> >> Context                 public
> >> Challenge Realm         auto_to
> >> RTP-IP                  192.168.0.125
> >> Ext-RTP-IP              98.118.151.30
> >> SIP-IP                  192.168.0.125
> >> Ext-SIP-IP              98.118.151.30
> >> URL                     sip:mod_sofia at 192.168.0.125:5080
> >> BIND-URL                sip:mod_sofia at 192.168.0.125:5080
> >> HOLD-MUSIC              local_stream://moh
> >> OUTBOUND-PROXY          N/A
> >> CODECS
> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h@20i,PCMU,PCMA,GSM
> >> TEL-EVENT               101
> >> DTMF-MODE               rfc2833
> >> CNG                     13
> >> SESSION-TO              0
> >> MAX-DIALOG              0
> >> NOMEDIA                 false
> >> LATE-NEG                false
> >> PROXY-MEDIA             false
> >> AGGRESSIVENAT           false
> >> STUN-ENABLED            true
> >> STUN-AUTO-DISABLE       false
> >> CALLS-IN                17
> >> FAILED-CALLS-IN         11
> >> CALLS-OUT               9
> >> FAILED-CALLS-OUT        9
> >>
> >> Registrations:
> >>
> >>
> =================================================================================================
> >> Call-ID:        MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
> >> User:           1000 at server1.altpressonline.com
> >> Contact:        "1000"
> >> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
> >> Agent:          X-Lite release 1103k stamp 53621
> >> Status:         Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
> >> Host:           server1.altpressonline.com
> >> IP:             69.204.30.67
> >> Port:           16006
> >> Auth-User:      1000
> >> Auth-Realm:     server1.altpressonline.com
> >>
> >>
> >>
> =================================================================================================
> >>
> >> sorry, i don't know how to login to pastebin
> >>
> >> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org>
> >> wrote:
> >> > Could you give a few more details? For example, could you pastebin the
> >> > output of "sofia status profile external"?
> >> > -MC
> >> >
> >> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
> >> > <e.schmidbauer at gmail.com>
> >> > wrote:
> >> >>
> >> >> I am unable to call a user outside of our local area network. the
> user
> >> >> is registered on the external profile but there is no way to call the
> >> >> phone. does anyone have any suggestions how to do this?
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From tculjaga at gmail.com  Mon Aug 31 12:45:58 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Mon, 31 Aug 2009 21:45:58 +0200
Subject: [Freeswitch-users] T.38 with V.34 support
In-Reply-To: <4A9BE78D.5040000@coppice.org>
References: <4A9BE78D.5040000@coppice.org>
Message-ID: <65d96fc80908311245j51026fd5g3c93213c67ba7e73@mail.gmail.com>

Hi Steve,

what exactly do you need?

i was experimenting with V.17/V.29 <-> V.34 fax interworking....

well... I have V.34 machines ready over here... I just need to reconfigure
some CPE to point towards FS and thats it.... what setup do you need ?

T.


On Mon, Aug 31, 2009 at 5:09 PM, Steve Underwood <steveu at coppice.org> wrote:

> Hi,
>
> Does anyone have access to a system that supports T.38 FAXing at V.34
> rates. These things seem to be as rare as bugless software.
>
> Steve
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From juanma.v82 at gmail.com  Mon Aug 31 12:56:48 2009
From: juanma.v82 at gmail.com (Juan Manuel Vicente)
Date: Mon, 31 Aug 2009 16:56:48 -0300
Subject: [Freeswitch-users] Load Balacing
Message-ID: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>

Hello:

I am using two boxes, with same domain, to register about 1000 users working
together a DNS box doing round robin to resolve two differentes Ips one to
each box. This two boxes are using the same database (through unixodbc).

When I use the funcion sofia_contact or sofia profile status both boxes show
the status of the user. Does not matter where the resgistration was did.

But when I try to call the user this call fails, the call only works if the
call is sent through the same box where the user is registered.

Any ideas where i need to looking for ?

Thx in Advance.
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From e.schmidbauer at gmail.com  Mon Aug 31 13:18:56 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Mon, 31 Aug 2009 16:18:56 -0400
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
	<2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
	<ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>
Message-ID: <2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com>

here's the sip trace:
http://pastebin.freeswitch.org/10172



2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel
> sofia/external/Anonymous at anonymous.invalid entering state [terminated][487]
>
> The far end seems to be replying with 487 - Request Terminated...
>
> Nothing wrong on FS, seems to be a problem with your endpoints. Can you
> enable a sip trace?
>
> jmesquita
>
>
> On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <e.schmidbauer at gmail.com>
> wrote:
>>
>> thanks...heres the pastebin:
>> http://pastebin.freeswitch.org/10171
>>
>> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> > Check the password dialog. It will tell you what the username/password
>> > is.
>> >
>> > post the logs for a call as well, please.
>> >
>> > Regards,
>> >
>> > jmesquita
>> >
>> > On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer <e.schmidbauer at gmail.com>
>> > wrote:
>> >>
>> >>
>> >>
>> >> =================================================================================================
>> >> Name ? ? ? ? ? ? ? ? ? ?external
>> >> Domain Name ? ? ? ? ? ? N/A
>> >> DBName ? ? ? ? ? ? ? ? ?sofia_reg_external
>> >> Pres Hosts
>> >> Dialplan ? ? ? ? ? ? ? ?XML
>> >> Context ? ? ? ? ? ? ? ? public
>> >> Challenge Realm ? ? ? ? auto_to
>> >> RTP-IP ? ? ? ? ? ? ? ? ?192.168.0.125
>> >> Ext-RTP-IP ? ? ? ? ? ? ?98.118.151.30
>> >> SIP-IP ? ? ? ? ? ? ? ? ?192.168.0.125
>> >> Ext-SIP-IP ? ? ? ? ? ? ?98.118.151.30
>> >> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.0.125:5080
>> >> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.0.125:5080
>> >> HOLD-MUSIC ? ? ? ? ? ? ?local_stream://moh
>> >> OUTBOUND-PROXY ? ? ? ? ?N/A
>> >> CODECS
>> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h@20i,PCMU,PCMA,GSM
>> >> TEL-EVENT ? ? ? ? ? ? ? 101
>> >> DTMF-MODE ? ? ? ? ? ? ? rfc2833
>> >> CNG ? ? ? ? ? ? ? ? ? ? 13
>> >> SESSION-TO ? ? ? ? ? ? ?0
>> >> MAX-DIALOG ? ? ? ? ? ? ?0
>> >> NOMEDIA ? ? ? ? ? ? ? ? false
>> >> LATE-NEG ? ? ? ? ? ? ? ?false
>> >> PROXY-MEDIA ? ? ? ? ? ? false
>> >> AGGRESSIVENAT ? ? ? ? ? false
>> >> STUN-ENABLED ? ? ? ? ? ?true
>> >> STUN-AUTO-DISABLE ? ? ? false
>> >> CALLS-IN ? ? ? ? ? ? ? ?17
>> >> FAILED-CALLS-IN ? ? ? ? 11
>> >> CALLS-OUT ? ? ? ? ? ? ? 9
>> >> FAILED-CALLS-OUT ? ? ? ?9
>> >>
>> >> Registrations:
>> >>
>> >>
>> >> =================================================================================================
>> >> Call-ID: ? ? ? ?MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
>> >> User: ? ? ? ? ? 1000 at server1.altpressonline.com
>> >> Contact: ? ? ? ?"1000"
>> >> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
>> >> Agent: ? ? ? ? ?X-Lite release 1103k stamp 53621
>> >> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
>> >> Host: ? ? ? ? ? server1.altpressonline.com
>> >> IP: ? ? ? ? ? ? 69.204.30.67
>> >> Port: ? ? ? ? ? 16006
>> >> Auth-User: ? ? ?1000
>> >> Auth-Realm: ? ? server1.altpressonline.com
>> >>
>> >>
>> >>
>> >> =================================================================================================
>> >>
>> >> sorry, i don't know how to login to pastebin
>> >>
>> >> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org>
>> >> wrote:
>> >> > Could you give a few more details? For example, could you pastebin
>> >> > the
>> >> > output of "sofia status profile external"?
>> >> > -MC
>> >> >
>> >> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
>> >> > <e.schmidbauer at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >> I am unable to call a user outside of our local area network. the
>> >> >> user
>> >> >> is registered on the external profile but there is no way to call
>> >> >> the
>> >> >> phone. does anyone have any suggestions how to do this?
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From pjintheusa at gmail.com  Mon Aug 31 13:23:36 2009
From: pjintheusa at gmail.com (Phillip Jones)
Date: Mon, 31 Aug 2009 16:23:36 -0400
Subject: [Freeswitch-users] Can I stream a file to a parked call.
In-Reply-To: <191c3a030908281412h51b75088s8c40bd0a53688928@mail.gmail.com>
References: <367751820908271309g4ccec8dex5ab4b03c76875260@mail.gmail.com>
	<191c3a030908271330x2a8ea039i1b60c78618b1060@mail.gmail.com>
	<367751820908281313s170307a6ob3c191f55a537453@mail.gmail.com>
	<191c3a030908281333w39f001bcx7b9914f7bdda4f8f@mail.gmail.com>
	<191c3a030908281340m83d3c55q96f1257341250b86@mail.gmail.com>
	<367751820908281353l69b26639va13b3d63321b89e5@mail.gmail.com>
	<191c3a030908281412h51b75088s8c40bd0a53688928@mail.gmail.com>
Message-ID: <367751820908311323v3ce593fej6e5fbf13e75c4e61@mail.gmail.com>

That makes perfect sense - I now send the call to a managed app instead of
park and every thing works great. thanks.



On Fri, Aug 28, 2009 at 5:12 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> Taking over the session while it's parked like that from your code and
> asking it to play a file is making it do 2 things at once.
> The session's thread is already busy in the park loop so you would have to
> change it's state to something passive like soft_execute so
> it's thread was not doing anything or send it a message so the park loop
> would pick it up by using the uuid_broadcast app::arg
>
> there are variables you can set that begin group_confirm_* that you can set
> to do this all without even using a script.
>
>
> http://wiki.freeswitch.org/wiki/Channel_Variables#Answer_confirmation_variables
>
>
>
>
>
> On Fri, Aug 28, 2009 at 3:53 PM, Phillip Jones <pjintheusa at gmail.com>wrote:
>
>> The dialplan is is kicking off the C# script:
>>
>> <condition field="destination_number" expression="^(12125553666)$">
>>     * <action application="managed" data="MyFSProjects.ProcessCall"/>*
>> </condition>
>>
>> I launch a thread and specify the uuid to originate:
>>
>> fsApi.Execute("originate",
>> string.Format("{{ignore_early_media=true,absolute_codec_string='PCMU'}}[*
>> origination_uuid={0}*,origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>> &park", *call.Uuid*, call.OutgoingCallerID, call.NumberToDial));
>>
>> in the callback i try and play the wav file:
>>
>> FreeSWITCH.Native.ManagedSession blegSession = new
>> FreeSWITCH.Native.ManagedSession(callID);
>>
>> if (blegSession.Ready() && blegSession.mediaReady())
>> {
>>       blegSession.Answer();
>>       blegSession.Execute("playback",
>> "prompts/press-1-to-accept-call-from.wav");
>> }
>>
>> I go on to bridge the leg a and b - that bits works fine.
>>
>>
>>
>> On Fri, Aug 28, 2009 at 4:40 PM, Anthony Minessale<
>> anthony.minessale at gmail.com> wrote:
>> > nevermind,
>> >
>> > I didn't look closely at your code.
>> >
>> > how did you end up with blegSession ?
>> > Are you running this script from the CLI or via the application
>> interface.
>> >
>> >
>> >
>> >
>> >
>> >
>> >
>> > On Fri, Aug 28, 2009 at 3:33 PM, Anthony Minessale
>> > <anthony.minessale at gmail.com> wrote:
>> >>
>> >> the L16 codec is used because the wav files is raw PCM audio
>> >> which is being encoded to PCMU
>> >>
>> >> Have you tried doing this to a locally registered phone?
>> >>
>> >>
>> >>
>> >> On Fri, Aug 28, 2009 at 3:13 PM, Phillip Jones <pjintheusa at gmail.com>
>> >> wrote:
>> >>>
>> >>> thanks for the reply.
>> >>>
>> >>> Tried this and exactly the same result. The prompt is slowed - as if
>> >>> the person is drunk.
>> >>>
>> >>> I do notice that the originating call SDP shows:
>> >>>
>> >>> a=rtpmap:0 PCMU/8000
>> >>>
>> >>> but the play tries to use:
>> >>>
>> >>> EXECUTE sofia/external/6093693828
>> >>> playback(prompts/press-1-to-accept-call-from.wav)
>> >>> 2009-08-28 16:03:14.507375 [DEBUG] switch_ivr_play_say.c:1097 Codec
>> >>> Activated L16 at 8000hz 1 channels 20ms
>> >>>
>> >>>
>> >>> Should the codec be PCMU at 8000hz as opposed to L16 at 8000hz? If so of
>> >>> there a way of forcing this codec for play.
>> >>>
>> >>> Thanks
>> >>>
>> >>>
>> >>> Phillip
>> >>>
>> >>>
>> >>> Full code & logs
>> >>> http://pastebin.freeswitch.org/10147
>> >>> http://pastebin.freeswitch.org/10150
>> >>>
>> >>>
>> >>>
>> >>> On Thu, Aug 27, 2009 at 4:30 PM, Anthony
>> >>> Minessale<anthony.minessale at gmail.com> wrote:
>> >>> > try wedging {ignore_early_media=true} before the first [ in your
>> dial
>> >>> > string
>> >>> > and eliminate the code waiting for answer.
>> >>> >
>> >>> >
>> >>> > On Thu, Aug 27, 2009 at 3:09 PM, Phillip Jones <
>> pjintheusa at gmail.com>
>> >>> > wrote:
>> >>> >>
>> >>> >> Hi there,
>> >>> >>
>> >>> >> I know there are other ways of doing this. I am just trying get to
>> >>> >> know have fun with the FreeSWITCH API.
>> >>> >>
>> >>> >> I am using originate and park a call:
>> >>> >>
>> >>> >> fsApi.Execute("originate",
>> >>> >>
>> >>> >>
>> >>> >>
>> string.Format("[origination_uuid={0},origination_caller_id_number={1}]sofia/gateway/broadvox/{2}
>> >>> >> &park", blegSession.Uuid, OutgoingCallerID, NumberToDial));
>> >>> >>
>> >>> >> That works great, the phone rings. I want to play something to this
>> >>> >> called party when they pick up.
>> >>> >>
>> >>> >>  while (!blegSession.answered())
>> >>> >>  {
>> >>> >>        Log.WriteLine(LogLevel.Alert,
>> >>> >> "Inside::CallReturns:!Session.answered::Loop");
>> >>> >>        blegSession.sleep(500, 1);
>> >>> >>  }
>> >>> >>
>> >>> >> string promptFile = "prompts/whisper.wav";
>> >>> >> blegSession.StreamFile(promptFile, 0);
>> >>> >>
>> >>> >>
>> >>> >> This works - but the audio is choppy and slow.
>> >>> >>
>> >>> >> Is there something I need to do to that parked call before
>> streaming
>> >>> >> that
>> >>> >> file?
>> >>> >>
>> >>> >>
>> >>> >> Thanks for any input.
>> >>> >>
>> >>> >> Phil
>> >>> >>
>> >>> >> _______________________________________________
>> >>> >> FreeSWITCH-users mailing list
>> >>> >> FreeSWITCH-users at lists.freeswitch.org
>> >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >>
>> >>> >> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> >> http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>> >
>> >>> > --
>> >>> > Anthony Minessale II
>> >>> >
>> >>> > FreeSWITCH http://www.freeswitch.org/
>> >>> > ClueCon http://www.cluecon.com/
>> >>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >>> >
>> >>> > AIM: anthm
>> >>> > MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >>> > IRC: irc.freenode.net #freeswitch
>> >>> >
>> >>> > FreeSWITCH Developer Conference
>> >>> > sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >>> > iax:guest at conference.freeswitch.org/888
>> >>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >>> > pstn:213-799-1400
>> >>> >
>> >>> > _______________________________________________
>> >>> > FreeSWITCH-users mailing list
>> >>> > FreeSWITCH-users at lists.freeswitch.org
>> >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> >
>> >>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> > http://www.freeswitch.org
>> >>> >
>> >>> >
>> >>>
>> >>> _______________________________________________
>> >>> FreeSWITCH-users mailing list
>> >>> FreeSWITCH-users at lists.freeswitch.org
>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>> UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >>> http://www.freeswitch.org
>> >>
>> >>
>> >>
>> >> --
>> >> Anthony Minessale II
>> >>
>> >> FreeSWITCH http://www.freeswitch.org/
>> >> ClueCon http://www.cluecon.com/
>> >> Twitter: http://twitter.com/FreeSWITCH_wire
>> >>
>> >> AIM: anthm
>> >> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
>> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> >> IRC: irc.freenode.net #freeswitch
>> >>
>> >> FreeSWITCH Developer Conference
>> >> sip:888 at conference.freeswitch.org<sip%3A888 at conference.freeswitch.org>
>> >> iax:guest at conference.freeswitch.org/888
>> >> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> >> pstn:213-799-1400
>> >
>> >
>> >
>> > --
>> > Anthony Minessale II
>> >
>> > FreeSWITCH http://www.freeswitch.org/
>> > ClueCon http://www.cluecon.com/
>> > Twitter: http://twitter.com/FreeSWITCH_wire
>> >
>> > AIM: anthm
>> > MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> > IRC: irc.freenode.net #freeswitch
>> >
>> > FreeSWITCH Developer Conference
>> > sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
>> > iax:guest at conference.freeswitch.org/888
>> > googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> > pstn:213-799-1400
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:
>> http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
> ClueCon http://www.cluecon.com/
> Twitter: http://twitter.com/FreeSWITCH_wire
>
> AIM: anthm
> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
> IRC: irc.freenode.net #freeswitch
>
> FreeSWITCH Developer Conference
> sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
> iax:guest at conference.freeswitch.org/888
> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
> pstn:213-799-1400
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From mike at jerris.com  Mon Aug 31 14:01:05 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 31 Aug 2009 17:01:05 -0400
Subject: [Freeswitch-users] Freeswitch pre-compiled for Solaris 10/x86
In-Reply-To: <4a9c0516.19bd720a.23b1.27df@mx.google.com>
References: <4a7b530a.29578c0a.53a8.0450@mx.google.com>	<84131AE6-4BBE-453C-90E6-E26765A49C1B@jerris.com>	<4EF4BF1E8F43894386584BE36354494A13D90103@ZANEMS01.cc-ntd1.covad.com>	<4a80632f.1508c00a.4d3c.090d@mx.google.com>	<A9660A56-BB38-45C5-A58C-11A08CB073AD@halo2.pl>	<4a8464d3.02578c0a.0e5c.ffff9baf@mx.google.com>	<4A92566C.3030200@blueface.ie>	<4a9be20d.22ba720a.5469.ffffb6ad@mx.google.com>
	<4A9BFEFC.2090800@blueface.ie>
	<4a9c0516.19bd720a.23b1.27df@mx.google.com>
Message-ID: <CAF0BC3A-32AC-4B97-9035-B16534FA4749@jerris.com>

What was the error building (which lib) ?  We should probably push  
this in so configure supplies this to the build system


Mike

On Aug 31, 2009, at 1:16 PM, vmorales wrote:

> Using Sun Studio and exporting the following flag:
>
> 	export CFLAGS="-lresolv"
>
> Is how our sysadmin was able to get FS to build.
>
> Vladimir
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Bruce McAlister
> Sent: Monday, August 31, 2009 12:49 PM
> To: freeswitch-users at lists.freeswitch.org
> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
> 10/x86
>
> Hi Vladimir,
>
> Would you mind letting us know what your sysadmin did to achieve a
> successful build?
>
> I am still having issues building freeswitch.
>
> Thanks
> Bruce
>
> vmorales wrote:
>> Sorry for the late reply, I was out of the office for the past two
>> weeks.  I'm not sure if there were updates over the mailing list
>> (still catching up), but while I was out, one of our sys admins was
>> able to get this compiled and has been verified to be functional on
>> our Solaris 10/x86 server.
>>
>> Vladimir
>>
>> -----Original Message-----
>> From: freeswitch-users-bounces at lists.freeswitch.org
>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>> Bruce McAlister
>> Sent: Monday, August 24, 2009 4:59 AM
>> To: freeswitch-users at lists.freeswitch.org
>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>> 10/x86
>>
>> Hi Vladimir,
>>
>> Did you get an update on this at all?
>>
>> Thanks
>>
>>
>> vmorales wrote:
>>> Hi Michal,
>>>
>>> Just checking in to see if you've been able to take a stab at this.
>>>
>>> Thanks,
>>> Vladimir
>>>
>>>
>>> -----Original Message-----
>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
>>> Michal Bielicki
>>> Sent: Tuesday, August 11, 2009 5:33 AM
>>> To: freeswitch-users at lists.freeswitch.org
>>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for Solaris
>>> 10/x86
>>>
>>> I'll retst it later today and give you a link with instructions
>>>
>>> Am 10.08.2009 um 20:14 schrieb vmorales:
>>>
>>>> By "./compile" I was referring to "./configure"
>>>>
>>>> Vladimir
>>>>
>>>> -----Original Message-----
>>>> From: vmorales [mailto:email.list.subscriber at gmail.com]
>>>> Sent: Monday, August 10, 2009 11:49 AM
>>>> To: 'freeswitch-users at lists.freeswitch.org'
>>>> Subject: RE: [Freeswitch-users] Freeswitch pre-compiled for
> Solaris
>>>> 10/x86
>>>>
>>>> Thanks for the response(s):
>>>>
>>>> I ran the "./compile" script with a set PREFIX.  This took a few
>>>> attempts with errors before it was able to complete error-free, as
>> I
>>>> had to install libtool.
>>>>
>>>> Since then, I have tried running 'make', 'gmake', and
>>>> '/opt/gnu/bin/make', but each results with an error.  This is the
>>>> error when running 'make' or 'gmake':
>>>>
>>>> <snip>
>>>> make: Fatal error: Command failed for target `all-recursive'
>>>> Current working directory /home/vmorales/freeswitch-1.0.4
>>>> *** Error code 1
>>>> make: Fatal error: Command failed for target `all'
>>>> </snip>
>>>>
>>>>
>>>> This is the error when running '/opt/gnu/bin/make':
>>>>
>>>> <snip>
>>>> make[5]: *** [mod_amr.so] Error 1
>>>> make[4]: *** [all] Error 1
>>>> make[3]: *** [mod_amr-all] Error 1
>>>> make[2]: *** [all-recursive] Error 1
>>>> Making all in build
>>>> +-------- FreeSWITCH Build Complete -----------+
>>>> + FreeSWITCH has been successfully built.      +
>>>> + Install by running:                          +
>>>> +                                              +
>>>> +               /opt/gnu/bin/make install                   +
>>>> +----------------------------------------------+
>>>> make[1]: *** [all-recursive] Error 1
>>>> make: *** [all] Error 2
>>>> </snip>
>>>>
>>>>
>>>> I re-untar'd before each compile/make attempt.  Let me know if
> this
>>> is
>>>> something that I can resolve.
>>>>
>>>> Vladimir
>>>>
>>>>
>>>> -----Original Message-----
>>>> From: freeswitch-users-bounces at lists.freeswitch.org
>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf
> Of
>>>> Michael Jerris
>>>> Sent: Saturday, August 08, 2009 12:37 AM
>>>> To: freeswitch-users at lists.freeswitch.org
>>>> Subject: Re: [Freeswitch-users] Freeswitch pre-compiled for
> Solaris
>>>> 10/x86
>>>>
>>>> This is not currently a supported platform, it only builds on 64
>> bit
>>>> right now I think on solaris.
>>>>
>>>> Mike
>>>>
>>>> On Aug 6, 2009, at 6:03 PM, vmorales wrote:
>>>>
>>>>> Hello,
>>>>>
>>>>> Does anyone have, or know where to get, a pre-compiled copy of
>>>>> FreeSwitch for Solaris 10/x86?
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>>> rs
>>>> http://www.freeswitch.org
>>>>
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>>> rs
>>>> http://www.freeswitch.org
>>> Michal Bielicki
>>> Leiter der Niederlassung
>>> HaloKwadrat Sp. z o.o.
>>> Niederlassung Kleinmachnow
>>> Eingetragen im Handelsregister beim Amtsgericht Potsdam, HRB21422P
>>> Ust.Id.: DE261885536
>>> Geschaeftsfuehrer: Aleksander Wiercinski
>>> Meiereifeld 2b, 14532 Kleinmachnow
>>> t. +49 33203 263220
>>> f. +49 33203 263229 sip. info at halokwadrat.de
>>> e. michal.bielicki at halokwadrat.de | w. www.halokwadrat.de
>>> Hauptgesch?ftsstelle:
>>> Halo Kwadrat Sp. z o.o.
>>> ul. Polna 46/14
>>> 00-644 Warszawa, Polen
>>> EIngetragen im HRB Warszawa, KRS 0000153539
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>>> http://www.freeswitch.org
>>>
>>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
>> rs
>> http://www.freeswitch.org
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
>> http://www.freeswitch.org
>>
>>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use
> rs
> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From mike at jerris.com  Mon Aug 31 14:02:18 2009
From: mike at jerris.com (Michael Jerris)
Date: Mon, 31 Aug 2009 17:02:18 -0400
Subject: [Freeswitch-users] Load Balacing
In-Reply-To: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>
References: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>
Message-ID: <B84A71A8-9C66-44F1-8D11-FDC7BB99AB60@jerris.com>

nat mapping on the client box not allowing the invite from another ip?

Mike

On Aug 31, 2009, at 3:56 PM, Juan Manuel Vicente wrote:

> Hello:
>
> I am using two boxes, with same domain, to register about 1000 users  
> working together a DNS box doing round robin to resolve two  
> differentes Ips one to each box. This two boxes are using the same  
> database (through unixodbc).
>
> When I use the funcion sofia_contact or sofia profile status both  
> boxes show the status of the user. Does not matter where the  
> resgistration was did.
>
> But when I try to call the user this call fails, the call only works  
> if the call is sent through the same box where the user is registered.
>
> Any ideas where i need to looking for ?
>
> Thx in Advance.
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- 
> users
> http://www.freeswitch.org




From krice at freeswitch.org  Mon Aug 31 14:35:22 2009
From: krice at freeswitch.org (Ken Rice)
Date: Mon, 31 Aug 2009 16:35:22 -0500
Subject: [Freeswitch-users] Load Balacing
In-Reply-To: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>
Message-ID: <C6C1AC4A.5E4D4%krice@freeswitch.org>

You need to write you a small application to use the sip_registration table
from your odbc datasource and route the call to the appropriate registration
server and then have that server send the calls out... I am doing this exact
same thing now and this is what I had to do to get it to work properly with
a stack of clients behind nat



From: Juan Manuel Vicente <juanma.v82 at gmail.com>
Reply-To: <freeswitch-users at lists.freeswitch.org>
Date: Mon, 31 Aug 2009 16:56:48 -0300
To: <freeswitch-users at lists.freeswitch.org>
Subject: [Freeswitch-users] Load Balacing

Hello:

I am using two boxes, with same domain, to register about 1000 users working
together a DNS box doing round robin to resolve two differentes Ips one to
each box. This two boxes are using the same database (through unixodbc).

When I use the funcion sofia_contact or sofia profile status both boxes show
the status of the user. Does not matter where the resgistration was did.

But when I try to call the user this call fails, the call only works if the
call is sent through the same box where the user is registered.

Any ideas where i need to looking for ?

Thx in Advance.



_______________________________________________
FreeSWITCH-users mailing list
FreeSWITCH-users at lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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From andrew at hijacked.us  Mon Aug 31 14:57:29 2009
From: andrew at hijacked.us (Andrew Thompson)
Date: Mon, 31 Aug 2009 17:57:29 -0400
Subject: [Freeswitch-users] Load Balacing
In-Reply-To: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>
References: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>
Message-ID: <20090831215728.GD20919@hijacked.us>

On Mon, Aug 31, 2009 at 04:56:48PM -0300, Juan Manuel Vicente wrote:
> Hello:
> 
> I am using two boxes, with same domain, to register about 1000 users working
> together a DNS box doing round robin to resolve two differentes Ips one to
> each box. This two boxes are using the same database (through unixodbc).
> 
> When I use the funcion sofia_contact or sofia profile status both boxes show
> the status of the user. Does not matter where the resgistration was did.
> 
> But when I try to call the user this call fails, the call only works if the
> call is sent through the same box where the user is registered.
>

I had the same problem except that I was using the multicast module to
replicate registrations, I added a sofia config parameter to optionally
rewrite the fs_path part of the sip URI on replicated packets so I could
hairpin the sip messages through the original registrar. This doesn't
help you directly but maybe it'll give you an idea?

Andrew



From tculjaga at gmail.com  Mon Aug 31 15:48:50 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 1 Sep 2009 00:48:50 +0200
Subject: [Freeswitch-users] mod_opal
Message-ID: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>

hello,

i'm trying to build mod_opal and getting this error:



making all mod_logfile

making all mod_loopback

making all mod_native_file

making all mod_opal
Compiling mod_opal.cpp...
quiet_libtool: compile:  g++ -g -ggdb -I.
-I/home/tculjaga/freeswitch-trunk/src/include
-I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
-fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
-D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal
-DHAVE_CONFIG_H -c mod_opal.cpp  -fPIC -DPIC -o .libs/mod_opal.o
In file included from mod_opal.cpp:25:
mod_opal.h:151: error: conflicting return type specified for ?virtual
OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)?
/usr/include/opal/opal/localep.h:267: error:   overriding ?virtual
ptlib_virtual_function_changed_or_removed******
OpalLocalEndPoint::CreateConnection(OpalCall&, void*)?
mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&,
FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
switch_channel_t*)?:
mod_opal.cpp:564: error: no matching function for call to
?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)?
/usr/include/opal/opal/localep.h:290: note: candidates are:
OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&,
void*, unsigned int, OpalConnection::StringOptions*, char)
/usr/include/opal/opal/localep.h:276: note:
OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
/usr/include/opal/opal/patch.h: In member function ?switch_status_t
FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)?:
/usr/include/opal/opal/patch.h:272: error: ?virtual
ptlib_virtual_function_changed_or_removed******
OpalMediaPatch::OnPatchStart()? is private
mod_opal.cpp:1277: error: within this context
mod_opal.cpp:1277: warning: ?virtual
ptlib_virtual_function_changed_or_removed******
OpalMediaPatch::OnPatchStart()? is deprecated (declared at
/usr/include/opal/opal/patch.h:272)
mod_opal.cpp:1277: warning: ignoring return value of function declared with
attribute warn_unused_result
/usr/include/opal/opal/patch.h: In member function ?switch_status_t
FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)?:
/usr/include/opal/opal/patch.h:272: error: ?virtual
ptlib_virtual_function_changed_or_removed******
OpalMediaPatch::OnPatchStart()? is private
mod_opal.cpp:1399: error: within this context
mod_opal.cpp:1399: warning: ?virtual
ptlib_virtual_function_changed_or_removed******
OpalMediaPatch::OnPatchStart()? is deprecated (declared at
/usr/include/opal/opal/patch.h:272)
mod_opal.cpp:1399: warning: ignoring return value of function declared with
attribute warn_unused_result
make[5]: *** [mod_opal.lo] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_opal-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
tculjaga at nemesis:~/freeswitch-trunk$
tculjaga at nemesis:~/freeswitch-trunk$
tculjaga at nemesis:~/freeswitch-trunk$



what ptlib/opal/fs version did you use to build it?


I tried with trunk (ptlib, opal, fs)... and as you can see :)



T.
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From msc at freeswitch.org  Mon Aug 31 16:00:39 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 16:00:39 -0700
Subject: [Freeswitch-users] mod_opal
In-Reply-To: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>
References: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>
Message-ID: <87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com>

On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hello,
>
> i'm trying to build mod_opal and getting this error:
>
>
>
> making all mod_logfile
>
> making all mod_loopback
>
> making all mod_native_file
>
> making all mod_opal
> Compiling mod_opal.cpp...
> quiet_libtool: compile:  g++ -g -ggdb -I.
> -I/home/tculjaga/freeswitch-trunk/src/include
> -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
> -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal
> -DHAVE_CONFIG_H -c mod_opal.cpp  -fPIC -DPIC -o .libs/mod_opal.o
> In file included from mod_opal.cpp:25:
> mod_opal.h:151: error: conflicting return type specified for ?virtual
> OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)?
> /usr/include/opal/opal/localep.h:267: error:   overriding ?virtual
> ptlib_virtual_function_changed_or_removed******
> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)?
> mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&,
> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
> switch_channel_t*)?:
> mod_opal.cpp:564: error: no matching function for call to
> ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)?
> /usr/include/opal/opal/localep.h:290: note: candidates are:
> OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&,
> void*, unsigned int, OpalConnection::StringOptions*, char)
> /usr/include/opal/opal/localep.h:276: note:
> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
> /usr/include/opal/opal/patch.h: In member function ?switch_status_t
> FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)?:
> /usr/include/opal/opal/patch.h:272: error: ?virtual
> ptlib_virtual_function_changed_or_removed******
> OpalMediaPatch::OnPatchStart()? is private
> mod_opal.cpp:1277: error: within this context
> mod_opal.cpp:1277: warning: ?virtual
> ptlib_virtual_function_changed_or_removed******
> OpalMediaPatch::OnPatchStart()? is deprecated (declared at
> /usr/include/opal/opal/patch.h:272)
> mod_opal.cpp:1277: warning: ignoring return value of function declared with
> attribute warn_unused_result
> /usr/include/opal/opal/patch.h: In member function ?switch_status_t
> FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)?:
> /usr/include/opal/opal/patch.h:272: error: ?virtual
> ptlib_virtual_function_changed_or_removed******
> OpalMediaPatch::OnPatchStart()? is private
> mod_opal.cpp:1399: error: within this context
> mod_opal.cpp:1399: warning: ?virtual
> ptlib_virtual_function_changed_or_removed******
> OpalMediaPatch::OnPatchStart()? is deprecated (declared at
> /usr/include/opal/opal/patch.h:272)
> mod_opal.cpp:1399: warning: ignoring return value of function declared with
> attribute warn_unused_result
> make[5]: *** [mod_opal.lo] Error 1
> make[4]: *** [all] Error 1
> make[3]: *** [mod_opal-all] Error 1
> make[2]: *** [all-recursive] Error 1
> Making all in build
>  +-------- FreeSWITCH Build Complete -----------+
>  + FreeSWITCH has been successfully built.      +
>  + Install by running:                          +
>  +                                              +
>  +               make install                   +
>  +----------------------------------------------+
> make[1]: *** [all-recursive] Error 1
> make: *** [all] Error 2
> tculjaga at nemesis:~/freeswitch-trunk$
> tculjaga at nemesis:~/freeswitch-trunk$
> tculjaga at nemesis:~/freeswitch-trunk$
>
>
>
> what ptlib/opal/fs version did you use to build it?
>
>
> I tried with trunk (ptlib, opal, fs)... and as you can see :)
>

Did you run the buildopal.sh script in src/build ?
-MC
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From jmesquita at freeswitch.org  Mon Aug 31 16:37:46 2009
From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=)
Date: Mon, 31 Aug 2009 20:37:46 -0300
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
	<2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
	<ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>
	<2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com>
Message-ID: <ebbc2d550908311637q22ef76aaj6621a5d2345885e8@mail.gmail.com>

Problem is definetly on far end.

If you look at the siptrace, you have the following sequence:

1. Asterisk calls in
2. FreeSWITCH replies with a Trying(100) to complete call right away and
proceeds to dialplan
3. FreeSWITCH invites (calls) 7 times the final destination that never
responds.
4. Asterisk sends a CANCEL message

In all that, your final endpoint never responds to any message. Are you sure
you can reach it?

jmesquita

On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer <e.schmidbauer at gmail.com>wrote:

> here's the sip trace:
> http://pastebin.freeswitch.org/10172
>
>
>
> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel
> > sofia/external/Anonymous at anonymous.invalid entering state
> [terminated][487]
> >
> > The far end seems to be replying with 487 - Request Terminated...
> >
> > Nothing wrong on FS, seems to be a problem with your endpoints. Can you
> > enable a sip trace?
> >
> > jmesquita
> >
> >
> > On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <e.schmidbauer at gmail.com>
> > wrote:
> >>
> >> thanks...heres the pastebin:
> >> http://pastebin.freeswitch.org/10171
> >>
> >> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> >> > Check the password dialog. It will tell you what the username/password
> >> > is.
> >> >
> >> > post the logs for a call as well, please.
> >> >
> >> > Regards,
> >> >
> >> > jmesquita
> >> >
> >> > On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer <
> e.schmidbauer at gmail.com>
> >> > wrote:
> >> >>
> >> >>
> >> >>
> >> >>
> =================================================================================================
> >> >> Name                    external
> >> >> Domain Name             N/A
> >> >> DBName                  sofia_reg_external
> >> >> Pres Hosts
> >> >> Dialplan                XML
> >> >> Context                 public
> >> >> Challenge Realm         auto_to
> >> >> RTP-IP                  192.168.0.125
> >> >> Ext-RTP-IP              98.118.151.30
> >> >> SIP-IP                  192.168.0.125
> >> >> Ext-SIP-IP              98.118.151.30
> >> >> URL                     sip:mod_sofia at 192.168.0.125:5080
> >> >> BIND-URL                sip:mod_sofia at 192.168.0.125:5080
> >> >> HOLD-MUSIC              local_stream://moh
> >> >> OUTBOUND-PROXY          N/A
> >> >> CODECS
> >> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h
> @20i,PCMU,PCMA,GSM
> >> >> TEL-EVENT               101
> >> >> DTMF-MODE               rfc2833
> >> >> CNG                     13
> >> >> SESSION-TO              0
> >> >> MAX-DIALOG              0
> >> >> NOMEDIA                 false
> >> >> LATE-NEG                false
> >> >> PROXY-MEDIA             false
> >> >> AGGRESSIVENAT           false
> >> >> STUN-ENABLED            true
> >> >> STUN-AUTO-DISABLE       false
> >> >> CALLS-IN                17
> >> >> FAILED-CALLS-IN         11
> >> >> CALLS-OUT               9
> >> >> FAILED-CALLS-OUT        9
> >> >>
> >> >> Registrations:
> >> >>
> >> >>
> >> >>
> =================================================================================================
> >> >> Call-ID:        MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
> >> >> User:           1000 at server1.altpressonline.com
> >> >> Contact:        "1000"
> >> >> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
> >> >> Agent:          X-Lite release 1103k stamp 53621
> >> >> Status:         Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
> >> >> Host:           server1.altpressonline.com
> >> >> IP:             69.204.30.67
> >> >> Port:           16006
> >> >> Auth-User:      1000
> >> >> Auth-Realm:     server1.altpressonline.com
> >> >>
> >> >>
> >> >>
> >> >>
> =================================================================================================
> >> >>
> >> >> sorry, i don't know how to login to pastebin
> >> >>
> >> >> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org>
> >> >> wrote:
> >> >> > Could you give a few more details? For example, could you pastebin
> >> >> > the
> >> >> > output of "sofia status profile external"?
> >> >> > -MC
> >> >> >
> >> >> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
> >> >> > <e.schmidbauer at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >> I am unable to call a user outside of our local area network. the
> >> >> >> user
> >> >> >> is registered on the external profile but there is no way to call
> >> >> >> the
> >> >> >> phone. does anyone have any suggestions how to do this?
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From e.schmidbauer at gmail.com  Mon Aug 31 17:06:52 2009
From: e.schmidbauer at gmail.com (e schmidbauer)
Date: Mon, 31 Aug 2009 20:06:52 -0400
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <ebbc2d550908311637q22ef76aaj6621a5d2345885e8@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
	<2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
	<ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>
	<2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com>
	<ebbc2d550908311637q22ef76aaj6621a5d2345885e8@mail.gmail.com>
Message-ID: <2cef777b0908311706o4173312bw9af8b217aca47392@mail.gmail.com>

i cannot reach the remote endpoint. the remote endpoint can reach a
locally registered endpoint. any idea why this is happening? the
remote endpoint is behind a linksys with upnp enabled.

2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> Problem is definetly on far end.
>
> If you look at the siptrace, you have the following sequence:
>
> 1. Asterisk calls in
> 2. FreeSWITCH replies with a Trying(100) to complete call right away and
> proceeds to dialplan
> 3. FreeSWITCH invites (calls) 7 times the final destination that never
> responds.
> 4. Asterisk sends a CANCEL message
>
> In all that, your final endpoint never responds to any message. Are you sure
> you can reach it?
>
> jmesquita
>
> On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer <e.schmidbauer at gmail.com>
> wrote:
>>
>> here's the sip trace:
>> http://pastebin.freeswitch.org/10172
>>
>>
>>
>> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> > 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel
>> > sofia/external/Anonymous at anonymous.invalid entering state
>> > [terminated][487]
>> >
>> > The far end seems to be replying with 487 - Request Terminated...
>> >
>> > Nothing wrong on FS, seems to be a problem with your endpoints. Can you
>> > enable a sip trace?
>> >
>> > jmesquita
>> >
>> >
>> > On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <e.schmidbauer at gmail.com>
>> > wrote:
>> >>
>> >> thanks...heres the pastebin:
>> >> http://pastebin.freeswitch.org/10171
>> >>
>> >> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
>> >> > Check the password dialog. It will tell you what the
>> >> > username/password
>> >> > is.
>> >> >
>> >> > post the logs for a call as well, please.
>> >> >
>> >> > Regards,
>> >> >
>> >> > jmesquita
>> >> >
>> >> > On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer
>> >> > <e.schmidbauer at gmail.com>
>> >> > wrote:
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >> Name ? ? ? ? ? ? ? ? ? ?external
>> >> >> Domain Name ? ? ? ? ? ? N/A
>> >> >> DBName ? ? ? ? ? ? ? ? ?sofia_reg_external
>> >> >> Pres Hosts
>> >> >> Dialplan ? ? ? ? ? ? ? ?XML
>> >> >> Context ? ? ? ? ? ? ? ? public
>> >> >> Challenge Realm ? ? ? ? auto_to
>> >> >> RTP-IP ? ? ? ? ? ? ? ? ?192.168.0.125
>> >> >> Ext-RTP-IP ? ? ? ? ? ? ?98.118.151.30
>> >> >> SIP-IP ? ? ? ? ? ? ? ? ?192.168.0.125
>> >> >> Ext-SIP-IP ? ? ? ? ? ? ?98.118.151.30
>> >> >> URL ? ? ? ? ? ? ? ? ? ? sip:mod_sofia at 192.168.0.125:5080
>> >> >> BIND-URL ? ? ? ? ? ? ? ?sip:mod_sofia at 192.168.0.125:5080
>> >> >> HOLD-MUSIC ? ? ? ? ? ? ?local_stream://moh
>> >> >> OUTBOUND-PROXY ? ? ? ? ?N/A
>> >> >> CODECS
>> >> >>
>> >> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h@20i,PCMU,PCMA,GSM
>> >> >> TEL-EVENT ? ? ? ? ? ? ? 101
>> >> >> DTMF-MODE ? ? ? ? ? ? ? rfc2833
>> >> >> CNG ? ? ? ? ? ? ? ? ? ? 13
>> >> >> SESSION-TO ? ? ? ? ? ? ?0
>> >> >> MAX-DIALOG ? ? ? ? ? ? ?0
>> >> >> NOMEDIA ? ? ? ? ? ? ? ? false
>> >> >> LATE-NEG ? ? ? ? ? ? ? ?false
>> >> >> PROXY-MEDIA ? ? ? ? ? ? false
>> >> >> AGGRESSIVENAT ? ? ? ? ? false
>> >> >> STUN-ENABLED ? ? ? ? ? ?true
>> >> >> STUN-AUTO-DISABLE ? ? ? false
>> >> >> CALLS-IN ? ? ? ? ? ? ? ?17
>> >> >> FAILED-CALLS-IN ? ? ? ? 11
>> >> >> CALLS-OUT ? ? ? ? ? ? ? 9
>> >> >> FAILED-CALLS-OUT ? ? ? ?9
>> >> >>
>> >> >> Registrations:
>> >> >>
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >> Call-ID: ? ? ? ?MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
>> >> >> User: ? ? ? ? ? 1000 at server1.altpressonline.com
>> >> >> Contact: ? ? ? ?"1000"
>> >> >> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
>> >> >> Agent: ? ? ? ? ?X-Lite release 1103k stamp 53621
>> >> >> Status: ? ? ? ? Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
>> >> >> Host: ? ? ? ? ? server1.altpressonline.com
>> >> >> IP: ? ? ? ? ? ? 69.204.30.67
>> >> >> Port: ? ? ? ? ? 16006
>> >> >> Auth-User: ? ? ?1000
>> >> >> Auth-Realm: ? ? server1.altpressonline.com
>> >> >>
>> >> >>
>> >> >>
>> >> >>
>> >> >> =================================================================================================
>> >> >>
>> >> >> sorry, i don't know how to login to pastebin
>> >> >>
>> >> >> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<msc at freeswitch.org>
>> >> >> wrote:
>> >> >> > Could you give a few more details? For example, could you pastebin
>> >> >> > the
>> >> >> > output of "sofia status profile external"?
>> >> >> > -MC
>> >> >> >
>> >> >> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
>> >> >> > <e.schmidbauer at gmail.com>
>> >> >> > wrote:
>> >> >> >>
>> >> >> >> I am unable to call a user outside of our local area network. the
>> >> >> >> user
>> >> >> >> is registered on the external profile but there is no way to call
>> >> >> >> the
>> >> >> >> phone. does anyone have any suggestions how to do this?
>> >> >> >>
>> >> >> >> _______________________________________________
>> >> >> >> FreeSWITCH-users mailing list
>> >> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >>
>> >> >> >>
>> >> >> >>
>> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> >> http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >> > _______________________________________________
>> >> >> > FreeSWITCH-users mailing list
>> >> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >> >
>> >> >> >
>> >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> > http://www.freeswitch.org
>> >> >> >
>> >> >> >
>> >> >>
>> >> >> _______________________________________________
>> >> >> FreeSWITCH-users mailing list
>> >> >> FreeSWITCH-users at lists.freeswitch.org
>> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >>
>> >> >>
>> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> >> http://www.freeswitch.org
>> >> >
>> >> >
>> >> > _______________________________________________
>> >> > FreeSWITCH-users mailing list
>> >> > FreeSWITCH-users at lists.freeswitch.org
>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> >
>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> > http://www.freeswitch.org
>> >> >
>> >> >
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >>
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>> >
>> >
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>



From msc at freeswitch.org  Mon Aug 31 17:23:34 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 17:23:34 -0700
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <2cef777b0908311706o4173312bw9af8b217aca47392@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
	<2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
	<ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>
	<2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com>
	<ebbc2d550908311637q22ef76aaj6621a5d2345885e8@mail.gmail.com>
	<2cef777b0908311706o4173312bw9af8b217aca47392@mail.gmail.com>
Message-ID: <87f2f3b90908311723v5c41801l5a8c8edcf9bc9904@mail.gmail.com>

What kind of linksys? I had a 54WRT-G that didn't work until I used Tomato
firmware. After that it all worked very well.
-MC

On Mon, Aug 31, 2009 at 5:06 PM, e schmidbauer <e.schmidbauer at gmail.com>wrote:

> i cannot reach the remote endpoint. the remote endpoint can reach a
> locally registered endpoint. any idea why this is happening? the
> remote endpoint is behind a linksys with upnp enabled.
>
> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > Problem is definetly on far end.
> >
> > If you look at the siptrace, you have the following sequence:
> >
> > 1. Asterisk calls in
> > 2. FreeSWITCH replies with a Trying(100) to complete call right away and
> > proceeds to dialplan
> > 3. FreeSWITCH invites (calls) 7 times the final destination that never
> > responds.
> > 4. Asterisk sends a CANCEL message
> >
> > In all that, your final endpoint never responds to any message. Are you
> sure
> > you can reach it?
> >
> > jmesquita
> >
> > On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer <e.schmidbauer at gmail.com>
> > wrote:
> >>
> >> here's the sip trace:
> >> http://pastebin.freeswitch.org/10172
> >>
> >>
> >>
> >> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> >> > 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel
> >> > sofia/external/Anonymous at anonymous.invalid entering state
> >> > [terminated][487]
> >> >
> >> > The far end seems to be replying with 487 - Request Terminated...
> >> >
> >> > Nothing wrong on FS, seems to be a problem with your endpoints. Can
> you
> >> > enable a sip trace?
> >> >
> >> > jmesquita
> >> >
> >> >
> >> > On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <
> e.schmidbauer at gmail.com>
> >> > wrote:
> >> >>
> >> >> thanks...heres the pastebin:
> >> >> http://pastebin.freeswitch.org/10171
> >> >>
> >> >> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> >> >> > Check the password dialog. It will tell you what the
> >> >> > username/password
> >> >> > is.
> >> >> >
> >> >> > post the logs for a call as well, please.
> >> >> >
> >> >> > Regards,
> >> >> >
> >> >> > jmesquita
> >> >> >
> >> >> > On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer
> >> >> > <e.schmidbauer at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >> Name                    external
> >> >> >> Domain Name             N/A
> >> >> >> DBName                  sofia_reg_external
> >> >> >> Pres Hosts
> >> >> >> Dialplan                XML
> >> >> >> Context                 public
> >> >> >> Challenge Realm         auto_to
> >> >> >> RTP-IP                  192.168.0.125
> >> >> >> Ext-RTP-IP              98.118.151.30
> >> >> >> SIP-IP                  192.168.0.125
> >> >> >> Ext-SIP-IP              98.118.151.30
> >> >> >> URL                     sip:mod_sofia at 192.168.0.125:5080
> >> >> >> BIND-URL                sip:mod_sofia at 192.168.0.125:5080
> >> >> >> HOLD-MUSIC              local_stream://moh
> >> >> >> OUTBOUND-PROXY          N/A
> >> >> >> CODECS
> >> >> >>
> >> >> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h
> @20i,PCMU,PCMA,GSM
> >> >> >> TEL-EVENT               101
> >> >> >> DTMF-MODE               rfc2833
> >> >> >> CNG                     13
> >> >> >> SESSION-TO              0
> >> >> >> MAX-DIALOG              0
> >> >> >> NOMEDIA                 false
> >> >> >> LATE-NEG                false
> >> >> >> PROXY-MEDIA             false
> >> >> >> AGGRESSIVENAT           false
> >> >> >> STUN-ENABLED            true
> >> >> >> STUN-AUTO-DISABLE       false
> >> >> >> CALLS-IN                17
> >> >> >> FAILED-CALLS-IN         11
> >> >> >> CALLS-OUT               9
> >> >> >> FAILED-CALLS-OUT        9
> >> >> >>
> >> >> >> Registrations:
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >> Call-ID:        MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
> >> >> >> User:           1000 at server1.altpressonline.com
> >> >> >> Contact:        "1000"
> >> >> >> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
> >> >> >> Agent:          X-Lite release 1103k stamp 53621
> >> >> >> Status:         Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
> >> >> >> Host:           server1.altpressonline.com
> >> >> >> IP:             69.204.30.67
> >> >> >> Port:           16006
> >> >> >> Auth-User:      1000
> >> >> >> Auth-Realm:     server1.altpressonline.com
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >>
> >> >> >> sorry, i don't know how to login to pastebin
> >> >> >>
> >> >> >> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<
> msc at freeswitch.org>
> >> >> >> wrote:
> >> >> >> > Could you give a few more details? For example, could you
> pastebin
> >> >> >> > the
> >> >> >> > output of "sofia status profile external"?
> >> >> >> > -MC
> >> >> >> >
> >> >> >> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
> >> >> >> > <e.schmidbauer at gmail.com>
> >> >> >> > wrote:
> >> >> >> >>
> >> >> >> >> I am unable to call a user outside of our local area network.
> the
> >> >> >> >> user
> >> >> >> >> is registered on the external profile but there is no way to
> call
> >> >> >> >> the
> >> >> >> >> phone. does anyone have any suggestions how to do this?
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From jim at evolutiontel.net  Mon Aug 31 18:26:19 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Tue, 1 Sep 2009 11:26:19 +1000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <1251721344.6737.32.camel@dk-d820>
References: <C6C0AC6C.5E412%krice@freeswitch.org>
	<1251721344.6737.32.camel@dk-d820>
Message-ID: <ddad80610908311826v3f0ec02ai195cd7772095c41@mail.gmail.com>

Actually Dave, our idea's are very similar.

The difference being is that you fork the calls into 2 separate legs
at a B2BUA and sending each to an FS box, while I would duplicate
messages to an FS box acting in Standby mode using extra SIP headers
to let the standby machine know the RTP port number and the UUID.
These 2 FS boxes would essentially become the core switching units in
Active/Standby configuration and be used to handle switching logic and
billing.  Thus if FS2 became the Active box, it would have exactly the
same call information as FS1 had.  Theoretically there would nothing
stopping FS1 and FS2 being master and standby for each other at the
same time and both handling traffic, although this has the potential
to cause over load if something did fail.

i.e FS1 receives INVITE, handles call and forwards INVITE to Gateway
and then sends a second INVITE (Duplication of Original message,
utilising extra SIP headers to pass on required information) to FS2.
FS1 fails FS2 takes over and handles call.

Then you would look at way's to distribute functionality across other
boxes to allow scalability.  An example would be to use dedicated FS
boxes for TDM cards and Opensips to handle registrations, SIP proxy or
SBC controllers.

Of course this assumes you are looking to build a softswitch that can
handle large call numbers and have the $$$ to do so.  Sadly, I have
niether the money or the requirement.




On Mon, Aug 31, 2009 at 10:22 PM, David Knell<dave at 3c.co.uk> wrote:
> A random thought. ?It strikes me that adding support for live call
> failover to FreeSWITCH might well be the wrong thing to do - it's
> something which is probably not too difficult to achieve if starting
> from the ground up, but which is going to be a right PITA to do
> retrospectively. ?And, once you've done switching, there's conferencing,
> IVR stuff...
>
> Here's an alternative approach. ?Have two FS boxes (which I'll call FS1
> and FS2 - you can't fault *my* imagination) and a front-end B2BUA/RTP
> proxy with a bit of a difference. ?The B2BUA sends incoming stuff to
> both FS1 and FS2 as two separate call legs, and the RTP proxy forwards
> inbound RTP to both FS1 and FS2. ?Assuming that FS1 is the current
> "master", it forwards RTP and call state transitions from FS1 back to
> the caller. ?If it detects that FS1 has gone away, then - as FS2 should
> be in roughly the same state as FS1 - it flips to using FS2 as its
> source.
>
> The internal state of the B2BUA/RTP proxy should be relatively simple,
> and therefore should be able to be duplicated to a hot standby.
>
> I can see a collection of gotchas, such as registration (the B2BUA needs
> access to the user database), and I haven't even begun to try to think
> of how to deal with calls *from* the FS boxes. ?Actually, that's not
> true: I have, and they clearly need to go through the same sort of
> mechanism in reverse, which is fine until I start trying to work out how
> to pair up calls from FS1 and FS2 reliably, and then my head hurts.
>
> But I think that this idea has merit. ?Firstly, it abstracts the
> failover mechanism away into one place; secondly, it is general - FS1
> and FS2 could just as easily be Asterisk1 and Asterisk2.
>
> And I'll do it for just $98,650..
>
> --Dave
>
>> Your forgetting that with RTP theres the duplication of RTP that may be
>> required along with a mechanism that keeps media timers working properly...
>>
>> Where many things don't support RTP timers they sure come in handy on a
>> regular basis especially dealing with several of the other software based
>> platforms out there that just love to have calls end but never want to send
>> a BYE
>>
>>
>> > From: Jim Burke <jim at evolutiontel.net>
>> > Reply-To: <freeswitch-users at lists.freeswitch.org>
>> > Date: Mon, 31 Aug 2009 12:54:50 +1000
>> > To: <freeswitch-users at lists.freeswitch.org>
>> > Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
>> >
>> > Conceptually in a pure SIP installation of Freeswitch, hot standby or
>> > active standby could be achieved through duplication of messages
>> > (INVITE, 200OK and BYE for calls) between 2 boxes and then using VRRP
>> > to change the IP address when the box falls over. ?Going this way
>> > allows call state to be as up to date as possible. ?Obviously this
>> > would require logic added to sofia to transfer the RTP port
>> > information and UUID information between the FS instances (Easily
>> > achieved using additional SIP headers). ?It would also require that
>> > sofia does not forward duplicated messages to gateways or user agents.
>> > ?CDR information would also need to be generated correctly (My
>> > experience of Radius with Opensips is that it generates start and stop
>> > messages on INVITE/200OK then BYE respectively so this could be an
>> > easy solution to the CDR issue).
>> >
>> > Regards,
>> >
>> >
>> >
>> > On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
>> >> So, the first way would be to have a look on the sip stack, which is, in
>> >> fact, sofia ..., well, sound's like a nice, fun, long-going hobby project to
>> >> me :-)
>> >> I will definitly at least look at the sip code to check if it is something i
>> >> would myself willingly give over to ...
>> >> But I *really* do want ?a setup where it does not matter if one specific FS
>> >> instance is UP or NOT ...
>> >>
>> >> --
>> >> Raimund Sacherer
>> >> -
>> >> RunSolutions
>> >> ? ? Open Source It Consulting
>> >> -
>> >>
>> >> Parc Bit - Centro Empresarial Son Espanyol
>> >> Edificio Estel - Local 3D
>> >> 07121 - ?Palma de Mallorca
>> >> Baleares
>> >> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>> >>
>> >> The sip stack needs to be modified to spin that data up into the state
>> >> machine so that it can take over calls once the fail over takes place... its
>> >> not an easy task.
>> >> /b
>> >> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>> >>
>> >> Is itnpossible to have a db cluster know the state of each and every call
>> >> and then use Heartbeat on this db +
>> >> fs cluster so that clients see only one ip where as internally all fs boxes
>> >> refer db for call states, db again is under replication.
>> >> This in the thioery can be written, but I am sure if we think bit more on
>> >> this direction the problem seem to be getting addressed.
>> >> Other guys also chip in their 2 cents, we just need 50 of em to make a full
>> >> dollar.
>> >>
>> >> Thanks & Regards,
>> >> Mitul Limbani,
>> >> Founder & CEO,
>> >> Enterux Solutions Pvt. Ltd.,
>> >> The Enterprise Linux Company (r),
>> >> http://www.enterux.com
>> >> http://www.entVoice.com
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >> _______________________________________________
>> >> FreeSWITCH-users mailing list
>> >> FreeSWITCH-users at lists.freeswitch.org
>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> >> http://www.freeswitch.org
>> >>
>> >>
>> >
>> >
>> >
>> > --
>> > Jim Burke
>> > Director Evolutiontel.
>> > http://www.evolutiontel.net
>> >
>> > _______________________________________________
>> > FreeSWITCH-users mailing list
>> > FreeSWITCH-users at lists.freeswitch.org
>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> > http://www.freeswitch.org
>>
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
> --
> David Knell, Director, 3C Limited
> T: +44 20 3298 2000
> E: dave at 3c.co.uk
> W: http://www.3c.co.uk
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net



From jim at evolutiontel.net  Mon Aug 31 18:27:14 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Tue, 1 Sep 2009 11:27:14 +1000
Subject: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
In-Reply-To: <C99339FA-1E87-4A23-9C25-FB7D564B83CB@runsolutions.com>
References: <C6C0AC6C.5E412%krice@freeswitch.org>
	<1251721344.6737.32.camel@dk-d820>
	<C99339FA-1E87-4A23-9C25-FB7D564B83CB@runsolutions.com>
Message-ID: <ddad80610908311827u538f64f2ra0a1e6b8ce593af6@mail.gmail.com>

> And I'll do it for just $98,650..

Haha, that's really funny :-)

Dave is talking New Zealand Dollars ;)

On Mon, Aug 31, 2009 at 10:36 PM, Raimund Sacherer<rs at runsolutions.com> wrote:
> Hi David,
>
> On Aug 31, 2009, at 2:22 PM, David Knell wrote:
>
>> A random thought. ?It strikes me that adding support for live call
>> failover to FreeSWITCH might well be the wrong thing to do - it's
>> something which is probably not too difficult to achieve if starting
>> from the ground up, but which is going to be a right PITA to do
>> retrospectively. ?And, once you've done switching, there's
>> conferencing,
>> IVR stuff...
>>
>> Here's an alternative approach. ?Have two FS boxes (which I'll call
>> FS1
>> and FS2 - you can't fault *my* imagination) and a front-end B2BUA/RTP
>> proxy with a bit of a difference. ?The B2BUA sends incoming stuff to
>> both FS1 and FS2 as two separate call legs, and the RTP proxy forwards
>> inbound RTP to both FS1 and FS2. ?Assuming that FS1 is the current
>> "master", it forwards RTP and call state transitions from FS1 back to
>> the caller. ?If it detects that FS1 has gone away, then - as FS2
>> should
>> be in roughly the same state as FS1 - it flips to using FS2 as its
>> source.
>>
>> The internal state of the B2BUA/RTP proxy should be relatively simple,
>> and therefore should be able to be duplicated to a hot standby.
>>
>> I can see a collection of gotchas, such as registration (the B2BUA
>> needs
>> access to the user database), and I haven't even begun to try to think
>> of how to deal with calls *from* the FS boxes. ?Actually, that's not
>> true: I have, and they clearly need to go through the same sort of
>> mechanism in reverse, which is fine until I start trying to work out
>> how
>> to pair up calls from FS1 and FS2 reliably, and then my head hurts.
> Yeah, sounds like a real challenge
>
>>
>> But I think that this idea has merit. ?Firstly, it abstracts the
>> failover mechanism away into one place; secondly, it is general - FS1
>> and FS2 could just as easily be Asterisk1 and Asterisk2.
> this, in theory, sound quite interesting ...
> would be nice to have a proof of concept setup thought.
>
> -> Sound's not quite THAT nice than a full-built-in solution to FS ...
> but if it works ...
>
>
>>
>> And I'll do it for just $98,650..
> Haha, that's really funny :-)
>
>
>>
>> --Dave
>>
>>> Your forgetting that with RTP theres the duplication of RTP that
>>> may be
>>> required along with a mechanism that keeps media timers working
>>> properly...
>>>
>>> Where many things don't support RTP timers they sure come in handy
>>> on a
>>> regular basis especially dealing with several of the other software
>>> based
>>> platforms out there that just love to have calls end but never want
>>> to send
>>> a BYE
>>>
>>>
>>>> From: Jim Burke <jim at evolutiontel.net>
>>>> Reply-To: <freeswitch-users at lists.freeswitch.org>
>>>> Date: Mon, 31 Aug 2009 12:54:50 +1000
>>>> To: <freeswitch-users at lists.freeswitch.org>
>>>> Subject: Re: [Freeswitch-users] FreeSWITCH HA + Loadbalancing
>>>>
>>>> Conceptually in a pure SIP installation of Freeswitch, hot standby
>>>> or
>>>> active standby could be achieved through duplication of messages
>>>> (INVITE, 200OK and BYE for calls) between 2 boxes and then using
>>>> VRRP
>>>> to change the IP address when the box falls over. ?Going this way
>>>> allows call state to be as up to date as possible. ?Obviously this
>>>> would require logic added to sofia to transfer the RTP port
>>>> information and UUID information between the FS instances (Easily
>>>> achieved using additional SIP headers). ?It would also require that
>>>> sofia does not forward duplicated messages to gateways or user
>>>> agents.
>>>> CDR information would also need to be generated correctly (My
>>>> experience of Radius with Opensips is that it generates start and
>>>> stop
>>>> messages on INVITE/200OK then BYE respectively so this could be an
>>>> easy solution to the CDR issue).
>>>>
>>>> Regards,
>>>>
>>>>
>>>>
>>>> On Sun, Aug 30, 2009 at 11:19 PM, Raimund Sacherer<rs at runsolutions.com
>>>> > wrote:
>>>>> So, the first way would be to have a look on the sip stack, which
>>>>> is, in
>>>>> fact, sofia ..., well, sound's like a nice, fun, long-going hobby
>>>>> project to
>>>>> me :-)
>>>>> I will definitly at least look at the sip code to check if it is
>>>>> something i
>>>>> would myself willingly give over to ...
>>>>> But I *really* do want ?a setup where it does not matter if one
>>>>> specific FS
>>>>> instance is UP or NOT ...
>>>>>
>>>>> --
>>>>> Raimund Sacherer
>>>>> -
>>>>> RunSolutions
>>>>> ? ?Open Source It Consulting
>>>>> -
>>>>>
>>>>> Parc Bit - Centro Empresarial Son Espanyol
>>>>> Edificio Estel - Local 3D
>>>>> 07121 - ?Palma de Mallorca
>>>>> Baleares
>>>>> On Aug 29, 2009, at 10:04 PM, Brian West wrote:
>>>>>
>>>>> The sip stack needs to be modified to spin that data up into the
>>>>> state
>>>>> machine so that it can take over calls once the fail over takes
>>>>> place... its
>>>>> not an easy task.
>>>>> /b
>>>>> On Aug 29, 2009, at 2:56 PM, Mitul Limbani wrote:
>>>>>
>>>>> Is itnpossible to have a db cluster know the state of each and
>>>>> every call
>>>>> and then use Heartbeat on this db +
>>>>> fs cluster so that clients see only one ip where as internally
>>>>> all fs boxes
>>>>> refer db for call states, db again is under replication.
>>>>> This in the thioery can be written, but I am sure if we think bit
>>>>> more on
>>>>> this direction the problem seem to be getting addressed.
>>>>> Other guys also chip in their 2 cents, we just need 50 of em to
>>>>> make a full
>>>>> dollar.
>>>>>
>>>>> Thanks & Regards,
>>>>> Mitul Limbani,
>>>>> Founder & CEO,
>>>>> Enterux Solutions Pvt. Ltd.,
>>>>> The Enterprise Linux Company (r),
>>>>> http://www.enterux.com
>>>>> http://www.entVoice.com
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>>> freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>> _______________________________________________
>>>>> FreeSWITCH-users mailing list
>>>>> FreeSWITCH-users at lists.freeswitch.org
>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/
>>>>> freeswitch-users
>>>>> http://www.freeswitch.org
>>>>>
>>>>>
>>>>
>>>>
>>>>
>>>> --
>>>> Jim Burke
>>>> Director Evolutiontel.
>>>> http://www.evolutiontel.net
>>>>
>>>> _______________________________________________
>>>> FreeSWITCH-users mailing list
>>>> FreeSWITCH-users at lists.freeswitch.org
>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>>> users
>>>> http://www.freeswitch.org
>>>
>>>
>>>
>>> _______________________________________________
>>> FreeSWITCH-users mailing list
>>> FreeSWITCH-users at lists.freeswitch.org
>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>>> users
>>> http://www.freeswitch.org
>>>
>> --
>> David Knell, Director, 3C Limited
>> T: +44 20 3298 2000
>> E: dave at 3c.co.uk
>> W: http://www.3c.co.uk
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-
>> users
>> http://www.freeswitch.org
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net



From jim at evolutiontel.net  Mon Aug 31 18:40:13 2009
From: jim at evolutiontel.net (Jim Burke)
Date: Tue, 1 Sep 2009 11:40:13 +1000
Subject: [Freeswitch-users] Load Balacing
In-Reply-To: <20090831215728.GD20919@hijacked.us>
References: <26613bbf0908311256m617604f9tb91107cfe63202ae@mail.gmail.com>
	<20090831215728.GD20919@hijacked.us>
Message-ID: <ddad80610908311840u5bae139q38496c49507b897a@mail.gmail.com>

Could you use the server host saved in the database at registration
and then use it with path to get to the other FS box?

http://wiki.freeswitch.org/wiki/Sofia#Specifying_SIP_Proxy_With_fs_path

On Tue, Sep 1, 2009 at 7:57 AM, Andrew Thompson<andrew at hijacked.us> wrote:
> On Mon, Aug 31, 2009 at 04:56:48PM -0300, Juan Manuel Vicente wrote:
>> Hello:
>>
>> I am using two boxes, with same domain, to register about 1000 users working
>> together a DNS box doing round robin to resolve two differentes Ips one to
>> each box. This two boxes are using the same database (through unixodbc).
>>
>> When I use the funcion sofia_contact or sofia profile status both boxes show
>> the status of the user. Does not matter where the resgistration was did.
>>
>> But when I try to call the user this call fails, the call only works if the
>> call is sent through the same box where the user is registered.
>>
>
> I had the same problem except that I was using the multicast module to
> replicate registrations, I added a sofia config parameter to optionally
> rewrite the fs_path part of the sip URI on replicated packets so I could
> hairpin the sip messages through the original registrar. This doesn't
> help you directly but maybe it'll give you an idea?
>
> Andrew
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>



-- 
Jim Burke
Director Evolutiontel.
http://www.evolutiontel.net



From se02005-mec at yahoo.com  Mon Aug 31 19:53:12 2009
From: se02005-mec at yahoo.com (Merle J. Ebbert)
Date: Mon, 31 Aug 2009 19:53:12 -0700
Subject: [Freeswitch-users] Can an Instant Message (IM) be sent to a cell
 phone with FS under Windows?
Message-ID: <4A9C8C98.3020205@yahoo.com>

An HTML attachment was scrubbed...
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From help at pdscc.com  Mon Aug 31 22:43:44 2009
From: help at pdscc.com (Harondel J. Sibble)
Date: Mon, 31 Aug 2009 22:43:44 -0700
Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3
In-Reply-To: <20090823185152.D17845FE@sinclaire.sibble.net>
References: <20090823185152.D17845FE@sinclaire.sibble.net>
Message-ID: <20090901054337.7317F65B@sinclaire.sibble.net>

Haven't had any responses, anyone have any ideas on the problem with 
compiling the ESL modules as below?

On 23 Aug 2009 at 11:51, Harondel J. Sibble wrote:

> Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server, 
> then went to install FreePBX v3, I've gotten all the prerequisities in the
> wizard fixed except for ESL
> 
> As per
> 
> http://wiki.freeswitch.org/wiki/Event_Socket_Library
> http://wiki.freeswitch.org/wiki/Event_Socket
> 
> I go into my FS source dir
> 
> /home/sibbleh/freeswitch-1.0.4/libs/esl
> 
> Run make and then "sudo make phpmod-install" 
> 
> and I get 
> 
> 
> $ sudo make phpmod-install
> make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-
> I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> -ggdb 
> -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-
> variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
> CXXFLAGS="-
> I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> -ggdb 
> -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable" 
> CXX_CFLAGS="" -C php
> make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> g++  -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE
> -g 
> -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable -
> I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -
> I/usr/include/php5/Zend -I/usr/include/php5/ext -
> I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE -D_FILE_OFFSET_BITS=64
> -
> Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> cc1plus: warnings being treated as errors
> esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1047: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1073: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:1111: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:1141: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1172: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1198: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1234: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1269: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1294: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**,
> zval*, 
> int)':
> esl_wrap.cpp:1346: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1403: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1441: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1478: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1508: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1538: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1571: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_addHeader(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1611: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_delHeader(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1644: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_firstHeader(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1674: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLevent_nextHeader(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:1704: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_0(int, zval*,
> zval**, zval*, int)':
> esl_wrap.cpp:1744: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_1(int, zval*,
> zval**, zval*, int)':
> esl_wrap.cpp:1770: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_new_ESLconnection(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1803: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_socketDescriptor(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:1846: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_connected(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1872: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_getInfo(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1898: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_send(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:1931: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendRecv(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:1964: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_api(int, zval*, zval**,
> zval*, int)':
> esl_wrap.cpp:2007: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_bgapi(int, zval*,
> zval**, 
> zval*, int)':
> esl_wrap.cpp:2050: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendEvent(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2082: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEvent(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2108: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEventTimed(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:2141: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_filter(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2181: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_events(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2221: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_execute(int, zval*, 
> zval**, zval*, int)':
> esl_wrap.cpp:2272: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_executeAsync(int, zval*,
> zval**, zval*, int)':
> esl_wrap.cpp:2323: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_setAsyncExecute(int, 
> zval*, zval**, zval*, int)':
> esl_wrap.cpp:2356: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_setEventLock(int, zval*,
> zval**, zval*, int)':
> esl_wrap.cpp:2389: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_ESLconnection_disconnect(int, zval*,
> zval**, zval*, int)':
> esl_wrap.cpp:2415: error: format not a string literal and no format
> arguments
> esl_wrap.cpp: In function 'void _wrap_eslSetLogLevel(int, zval*, zval**, 
> zval*, int)':
> esl_wrap.cpp:2438: error: format not a string literal and no format
> arguments
> make[1]: *** [esl_wrap.o] Error 1
> make[1]: Leaving directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> make: *** [phpmod] Error 2
> 
> Same thing happens if I try sudo make everymod
> 
> Checking the list archives I found this thread
> 
> http://www.nabble.com/ESL-Wrapper-td22209991.html#a22222338
> 
> I've made sure that the php-dev packages are installed. Any suggestions on
> what to do next?
> -- 

-- 
Harondel J. Sibble 
Sibble Computer Consulting
Creating Solutions for the small and medium business computer user.
help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
(604) 739-3709 (voice)




From tculjaga at gmail.com  Mon Aug 31 23:09:00 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 1 Sep 2009 08:09:00 +0200
Subject: [Freeswitch-users] mod_opal
In-Reply-To: <87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com>
References: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>
	<87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com>
Message-ID: <65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>

hhmmm :))

is there any doc following up mod_opal ?
I really don't want to waste your time :)

T.


On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins <msc at freeswitch.org> wrote:

>
>
> On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>
>> hello,
>>
>> i'm trying to build mod_opal and getting this error:
>>
>>
>>
>> making all mod_logfile
>>
>> making all mod_loopback
>>
>> making all mod_native_file
>>
>> making all mod_opal
>> Compiling mod_opal.cpp...
>> quiet_libtool: compile:  g++ -g -ggdb -I.
>> -I/home/tculjaga/freeswitch-trunk/src/include
>> -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
>> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
>> -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal
>> -DHAVE_CONFIG_H -c mod_opal.cpp  -fPIC -DPIC -o .libs/mod_opal.o
>> In file included from mod_opal.cpp:25:
>> mod_opal.h:151: error: conflicting return type specified for ?virtual
>> OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)?
>> /usr/include/opal/opal/localep.h:267: error:   overriding ?virtual
>> ptlib_virtual_function_changed_or_removed******
>> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)?
>> mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&,
>> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
>> switch_channel_t*)?:
>> mod_opal.cpp:564: error: no matching function for call to
>> ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)?
>> /usr/include/opal/opal/localep.h:290: note: candidates are:
>> OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&,
>> void*, unsigned int, OpalConnection::StringOptions*, char)
>> /usr/include/opal/opal/localep.h:276: note:
>> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
>> /usr/include/opal/opal/patch.h: In member function ?switch_status_t
>> FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)?:
>> /usr/include/opal/opal/patch.h:272: error: ?virtual
>> ptlib_virtual_function_changed_or_removed******
>> OpalMediaPatch::OnPatchStart()? is private
>> mod_opal.cpp:1277: error: within this context
>> mod_opal.cpp:1277: warning: ?virtual
>> ptlib_virtual_function_changed_or_removed******
>> OpalMediaPatch::OnPatchStart()? is deprecated (declared at
>> /usr/include/opal/opal/patch.h:272)
>> mod_opal.cpp:1277: warning: ignoring return value of function declared
>> with attribute warn_unused_result
>> /usr/include/opal/opal/patch.h: In member function ?switch_status_t
>> FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)?:
>> /usr/include/opal/opal/patch.h:272: error: ?virtual
>> ptlib_virtual_function_changed_or_removed******
>> OpalMediaPatch::OnPatchStart()? is private
>> mod_opal.cpp:1399: error: within this context
>> mod_opal.cpp:1399: warning: ?virtual
>> ptlib_virtual_function_changed_or_removed******
>> OpalMediaPatch::OnPatchStart()? is deprecated (declared at
>> /usr/include/opal/opal/patch.h:272)
>> mod_opal.cpp:1399: warning: ignoring return value of function declared
>> with attribute warn_unused_result
>> make[5]: *** [mod_opal.lo] Error 1
>> make[4]: *** [all] Error 1
>> make[3]: *** [mod_opal-all] Error 1
>> make[2]: *** [all-recursive] Error 1
>> Making all in build
>>  +-------- FreeSWITCH Build Complete -----------+
>>  + FreeSWITCH has been successfully built.      +
>>  + Install by running:                          +
>>  +                                              +
>>  +               make install                   +
>>  +----------------------------------------------+
>> make[1]: *** [all-recursive] Error 1
>> make: *** [all] Error 2
>> tculjaga at nemesis:~/freeswitch-trunk$
>> tculjaga at nemesis:~/freeswitch-trunk$
>> tculjaga at nemesis:~/freeswitch-trunk$
>>
>>
>>
>> what ptlib/opal/fs version did you use to build it?
>>
>>
>> I tried with trunk (ptlib, opal, fs)... and as you can see :)
>>
>
> Did you run the buildopal.sh script in src/build ?
> -MC
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
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From tculjaga at gmail.com  Mon Aug 31 23:29:24 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 1 Sep 2009 08:29:24 +0200
Subject: [Freeswitch-users] mod_opal
In-Reply-To: <65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>
References: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>
	<87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com>
	<65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>
Message-ID: <65d96fc80908312329l7fb63744q97adaa5ea7d18997@mail.gmail.com>

hi, It went well

obviously FS needs v3_6 opal :)

thx.



On Tue, Sep 1, 2009 at 8:09 AM, Tihomir Culjaga <tculjaga at gmail.com> wrote:

> hhmmm :))
>
> is there any doc following up mod_opal ?
> I really don't want to waste your time :)
>
> T.
>
>
> On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins <msc at freeswitch.org>wrote:
>
>>
>>
>> On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:
>>
>>> hello,
>>>
>>> i'm trying to build mod_opal and getting this error:
>>>
>>>
>>>
>>> making all mod_logfile
>>>
>>> making all mod_loopback
>>>
>>> making all mod_native_file
>>>
>>> making all mod_opal
>>> Compiling mod_opal.cpp...
>>> quiet_libtool: compile:  g++ -g -ggdb -I.
>>> -I/home/tculjaga/freeswitch-trunk/src/include
>>> -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC
>>> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2
>>> -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal
>>> -DHAVE_CONFIG_H -c mod_opal.cpp  -fPIC -DPIC -o .libs/mod_opal.o
>>> In file included from mod_opal.cpp:25:
>>> mod_opal.h:151: error: conflicting return type specified for ?virtual
>>> OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)?
>>> /usr/include/opal/opal/localep.h:267: error:   overriding ?virtual
>>> ptlib_virtual_function_changed_or_removed******
>>> OpalLocalEndPoint::CreateConnection(OpalCall&, void*)?
>>> mod_opal.cpp: In constructor ?FSConnection::FSConnection(OpalCall&,
>>> FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*,
>>> switch_channel_t*)?:
>>> mod_opal.cpp:564: error: no matching function for call to
>>> ?OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)?
>>> /usr/include/opal/opal/localep.h:290: note: candidates are:
>>> OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&,
>>> void*, unsigned int, OpalConnection::StringOptions*, char)
>>> /usr/include/opal/opal/localep.h:276: note:
>>> OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
>>> /usr/include/opal/opal/patch.h: In member function ?switch_status_t
>>> FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)?:
>>> /usr/include/opal/opal/patch.h:272: error: ?virtual
>>> ptlib_virtual_function_changed_or_removed******
>>> OpalMediaPatch::OnPatchStart()? is private
>>> mod_opal.cpp:1277: error: within this context
>>> mod_opal.cpp:1277: warning: ?virtual
>>> ptlib_virtual_function_changed_or_removed******
>>> OpalMediaPatch::OnPatchStart()? is deprecated (declared at
>>> /usr/include/opal/opal/patch.h:272)
>>> mod_opal.cpp:1277: warning: ignoring return value of function declared
>>> with attribute warn_unused_result
>>> /usr/include/opal/opal/patch.h: In member function ?switch_status_t
>>> FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)?:
>>> /usr/include/opal/opal/patch.h:272: error: ?virtual
>>> ptlib_virtual_function_changed_or_removed******
>>> OpalMediaPatch::OnPatchStart()? is private
>>> mod_opal.cpp:1399: error: within this context
>>> mod_opal.cpp:1399: warning: ?virtual
>>> ptlib_virtual_function_changed_or_removed******
>>> OpalMediaPatch::OnPatchStart()? is deprecated (declared at
>>> /usr/include/opal/opal/patch.h:272)
>>> mod_opal.cpp:1399: warning: ignoring return value of function declared
>>> with attribute warn_unused_result
>>> make[5]: *** [mod_opal.lo] Error 1
>>> make[4]: *** [all] Error 1
>>> make[3]: *** [mod_opal-all] Error 1
>>> make[2]: *** [all-recursive] Error 1
>>> Making all in build
>>>  +-------- FreeSWITCH Build Complete -----------+
>>>  + FreeSWITCH has been successfully built.      +
>>>  + Install by running:                          +
>>>  +                                              +
>>>  +               make install                   +
>>>  +----------------------------------------------+
>>> make[1]: *** [all-recursive] Error 1
>>> make: *** [all] Error 2
>>> tculjaga at nemesis:~/freeswitch-trunk$
>>> tculjaga at nemesis:~/freeswitch-trunk$
>>> tculjaga at nemesis:~/freeswitch-trunk$
>>>
>>>
>>>
>>> what ptlib/opal/fs version did you use to build it?
>>>
>>>
>>> I tried with trunk (ptlib, opal, fs)... and as you can see :)
>>>
>>
>> Did you run the buildopal.sh script in src/build ?
>> -MC
>>
>>
>> _______________________________________________
>> FreeSWITCH-users mailing list
>> FreeSWITCH-users at lists.freeswitch.org
>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>>
>>
>
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From msc at freeswitch.org  Mon Aug 31 23:29:28 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 23:29:28 -0700
Subject: [Freeswitch-users] mod_opal
In-Reply-To: <65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>
References: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>
	<87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com>
	<65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>
Message-ID: <87f2f3b90908312329g17b642adj9c0073a863bf28d7@mail.gmail.com>

On Mon, Aug 31, 2009 at 11:09 PM, Tihomir Culjaga <tculjaga at gmail.com>wrote:

> hhmmm :))
>
> is there any doc following up mod_opal ?
> I really don't want to waste your time :)
>
> T.
>
Last I heard this was it:
http://jira.freeswitch.org/browse/MODOPAL-10
-MC
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From tculjaga at gmail.com  Mon Aug 31 23:32:00 2009
From: tculjaga at gmail.com (Tihomir Culjaga)
Date: Tue, 1 Sep 2009 08:32:00 +0200
Subject: [Freeswitch-users] remote endpoints
In-Reply-To: <2cef777b0908311706o4173312bw9af8b217aca47392@mail.gmail.com>
References: <2cef777b0908311020g1ff7be3fh1d3625312ecee758@mail.gmail.com>
	<87f2f3b90908311042n6be40a0v7fffce9f85e263b2@mail.gmail.com>
	<2cef777b0908311154n7b83411q40a8b31da0a44525@mail.gmail.com>
	<ebbc2d550908311201r5d576a6ejfb998b2dbb77a2b0@mail.gmail.com>
	<2cef777b0908311207n71073d46x56d687f15523a69b@mail.gmail.com>
	<ebbc2d550908311230g3704ce45ua2050d1e13dd3b30@mail.gmail.com>
	<2cef777b0908311318x11e82e60s5160f54e237ba422@mail.gmail.com>
	<ebbc2d550908311637q22ef76aaj6621a5d2345885e8@mail.gmail.com>
	<2cef777b0908311706o4173312bw9af8b217aca47392@mail.gmail.com>
Message-ID: <65d96fc80908312332h48c92eden4cdd28ae2f66e4f7@mail.gmail.com>

oh good, on remote router/dsl modem (whatever doing NAT) never use upnp,
never use ALG, just do a simple NAT and it is alway gonna work!

T.

On Tue, Sep 1, 2009 at 2:06 AM, e schmidbauer <e.schmidbauer at gmail.com>wrote:

> i cannot reach the remote endpoint. the remote endpoint can reach a
> locally registered endpoint. any idea why this is happening? the
> remote endpoint is behind a linksys with upnp enabled.
>
> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> > Problem is definetly on far end.
> >
> > If you look at the siptrace, you have the following sequence:
> >
> > 1. Asterisk calls in
> > 2. FreeSWITCH replies with a Trying(100) to complete call right away and
> > proceeds to dialplan
> > 3. FreeSWITCH invites (calls) 7 times the final destination that never
> > responds.
> > 4. Asterisk sends a CANCEL message
> >
> > In all that, your final endpoint never responds to any message. Are you
> sure
> > you can reach it?
> >
> > jmesquita
> >
> > On Mon, Aug 31, 2009 at 5:18 PM, e schmidbauer <e.schmidbauer at gmail.com>
> > wrote:
> >>
> >> here's the sip trace:
> >> http://pastebin.freeswitch.org/10172
> >>
> >>
> >>
> >> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> >> > 2009-08-31 15:06:26.142940 [DEBUG] sofia.c:3300 Channel
> >> > sofia/external/Anonymous at anonymous.invalid entering state
> >> > [terminated][487]
> >> >
> >> > The far end seems to be replying with 487 - Request Terminated...
> >> >
> >> > Nothing wrong on FS, seems to be a problem with your endpoints. Can
> you
> >> > enable a sip trace?
> >> >
> >> > jmesquita
> >> >
> >> >
> >> > On Mon, Aug 31, 2009 at 4:07 PM, e schmidbauer <
> e.schmidbauer at gmail.com>
> >> > wrote:
> >> >>
> >> >> thanks...heres the pastebin:
> >> >> http://pastebin.freeswitch.org/10171
> >> >>
> >> >> 2009/8/31 Jo?o Mesquita <jmesquita at freeswitch.org>:
> >> >> > Check the password dialog. It will tell you what the
> >> >> > username/password
> >> >> > is.
> >> >> >
> >> >> > post the logs for a call as well, please.
> >> >> >
> >> >> > Regards,
> >> >> >
> >> >> > jmesquita
> >> >> >
> >> >> > On Mon, Aug 31, 2009 at 3:54 PM, e schmidbauer
> >> >> > <e.schmidbauer at gmail.com>
> >> >> > wrote:
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >> Name                    external
> >> >> >> Domain Name             N/A
> >> >> >> DBName                  sofia_reg_external
> >> >> >> Pres Hosts
> >> >> >> Dialplan                XML
> >> >> >> Context                 public
> >> >> >> Challenge Realm         auto_to
> >> >> >> RTP-IP                  192.168.0.125
> >> >> >> Ext-RTP-IP              98.118.151.30
> >> >> >> SIP-IP                  192.168.0.125
> >> >> >> Ext-SIP-IP              98.118.151.30
> >> >> >> URL                     sip:mod_sofia at 192.168.0.125:5080
> >> >> >> BIND-URL                sip:mod_sofia at 192.168.0.125:5080
> >> >> >> HOLD-MUSIC              local_stream://moh
> >> >> >> OUTBOUND-PROXY          N/A
> >> >> >> CODECS
> >> >> >>
> >> >> >> CELT at 48000h,CELT at 32000h,speex at 32000h@20i,speex at 16000h
> @20i,PCMU,PCMA,GSM
> >> >> >> TEL-EVENT               101
> >> >> >> DTMF-MODE               rfc2833
> >> >> >> CNG                     13
> >> >> >> SESSION-TO              0
> >> >> >> MAX-DIALOG              0
> >> >> >> NOMEDIA                 false
> >> >> >> LATE-NEG                false
> >> >> >> PROXY-MEDIA             false
> >> >> >> AGGRESSIVENAT           false
> >> >> >> STUN-ENABLED            true
> >> >> >> STUN-AUTO-DISABLE       false
> >> >> >> CALLS-IN                17
> >> >> >> FAILED-CALLS-IN         11
> >> >> >> CALLS-OUT               9
> >> >> >> FAILED-CALLS-OUT        9
> >> >> >>
> >> >> >> Registrations:
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >> Call-ID:        MTc1NThiMWI5YWZkYWEzNzZmZTFmNTJiNDIwMzVhNTI.
> >> >> >> User:           1000 at server1.altpressonline.com
> >> >> >> Contact:        "1000"
> >> >> >> <sip:1000 at 69.204.30.67:16006;rinstance=50cbd6140e1ab991>
> >> >> >> Agent:          X-Lite release 1103k stamp 53621
> >> >> >> Status:         Registered(UDP)(unknown) EXP(2009-08-31 16:24:11)
> >> >> >> Host:           server1.altpressonline.com
> >> >> >> IP:             69.204.30.67
> >> >> >> Port:           16006
> >> >> >> Auth-User:      1000
> >> >> >> Auth-Realm:     server1.altpressonline.com
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> >> >> >>
> =================================================================================================
> >> >> >>
> >> >> >> sorry, i don't know how to login to pastebin
> >> >> >>
> >> >> >> On Mon, Aug 31, 2009 at 1:42 PM, Michael Collins<
> msc at freeswitch.org>
> >> >> >> wrote:
> >> >> >> > Could you give a few more details? For example, could you
> pastebin
> >> >> >> > the
> >> >> >> > output of "sofia status profile external"?
> >> >> >> > -MC
> >> >> >> >
> >> >> >> > On Mon, Aug 31, 2009 at 10:20 AM, e schmidbauer
> >> >> >> > <e.schmidbauer at gmail.com>
> >> >> >> > wrote:
> >> >> >> >>
> >> >> >> >> I am unable to call a user outside of our local area network.
> the
> >> >> >> >> user
> >> >> >> >> is registered on the external profile but there is no way to
> call
> >> >> >> >> the
> >> >> >> >> phone. does anyone have any suggestions how to do this?
> >> >> >> >>
> >> >> >> >> _______________________________________________
> >> >> >> >> FreeSWITCH-users mailing list
> >> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >>
> >> >> >> >>
> >> >> >> >>
> >> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> >> http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >> > _______________________________________________
> >> >> >> > FreeSWITCH-users mailing list
> >> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >> >
> >> >> >> >
> >> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> > http://www.freeswitch.org
> >> >> >> >
> >> >> >> >
> >> >> >>
> >> >> >> _______________________________________________
> >> >> >> FreeSWITCH-users mailing list
> >> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >>
> >> >> >>
> >> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> >> http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >> > _______________________________________________
> >> >> > FreeSWITCH-users mailing list
> >> >> > FreeSWITCH-users at lists.freeswitch.org
> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >> >
> >> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> > http://www.freeswitch.org
> >> >> >
> >> >> >
> >> >>
> >> >> _______________________________________________
> >> >> FreeSWITCH-users mailing list
> >> >> FreeSWITCH-users at lists.freeswitch.org
> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> >>
> >> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> >> http://www.freeswitch.org
> >> >
> >> >
> >> > _______________________________________________
> >> > FreeSWITCH-users mailing list
> >> > FreeSWITCH-users at lists.freeswitch.org
> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> > UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> > http://www.freeswitch.org
> >> >
> >> >
> >>
> >> _______________________________________________
> >> FreeSWITCH-users mailing list
> >> FreeSWITCH-users at lists.freeswitch.org
> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> >> UNSUBSCRIBE:
> http://lists.freeswitch.org/mailman/options/freeswitch-users
> >> http://www.freeswitch.org
> >
> >
> > _______________________________________________
> > FreeSWITCH-users mailing list
> > FreeSWITCH-users at lists.freeswitch.org
> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> > http://www.freeswitch.org
> >
> >
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From msc at freeswitch.org  Mon Aug 31 23:33:50 2009
From: msc at freeswitch.org (Michael Collins)
Date: Mon, 31 Aug 2009 23:33:50 -0700
Subject: [Freeswitch-users] problem compiling esl for use with freepbx v3
In-Reply-To: <20090901054337.7317F65B@sinclaire.sibble.net>
References: <20090823185152.D17845FE@sinclaire.sibble.net>
	<20090901054337.7317F65B@sinclaire.sibble.net>
Message-ID: <87f2f3b90908312333s5bf47d6et94ec24fbc51d6d24@mail.gmail.com>

Did the simple "make" in the libs/esl directory run properly? Just curious.
I'll have to defer to the Ubuntu gurus out there for thoughts on what else
could be wrong.
-MC

On Mon, Aug 31, 2009 at 10:43 PM, Harondel J. Sibble <help at pdscc.com> wrote:

> Haven't had any responses, anyone have any ideas on the problem with
> compiling the ESL modules as below?
>
> On 23 Aug 2009 at 11:51, Harondel J. Sibble wrote:
>
> > Okay, just finished installing 1.0.4 from tarball on Ubuntu 9.0.4 server,
> > then went to install FreePBX v3, I've gotten all the prerequisities in
> the
> > wizard fixed except for ESL
> >
> > As per
> >
> > http://wiki.freeswitch.org/wiki/Event_Socket_Library
> > http://wiki.freeswitch.org/wiki/Event_Socket
> >
> > I go into my FS source dir
> >
> > /home/sibbleh/freeswitch-1.0.4/libs/esl
> >
> > Run make and then "sudo make phpmod-install"
> >
> > and I get
> >
> >
> > $ sudo make phpmod-install
> > make MYLIB="../libesl.a" SOLINK="-shared -Xlinker -x" CFLAGS="-
> > I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> > -ggdb
> > -I../../libs/libedit/src/ -fPIC -O2 -ffast-math -Wall -Werror -Wunused-
> > variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
> > CXXFLAGS="-
> > I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include -DHAVE_EDITLINE -g
> > -ggdb
> > -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable"
> > CXX_CFLAGS="" -C php
> > make[1]: Entering directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> > g++  -I/home/sibbleh/freeswitch-1.0.4/libs/esl/src/include
> -DHAVE_EDITLINE
> > -g
> > -ggdb -I../../libs/libedit/src/ -fPIC -Wall -Werror -Wno-unused-variable
> -
> > I/usr/include/php5 -I/usr/include/php5/main -I/usr/include/php5/TSRM -
> > I/usr/include/php5/Zend -I/usr/include/php5/ext -
> > I/usr/include/php5/ext/date/lib -D_LARGEFILE_SOURCE
> -D_FILE_OFFSET_BITS=64
> > -
> > Wno-unused-label -Wno-unused-function -c esl_wrap.cpp -o esl_wrap.o
> > cc1plus: warnings being treated as errors
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_event_set(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1047: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_event_get(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1073: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_set(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:1111: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_serialized_string_get(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:1141: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_set(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1172: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_mine_get(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1198: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_0(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1234: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_1(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1269: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent__SWIG_2(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1294: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLevent(int, zval*, zval**,
> > zval*,
> > int)':
> > esl_wrap.cpp:1346: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_serialize(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1403: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_setPriority(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1441: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_getHeader(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1478: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_getBody(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1508: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_getType(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1538: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_addBody(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1571: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_addHeader(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1611: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_delHeader(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1644: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_firstHeader(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1674: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLevent_nextHeader(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:1704: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_0(int,
> zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1744: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection__SWIG_1(int,
> zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1770: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_new_ESLconnection(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1803: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_socketDescriptor(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:1846: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_connected(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1872: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_getInfo(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1898: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_send(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:1931: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendRecv(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:1964: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_api(int, zval*,
> zval**,
> > zval*, int)':
> > esl_wrap.cpp:2007: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_bgapi(int, zval*,
> > zval**,
> > zval*, int)':
> > esl_wrap.cpp:2050: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_sendEvent(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2082: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEvent(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2108: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_recvEventTimed(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:2141: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_filter(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2181: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_events(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2221: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_execute(int, zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2272: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_executeAsync(int,
> zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2323: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_setAsyncExecute(int,
> > zval*, zval**, zval*, int)':
> > esl_wrap.cpp:2356: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_setEventLock(int,
> zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2389: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_ESLconnection_disconnect(int,
> zval*,
> > zval**, zval*, int)':
> > esl_wrap.cpp:2415: error: format not a string literal and no format
> > arguments
> > esl_wrap.cpp: In function 'void _wrap_eslSetLogLevel(int, zval*, zval**,
> > zval*, int)':
> > esl_wrap.cpp:2438: error: format not a string literal and no format
> > arguments
> > make[1]: *** [esl_wrap.o] Error 1
> > make[1]: Leaving directory `/home/sibbleh/freeswitch-1.0.4/libs/esl/php'
> > make: *** [phpmod] Error 2
> >
> > Same thing happens if I try sudo make everymod
> >
> > Checking the list archives I found this thread
> >
> > http://www.nabble.com/ESL-Wrapper-td22209991.html#a22222338
> >
> > I've made sure that the php-dev packages are installed. Any suggestions
> on
> > what to do next?
> > --
>
> --
> Harondel J. Sibble
> Sibble Computer Consulting
> Creating Solutions for the small and medium business computer user.
> help at pdscc.com (use pgp keyid 0x3AD5C11D) http://www.pdscc.com
> (604) 739-3709 (voice)
>
>
> _______________________________________________
> FreeSWITCH-users mailing list
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
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From peter.olsson at visionutveckling.se  Mon Aug 31 23:37:41 2009
From: peter.olsson at visionutveckling.se (Peter Olsson)
Date: Tue, 1 Sep 2009 08:37:41 +0200
Subject: [Freeswitch-users] mod_opal
In-Reply-To: <65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>
References: <65d96fc80908311548r27521b27if78b05fac2d0d42d@mail.gmail.com>
	<87f2f3b90908311600h43468c1ao70719aa64479d226@mail.gmail.com>
	<65d96fc80908312309n204f9e1ew983f77c505326ba1@mail.gmail.com>
Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C52ED0F4743@cooper>

Please look into MODOPAL-10 in jira. You need to apply a patch if you're using latest opal trunk, ro else you need to use the latest stable version of opal. However, I'm not sure how automated this is in the build process in Linux. I've only done this on Windows builds lately.

/Peter

Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Tihomir Culjaga
Skickat: den 1 september 2009 08:09
Till: freeswitch-users at lists.freeswitch.org
?mne: Re: [Freeswitch-users] mod_opal

hhmmm :))

is there any doc following up mod_opal ?
I really don't want to waste your time :)

T.

On Tue, Sep 1, 2009 at 1:00 AM, Michael Collins <msc at freeswitch.org<mailto:msc at freeswitch.org>> wrote:

On Mon, Aug 31, 2009 at 3:48 PM, Tihomir Culjaga <tculjaga at gmail.com<mailto:tculjaga at gmail.com>> wrote:
hello,

i'm trying to build mod_opal and getting this error:



making all mod_logfile

making all mod_loopback

making all mod_native_file

making all mod_opal
Compiling mod_opal.cpp...
quiet_libtool: compile:  g++ -g -ggdb -I. -I/home/tculjaga/freeswitch-trunk/src/include -I/home/tculjaga/freeswitch-trunk/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -O2 -D_GNU_SOURCE -DPTRACING=1 -D_REENTRANT -fno-exceptions -I/usr/include/opal -DHAVE_CONFIG_H -c mod_opal.cpp  -fPIC -DPIC -o .libs/mod_opal.o
In file included from mod_opal.cpp:25:
mod_opal.h:151: error: conflicting return type specified for 'virtual OpalLocalConnection* FSEndPoint::CreateConnection(OpalCall&, void*)'
/usr/include/opal/opal/localep.h:267: error:   overriding 'virtual ptlib_virtual_function_changed_or_removed****** OpalLocalEndPoint::CreateConnection(OpalCall&, void*)'
mod_opal.cpp: In constructor 'FSConnection::FSConnection(OpalCall&, FSEndPoint&, switch_caller_profile_t*, switch_core_session_t*, switch_channel_t*)':
mod_opal.cpp:564: error: no matching function for call to 'OpalLocalConnection::OpalLocalConnection(OpalCall&, FSEndPoint&, NULL)'
/usr/include/opal/opal/localep.h:290: note: candidates are: OpalLocalConnection::OpalLocalConnection(OpalCall&, OpalLocalEndPoint&, void*, unsigned int, OpalConnection::StringOptions*, char)
/usr/include/opal/opal/localep.h:276: note:                 OpalLocalConnection::OpalLocalConnection(const OpalLocalConnection&)
/usr/include/opal/opal/patch.h: In member function 'switch_status_t FSMediaStream::read_frame(switch_frame_t**, switch_io_flag_t)':
/usr/include/opal/opal/patch.h:272: error: 'virtual ptlib_virtual_function_changed_or_removed****** OpalMediaPatch::OnPatchStart()' is private
mod_opal.cpp:1277: error: within this context
mod_opal.cpp:1277: warning: 'virtual ptlib_virtual_function_changed_or_removed****** OpalMediaPatch::OnPatchStart()' is deprecated (declared at /usr/include/opal/opal/patch.h:272)
mod_opal.cpp:1277: warning: ignoring return value of function declared with attribute warn_unused_result
/usr/include/opal/opal/patch.h: In member function 'switch_status_t FSMediaStream::write_frame(const switch_frame_t*, switch_io_flag_t)':
/usr/include/opal/opal/patch.h:272: error: 'virtual ptlib_virtual_function_changed_or_removed****** OpalMediaPatch::OnPatchStart()' is private
mod_opal.cpp:1399: error: within this context
mod_opal.cpp:1399: warning: 'virtual ptlib_virtual_function_changed_or_removed****** OpalMediaPatch::OnPatchStart()' is deprecated (declared at /usr/include/opal/opal/patch.h:272)
mod_opal.cpp:1399: warning: ignoring return value of function declared with attribute warn_unused_result
make[5]: *** [mod_opal.lo] Error 1
make[4]: *** [all] Error 1
make[3]: *** [mod_opal-all] Error 1
make[2]: *** [all-recursive] Error 1
Making all in build
 +-------- FreeSWITCH Build Complete -----------+
 + FreeSWITCH has been successfully built.      +
 + Install by running:                          +
 +                                              +
 +               make install                   +
 +----------------------------------------------+
make[1]: *** [all-recursive] Error 1
make: *** [all] Error 2
tculjaga at nemesis:~/freeswitch-trunk$
tculjaga at nemesis:~/freeswitch-trunk$
tculjaga at nemesis:~/freeswitch-trunk$



what ptlib/opal/fs version did you use to build it?


I tried with trunk (ptlib, opal, fs)... and as you can see :)

Did you run the buildopal.sh script in src/build ?
-MC


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